From 2998369fb93f129b953aeb2984ae01e26c4fdf69 Mon Sep 17 00:00:00 2001 From: Rohit kumar Date: Fri, 14 Dec 2018 15:31:43 +0530 Subject: [PATCH 01/67] ASoC: sdm845: set jack only for a specific backend Headset codec is connected over PRIMARY_MI2S interface. Call set_jack for codec associated with Primary Mi2s interface. Also, set_jack to NULL when jack is freed. Signed-off-by: Rohit kumar Signed-off-by: Mark Brown --- sound/soc/qcom/sdm845.c | 31 ++++++++++++++++++++++--------- 1 file changed, 22 insertions(+), 9 deletions(-) diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index 1db8ef668223..6f66a58e23ca 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -158,17 +158,24 @@ static int sdm845_snd_hw_params(struct snd_pcm_substream *substream, return ret; } +static void sdm845_jack_free(struct snd_jack *jack) +{ + struct snd_soc_component *component = jack->private_data; + + snd_soc_component_set_jack(component, NULL, NULL); +} + static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_component *component; - struct snd_soc_dai_link *dai_link = rtd->dai_link; struct snd_soc_card *card = rtd->card; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct sdm845_snd_data *pdata = snd_soc_card_get_drvdata(card); - int i, rval; + struct snd_jack *jack; + int rval; if (!pdata->jack_setup) { - struct snd_jack *jack; - rval = snd_soc_card_jack_new(card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_HEADPHONE | @@ -190,16 +197,22 @@ static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd) pdata->jack_setup = true; } - for (i = 0 ; i < dai_link->num_codecs; i++) { - struct snd_soc_dai *dai = rtd->codec_dais[i]; + switch (cpu_dai->id) { + case PRIMARY_MI2S_RX: + jack = pdata->jack.jack; + component = codec_dai->component; - component = dai->component; - rval = snd_soc_component_set_jack( - component, &pdata->jack, NULL); + jack->private_data = component; + jack->private_free = sdm845_jack_free; + rval = snd_soc_component_set_jack(component, + &pdata->jack, NULL); if (rval != 0 && rval != -ENOTSUPP) { dev_warn(card->dev, "Failed to set jack: %d\n", rval); return rval; } + break; + default: + break; } return 0; From 02a07872f84fc5fe61fa65310ff23bcad166a4f5 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Tue, 18 Dec 2018 11:18:10 +0300 Subject: [PATCH 02/67] ASoC: dma-sh7760: cleanup a debug printk The intent was to print the address as a hexadecimal but there is an extra "u" in the "0x%08ulx" format specification so it is displayed as decimal. Fixes: aef3b06ac697 ("[ALSA] SH7760 ASoC support") Signed-off-by: Dan Carpenter Signed-off-by: Mark Brown --- sound/soc/sh/dma-sh7760.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c index 922fb6aa3ed1..5aee11c94f2a 100644 --- a/sound/soc/sh/dma-sh7760.c +++ b/sound/soc/sh/dma-sh7760.c @@ -202,7 +202,7 @@ static int camelot_prepare(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct camelot_pcm *cam = &cam_pcm_data[rtd->cpu_dai->id]; - pr_debug("PCM data: addr 0x%08ulx len %d\n", + pr_debug("PCM data: addr 0x%08lx len %d\n", (u32)runtime->dma_addr, runtime->dma_bytes); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { From 6cb6746e95576878835cd27f634194bbd771c3f2 Mon Sep 17 00:00:00 2001 From: Geert Uytterhoeven Date: Tue, 18 Dec 2018 14:47:43 +0100 Subject: [PATCH 03/67] ASoC: xlnx: Grammar s/the the/the/ Fixes: 33f8db9a89200c18 ("ASoC: xlnx: enable i2s driver build") Signed-off-by: Geert Uytterhoeven Signed-off-by: Mark Brown --- sound/soc/xilinx/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/xilinx/Kconfig b/sound/soc/xilinx/Kconfig index 25e287feb58c..723a583a8d57 100644 --- a/sound/soc/xilinx/Kconfig +++ b/sound/soc/xilinx/Kconfig @@ -1,5 +1,5 @@ config SND_SOC_XILINX_I2S - tristate "Audio support for the the Xilinx I2S" + tristate "Audio support for the Xilinx I2S" help Select this option to enable Xilinx I2S Audio. This enables I2S playback and capture using xilinx soft IP. In transmitter From 906a9abc5de73c383af518f5a806f4be2993a0c7 Mon Sep 17 00:00:00 2001 From: Rander Wang Date: Tue, 18 Dec 2018 16:24:54 +0800 Subject: [PATCH 04/67] ASoC: Intel: Haswell/Broadwell: fix setting for .dynamic field For some reason this field was set to zero when all other drivers use .dynamic = 1 for front-ends. This change was tested on Dell XPS13 and has no impact with the existing legacy driver. The SOF driver also works with this change which enables it to override the fixed topology. Signed-off-by: Rander Wang Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/broadwell.c | 2 +- sound/soc/intel/boards/haswell.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c index 68e6543e6cb0..99f2a0156ae8 100644 --- a/sound/soc/intel/boards/broadwell.c +++ b/sound/soc/intel/boards/broadwell.c @@ -192,7 +192,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { .stream_name = "Loopback", .cpu_dai_name = "Loopback Pin", .platform_name = "haswell-pcm-audio", - .dynamic = 0, + .dynamic = 1, .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c index eab1f439dd3f..a4022983a7ce 100644 --- a/sound/soc/intel/boards/haswell.c +++ b/sound/soc/intel/boards/haswell.c @@ -146,7 +146,7 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = { .stream_name = "Loopback", .cpu_dai_name = "Loopback Pin", .platform_name = "haswell-pcm-audio", - .dynamic = 0, + .dynamic = 1, .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, From fd270fca2001bcdac0658eb673c20920baeed86c Mon Sep 17 00:00:00 2001 From: Maruthi Srinivas Bayyavarapu Date: Wed, 19 Dec 2018 15:10:40 +0530 Subject: [PATCH 05/67] ASoC: xlnx: change license header format style Changed License header from C to C++ style comment block. Signed-off-by: Maruthi Srinivas Bayyavarapu Signed-off-by: Mark Brown --- sound/soc/xilinx/xlnx_i2s.c | 15 +++++++-------- 1 file changed, 7 insertions(+), 8 deletions(-) diff --git a/sound/soc/xilinx/xlnx_i2s.c b/sound/soc/xilinx/xlnx_i2s.c index d4ae9eff41ce..8b353166ad44 100644 --- a/sound/soc/xilinx/xlnx_i2s.c +++ b/sound/soc/xilinx/xlnx_i2s.c @@ -1,12 +1,11 @@ // SPDX-License-Identifier: GPL-2.0 -/* - * Xilinx ASoC I2S audio support - * - * Copyright (C) 2018 Xilinx, Inc. - * - * Author: Praveen Vuppala - * Author: Maruthi Srinivas Bayyavarapu - */ +// +// Xilinx ASoC I2S audio support +// +// Copyright (C) 2018 Xilinx, Inc. +// +// Author: Praveen Vuppala +// Author: Maruthi Srinivas Bayyavarapu #include #include From 28b698b7342c7d5300cfe217cd77ff7d2a55e03d Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Fri, 21 Dec 2018 12:11:20 +0300 Subject: [PATCH 06/67] ASoC: pcm512x: Fix a double unlock in pcm512x_digital_mute() We accidentally call mutex_unlock(&pcm512x->mutex); twice in a row. I re-wrote the error handling to use "goto unlock;" instead of returning directly. Hopefully, it makes the code a little simpler. Fixes: 3500f1c589e9 ("ASoC: pcm512x: Implement the digital_mute interface") Signed-off-by: Dan Carpenter Reviwed-by: Dimitris Papavasiliou Signed-off-by: Mark Brown --- sound/soc/codecs/pcm512x.c | 11 ++++------- 1 file changed, 4 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 6cb1653be804..4cc24a5d5c31 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -1400,24 +1400,20 @@ static int pcm512x_digital_mute(struct snd_soc_dai *dai, int mute) if (ret != 0) { dev_err(component->dev, "Failed to set digital mute: %d\n", ret); - mutex_unlock(&pcm512x->mutex); - return ret; + goto unlock; } regmap_read_poll_timeout(pcm512x->regmap, PCM512x_ANALOG_MUTE_DET, mute_det, (mute_det & 0x3) == 0, 200, 10000); - - mutex_unlock(&pcm512x->mutex); } else { pcm512x->mute &= ~0x1; ret = pcm512x_update_mute(pcm512x); if (ret != 0) { dev_err(component->dev, "Failed to update digital mute: %d\n", ret); - mutex_unlock(&pcm512x->mutex); - return ret; + goto unlock; } regmap_read_poll_timeout(pcm512x->regmap, @@ -1428,9 +1424,10 @@ static int pcm512x_digital_mute(struct snd_soc_dai *dai, int mute) 200, 10000); } +unlock: mutex_unlock(&pcm512x->mutex); - return 0; + return ret; } static const struct snd_soc_dai_ops pcm512x_dai_ops = { From eef08e5350618b7a9fdc8ac5b821a925366c8f3f Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Fri, 21 Dec 2018 12:04:42 +0300 Subject: [PATCH 07/67] ASoC: qdsp6: q6asm-dai: Off by one in of_q6asm_parse_dai_data() The q6asm_fe_dais[] array has MAX_SESSIONS (8) elements so the > comparison should be >= or we access one element beyond the end of the array. Fixes: 22930c79ac5c ("ASoC: qdsp6: q6asm-dai: Add support to compress offload") Signed-off-by: Dan Carpenter Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6asm-dai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index 5b986b74dd36..9d738b4c1e05 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -874,7 +874,7 @@ static int of_q6asm_parse_dai_data(struct device *dev, for_each_child_of_node(dev->of_node, node) { ret = of_property_read_u32(node, "reg", &id); - if (ret || id > MAX_SESSIONS || id < 0) { + if (ret || id >= MAX_SESSIONS || id < 0) { dev_err(dev, "valid dai id not found:%d\n", ret); continue; } From 3391034e18b35bba8904cae457598ac276ac685a Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Fri, 21 Dec 2018 12:05:16 +0300 Subject: [PATCH 08/67] ASoC: qdsp6: q6asm-dai: Fix a NULL vs IS_ERR() bug The q6asm_audio_client_alloc() doesn't return NULL, it returns error pointers. Fixes: 22930c79ac5c ("ASoC: qdsp6: q6asm-dai: Add support to compress offload") Signed-off-by: Dan Carpenter Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6asm-dai.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index 9d738b4c1e05..3407e51b8861 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -570,10 +570,11 @@ static int q6asm_dai_compr_open(struct snd_compr_stream *stream) prtd->audio_client = q6asm_audio_client_alloc(dev, (q6asm_cb)compress_event_handler, prtd, stream_id, LEGACY_PCM_MODE); - if (!prtd->audio_client) { + if (IS_ERR(prtd->audio_client)) { dev_err(dev, "Could not allocate memory\n"); + ret = PTR_ERR(prtd->audio_client); kfree(prtd); - return -ENOMEM; + return ret; } size = COMPR_PLAYBACK_MAX_FRAGMENT_SIZE * From a41d9dbf5dac5b6a1283ee8001f22807d18352ea Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Fri, 21 Dec 2018 12:06:10 +0300 Subject: [PATCH 09/67] ASoC: qdsp6: q6asm-dai: Fix a small memory leak We can't return directly if snd_dma_alloc_pages() fails; we first need to free prtd->audio_client and prtd. Fixes: 22930c79ac5c ("ASoC: qdsp6: q6asm-dai: Add support to compress offload") Signed-off-by: Dan Carpenter Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6asm-dai.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index 3407e51b8861..548eb4fa2da6 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -573,8 +573,7 @@ static int q6asm_dai_compr_open(struct snd_compr_stream *stream) if (IS_ERR(prtd->audio_client)) { dev_err(dev, "Could not allocate memory\n"); ret = PTR_ERR(prtd->audio_client); - kfree(prtd); - return ret; + goto free_prtd; } size = COMPR_PLAYBACK_MAX_FRAGMENT_SIZE * @@ -583,7 +582,7 @@ static int q6asm_dai_compr_open(struct snd_compr_stream *stream) &prtd->dma_buffer); if (ret) { dev_err(dev, "Cannot allocate buffer(s)\n"); - return ret; + goto free_client; } if (pdata->sid < 0) @@ -596,6 +595,13 @@ static int q6asm_dai_compr_open(struct snd_compr_stream *stream) runtime->private_data = prtd; return 0; + +free_client: + q6asm_audio_client_free(prtd->audio_client); +free_prtd: + kfree(prtd); + + return ret; } static int q6asm_dai_compr_free(struct snd_compr_stream *stream) From 678e2b44c8e3fec3afc7202f1996a4500a50be93 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Fri, 21 Dec 2018 12:06:58 +0300 Subject: [PATCH 10/67] ALSA: compress: prevent potential divide by zero bugs The problem is seen in the q6asm_dai_compr_set_params() function: ret = q6asm_map_memory_regions(dir, prtd->audio_client, prtd->phys, (prtd->pcm_size / prtd->periods), ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ prtd->periods); In this code prtd->pcm_size is the buffer_size and prtd->periods comes from params->buffer.fragments. If we allow the number of fragments to be zero then it results in a divide by zero bug. One possible fix would be to use prtd->pcm_count directly instead of using the division to re-calculate it. But I decided that it doesn't really make sense to allow zero fragments. Signed-off-by: Dan Carpenter Signed-off-by: Mark Brown --- sound/core/compress_offload.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index a5b09e75e787..f7d2b373da0a 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -541,7 +541,8 @@ static int snd_compress_check_input(struct snd_compr_params *params) { /* first let's check the buffer parameter's */ if (params->buffer.fragment_size == 0 || - params->buffer.fragments > INT_MAX / params->buffer.fragment_size) + params->buffer.fragments > INT_MAX / params->buffer.fragment_size || + params->buffer.fragments == 0) return -EINVAL; /* now codec parameters */ From a3d9036078715385ba156373e6cbc1a0b1deb075 Mon Sep 17 00:00:00 2001 From: Sinan Kaya Date: Wed, 2 Jan 2019 18:10:35 +0000 Subject: [PATCH 11/67] ASoC: Intel: atom: Make PCI dependency explicit After 'commit 5d32a66541c4 ("PCI/ACPI: Allow ACPI to be built without CONFIG_PCI set")' dependencies on CONFIG_PCI that previously were satisfied implicitly through dependencies on CONFIG_ACPI have to be specified directly. This code relies on IOSF_MBI and IOSF_MBI depends on PCI. For this reason, add a direct dependency on CONFIG_PCI to the IOSF_MBI driver. Fixes: 5d32a66541c46 ("PCI/ACPI: Allow ACPI to be built without CONFIG_PCI set") Signed-off-by: Sinan Kaya Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 99a62ba409df..bd9fd2035c55 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -91,7 +91,7 @@ config SND_SST_ATOM_HIFI2_PLATFORM_PCI config SND_SST_ATOM_HIFI2_PLATFORM_ACPI tristate "ACPI HiFi2 (Baytrail, Cherrytrail) Platforms" default ACPI - depends on X86 && ACPI + depends on X86 && ACPI && PCI select SND_SST_IPC_ACPI select SND_SST_ATOM_HIFI2_PLATFORM select SND_SOC_ACPI_INTEL_MATCH From 22c7d5e7bad1fb2d8b9c611acb55a389f5d848d8 Mon Sep 17 00:00:00 2001 From: Shuming Fan Date: Wed, 2 Jan 2019 17:18:56 +0800 Subject: [PATCH 12/67] ASoC: rt5682: Fix recording no sound issue The ADC mixer setting needs to restore to default value after calibration. Signed-off-by: Shuming Fan Signed-off-by: Mark Brown --- sound/soc/codecs/rt5682.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 34cfaf8f6f34..89c43b26c379 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -2512,6 +2512,7 @@ static void rt5682_calibrate(struct rt5682_priv *rt5682) regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0000); regmap_write(rt5682->regmap, RT5682_CHOP_DAC, 0x2000); regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x2005); + regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0xc0c4); mutex_unlock(&rt5682->calibrate_mutex); From 0f68c396f6048cf87c662aab1ef9c9aa237153a8 Mon Sep 17 00:00:00 2001 From: Alexander Shiyan Date: Thu, 20 Dec 2018 10:36:12 +0300 Subject: [PATCH 13/67] ASoC: cs4341: Add driver for CS4341 DAC This patch adds Cirrus Logic CS4341. This is a very simple, playback only, stereo DAC. Signed-off-by: Alexander Shiyan Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs4341.c | 346 ++++++++++++++++++++++++++++++++++++++ 3 files changed, 355 insertions(+) create mode 100644 sound/soc/codecs/cs4341.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 62bdb7e333b8..3f742753abd1 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -65,6 +65,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS4271_SPI if SPI_MASTER select SND_SOC_CS42XX8_I2C if I2C select SND_SOC_CS43130 if I2C + select SND_SOC_CS4341 if SND_SOC_I2C_AND_SPI select SND_SOC_CS4349 if I2C select SND_SOC_CS47L24 if MFD_CS47L24 select SND_SOC_CS53L30 if I2C @@ -542,6 +543,12 @@ config SND_SOC_CS43130 tristate "Cirrus Logic CS43130 CODEC" depends on I2C +config SND_SOC_CS4341 + tristate "Cirrus Logic CS4341 CODEC" + depends on I2C || SPI_MASTER + select REGMAP_I2C if I2C + select REGMAP_SPI if SPI_MASTER + # Cirrus Logic CS4349 HiFi DAC config SND_SOC_CS4349 tristate "Cirrus Logic CS4349 CODEC" diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 66f55d185620..fbe36e6177b0 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -60,6 +60,7 @@ snd-soc-cs4271-spi-objs := cs4271-spi.o snd-soc-cs42xx8-objs := cs42xx8.o snd-soc-cs42xx8-i2c-objs := cs42xx8-i2c.o snd-soc-cs43130-objs := cs43130.o +snd-soc-cs4341-objs := cs4341.o snd-soc-cs4349-objs := cs4349.o snd-soc-cs47l24-objs := cs47l24.o snd-soc-cs53l30-objs := cs53l30.o @@ -326,6 +327,7 @@ obj-$(CONFIG_SND_SOC_CS4271_SPI) += snd-soc-cs4271-spi.o obj-$(CONFIG_SND_SOC_CS42XX8) += snd-soc-cs42xx8.o obj-$(CONFIG_SND_SOC_CS42XX8_I2C) += snd-soc-cs42xx8-i2c.o obj-$(CONFIG_SND_SOC_CS43130) += snd-soc-cs43130.o +obj-$(CONFIG_SND_SOC_CS4341) += snd-soc-cs4341.o obj-$(CONFIG_SND_SOC_CS4349) += snd-soc-cs4349.o obj-$(CONFIG_SND_SOC_CS47L24) += snd-soc-cs47l24.o obj-$(CONFIG_SND_SOC_CS53L30) += snd-soc-cs53l30.o diff --git a/sound/soc/codecs/cs4341.c b/sound/soc/codecs/cs4341.c new file mode 100644 index 000000000000..d2e616a89fd4 --- /dev/null +++ b/sound/soc/codecs/cs4341.c @@ -0,0 +1,346 @@ +/* SPDX-License-Identifier: GPL-2.0+ */ +/* + * Cirrus Logic CS4341A ALSA SoC Codec Driver + * Author: Alexander Shiyan + */ + +#include +#include +#include +#include +#include + +#include +#include +#include +#include + +#define CS4341_REG_MODE1 0x00 +#define CS4341_REG_MODE2 0x01 +#define CS4341_REG_MIX 0x02 +#define CS4341_REG_VOLA 0x03 +#define CS4341_REG_VOLB 0x04 + +#define CS4341_MODE2_DIF (7 << 4) +#define CS4341_MODE2_DIF_I2S_24 (0 << 4) +#define CS4341_MODE2_DIF_I2S_16 (1 << 4) +#define CS4341_MODE2_DIF_LJ_24 (2 << 4) +#define CS4341_MODE2_DIF_RJ_24 (3 << 4) +#define CS4341_MODE2_DIF_RJ_16 (5 << 4) +#define CS4341_VOLX_MUTE (1 << 7) + +struct cs4341_priv { + unsigned int fmt; + struct regmap *regmap; + struct regmap_config regcfg; +}; + +static const struct reg_default cs4341_reg_defaults[] = { + { CS4341_REG_MODE1, 0x00 }, + { CS4341_REG_MODE2, 0x82 }, + { CS4341_REG_MIX, 0x49 }, + { CS4341_REG_VOLA, 0x80 }, + { CS4341_REG_VOLB, 0x80 }, +}; + +static int cs4341_set_fmt(struct snd_soc_dai *dai, unsigned int format) +{ + struct snd_soc_component *component = dai->component; + struct cs4341_priv *cs4341 = snd_soc_component_get_drvdata(component); + + switch (format & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (format & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } + + switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_RIGHT_J: + cs4341->fmt = format & SND_SOC_DAIFMT_FORMAT_MASK; + break; + default: + return -EINVAL; + } + + return 0; +} + +static int cs4341_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct cs4341_priv *cs4341 = snd_soc_component_get_drvdata(component); + unsigned int mode = 0; + int b24 = 0; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S24_LE: + b24 = 1; + break; + case SNDRV_PCM_FORMAT_S16_LE: + break; + default: + dev_err(component->dev, "Unsupported PCM format 0x%08x.\n", + params_format(params)); + return -EINVAL; + } + + switch (cs4341->fmt) { + case SND_SOC_DAIFMT_I2S: + mode = b24 ? CS4341_MODE2_DIF_I2S_24 : CS4341_MODE2_DIF_I2S_16; + break; + case SND_SOC_DAIFMT_LEFT_J: + mode = CS4341_MODE2_DIF_LJ_24; + break; + case SND_SOC_DAIFMT_RIGHT_J: + mode = b24 ? CS4341_MODE2_DIF_RJ_24 : CS4341_MODE2_DIF_RJ_16; + break; + default: + dev_err(component->dev, "Unsupported DAI format 0x%08x.\n", + cs4341->fmt); + return -EINVAL; + } + + return snd_soc_component_update_bits(component, CS4341_REG_MODE2, + CS4341_MODE2_DIF, mode); +} + +static int cs4341_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_component *component = dai->component; + int ret; + + ret = snd_soc_component_update_bits(component, CS4341_REG_VOLA, + CS4341_VOLX_MUTE, + mute ? CS4341_VOLX_MUTE : 0); + if (ret < 0) + return ret; + + return snd_soc_component_update_bits(component, CS4341_REG_VOLB, + CS4341_VOLX_MUTE, + mute ? CS4341_VOLX_MUTE : 0); +} + +static DECLARE_TLV_DB_SCALE(out_tlv, -9000, 100, 0); + +static const char * const deemph[] = { + "None", "44.1k", "48k", "32k", +}; + +static const struct soc_enum deemph_enum = + SOC_ENUM_SINGLE(CS4341_REG_MODE2, 2, 4, deemph); + +static const char * const srzc[] = { + "Immediate", "Zero Cross", "Soft Ramp", "SR on ZC", +}; + +static const struct soc_enum srzc_enum = + SOC_ENUM_SINGLE(CS4341_REG_MIX, 5, 4, srzc); + + +static const struct snd_soc_dapm_widget cs4341_dapm_widgets[] = { + SND_SOC_DAPM_DAC("HiFi DAC", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_OUTPUT("OutA"), + SND_SOC_DAPM_OUTPUT("OutB"), +}; + +static const struct snd_soc_dapm_route cs4341_routes[] = { + { "OutA", NULL, "HiFi DAC" }, + { "OutB", NULL, "HiFi DAC" }, + { "DAC Playback", NULL, "OutA" }, + { "DAC Playback", NULL, "OutB" }, +}; + +static const struct snd_kcontrol_new cs4341_controls[] = { + SOC_DOUBLE_R_TLV("Master Playback Volume", + CS4341_REG_VOLA, CS4341_REG_VOLB, 0, 90, 1, out_tlv), + SOC_ENUM("De-Emphasis Control", deemph_enum), + SOC_ENUM("Soft Ramp Zero Cross Control", srzc_enum), + SOC_SINGLE("Auto-Mute Switch", CS4341_REG_MODE2, 7, 1, 0), + SOC_SINGLE("Popguard Transient Switch", CS4341_REG_MODE2, 1, 1, 0), +}; + +static const struct snd_soc_dai_ops cs4341_dai_ops = { + .set_fmt = cs4341_set_fmt, + .hw_params = cs4341_hw_params, + .digital_mute = cs4341_digital_mute, +}; + +static struct snd_soc_dai_driver cs4341_dai = { + .name = "cs4341a-hifi", + .playback = { + .stream_name = "DAC Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + }, + .ops = &cs4341_dai_ops, + .symmetric_rates = 1, +}; + +static const struct snd_soc_component_driver soc_component_cs4341 = { + .controls = cs4341_controls, + .num_controls = ARRAY_SIZE(cs4341_controls), + .dapm_widgets = cs4341_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs4341_dapm_widgets), + .dapm_routes = cs4341_routes, + .num_dapm_routes = ARRAY_SIZE(cs4341_routes), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static const struct of_device_id __maybe_unused cs4341_dt_ids[] = { + { .compatible = "cirrus,cs4341a", }, + { } +}; +MODULE_DEVICE_TABLE(of, cs4341_dt_ids); + +static int cs4341_probe(struct device *dev) +{ + struct cs4341_priv *cs4341 = dev_get_drvdata(dev); + int i; + + for (i = 0; i < ARRAY_SIZE(cs4341_reg_defaults); i++) + regmap_write(cs4341->regmap, cs4341_reg_defaults[i].reg, + cs4341_reg_defaults[i].def); + + return devm_snd_soc_register_component(dev, &soc_component_cs4341, + &cs4341_dai, 1); +} + +#if defined(CONFIG_I2C) +static int cs4341_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct cs4341_priv *cs4341; + + cs4341 = devm_kzalloc(&i2c->dev, sizeof(*cs4341), GFP_KERNEL); + if (!cs4341) + return -ENOMEM; + + i2c_set_clientdata(i2c, cs4341); + + cs4341->regcfg.reg_bits = 8; + cs4341->regcfg.val_bits = 8; + cs4341->regcfg.max_register = CS4341_REG_VOLB; + cs4341->regcfg.cache_type = REGCACHE_FLAT; + cs4341->regcfg.reg_defaults = cs4341_reg_defaults; + cs4341->regcfg.num_reg_defaults = ARRAY_SIZE(cs4341_reg_defaults); + cs4341->regmap = devm_regmap_init_i2c(i2c, &cs4341->regcfg); + if (IS_ERR(cs4341->regmap)) + return PTR_ERR(cs4341->regmap); + + return cs4341_probe(&i2c->dev); +} + +static const struct i2c_device_id cs4341_i2c_id[] = { + { "cs4341", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, cs4341_i2c_id); + +static struct i2c_driver cs4341_i2c_driver = { + .driver = { + .name = "cs4341-i2c", + .of_match_table = of_match_ptr(cs4341_dt_ids), + }, + .probe = cs4341_i2c_probe, + .id_table = cs4341_i2c_id, +}; +#endif + +#if defined(CONFIG_SPI_MASTER) +static bool cs4341_reg_readable(struct device *dev, unsigned int reg) +{ + return false; +} + +static int cs4341_spi_probe(struct spi_device *spi) +{ + struct cs4341_priv *cs4341; + int ret; + + cs4341 = devm_kzalloc(&spi->dev, sizeof(*cs4341), GFP_KERNEL); + if (!cs4341) + return -ENOMEM; + + if (!spi->bits_per_word) + spi->bits_per_word = 8; + if (!spi->max_speed_hz) + spi->max_speed_hz = 6000000; + ret = spi_setup(spi); + if (ret) + return ret; + + spi_set_drvdata(spi, cs4341); + + cs4341->regcfg.reg_bits = 16; + cs4341->regcfg.val_bits = 8; + cs4341->regcfg.write_flag_mask = 0x20; + cs4341->regcfg.max_register = CS4341_REG_VOLB; + cs4341->regcfg.cache_type = REGCACHE_FLAT; + cs4341->regcfg.readable_reg = cs4341_reg_readable; + cs4341->regcfg.reg_defaults = cs4341_reg_defaults; + cs4341->regcfg.num_reg_defaults = ARRAY_SIZE(cs4341_reg_defaults); + cs4341->regmap = devm_regmap_init_spi(spi, &cs4341->regcfg); + if (IS_ERR(cs4341->regmap)) + return PTR_ERR(cs4341->regmap); + + return cs4341_probe(&spi->dev); +} + +static struct spi_driver cs4341_spi_driver = { + .driver = { + .name = "cs4341-spi", + .of_match_table = of_match_ptr(cs4341_dt_ids), + }, + .probe = cs4341_spi_probe, +}; +#endif + +static int __init cs4341_init(void) +{ + int ret = 0; + +#if defined(CONFIG_I2C) + ret = i2c_add_driver(&cs4341_i2c_driver); + if (ret) + return ret; +#endif +#if defined(CONFIG_SPI_MASTER) + ret = spi_register_driver(&cs4341_spi_driver); +#endif + + return ret; +} +module_init(cs4341_init); + +static void __exit cs4341_exit(void) +{ +#if defined(CONFIG_I2C) + i2c_del_driver(&cs4341_i2c_driver); +#endif +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&cs4341_spi_driver); +#endif +} +module_exit(cs4341_exit); + +MODULE_AUTHOR("Alexander Shiyan "); +MODULE_DESCRIPTION("Cirrus Logic CS4341 ALSA SoC Codec Driver"); +MODULE_LICENSE("GPL"); From 0ddb46080a465fad99cff838682744f1f4848a4b Mon Sep 17 00:00:00 2001 From: Alexander Shiyan Date: Thu, 20 Dec 2018 10:37:08 +0300 Subject: [PATCH 14/67] ASoC: cs4341: Add DT bindings documentation for CS4341 DAC This patch adds DT bindings documentation for Cirrus Logic CS4341 DAC. Signed-off-by: Alexander Shiyan Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/cs4341.txt | 22 +++++++++++++++++++ 1 file changed, 22 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs4341.txt diff --git a/Documentation/devicetree/bindings/sound/cs4341.txt b/Documentation/devicetree/bindings/sound/cs4341.txt new file mode 100644 index 000000000000..12b4aa8ef0db --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs4341.txt @@ -0,0 +1,22 @@ +Cirrus Logic CS4341 audio DAC + +This device supports both I2C and SPI (configured with pin strapping +on the board). + +Required properties: + - compatible: "cirrus,cs4341a" + - reg : the I2C address of the device for I2C, the chip select + number for SPI. + +For required properties on I2C-bus, please consult +Documentation/devicetree/bindings/i2c/i2c.txt +For required properties on SPI-bus, please consult +Documentation/devicetree/bindings/spi/spi-bus.txt + +Example: + codec: cs4341@0 { + #sound-dai-cells = <0>; + compatible = "cirrus,cs4341a"; + reg = <0>; + spi-max-frequency = <6000000>; + }; From 2bb853f6f93775dc4dd4683a42f6934700d90d07 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Micha=C5=82=20Miros=C5=82aw?= Date: Wed, 19 Dec 2018 21:11:15 +0100 Subject: [PATCH 15/67] ASoC: wm8904: make the driver visible in Kconfig MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit For platforms that use the audio-graph-card driver, the codec is not selected by SoC-platform driver. Make it available. Signed-off-by: Michał Mirosław Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3f742753abd1..d46de3e04ff6 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1218,7 +1218,8 @@ config SND_SOC_WM8903 depends on I2C config SND_SOC_WM8904 - tristate + tristate "Wolfson Microelectronics WM8904 CODEC" + depends on I2C config SND_SOC_WM8940 tristate From fb82c6ed31902e651cc9324108f507babfabc890 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Micha=C5=82=20Miros=C5=82aw?= Date: Wed, 19 Dec 2018 21:11:16 +0100 Subject: [PATCH 16/67] ASoC: wm8904: save model id directly in of_device_id.data MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Save 2x unsigned int of .rodata. Signed-off-by: Michał Mirosław Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8904.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 2a3e5fbd04e4..9283a2dc70aa 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2108,16 +2108,13 @@ static const struct regmap_config wm8904_regmap = { }; #ifdef CONFIG_OF -static enum wm8904_type wm8904_data = WM8904; -static enum wm8904_type wm8912_data = WM8912; - static const struct of_device_id wm8904_of_match[] = { { .compatible = "wlf,wm8904", - .data = &wm8904_data, + .data = (void *)WM8904, }, { .compatible = "wlf,wm8912", - .data = &wm8912_data, + .data = (void *)WM8912, }, { /* sentinel */ } @@ -2158,7 +2155,7 @@ static int wm8904_i2c_probe(struct i2c_client *i2c, match = of_match_node(wm8904_of_match, i2c->dev.of_node); if (match == NULL) return -EINVAL; - wm8904->devtype = *((enum wm8904_type *)match->data); + wm8904->devtype = (enum wm8904_type)match->data; } else { wm8904->devtype = id->driver_data; } From 5489e81f981b1fb7c2fdaba332122fff3290e9a4 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Micha=C5=82=20Miros=C5=82aw?= Date: Wed, 19 Dec 2018 21:11:16 +0100 Subject: [PATCH 17/67] ASoC: wm8904: enable MCLK in STANDBY MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit MCLK input is needed when accessing any register after enabling SYSCLK. This also fixes imbalance of clk_enable / clk_disable when transitioning between ON -> STANDBY -> ON bias levels. Signed-off-by: Michał Mirosław Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8904.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 9283a2dc70aa..9e0f96e0f8ec 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1837,9 +1837,6 @@ static int wm8904_set_bias_level(struct snd_soc_component *component, switch (level) { case SND_SOC_BIAS_ON: - ret = clk_prepare_enable(wm8904->mclk); - if (ret) - return ret; break; case SND_SOC_BIAS_PREPARE: @@ -1864,6 +1861,15 @@ static int wm8904_set_bias_level(struct snd_soc_component *component, return ret; } + ret = clk_prepare_enable(wm8904->mclk); + if (ret) { + dev_err(component->dev, + "Failed to enable MCLK: %d\n", ret); + regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), + wm8904->supplies); + return ret; + } + regcache_cache_only(wm8904->regmap, false); regcache_sync(wm8904->regmap); From 431b67c27c57bc6a752482727c87f6dda988aae5 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Sun, 16 Dec 2018 16:49:02 -0600 Subject: [PATCH 18/67] ASoC: Intel: Skylake: remove useless cast Detected with Coccinelle sound/soc/intel/skylake/skl-topology.c:3106:16-20: WARNING: casting value returned by memory allocation function to (char *) is useless. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index cf8848b779dc..389f1862bc43 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -3103,7 +3103,7 @@ static int skl_init_algo_data(struct device *dev, struct soc_bytes_ext *be, ac->size = dfw_ac->max; if (ac->max) { - ac->params = (char *) devm_kzalloc(dev, ac->max, GFP_KERNEL); + ac->params = devm_kzalloc(dev, ac->max, GFP_KERNEL); if (!ac->params) return -ENOMEM; From d8747d30aa7f9e7dc6123709d7ca1d8429d648b0 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Sun, 16 Dec 2018 16:49:03 -0600 Subject: [PATCH 19/67] ASoC: Intel: Skylake: simplify boolean tests Detected with Coccinelle skl-messages.c:419:5-32: WARNING: Comparison to bool skl-pcm.c:1426:6-33: WARNING: Comparison to bool Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl-messages.c | 2 +- sound/soc/intel/skylake/skl-pcm.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c index b0e6fb93eaf8..28c4806b196a 100644 --- a/sound/soc/intel/skylake/skl-messages.c +++ b/sound/soc/intel/skylake/skl-messages.c @@ -416,7 +416,7 @@ int skl_resume_dsp(struct skl *skl) snd_hdac_ext_bus_ppcap_int_enable(bus, true); /* check if DSP 1st boot is done */ - if (skl->skl_sst->is_first_boot == true) + if (skl->skl_sst->is_first_boot) return 0; /* diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 557f80c0bfe5..8e589d698c58 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -1423,7 +1423,7 @@ static int skl_platform_soc_probe(struct snd_soc_component *component) if (!ops) return -EIO; - if (skl->skl_sst->is_first_boot == false) { + if (!skl->skl_sst->is_first_boot) { dev_err(component->dev, "DSP reports first boot done!!!\n"); return -EIO; } From 6c5414589721d696fe300dc0b8720e0368e3907a Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Sun, 16 Dec 2018 16:49:04 -0600 Subject: [PATCH 20/67] ASoC: Intel: Haswell: remove unneeded semicolon Detected with Coccinelle Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/haswell/sst-haswell-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/haswell/sst-haswell-pcm.c b/sound/soc/intel/haswell/sst-haswell-pcm.c index fe2c826e710c..fb9b8608eb3b 100644 --- a/sound/soc/intel/haswell/sst-haswell-pcm.c +++ b/sound/soc/intel/haswell/sst-haswell-pcm.c @@ -544,7 +544,7 @@ static int hsw_pcm_hw_params(struct snd_pcm_substream *substream, dev_err(rtd->dev, "error: invalid DAI ID %d\n", rtd->cpu_dai->id); return -EINVAL; - }; + } ret = sst_hsw_stream_format(hsw, pcm_data->stream, path_id, stream_type, SST_HSW_STREAM_FORMAT_PCM_FORMAT); From bf88b3c3c277c8138d688a0fc3199b57fecfaf56 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Sun, 16 Dec 2018 16:49:05 -0600 Subject: [PATCH 21/67] ASoC: Intel: Haswell: assign booleans to true/false Detected with Coccinelle Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/haswell/sst-haswell-ipc.c | 2 +- sound/soc/intel/haswell/sst-haswell-pcm.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/haswell/sst-haswell-ipc.c b/sound/soc/intel/haswell/sst-haswell-ipc.c index d33bdaf92c57..31fcdf12c67d 100644 --- a/sound/soc/intel/haswell/sst-haswell-ipc.c +++ b/sound/soc/intel/haswell/sst-haswell-ipc.c @@ -1216,7 +1216,7 @@ int sst_hsw_stream_commit(struct sst_hsw *hsw, struct sst_hsw_stream *stream) return ret; } - stream->commited = 1; + stream->commited = true; trace_hsw_stream_alloc_reply(stream); return 0; diff --git a/sound/soc/intel/haswell/sst-haswell-pcm.c b/sound/soc/intel/haswell/sst-haswell-pcm.c index fb9b8608eb3b..2debcc2ed99a 100644 --- a/sound/soc/intel/haswell/sst-haswell-pcm.c +++ b/sound/soc/intel/haswell/sst-haswell-pcm.c @@ -861,7 +861,7 @@ static int hsw_pcm_close(struct snd_pcm_substream *substream) dev_dbg(rtd->dev, "error: free stream failed %d\n", ret); goto out; } - pcm_data->allocated = 0; + pcm_data->allocated = false; pcm_data->stream = NULL; out: From 060d35be2dfa9202d37f967fd20f133c530505d2 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Sun, 16 Dec 2018 16:49:06 -0600 Subject: [PATCH 22/67] ASoC: Intel: Baytrail: remove unneeded variable Detected with Coccinelle Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/baytrail/sst-baytrail-ipc.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/intel/baytrail/sst-baytrail-ipc.c b/sound/soc/intel/baytrail/sst-baytrail-ipc.c index 260447da32b8..2cd8f9668b50 100644 --- a/sound/soc/intel/baytrail/sst-baytrail-ipc.c +++ b/sound/soc/intel/baytrail/sst-baytrail-ipc.c @@ -278,7 +278,6 @@ static int sst_byt_process_notification(struct sst_byt *byt, struct sst_byt_stream *stream; u64 header; u8 msg_id, stream_id; - int handled = 1; header = sst_dsp_shim_read64_unlocked(sst, SST_IPCD); msg_id = sst_byt_header_msg_id(header); @@ -298,7 +297,7 @@ static int sst_byt_process_notification(struct sst_byt *byt, break; } - return handled; + return 1; } static irqreturn_t sst_byt_irq_thread(int irq, void *context) From e295450dd86d974861e1e9e302d67b0a23457ea8 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Sun, 16 Dec 2018 16:49:07 -0600 Subject: [PATCH 23/67] ASoC: Intel: Baytrail: simplify boolean test Detected with Coccinelle Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/baytrail/sst-baytrail-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/baytrail/sst-baytrail-pcm.c b/sound/soc/intel/baytrail/sst-baytrail-pcm.c index aabb35bf6b96..498fb5346f1a 100644 --- a/sound/soc/intel/baytrail/sst-baytrail-pcm.c +++ b/sound/soc/intel/baytrail/sst-baytrail-pcm.c @@ -188,7 +188,7 @@ static int sst_byt_pcm_trigger(struct snd_pcm_substream *substream, int cmd) sst_byt_stream_start(byt, pcm_data->stream, 0); break; case SNDRV_PCM_TRIGGER_RESUME: - if (pdata->restore_stream == true) + if (pdata->restore_stream) schedule_work(&pcm_data->work); else sst_byt_stream_resume(byt, pcm_data->stream); From 10583cdac237b32c0d3f6027b06c5eec8bf91211 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Sun, 16 Dec 2018 16:49:08 -0600 Subject: [PATCH 24/67] ASoC: Intel: Atom: simplify boolean tests Detected with Coccinelle Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/atom/sst-atom-controls.c | 2 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +- sound/soc/intel/atom/sst/sst_acpi.c | 2 +- sound/soc/intel/atom/sst/sst_drv_interface.c | 2 +- sound/soc/intel/atom/sst/sst_loader.c | 2 +- 5 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c index 3672d36b4b66..d1207ea53523 100644 --- a/sound/soc/intel/atom/sst-atom-controls.c +++ b/sound/soc/intel/atom/sst-atom-controls.c @@ -647,7 +647,7 @@ static int sst_swm_mixer_event(struct snd_soc_dapm_widget *w, set_mixer = false; } - if (set_mixer == false) + if (!set_mixer) return 0; if (SND_SOC_DAPM_EVENT_ON(event) || diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index afc559866095..aefa5ce4cb59 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -190,7 +190,7 @@ int sst_fill_stream_params(void *substream, map = ctx->pdata->pdev_strm_map; map_size = ctx->pdata->strm_map_size; - if (is_compress == true) + if (is_compress) cstream = (struct snd_compr_stream *)substream; else pstream = (struct snd_pcm_substream *)substream; diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index ac542535b9d5..3a95ebbfc45d 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -334,7 +334,7 @@ static int sst_acpi_probe(struct platform_device *pdev) return ret; ret = is_byt_cr(dev, &bytcr); - if (!((ret < 0) || (bytcr == false))) { + if (!(ret < 0 || !bytcr)) { dev_info(dev, "Detected Baytrail-CR platform\n"); /* override resource info */ diff --git a/sound/soc/intel/atom/sst/sst_drv_interface.c b/sound/soc/intel/atom/sst/sst_drv_interface.c index 5455d6e0ab53..a592df06aa58 100644 --- a/sound/soc/intel/atom/sst/sst_drv_interface.c +++ b/sound/soc/intel/atom/sst/sst_drv_interface.c @@ -146,7 +146,7 @@ static int sst_power_control(struct device *dev, bool state) int ret = 0; int usage_count = 0; - if (state == true) { + if (state) { ret = pm_runtime_get_sync(dev); usage_count = GET_USAGE_COUNT(dev); dev_dbg(ctx->dev, "Enable: pm usage count: %d\n", usage_count); diff --git a/sound/soc/intel/atom/sst/sst_loader.c b/sound/soc/intel/atom/sst/sst_loader.c index b8c456753f01..321c783cf833 100644 --- a/sound/soc/intel/atom/sst/sst_loader.c +++ b/sound/soc/intel/atom/sst/sst_loader.c @@ -269,7 +269,7 @@ static void sst_do_memcpy(struct list_head *memcpy_list) struct sst_memcpy_list *listnode; list_for_each_entry(listnode, memcpy_list, memcpylist) { - if (listnode->is_io == true) + if (listnode->is_io) memcpy32_toio((void __iomem *)listnode->dstn, listnode->src, listnode->size); else From 4e88068ed0888549acd1cbb2f6e271b007051203 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Sun, 16 Dec 2018 16:49:10 -0600 Subject: [PATCH 25/67] ASoC: Intel: boards: use snd_mask_set_format in all machine drivers Fix Sparse warnings with two machine drivers which weren't updated Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/glk_rt5682_max98357a.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 4 ++-- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c index c74c4f17316f..0739e3a75083 100644 --- a/sound/soc/intel/boards/glk_rt5682_max98357a.c +++ b/sound/soc/intel/boards/glk_rt5682_max98357a.c @@ -164,7 +164,7 @@ static int geminilake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, /* set SSP to 24 bit */ snd_mask_none(fmt); - snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); return 0; } diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index 723a4935ed76..6dd5c69671b3 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -221,7 +221,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, rate->min = rate->max = 48000; channels->min = channels->max = 2; snd_mask_none(fmt); - snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); } /* @@ -229,7 +229,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, * thus changing the mask here */ if (!strcmp(be_dai_link->name, "SSP0-Codec")) - snd_mask_set(fmt, SNDRV_PCM_FORMAT_S16_LE); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); return 0; } From a0c426fe143328760c9fd565cd203a37a7b4fde8 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 20 Dec 2018 10:45:42 +0900 Subject: [PATCH 26/67] ASoC: simple-card-utils: check "reg" property on asoc_simple_card_get_dai_id() We will get DAI ID from "reg" property if it has on DT, otherwise get it by counting port/endpoint. But in below case, we need to get DAI ID = 0 via port reg = <0>, but current implementation returns ID = 1, because it can't judge ID = 0 was from "non reg" or "reg = <0>". Thus, it will count port/endpoint number as "non reg" case. of_graph_parse_endpoint() implementation itself is not a problem, but because asoc_simple_card_get_dai_id() need to count port/endpoint number when "non reg" case, it need to know ID = 0 was from "non reg" or "reg = <0>". This patch fix this issue. port { reg = <0>; xxxx: endpoint@0 { }; => xxxx: endpoint@1 { }; }; Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/simple-card-utils.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index b807a47515eb..336895f7fd1e 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -283,12 +283,20 @@ static int asoc_simple_card_get_dai_id(struct device_node *ep) /* use endpoint/port reg if exist */ ret = of_graph_parse_endpoint(ep, &info); if (ret == 0) { - if (info.id) + /* + * Because it will count port/endpoint if it doesn't have "reg". + * But, we can't judge whether it has "no reg", or "reg = <0>" + * only of_graph_parse_endpoint(). + * We need to check "reg" property + */ + if (of_get_property(ep, "reg", NULL)) return info.id; - if (info.port) + + node = of_get_parent(ep); + of_node_put(node); + if (of_get_property(node, "reg", NULL)) return info.port; } - node = of_graph_get_port_parent(ep); /* From 40dfae169ad047535d566a4791daae3b08f71c0c Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 20 Dec 2018 10:45:48 +0900 Subject: [PATCH 27/67] ASoC: audio-graph-card: add asoc_graph_card_get_conversion() audio-graph-card is now supporting normal sound and DPCM sound. For DPCM sound, original sound card (= audio-graph-scu) had been supported 1 CPU : 1 Codec connection which uses hw_params_fixup() for convert-rate/channel. But, merged audio-graph-card is completely forgeting about it. To re-support 1 CPU : 1 Codec DPCM for hw_params_fixup(), it need to judge whether it is DPCM by checking convert-rate/channel. For this purpose, this patch adds asoc_graph_card_get_conversion() as preparation Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/audio-graph-card.c | 22 +++++++++++++++++----- 1 file changed, 17 insertions(+), 5 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 0d6144560a1e..c3e80bc27e80 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -169,6 +169,22 @@ static int asoc_graph_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } +static void asoc_graph_card_get_conversion(struct device *dev, + struct device_node *ep, + struct asoc_simple_card_data *adata) +{ + struct device_node *top = dev->of_node; + struct device_node *port = of_get_parent(ep); + struct device_node *ports = of_get_parent(port); + struct device_node *node = of_graph_get_port_parent(ep); + + asoc_simple_card_parse_convert(dev, top, NULL, adata); + asoc_simple_card_parse_convert(dev, node, PREFIX, adata); + asoc_simple_card_parse_convert(dev, ports, NULL, adata); + asoc_simple_card_parse_convert(dev, port, NULL, adata); + asoc_simple_card_parse_convert(dev, ep, NULL, adata); +} + static int asoc_graph_card_dai_link_of_dpcm(struct device_node *top, struct device_node *cpu_ep, struct device_node *codec_ep, @@ -194,11 +210,7 @@ static int asoc_graph_card_dai_link_of_dpcm(struct device_node *top, of_property_read_u32(port, "mclk-fs", &dai_props->mclk_fs); of_property_read_u32(ep, "mclk-fs", &dai_props->mclk_fs); - asoc_simple_card_parse_convert(dev, top, NULL, &dai_props->adata); - asoc_simple_card_parse_convert(dev, node, PREFIX, &dai_props->adata); - asoc_simple_card_parse_convert(dev, ports, NULL, &dai_props->adata); - asoc_simple_card_parse_convert(dev, port, NULL, &dai_props->adata); - asoc_simple_card_parse_convert(dev, ep, NULL, &dai_props->adata); + asoc_graph_card_get_conversion(dev, ep, &dai_props->adata); of_node_put(ports); of_node_put(port); From e4f4fdfc57d9c846862ea6109e356b3a4542df5b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 18 Dec 2018 11:50:27 +0900 Subject: [PATCH 28/67] ASoC: audio-graph-scu-card: remove audio-graph-scu-card on Doc It is already merged into audio-graph-card. audio-graph-scu-card is no longer needed. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- .../bindings/sound/audio-graph-scu-card.txt | 123 ------------------ 1 file changed, 123 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt diff --git a/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt b/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt deleted file mode 100644 index 62d42768a00b..000000000000 --- a/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt +++ /dev/null @@ -1,123 +0,0 @@ -Audio-Graph-SCU-Card: - -Audio-Graph-SCU-Card is "Audio-Graph-Card" + "ALSA DPCM". - -It is based on common bindings for device graphs. -see ${LINUX}/Documentation/devicetree/bindings/graph.txt - -Basically, Audio-Graph-SCU-Card property is same as -Simple-Card / Simple-SCU-Card / Audio-Graph-Card. -see ${LINUX}/Documentation/devicetree/bindings/sound/simple-card.txt - ${LINUX}/Documentation/devicetree/bindings/sound/simple-scu-card.txt - ${LINUX}/Documentation/devicetree/bindings/sound/audio-graph-card.txt - -Below are same as Simple-Card / Audio-Graph-Card. - -- label -- dai-format -- frame-master -- bitclock-master -- bitclock-inversion -- frame-inversion -- dai-tdm-slot-num -- dai-tdm-slot-width -- clocks / system-clock-frequency - -Below are same as Simple-SCU-Card. - -- convert-rate -- convert-channels -- prefix -- routing - -Required properties: - -- compatible : "audio-graph-scu-card"; -- dais : list of CPU DAI port{s} - -Example 1. Sampling Rate Conversion - - sound_card { - compatible = "audio-graph-scu-card"; - - label = "sound-card"; - prefix = "codec"; - routing = "codec Playback", "DAI0 Playback", - "DAI0 Capture", "codec Capture"; - convert-rate = <48000>; - - dais = <&cpu_port>; - }; - - audio-codec { - ... - - port { - codec_endpoint: endpoint { - remote-endpoint = <&cpu_endpoint>; - }; - }; - }; - - dai-controller { - ... - cpu_port: port { - cpu_endpoint: endpoint { - remote-endpoint = <&codec_endpoint>; - - dai-format = "left_j"; - ... - }; - }; - }; - -Example 2. 2 CPU 1 Codec (Mixing) - - sound_card { - compatible = "audio-graph-scu-card"; - - label = "sound-card"; - routing = "codec Playback", "DAI0 Playback", - "codec Playback", "DAI1 Playback", - "DAI0 Capture", "codec Capture"; - - dais = <&cpu_port0 - &cpu_port1>; - }; - - audio-codec { - ... - - audio-graph-card,prefix = "codec"; - audio-graph-card,convert-rate = <48000>; - port { - codec_endpoint0: endpoint { - remote-endpoint = <&cpu_endpoint0>; - }; - codec_endpoint1: endpoint { - remote-endpoint = <&cpu_endpoint1>; - }; - }; - }; - - dai-controller { - ... - ports { - cpu_port0: port { - cpu_endpoint0: endpoint { - remote-endpoint = <&codec_endpoint0>; - - dai-format = "left_j"; - ... - }; - }; - cpu_port1: port { - cpu_endpoint1: endpoint { - remote-endpoint = <&codec_endpoint1>; - - dai-format = "left_j"; - ... - }; - }; - }; - }; From 61c263ac27a307cdf7f46aaee4810619103effad Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 18 Dec 2018 11:50:32 +0900 Subject: [PATCH 29/67] ASoC: audio-graph-scu-card: remove audio-graph-scu-card It is already merged into audio-graph-card. audio-graph-scu-card is no longer needed. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/Kconfig | 9 - sound/soc/generic/Makefile | 2 - sound/soc/generic/audio-graph-scu-card.c | 501 ----------------------- 3 files changed, 512 deletions(-) delete mode 100644 sound/soc/generic/audio-graph-scu-card.c diff --git a/sound/soc/generic/Kconfig b/sound/soc/generic/Kconfig index 92c2cf06f40a..59190f42fc08 100644 --- a/sound/soc/generic/Kconfig +++ b/sound/soc/generic/Kconfig @@ -24,12 +24,3 @@ config SND_AUDIO_GRAPH_CARD This option enables generic simple sound card support with OF-graph DT bindings. It also support DPCM of multi CPU single Codec ststem. - -config SND_AUDIO_GRAPH_SCU_CARD - tristate "ASoC Audio Graph SCU sound card support" - depends on OF - select SND_SIMPLE_CARD_UTILS - help - This option enables generic simple SCU sound card support - with OF-graph DT bindings. - It supports DPCM of multi CPU single Codec ststem. diff --git a/sound/soc/generic/Makefile b/sound/soc/generic/Makefile index 9dec293a4c4d..9fbfdd524b24 100644 --- a/sound/soc/generic/Makefile +++ b/sound/soc/generic/Makefile @@ -3,10 +3,8 @@ snd-soc-simple-card-utils-objs := simple-card-utils.o snd-soc-simple-card-objs := simple-card.o snd-soc-simple-scu-card-objs := simple-scu-card.o snd-soc-audio-graph-card-objs := audio-graph-card.o -snd-soc-audio-graph-scu-card-objs := audio-graph-scu-card.o obj-$(CONFIG_SND_SIMPLE_CARD_UTILS) += snd-soc-simple-card-utils.o obj-$(CONFIG_SND_SIMPLE_CARD) += snd-soc-simple-card.o obj-$(CONFIG_SND_SIMPLE_SCU_CARD) += snd-soc-simple-scu-card.o obj-$(CONFIG_SND_AUDIO_GRAPH_CARD) += snd-soc-audio-graph-card.o -obj-$(CONFIG_SND_AUDIO_GRAPH_SCU_CARD) += snd-soc-audio-graph-scu-card.o diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c deleted file mode 100644 index e1b192ea147b..000000000000 --- a/sound/soc/generic/audio-graph-scu-card.c +++ /dev/null @@ -1,501 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0 -// -// ASoC audio graph SCU sound card support -// -// Copyright (C) 2017 Renesas Solutions Corp. -// Kuninori Morimoto -// -// based on -// ${LINUX}/sound/soc/generic/simple-scu-card.c -// ${LINUX}/sound/soc/generic/audio-graph-card.c - -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include - -struct graph_card_data { - struct snd_soc_card snd_card; - struct graph_dai_props { - struct asoc_simple_dai *cpu_dai; - struct asoc_simple_dai *codec_dai; - struct snd_soc_dai_link_component codecs; - struct snd_soc_dai_link_component platform; - struct asoc_simple_card_data adata; - struct snd_soc_codec_conf *codec_conf; - } *dai_props; - struct snd_soc_dai_link *dai_link; - struct asoc_simple_dai *dais; - struct asoc_simple_card_data adata; - struct snd_soc_codec_conf *codec_conf; -}; - -#define graph_priv_to_card(priv) (&(priv)->snd_card) -#define graph_priv_to_props(priv, i) ((priv)->dai_props + (i)) -#define graph_priv_to_dev(priv) (graph_priv_to_card(priv)->dev) -#define graph_priv_to_link(priv, i) (graph_priv_to_card(priv)->dai_link + (i)) - -#define PREFIX "audio-graph-card," - -static int asoc_graph_card_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); - int ret = 0; - - ret = asoc_simple_card_clk_enable(dai_props->cpu_dai); - if (ret) - return ret; - - ret = asoc_simple_card_clk_enable(dai_props->codec_dai); - if (ret) - asoc_simple_card_clk_disable(dai_props->cpu_dai); - - return ret; -} - -static void asoc_graph_card_shutdown(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); - - asoc_simple_card_clk_disable(dai_props->cpu_dai); - - asoc_simple_card_clk_disable(dai_props->codec_dai); -} - -static const struct snd_soc_ops asoc_graph_card_ops = { - .startup = asoc_graph_card_startup, - .shutdown = asoc_graph_card_shutdown, -}; - -static int asoc_graph_card_dai_init(struct snd_soc_pcm_runtime *rtd) -{ - struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); - int ret = 0; - - ret = asoc_simple_card_init_dai(rtd->codec_dai, - dai_props->codec_dai); - if (ret < 0) - return ret; - - ret = asoc_simple_card_init_dai(rtd->cpu_dai, - dai_props->cpu_dai); - if (ret < 0) - return ret; - - return 0; -} - -static int asoc_graph_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) -{ - struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); - - asoc_simple_card_convert_fixup(&dai_props->adata, params); - - /* overwrite by top level adata if exist */ - asoc_simple_card_convert_fixup(&priv->adata, params); - - return 0; -} - -static int asoc_graph_card_dai_link_of(struct device_node *cpu_ep, - struct device_node *codec_ep, - struct graph_card_data *priv, - int *dai_idx, int link_idx, - int *conf_idx, int is_fe) -{ - struct device *dev = graph_priv_to_dev(priv); - struct snd_soc_dai_link *dai_link = graph_priv_to_link(priv, link_idx); - struct graph_dai_props *dai_props = graph_priv_to_props(priv, link_idx); - struct snd_soc_card *card = graph_priv_to_card(priv); - struct device_node *ep = is_fe ? cpu_ep : codec_ep; - struct device_node *node = of_graph_get_port_parent(ep); - struct asoc_simple_dai *dai; - int ret; - - if (is_fe) { - struct snd_soc_dai_link_component *codecs; - - /* BE is dummy */ - codecs = dai_link->codecs; - codecs->of_node = NULL; - codecs->dai_name = "snd-soc-dummy-dai"; - codecs->name = "snd-soc-dummy"; - - /* FE settings */ - dai_link->dynamic = 1; - dai_link->dpcm_merged_format = 1; - - dai = - dai_props->cpu_dai = &priv->dais[(*dai_idx)++]; - - ret = asoc_simple_card_parse_graph_cpu(ep, dai_link); - if (ret) - return ret; - - ret = asoc_simple_card_parse_clk_cpu(dev, ep, dai_link, dai); - if (ret < 0) - return ret; - - ret = asoc_simple_card_set_dailink_name(dev, dai_link, - "fe.%s", - dai_link->cpu_dai_name); - if (ret < 0) - return ret; - - /* card->num_links includes Codec */ - asoc_simple_card_canonicalize_cpu(dai_link, - of_graph_get_endpoint_count(dai_link->cpu_of_node) == 1); - } else { - struct snd_soc_codec_conf *cconf; - - /* FE is dummy */ - dai_link->cpu_of_node = NULL; - dai_link->cpu_dai_name = "snd-soc-dummy-dai"; - dai_link->cpu_name = "snd-soc-dummy"; - - /* BE settings */ - dai_link->no_pcm = 1; - dai_link->be_hw_params_fixup = asoc_graph_card_be_hw_params_fixup; - - dai = - dai_props->codec_dai = &priv->dais[(*dai_idx)++]; - - cconf = - dai_props->codec_conf = &priv->codec_conf[(*conf_idx)++]; - - ret = asoc_simple_card_parse_graph_codec(ep, dai_link); - if (ret < 0) - return ret; - - ret = asoc_simple_card_parse_clk_codec(dev, ep, dai_link, dai); - if (ret < 0) - return ret; - - ret = asoc_simple_card_set_dailink_name(dev, dai_link, - "be.%s", - dai_link->codecs->dai_name); - if (ret < 0) - return ret; - - /* check "prefix" from top node */ - snd_soc_of_parse_audio_prefix(card, cconf, - dai_link->codecs->of_node, - "prefix"); - /* check "prefix" from each node if top doesn't have */ - if (!cconf->of_node) - snd_soc_of_parse_node_prefix(node, cconf, - dai_link->codecs->of_node, - PREFIX "prefix"); - } - - asoc_simple_card_parse_convert(dev, node, PREFIX, &dai_props->adata); - - ret = asoc_simple_card_of_parse_tdm(ep, dai); - if (ret) - return ret; - - ret = asoc_simple_card_canonicalize_dailink(dai_link); - if (ret < 0) - return ret; - - ret = asoc_simple_card_parse_daifmt(dev, cpu_ep, codec_ep, - NULL, &dai_link->dai_fmt); - if (ret < 0) - return ret; - - dai_link->dpcm_playback = 1; - dai_link->dpcm_capture = 1; - dai_link->ops = &asoc_graph_card_ops; - dai_link->init = asoc_graph_card_dai_init; - - return 0; -} - -static int asoc_graph_card_parse_of(struct graph_card_data *priv) -{ - struct of_phandle_iterator it; - struct device *dev = graph_priv_to_dev(priv); - struct snd_soc_card *card = graph_priv_to_card(priv); - struct device_node *node = dev->of_node; - struct device_node *cpu_port; - struct device_node *cpu_ep; - struct device_node *codec_ep; - struct device_node *codec_port; - struct device_node *codec_port_old; - int dai_idx, link_idx, conf_idx, ret; - int rc, codec; - - if (!node) - return -EINVAL; - - /* - * we need to consider "widgets", "mclk-fs" around here - * see simple-card - */ - - ret = asoc_simple_card_of_parse_routing(card, NULL); - if (ret < 0) - return ret; - - asoc_simple_card_parse_convert(dev, node, NULL, &priv->adata); - - /* - * it supports multi CPU, single CODEC only here - * see asoc_graph_get_dais_count - */ - - link_idx = 0; - dai_idx = 0; - conf_idx = 0; - codec_port_old = NULL; - for (codec = 0; codec < 2; codec++) { - /* - * To listup valid sounds continuously, - * detect all CPU-dummy first, and - * detect all dummy-Codec second - */ - of_for_each_phandle(&it, rc, node, "dais", NULL, 0) { - cpu_port = it.node; - cpu_ep = of_get_next_child(cpu_port, NULL); - codec_ep = of_graph_get_remote_endpoint(cpu_ep); - codec_port = of_graph_get_port_parent(codec_ep); - - of_node_put(cpu_ep); - of_node_put(codec_ep); - of_node_put(cpu_port); - of_node_put(codec_port); - it.node = NULL; - - if (codec) { - if (codec_port_old == codec_port) - continue; - - codec_port_old = codec_port; - } - - ret = asoc_graph_card_dai_link_of(cpu_ep, codec_ep, - priv, &dai_idx, - link_idx++, &conf_idx, - !codec); - if (ret < 0) - goto parse_of_err; - } - } - - ret = asoc_simple_card_parse_card_name(card, NULL); - if (ret) - goto parse_of_err; - - if ((card->num_links != link_idx) || - (card->num_configs != conf_idx)) { - dev_err(dev, "dai_link or codec_config wrong (%d/%d, %d/%d)\n", - card->num_links, link_idx, card->num_configs, conf_idx); - ret = -EINVAL; - goto parse_of_err; - } - - ret = 0; - -parse_of_err: - return ret; -} - -static void asoc_graph_get_dais_count(struct device *dev, - int *link_num, - int *dais_num, - int *ccnf_num) -{ - struct of_phandle_iterator it; - struct device_node *node = dev->of_node; - struct device_node *cpu_port; - struct device_node *cpu_ep; - struct device_node *codec_ep; - struct device_node *codec_port; - struct device_node *codec_port_old; - struct device_node *codec_port_old2; - int rc; - - /* - * link_num : number of links. - * CPU-Codec / CPU-dummy / dummy-Codec - * dais_num : number of DAIs - * ccnf_num : number of codec_conf - * same number for dummy-Codec - * - * ex1) - * CPU0 --- Codec0 link : 5 - * CPU1 --- Codec1 dais : 7 - * CPU2 -/ ccnf : 1 - * CPU3 --- Codec2 - * - * => 5 links = 2xCPU-Codec + 2xCPU-dummy + 1xdummy-Codec - * => 7 DAIs = 4xCPU + 3xCodec - * => 1 ccnf = 1xdummy-Codec - * - * ex2) - * CPU0 --- Codec0 link : 5 - * CPU1 --- Codec1 dais : 6 - * CPU2 -/ ccnf : 1 - * CPU3 -/ - * - * => 5 links = 1xCPU-Codec + 3xCPU-dummy + 1xdummy-Codec - * => 6 DAIs = 4xCPU + 2xCodec - * => 1 ccnf = 1xdummy-Codec - * - * ex3) - * CPU0 --- Codec0 link : 6 - * CPU1 -/ dais : 6 - * CPU2 --- Codec1 ccnf : 2 - * CPU3 -/ - * - * => 6 links = 0xCPU-Codec + 4xCPU-dummy + 2xdummy-Codec - * => 6 DAIs = 4xCPU + 2xCodec - * => 2 ccnf = 2xdummy-Codec - */ - codec_port_old = NULL; - codec_port_old2 = NULL; - of_for_each_phandle(&it, rc, node, "dais", NULL, 0) { - cpu_port = it.node; - cpu_ep = of_get_next_child(cpu_port, NULL); - codec_ep = of_graph_get_remote_endpoint(cpu_ep); - codec_port = of_graph_get_port_parent(codec_ep); - - of_node_put(cpu_ep); - of_node_put(codec_ep); - of_node_put(codec_port); - - (*link_num)++; - (*dais_num)++; - - if (codec_port_old == codec_port) { - if (codec_port_old2 != codec_port_old) { - (*link_num)++; - (*ccnf_num)++; - } - - codec_port_old2 = codec_port_old; - continue; - } - - (*dais_num)++; - codec_port_old = codec_port; - } -} - -static int asoc_graph_card_probe(struct platform_device *pdev) -{ - struct graph_card_data *priv; - struct snd_soc_dai_link *dai_link; - struct graph_dai_props *dai_props; - struct asoc_simple_dai *dais; - struct device *dev = &pdev->dev; - struct snd_soc_card *card; - struct snd_soc_codec_conf *cconf; - int lnum = 0, dnum = 0, cnum = 0; - int ret, i; - - /* Allocate the private data and the DAI link array */ - priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); - if (!priv) - return -ENOMEM; - - asoc_graph_get_dais_count(dev, &lnum, &dnum, &cnum); - if (!lnum || !dnum) - return -EINVAL; - - dai_props = devm_kcalloc(dev, lnum, sizeof(*dai_props), GFP_KERNEL); - dai_link = devm_kcalloc(dev, lnum, sizeof(*dai_link), GFP_KERNEL); - dais = devm_kcalloc(dev, dnum, sizeof(*dais), GFP_KERNEL); - cconf = devm_kcalloc(dev, cnum, sizeof(*cconf), GFP_KERNEL); - if (!dai_props || !dai_link || !dais) - return -ENOMEM; - - /* - * Use snd_soc_dai_link_component instead of legacy style - * It is codec only. but cpu/platform will be supported in the future. - * see - * soc-core.c :: snd_soc_init_multicodec() - */ - for (i = 0; i < lnum; i++) { - dai_link[i].codecs = &dai_props[i].codecs; - dai_link[i].num_codecs = 1; - dai_link[i].platform = &dai_props[i].platform; - } - - priv->dai_props = dai_props; - priv->dai_link = dai_link; - priv->dais = dais; - priv->codec_conf = cconf; - - /* Init snd_soc_card */ - card = graph_priv_to_card(priv); - card->owner = THIS_MODULE; - card->dev = dev; - card->dai_link = priv->dai_link; - card->num_links = lnum; - card->codec_conf = cconf; - card->num_configs = cnum; - - ret = asoc_graph_card_parse_of(priv); - if (ret < 0) { - if (ret != -EPROBE_DEFER) - dev_err(dev, "parse error %d\n", ret); - goto err; - } - - snd_soc_card_set_drvdata(card, priv); - - ret = devm_snd_soc_register_card(dev, card); - if (ret < 0) - goto err; - - return 0; -err: - asoc_simple_card_clean_reference(card); - - return ret; -} - -static int asoc_graph_card_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - return asoc_simple_card_clean_reference(card); -} - -static const struct of_device_id asoc_graph_of_match[] = { - { .compatible = "audio-graph-scu-card", }, - {}, -}; -MODULE_DEVICE_TABLE(of, asoc_graph_of_match); - -static struct platform_driver asoc_graph_card = { - .driver = { - .name = "asoc-audio-graph-scu-card", - .pm = &snd_soc_pm_ops, - .of_match_table = asoc_graph_of_match, - }, - .probe = asoc_graph_card_probe, - .remove = asoc_graph_card_remove, -}; -module_platform_driver(asoc_graph_card); - -MODULE_ALIAS("platform:asoc-audio-graph-scu-card"); -MODULE_LICENSE("GPL v2"); -MODULE_DESCRIPTION("ASoC Audio Graph SCU Sound Card"); -MODULE_AUTHOR("Kuninori Morimoto "); From bb93487b85012b2232149888d260f935e4da680d Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 18 Dec 2018 11:50:37 +0900 Subject: [PATCH 30/67] ASoC: simple-scu-card: remove simple-scu-card on Doc It is already merged into simple-card. simple-scu-card is no longer needed. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- .../bindings/sound/simple-scu-card.txt | 94 ------------------- 1 file changed, 94 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/simple-scu-card.txt diff --git a/Documentation/devicetree/bindings/sound/simple-scu-card.txt b/Documentation/devicetree/bindings/sound/simple-scu-card.txt deleted file mode 100644 index 3a2f71616cda..000000000000 --- a/Documentation/devicetree/bindings/sound/simple-scu-card.txt +++ /dev/null @@ -1,94 +0,0 @@ -ASoC Simple SCU Sound Card - -Simple SCU Sound Card is "Simple Sound Card" + "ALSA DPCM". -For example, you can use this driver if you want to exchange sampling rate convert, -Mixing, etc... - -Required properties: - -- compatible : "simple-scu-audio-card" - "renesas,rsrc-card" -Optional properties: - -- simple-audio-card,name : see simple-audio-card.txt -- simple-audio-card,cpu : see simple-audio-card.txt -- simple-audio-card,codec : see simple-audio-card.txt - -Optional subnode properties: - -- simple-audio-card,format : see simple-audio-card.txt -- simple-audio-card,frame-master : see simple-audio-card.txt -- simple-audio-card,bitclock-master : see simple-audio-card.txt -- simple-audio-card,bitclock-inversion : see simple-audio-card.txt -- simple-audio-card,frame-inversion : see simple-audio-card.txt -- simple-audio-card,convert-rate : platform specified sampling rate convert -- simple-audio-card,convert-channels : platform specified converted channel size (2 - 8 ch) -- simple-audio-card,prefix : see routing -- simple-audio-card,widgets : Please refer to widgets.txt. -- simple-audio-card,routing : A list of the connections between audio components. - Each entry is a pair of strings, the first being the connection's sink, - the second being the connection's source. Valid names for sources. - use audio-prefix if some components is using same sink/sources naming. - it can be used if compatible was "renesas,rsrc-card"; - -Required CPU/CODEC subnodes properties: - -- sound-dai : see simple-audio-card.txt - -Optional CPU/CODEC subnodes properties: - -- clocks / system-clock-frequency : see simple-audio-card.txt - -Example 1. Sampling Rate Conversion - -sound { - compatible = "simple-scu-audio-card"; - - simple-audio-card,name = "rsnd-ak4643"; - simple-audio-card,format = "left_j"; - simple-audio-card,bitclock-master = <&sndcodec>; - simple-audio-card,frame-master = <&sndcodec>; - - simple-audio-card,convert-rate = <48000>; - - simple-audio-card,prefix = "ak4642"; - simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", - "DAI0 Capture", "ak4642 Capture"; - - sndcpu: simple-audio-card,cpu { - sound-dai = <&rcar_sound>; - }; - - sndcodec: simple-audio-card,codec { - sound-dai = <&ak4643>; - system-clock-frequency = <11289600>; - }; -}; - -Example 2. 2 CPU 1 Codec (Mixing) - -sound { - compatible = "simple-scu-audio-card"; - - simple-audio-card,name = "rsnd-ak4643"; - simple-audio-card,format = "left_j"; - simple-audio-card,bitclock-master = <&dpcmcpu>; - simple-audio-card,frame-master = <&dpcmcpu>; - - simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", - "ak4642 Playback", "DAI1 Playback"; - - dpcmcpu: cpu@0 { - sound-dai = <&rcar_sound 0>; - }; - - cpu@1 { - sound-dai = <&rcar_sound 1>; - }; - - codec { - prefix = "ak4642"; - sound-dai = <&ak4643>; - clocks = <&audio_clock>; - }; -}; From c8ed6aca6b824018a39702a563f2f6591de20d64 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 18 Dec 2018 11:50:42 +0900 Subject: [PATCH 31/67] ASoC: simple-scu-card: remove simple-scu-card It is already merged into simple-card. simple-scu-card is no longer needed. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/Kconfig | 8 - sound/soc/generic/Makefile | 2 - sound/soc/generic/simple-scu-card.c | 474 ---------------------------- 3 files changed, 484 deletions(-) delete mode 100644 sound/soc/generic/simple-scu-card.c diff --git a/sound/soc/generic/Kconfig b/sound/soc/generic/Kconfig index 59190f42fc08..83f1243145b0 100644 --- a/sound/soc/generic/Kconfig +++ b/sound/soc/generic/Kconfig @@ -8,14 +8,6 @@ config SND_SIMPLE_CARD This option enables generic simple sound card support It also support DPCM of multi CPU single Codec ststem. -config SND_SIMPLE_SCU_CARD - tristate "ASoC Simple SCU sound card support" - depends on OF - select SND_SIMPLE_CARD_UTILS - help - This option enables generic simple SCU sound card support. - It supports DPCM of multi CPU single Codec system. - config SND_AUDIO_GRAPH_CARD tristate "ASoC Audio Graph sound card support" depends on OF diff --git a/sound/soc/generic/Makefile b/sound/soc/generic/Makefile index 9fbfdd524b24..21c29e5e0671 100644 --- a/sound/soc/generic/Makefile +++ b/sound/soc/generic/Makefile @@ -1,10 +1,8 @@ # SPDX-License-Identifier: GPL-2.0 snd-soc-simple-card-utils-objs := simple-card-utils.o snd-soc-simple-card-objs := simple-card.o -snd-soc-simple-scu-card-objs := simple-scu-card.o snd-soc-audio-graph-card-objs := audio-graph-card.o obj-$(CONFIG_SND_SIMPLE_CARD_UTILS) += snd-soc-simple-card-utils.o obj-$(CONFIG_SND_SIMPLE_CARD) += snd-soc-simple-card.o -obj-$(CONFIG_SND_SIMPLE_SCU_CARD) += snd-soc-simple-scu-card.o obj-$(CONFIG_SND_AUDIO_GRAPH_CARD) += snd-soc-audio-graph-card.o diff --git a/sound/soc/generic/simple-scu-card.c b/sound/soc/generic/simple-scu-card.c deleted file mode 100644 index 9d7299d536a8..000000000000 --- a/sound/soc/generic/simple-scu-card.c +++ /dev/null @@ -1,474 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0 -// -// ASoC simple SCU sound card support -// -// Copyright (C) 2015 Renesas Solutions Corp. -// Kuninori Morimoto -// -// based on ${LINUX}/sound/soc/generic/simple-card.c - -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include - -struct simple_card_data { - struct snd_soc_card snd_card; - struct simple_dai_props { - struct asoc_simple_dai *cpu_dai; - struct asoc_simple_dai *codec_dai; - struct snd_soc_dai_link_component codecs; - struct snd_soc_dai_link_component platform; - struct asoc_simple_card_data adata; - struct snd_soc_codec_conf *codec_conf; - } *dai_props; - struct snd_soc_dai_link *dai_link; - struct asoc_simple_dai *dais; - struct asoc_simple_card_data adata; - struct snd_soc_codec_conf *codec_conf; -}; - -#define simple_priv_to_card(priv) (&(priv)->snd_card) -#define simple_priv_to_props(priv, i) ((priv)->dai_props + (i)) -#define simple_priv_to_dev(priv) (simple_priv_to_card(priv)->dev) -#define simple_priv_to_link(priv, i) (simple_priv_to_card(priv)->dai_link + (i)) - -#define DAI "sound-dai" -#define CELL "#sound-dai-cells" -#define PREFIX "simple-audio-card," - -static int asoc_simple_card_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct simple_dai_props *dai_props = - simple_priv_to_props(priv, rtd->num); - int ret; - - ret = asoc_simple_card_clk_enable(dai_props->cpu_dai); - if (ret) - return ret; - - ret = asoc_simple_card_clk_enable(dai_props->codec_dai); - if (ret) - asoc_simple_card_clk_disable(dai_props->cpu_dai); - - return ret; -} - -static void asoc_simple_card_shutdown(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct simple_dai_props *dai_props = - simple_priv_to_props(priv, rtd->num); - - asoc_simple_card_clk_disable(dai_props->cpu_dai); - - asoc_simple_card_clk_disable(dai_props->codec_dai); -} - -static const struct snd_soc_ops asoc_simple_card_ops = { - .startup = asoc_simple_card_startup, - .shutdown = asoc_simple_card_shutdown, -}; - -static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) -{ - struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); - int ret; - - ret = asoc_simple_card_init_dai(rtd->codec_dai, - dai_props->codec_dai); - if (ret < 0) - return ret; - - ret = asoc_simple_card_init_dai(rtd->cpu_dai, - dai_props->cpu_dai); - if (ret < 0) - return ret; - - return 0; -} - -static int asoc_simple_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) -{ - struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); - - asoc_simple_card_convert_fixup(&dai_props->adata, params); - - /* overwrite by top level adata if exist */ - asoc_simple_card_convert_fixup(&priv->adata, params); - - return 0; -} - -static int asoc_simple_card_dai_link_of(struct device_node *link, - struct device_node *np, - struct device_node *codec, - struct simple_card_data *priv, - int *dai_idx, int link_idx, - int *conf_idx, int is_fe, - bool is_top_level_node) -{ - struct device *dev = simple_priv_to_dev(priv); - struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, link_idx); - struct simple_dai_props *dai_props = simple_priv_to_props(priv, link_idx); - struct snd_soc_card *card = simple_priv_to_card(priv); - struct asoc_simple_dai *dai; - char *prefix = ""; - int ret; - - /* For single DAI link & old style of DT node */ - if (is_top_level_node) - prefix = PREFIX; - - if (is_fe) { - int is_single_links = 0; - struct snd_soc_dai_link_component *codecs; - - /* BE is dummy */ - codecs = dai_link->codecs; - codecs->of_node = NULL; - codecs->dai_name = "snd-soc-dummy-dai"; - codecs->name = "snd-soc-dummy"; - - /* FE settings */ - dai_link->dynamic = 1; - dai_link->dpcm_merged_format = 1; - - dai = - dai_props->cpu_dai = &priv->dais[(*dai_idx)++]; - - ret = asoc_simple_card_parse_cpu(np, dai_link, DAI, CELL, - &is_single_links); - if (ret) - return ret; - - ret = asoc_simple_card_parse_clk_cpu(dev, np, dai_link, dai); - if (ret < 0) - return ret; - - ret = asoc_simple_card_set_dailink_name(dev, dai_link, - "fe.%s", - dai_link->cpu_dai_name); - if (ret < 0) - return ret; - - asoc_simple_card_canonicalize_cpu(dai_link, is_single_links); - } else { - struct snd_soc_codec_conf *cconf; - - /* FE is dummy */ - dai_link->cpu_of_node = NULL; - dai_link->cpu_dai_name = "snd-soc-dummy-dai"; - dai_link->cpu_name = "snd-soc-dummy"; - - /* BE settings */ - dai_link->no_pcm = 1; - dai_link->be_hw_params_fixup = asoc_simple_card_be_hw_params_fixup; - - dai = - dai_props->codec_dai = &priv->dais[(*dai_idx)++]; - - cconf = - dai_props->codec_conf = &priv->codec_conf[(*conf_idx)++]; - - ret = asoc_simple_card_parse_codec(np, dai_link, DAI, CELL); - if (ret < 0) - return ret; - - ret = asoc_simple_card_parse_clk_codec(dev, np, dai_link, dai); - if (ret < 0) - return ret; - - ret = asoc_simple_card_set_dailink_name(dev, dai_link, - "be.%s", - dai_link->codecs->dai_name); - if (ret < 0) - return ret; - - /* check "prefix" from top node */ - snd_soc_of_parse_audio_prefix(card, cconf, - dai_link->codecs->of_node, - PREFIX "prefix"); - /* check "prefix" from each node if top doesn't have */ - if (!cconf->of_node) - snd_soc_of_parse_node_prefix(np, cconf, - dai_link->codecs->of_node, - "prefix"); - } - - asoc_simple_card_parse_convert(dev, link, prefix, &dai_props->adata); - - ret = asoc_simple_card_of_parse_tdm(np, dai); - if (ret) - return ret; - - ret = asoc_simple_card_canonicalize_dailink(dai_link); - if (ret < 0) - return ret; - - ret = asoc_simple_card_parse_daifmt(dev, link, codec, - prefix, &dai_link->dai_fmt); - if (ret < 0) - return ret; - - dai_link->dpcm_playback = 1; - dai_link->dpcm_capture = 1; - dai_link->ops = &asoc_simple_card_ops; - dai_link->init = asoc_simple_card_dai_init; - - return 0; -} - -static int asoc_simple_card_parse_of(struct simple_card_data *priv) - -{ - struct device *dev = simple_priv_to_dev(priv); - struct device_node *top = dev->of_node; - struct device_node *node; - struct device_node *np; - struct device_node *codec; - struct snd_soc_card *card = simple_priv_to_card(priv); - bool is_fe; - int ret, loop; - int dai_idx, link_idx, conf_idx; - - if (!top) - return -EINVAL; - - ret = asoc_simple_card_of_parse_widgets(card, PREFIX); - if (ret < 0) - return ret; - - ret = asoc_simple_card_of_parse_routing(card, PREFIX); - if (ret < 0) - return ret; - - asoc_simple_card_parse_convert(dev, top, PREFIX, &priv->adata); - - loop = 1; - link_idx = 0; - dai_idx = 0; - conf_idx = 0; - node = of_get_child_by_name(top, PREFIX "dai-link"); - if (!node) { - node = dev->of_node; - loop = 0; - } - - do { - codec = of_get_child_by_name(node, - loop ? "codec" : PREFIX "codec"); - if (!codec) - return -ENODEV; - - for_each_child_of_node(node, np) { - is_fe = (np != codec); - - ret = asoc_simple_card_dai_link_of(node, np, codec, priv, - &dai_idx, link_idx++, - &conf_idx, - is_fe, !loop); - if (ret < 0) - return ret; - } - node = of_get_next_child(top, node); - } while (loop && node); - - ret = asoc_simple_card_parse_card_name(card, PREFIX); - if (ret < 0) - return ret; - - return 0; -} - -static void asoc_simple_card_get_dais_count(struct device *dev, - int *link_num, - int *dais_num, - int *ccnf_num) -{ - struct device_node *top = dev->of_node; - struct device_node *node; - int loop; - int num; - - /* - * link_num : number of links. - * CPU-Codec / CPU-dummy / dummy-Codec - * dais_num : number of DAIs - * ccnf_num : number of codec_conf - * same number for "dummy-Codec" - * - * ex1) - * CPU0 --- Codec0 link : 5 - * CPU1 --- Codec1 dais : 7 - * CPU2 -/ ccnf : 1 - * CPU3 --- Codec2 - * - * => 5 links = 2xCPU-Codec + 2xCPU-dummy + 1xdummy-Codec - * => 7 DAIs = 4xCPU + 3xCodec - * => 1 ccnf = 1xdummy-Codec - * - * ex2) - * CPU0 --- Codec0 link : 5 - * CPU1 --- Codec1 dais : 6 - * CPU2 -/ ccnf : 1 - * CPU3 -/ - * - * => 5 links = 1xCPU-Codec + 3xCPU-dummy + 1xdummy-Codec - * => 6 DAIs = 4xCPU + 2xCodec - * => 1 ccnf = 1xdummy-Codec - * - * ex3) - * CPU0 --- Codec0 link : 6 - * CPU1 -/ dais : 6 - * CPU2 --- Codec1 ccnf : 2 - * CPU3 -/ - * - * => 6 links = 0xCPU-Codec + 4xCPU-dummy + 2xdummy-Codec - * => 6 DAIs = 4xCPU + 2xCodec - * => 2 ccnf = 2xdummy-Codec - */ - if (!top) { - (*link_num) = 1; - (*dais_num) = 2; - (*ccnf_num) = 0; - return; - } - - loop = 1; - node = of_get_child_by_name(top, PREFIX "dai-link"); - if (!node) { - node = top; - loop = 0; - } - - do { - num = of_get_child_count(node); - (*dais_num) += num; - if (num > 2) { - (*link_num) += num; - (*ccnf_num)++; - } else { - (*link_num)++; - } - node = of_get_next_child(top, node); - } while (loop && node); -} - -static int asoc_simple_card_probe(struct platform_device *pdev) -{ - struct simple_card_data *priv; - struct snd_soc_dai_link *dai_link; - struct simple_dai_props *dai_props; - struct asoc_simple_dai *dais; - struct snd_soc_card *card; - struct snd_soc_codec_conf *cconf; - struct device *dev = &pdev->dev; - int ret, i; - int lnum = 0, dnum = 0, cnum = 0; - - /* Allocate the private data */ - priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); - if (!priv) - return -ENOMEM; - - asoc_simple_card_get_dais_count(dev, &lnum, &dnum, &cnum); - if (!lnum || !dnum) - return -EINVAL; - - dai_props = devm_kcalloc(dev, lnum, sizeof(*dai_props), GFP_KERNEL); - dai_link = devm_kcalloc(dev, lnum, sizeof(*dai_link), GFP_KERNEL); - dais = devm_kcalloc(dev, dnum, sizeof(*dais), GFP_KERNEL); - cconf = devm_kcalloc(dev, cnum, sizeof(*cconf), GFP_KERNEL); - if (!dai_props || !dai_link || !dais) - return -ENOMEM; - - /* - * Use snd_soc_dai_link_component instead of legacy style - * It is codec only. but cpu/platform will be supported in the future. - * see - * soc-core.c :: snd_soc_init_multicodec() - */ - for (i = 0; i < lnum; i++) { - dai_link[i].codecs = &dai_props[i].codecs; - dai_link[i].num_codecs = 1; - dai_link[i].platform = &dai_props[i].platform; - } - - priv->dai_props = dai_props; - priv->dai_link = dai_link; - priv->dais = dais; - priv->codec_conf = cconf; - - /* Init snd_soc_card */ - card = simple_priv_to_card(priv); - card->owner = THIS_MODULE; - card->dev = dev; - card->dai_link = priv->dai_link; - card->num_links = lnum; - card->codec_conf = cconf; - card->num_configs = cnum; - - ret = asoc_simple_card_parse_of(priv); - if (ret < 0) { - if (ret != -EPROBE_DEFER) - dev_err(dev, "parse error %d\n", ret); - goto err; - } - - snd_soc_card_set_drvdata(card, priv); - - ret = devm_snd_soc_register_card(dev, card); - if (ret < 0) - goto err; - - return 0; -err: - asoc_simple_card_clean_reference(card); - - return ret; -} - -static int asoc_simple_card_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - return asoc_simple_card_clean_reference(card); -} - -static const struct of_device_id asoc_simple_of_match[] = { - { .compatible = "renesas,rsrc-card", }, - { .compatible = "simple-scu-audio-card", }, - {}, -}; -MODULE_DEVICE_TABLE(of, asoc_simple_of_match); - -static struct platform_driver asoc_simple_card = { - .driver = { - .name = "simple-scu-audio-card", - .pm = &snd_soc_pm_ops, - .of_match_table = asoc_simple_of_match, - }, - .probe = asoc_simple_card_probe, - .remove = asoc_simple_card_remove, -}; - -module_platform_driver(asoc_simple_card); - -MODULE_ALIAS("platform:asoc-simple-scu-card"); -MODULE_LICENSE("GPL v2"); -MODULE_DESCRIPTION("ASoC Simple SCU Sound Card"); -MODULE_AUTHOR("Kuninori Morimoto "); From e3e12ec09a18ad779b637f4a006a908cb6045aa7 Mon Sep 17 00:00:00 2001 From: Maruthi Srinivas Bayyavarapu Date: Fri, 21 Dec 2018 14:27:27 +0530 Subject: [PATCH 32/67] dt-bindings: ASoC: xlnx, audio-formatter: Document audio formatter bindings Added documentation for audio formatter IP core DT bindings. Signed-off-by: Maruthi Srinivas Bayyavarapu Signed-off-by: Mark Brown --- .../bindings/sound/xlnx,audio-formatter.txt | 29 +++++++++++++++++++ 1 file changed, 29 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/xlnx,audio-formatter.txt diff --git a/Documentation/devicetree/bindings/sound/xlnx,audio-formatter.txt b/Documentation/devicetree/bindings/sound/xlnx,audio-formatter.txt new file mode 100644 index 000000000000..cbc93c8f4963 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/xlnx,audio-formatter.txt @@ -0,0 +1,29 @@ +Device-Tree bindings for Xilinx PL audio formatter + +The IP core supports DMA, data formatting(AES<->PCM conversion) +of audio samples. + +Required properties: + - compatible: "xlnx,audio-formatter-1.0" + - interrupt-names: Names specified to list of interrupts in same + order mentioned under "interrupts". + List of supported interrupt names are: + "irq_mm2s" : interrupt from MM2S block + "irq_s2mm" : interrupt from S2MM block + - interrupts-parent: Phandle for interrupt controller. + - interrupts: List of Interrupt numbers. + - reg: Base address and size of the IP core instance. + - clock-names: List of input clocks. + Required elements: "s_axi_lite_aclk", "aud_mclk" + - clocks: Input clock specifier. Refer to common clock bindings. + +Example: + audio_ss_0_audio_formatter_0: audio_formatter@80010000 { + compatible = "xlnx,audio-formatter-1.0"; + interrupt-names = "irq_mm2s", "irq_s2mm"; + interrupt-parent = <&gic>; + interrupts = <0 104 4>, <0 105 4>; + reg = <0x0 0x80010000 0x0 0x1000>; + clock-names = "s_axi_lite_aclk", "aud_mclk"; + clocks = <&clk 71>, <&clk_wiz_1 0>; + }; From 6f6c3c36f0917be24587eeba818ab4fdfcb5465a Mon Sep 17 00:00:00 2001 From: Maruthi Srinivas Bayyavarapu Date: Fri, 21 Dec 2018 14:27:28 +0530 Subject: [PATCH 33/67] ASoC: xlnx: add pcm formatter platform driver The audio formatter PL IP supports DMA of two streams - mm2s and s2mm for playback and capture respectively. Apart from DMA, IP also does conversions like PCM to AES and viceversa. This patch adds DMA component driver for the IP. Signed-off-by: Maruthi Srinivas Bayyavarapu Signed-off-by: Mark Brown --- sound/soc/xilinx/xlnx_formatter_pcm.c | 565 ++++++++++++++++++++++++++ 1 file changed, 565 insertions(+) create mode 100644 sound/soc/xilinx/xlnx_formatter_pcm.c diff --git a/sound/soc/xilinx/xlnx_formatter_pcm.c b/sound/soc/xilinx/xlnx_formatter_pcm.c new file mode 100644 index 000000000000..f7235f7664d7 --- /dev/null +++ b/sound/soc/xilinx/xlnx_formatter_pcm.c @@ -0,0 +1,565 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Xilinx ASoC audio formatter support +// +// Copyright (C) 2018 Xilinx, Inc. +// +// Author: Maruthi Srinivas Bayyavarapu + +#include +#include +#include +#include +#include +#include + +#include +#include + +#define DRV_NAME "xlnx_formatter_pcm" + +#define XLNX_S2MM_OFFSET 0 +#define XLNX_MM2S_OFFSET 0x100 + +#define XLNX_AUD_CORE_CONFIG 0x4 +#define XLNX_AUD_CTRL 0x10 +#define XLNX_AUD_STS 0x14 + +#define AUD_CTRL_RESET_MASK BIT(1) +#define AUD_CFG_MM2S_MASK BIT(15) +#define AUD_CFG_S2MM_MASK BIT(31) + +#define XLNX_AUD_FS_MULTIPLIER 0x18 +#define XLNX_AUD_PERIOD_CONFIG 0x1C +#define XLNX_AUD_BUFF_ADDR_LSB 0x20 +#define XLNX_AUD_BUFF_ADDR_MSB 0x24 +#define XLNX_AUD_XFER_COUNT 0x28 +#define XLNX_AUD_CH_STS_START 0x2C +#define XLNX_BYTES_PER_CH 0x44 + +#define AUD_STS_IOC_IRQ_MASK BIT(31) +#define AUD_STS_CH_STS_MASK BIT(29) +#define AUD_CTRL_IOC_IRQ_MASK BIT(13) +#define AUD_CTRL_TOUT_IRQ_MASK BIT(14) +#define AUD_CTRL_DMA_EN_MASK BIT(0) + +#define CFG_MM2S_CH_MASK GENMASK(11, 8) +#define CFG_MM2S_CH_SHIFT 8 +#define CFG_MM2S_XFER_MASK GENMASK(14, 13) +#define CFG_MM2S_XFER_SHIFT 13 +#define CFG_MM2S_PKG_MASK BIT(12) + +#define CFG_S2MM_CH_MASK GENMASK(27, 24) +#define CFG_S2MM_CH_SHIFT 24 +#define CFG_S2MM_XFER_MASK GENMASK(30, 29) +#define CFG_S2MM_XFER_SHIFT 29 +#define CFG_S2MM_PKG_MASK BIT(28) + +#define AUD_CTRL_DATA_WIDTH_SHIFT 16 +#define AUD_CTRL_ACTIVE_CH_SHIFT 19 +#define PERIOD_CFG_PERIODS_SHIFT 16 + +#define PERIODS_MIN 2 +#define PERIODS_MAX 6 +#define PERIOD_BYTES_MIN 192 +#define PERIOD_BYTES_MAX (50 * 1024) + +enum bit_depth { + BIT_DEPTH_8, + BIT_DEPTH_16, + BIT_DEPTH_20, + BIT_DEPTH_24, + BIT_DEPTH_32, +}; + +struct xlnx_pcm_drv_data { + void __iomem *mmio; + bool s2mm_presence; + bool mm2s_presence; + unsigned int s2mm_irq; + unsigned int mm2s_irq; + struct snd_pcm_substream *play_stream; + struct snd_pcm_substream *capture_stream; + struct clk *axi_clk; +}; + +/* + * struct xlnx_pcm_stream_param - stream configuration + * @mmio: base address offset + * @interleaved: audio channels arrangement in buffer + * @xfer_mode: data formatting mode during transfer + * @ch_limit: Maximum channels supported + * @buffer_size: stream ring buffer size + */ +struct xlnx_pcm_stream_param { + void __iomem *mmio; + bool interleaved; + u32 xfer_mode; + u32 ch_limit; + u64 buffer_size; +}; + +static const struct snd_pcm_hardware xlnx_pcm_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_BATCH | SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME, + .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .rate_min = 8000, + .rate_max = 192000, + .buffer_bytes_max = PERIODS_MAX * PERIOD_BYTES_MAX, + .period_bytes_min = PERIOD_BYTES_MIN, + .period_bytes_max = PERIOD_BYTES_MAX, + .periods_min = PERIODS_MIN, + .periods_max = PERIODS_MAX, +}; + +static int xlnx_formatter_pcm_reset(void __iomem *mmio_base) +{ + u32 val, retries = 0; + + val = readl(mmio_base + XLNX_AUD_CTRL); + val |= AUD_CTRL_RESET_MASK; + writel(val, mmio_base + XLNX_AUD_CTRL); + + val = readl(mmio_base + XLNX_AUD_CTRL); + /* Poll for maximum timeout of approximately 100ms (1 * 100)*/ + while ((val & AUD_CTRL_RESET_MASK) && (retries < 100)) { + mdelay(1); + retries++; + val = readl(mmio_base + XLNX_AUD_CTRL); + } + if (val & AUD_CTRL_RESET_MASK) + return -ENODEV; + + return 0; +} + +static void xlnx_formatter_disable_irqs(void __iomem *mmio_base, int stream) +{ + u32 val; + + val = readl(mmio_base + XLNX_AUD_CTRL); + val &= ~AUD_CTRL_IOC_IRQ_MASK; + if (stream == SNDRV_PCM_STREAM_CAPTURE) + val &= ~AUD_CTRL_TOUT_IRQ_MASK; + + writel(val, mmio_base + XLNX_AUD_CTRL); +} + +static irqreturn_t xlnx_mm2s_irq_handler(int irq, void *arg) +{ + u32 val; + void __iomem *reg; + struct device *dev = arg; + struct xlnx_pcm_drv_data *adata = dev_get_drvdata(dev); + + reg = adata->mmio + XLNX_MM2S_OFFSET + XLNX_AUD_STS; + val = readl(reg); + if (val & AUD_STS_IOC_IRQ_MASK) { + writel(val & AUD_STS_IOC_IRQ_MASK, reg); + if (adata->play_stream) + snd_pcm_period_elapsed(adata->play_stream); + return IRQ_HANDLED; + } + + return IRQ_NONE; +} + +static irqreturn_t xlnx_s2mm_irq_handler(int irq, void *arg) +{ + u32 val; + void __iomem *reg; + struct device *dev = arg; + struct xlnx_pcm_drv_data *adata = dev_get_drvdata(dev); + + reg = adata->mmio + XLNX_S2MM_OFFSET + XLNX_AUD_STS; + val = readl(reg); + if (val & AUD_STS_IOC_IRQ_MASK) { + writel(val & AUD_STS_IOC_IRQ_MASK, reg); + if (adata->capture_stream) + snd_pcm_period_elapsed(adata->capture_stream); + return IRQ_HANDLED; + } + + return IRQ_NONE; +} + +static int xlnx_formatter_pcm_open(struct snd_pcm_substream *substream) +{ + int err; + u32 val, data_format_mode; + u32 ch_count_mask, ch_count_shift, data_xfer_mode, data_xfer_shift; + struct xlnx_pcm_stream_param *stream_data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *prtd = substream->private_data; + struct snd_soc_component *component = snd_soc_rtdcom_lookup(prtd, + DRV_NAME); + struct xlnx_pcm_drv_data *adata = dev_get_drvdata(component->dev); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && + !adata->mm2s_presence) + return -ENODEV; + else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE && + !adata->s2mm_presence) + return -ENODEV; + + stream_data = kzalloc(sizeof(*stream_data), GFP_KERNEL); + if (!stream_data) + return -ENOMEM; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + ch_count_mask = CFG_MM2S_CH_MASK; + ch_count_shift = CFG_MM2S_CH_SHIFT; + data_xfer_mode = CFG_MM2S_XFER_MASK; + data_xfer_shift = CFG_MM2S_XFER_SHIFT; + data_format_mode = CFG_MM2S_PKG_MASK; + stream_data->mmio = adata->mmio + XLNX_MM2S_OFFSET; + adata->play_stream = substream; + + } else { + ch_count_mask = CFG_S2MM_CH_MASK; + ch_count_shift = CFG_S2MM_CH_SHIFT; + data_xfer_mode = CFG_S2MM_XFER_MASK; + data_xfer_shift = CFG_S2MM_XFER_SHIFT; + data_format_mode = CFG_S2MM_PKG_MASK; + stream_data->mmio = adata->mmio + XLNX_S2MM_OFFSET; + adata->capture_stream = substream; + } + + val = readl(adata->mmio + XLNX_AUD_CORE_CONFIG); + + if (!(val & data_format_mode)) + stream_data->interleaved = true; + + stream_data->xfer_mode = (val & data_xfer_mode) >> data_xfer_shift; + stream_data->ch_limit = (val & ch_count_mask) >> ch_count_shift; + dev_info(component->dev, + "stream %d : format = %d mode = %d ch_limit = %d\n", + substream->stream, stream_data->interleaved, + stream_data->xfer_mode, stream_data->ch_limit); + + snd_soc_set_runtime_hwparams(substream, &xlnx_pcm_hardware); + runtime->private_data = stream_data; + + /* Resize the period size divisible by 64 */ + err = snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64); + if (err) { + dev_err(component->dev, + "unable to set constraint on period bytes\n"); + return err; + } + + /* enable DMA IOC irq */ + val = readl(stream_data->mmio + XLNX_AUD_CTRL); + val |= AUD_CTRL_IOC_IRQ_MASK; + writel(val, stream_data->mmio + XLNX_AUD_CTRL); + + return 0; +} + +static int xlnx_formatter_pcm_close(struct snd_pcm_substream *substream) +{ + int ret; + struct xlnx_pcm_stream_param *stream_data = + substream->runtime->private_data; + struct snd_soc_pcm_runtime *prtd = substream->private_data; + struct snd_soc_component *component = snd_soc_rtdcom_lookup(prtd, + DRV_NAME); + + ret = xlnx_formatter_pcm_reset(stream_data->mmio); + if (ret) { + dev_err(component->dev, "audio formatter reset failed\n"); + goto err_reset; + } + xlnx_formatter_disable_irqs(stream_data->mmio, substream->stream); + +err_reset: + kfree(stream_data); + return 0; +} + +static snd_pcm_uframes_t +xlnx_formatter_pcm_pointer(struct snd_pcm_substream *substream) +{ + u32 pos; + struct snd_pcm_runtime *runtime = substream->runtime; + struct xlnx_pcm_stream_param *stream_data = runtime->private_data; + + pos = readl(stream_data->mmio + XLNX_AUD_XFER_COUNT); + + if (pos >= stream_data->buffer_size) + pos = 0; + + return bytes_to_frames(runtime, pos); +} + +static int xlnx_formatter_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + u32 low, high, active_ch, val, bytes_per_ch, bits_per_sample; + int status; + u64 size; + struct snd_pcm_runtime *runtime = substream->runtime; + struct xlnx_pcm_stream_param *stream_data = runtime->private_data; + + active_ch = params_channels(params); + if (active_ch > stream_data->ch_limit) + return -EINVAL; + + size = params_buffer_bytes(params); + status = snd_pcm_lib_malloc_pages(substream, size); + if (status < 0) + return status; + + stream_data->buffer_size = size; + + low = lower_32_bits(substream->dma_buffer.addr); + high = upper_32_bits(substream->dma_buffer.addr); + writel(low, stream_data->mmio + XLNX_AUD_BUFF_ADDR_LSB); + writel(high, stream_data->mmio + XLNX_AUD_BUFF_ADDR_MSB); + + val = readl(stream_data->mmio + XLNX_AUD_CTRL); + bits_per_sample = params_width(params); + switch (bits_per_sample) { + case 8: + val |= (BIT_DEPTH_8 << AUD_CTRL_DATA_WIDTH_SHIFT); + break; + case 16: + val |= (BIT_DEPTH_16 << AUD_CTRL_DATA_WIDTH_SHIFT); + break; + case 20: + val |= (BIT_DEPTH_20 << AUD_CTRL_DATA_WIDTH_SHIFT); + break; + case 24: + val |= (BIT_DEPTH_24 << AUD_CTRL_DATA_WIDTH_SHIFT); + break; + case 32: + val |= (BIT_DEPTH_32 << AUD_CTRL_DATA_WIDTH_SHIFT); + break; + default: + return -EINVAL; + } + + val |= active_ch << AUD_CTRL_ACTIVE_CH_SHIFT; + writel(val, stream_data->mmio + XLNX_AUD_CTRL); + + val = (params_periods(params) << PERIOD_CFG_PERIODS_SHIFT) + | params_period_bytes(params); + writel(val, stream_data->mmio + XLNX_AUD_PERIOD_CONFIG); + bytes_per_ch = DIV_ROUND_UP(params_period_bytes(params), active_ch); + writel(bytes_per_ch, stream_data->mmio + XLNX_BYTES_PER_CH); + + return 0; +} + +static int xlnx_formatter_pcm_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static int xlnx_formatter_pcm_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + u32 val; + struct xlnx_pcm_stream_param *stream_data = + substream->runtime->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: + val = readl(stream_data->mmio + XLNX_AUD_CTRL); + val |= AUD_CTRL_DMA_EN_MASK; + writel(val, stream_data->mmio + XLNX_AUD_CTRL); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: + val = readl(stream_data->mmio + XLNX_AUD_CTRL); + val &= ~AUD_CTRL_DMA_EN_MASK; + writel(val, stream_data->mmio + XLNX_AUD_CTRL); + break; + } + + return 0; +} + +static int xlnx_formatter_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd, + DRV_NAME); + return snd_pcm_lib_preallocate_pages_for_all(rtd->pcm, + SNDRV_DMA_TYPE_DEV, component->dev, + xlnx_pcm_hardware.buffer_bytes_max, + xlnx_pcm_hardware.buffer_bytes_max); +} + +static const struct snd_pcm_ops xlnx_formatter_pcm_ops = { + .open = xlnx_formatter_pcm_open, + .close = xlnx_formatter_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = xlnx_formatter_pcm_hw_params, + .hw_free = xlnx_formatter_pcm_hw_free, + .trigger = xlnx_formatter_pcm_trigger, + .pointer = xlnx_formatter_pcm_pointer, +}; + +static const struct snd_soc_component_driver xlnx_asoc_component = { + .name = DRV_NAME, + .ops = &xlnx_formatter_pcm_ops, + .pcm_new = xlnx_formatter_pcm_new, +}; + +static int xlnx_formatter_pcm_probe(struct platform_device *pdev) +{ + int ret; + u32 val; + struct xlnx_pcm_drv_data *aud_drv_data; + struct resource *res; + struct device *dev = &pdev->dev; + + aud_drv_data = devm_kzalloc(dev, sizeof(*aud_drv_data), GFP_KERNEL); + if (!aud_drv_data) + return -ENOMEM; + + aud_drv_data->axi_clk = devm_clk_get(dev, "s_axi_lite_aclk"); + if (IS_ERR(aud_drv_data->axi_clk)) { + ret = PTR_ERR(aud_drv_data->axi_clk); + dev_err(dev, "failed to get s_axi_lite_aclk(%d)\n", ret); + return ret; + } + ret = clk_prepare_enable(aud_drv_data->axi_clk); + if (ret) { + dev_err(dev, + "failed to enable s_axi_lite_aclk(%d)\n", ret); + return ret; + } + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(dev, "audio formatter node:addr to resource failed\n"); + ret = -ENXIO; + goto clk_err; + } + aud_drv_data->mmio = devm_ioremap_resource(dev, res); + if (IS_ERR(aud_drv_data->mmio)) { + dev_err(dev, "audio formatter ioremap failed\n"); + ret = PTR_ERR(aud_drv_data->mmio); + goto clk_err; + } + + val = readl(aud_drv_data->mmio + XLNX_AUD_CORE_CONFIG); + if (val & AUD_CFG_MM2S_MASK) { + aud_drv_data->mm2s_presence = true; + ret = xlnx_formatter_pcm_reset(aud_drv_data->mmio + + XLNX_MM2S_OFFSET); + if (ret) { + dev_err(dev, "audio formatter reset failed\n"); + goto clk_err; + } + xlnx_formatter_disable_irqs(aud_drv_data->mmio + + XLNX_MM2S_OFFSET, + SNDRV_PCM_STREAM_PLAYBACK); + + aud_drv_data->mm2s_irq = platform_get_irq_byname(pdev, + "irq_mm2s"); + if (aud_drv_data->mm2s_irq < 0) { + dev_err(dev, "xlnx audio mm2s irq resource failed\n"); + ret = aud_drv_data->mm2s_irq; + goto clk_err; + } + ret = devm_request_irq(dev, aud_drv_data->mm2s_irq, + xlnx_mm2s_irq_handler, 0, + "xlnx_formatter_pcm_mm2s_irq", dev); + if (ret) { + dev_err(dev, "xlnx audio mm2s irq request failed\n"); + goto clk_err; + } + } + if (val & AUD_CFG_S2MM_MASK) { + aud_drv_data->s2mm_presence = true; + ret = xlnx_formatter_pcm_reset(aud_drv_data->mmio + + XLNX_S2MM_OFFSET); + if (ret) { + dev_err(dev, "audio formatter reset failed\n"); + goto clk_err; + } + xlnx_formatter_disable_irqs(aud_drv_data->mmio + + XLNX_S2MM_OFFSET, + SNDRV_PCM_STREAM_CAPTURE); + + aud_drv_data->s2mm_irq = platform_get_irq_byname(pdev, + "irq_s2mm"); + if (aud_drv_data->s2mm_irq < 0) { + dev_err(dev, "xlnx audio s2mm irq resource failed\n"); + ret = aud_drv_data->s2mm_irq; + goto clk_err; + } + ret = devm_request_irq(dev, aud_drv_data->s2mm_irq, + xlnx_s2mm_irq_handler, 0, + "xlnx_formatter_pcm_s2mm_irq", + dev); + if (ret) { + dev_err(dev, "xlnx audio s2mm irq request failed\n"); + goto clk_err; + } + } + + dev_set_drvdata(dev, aud_drv_data); + + ret = devm_snd_soc_register_component(dev, &xlnx_asoc_component, + NULL, 0); + if (ret) { + dev_err(dev, "pcm platform device register failed\n"); + goto clk_err; + } + + return 0; + +clk_err: + clk_disable_unprepare(aud_drv_data->axi_clk); + return ret; +} + +static int xlnx_formatter_pcm_remove(struct platform_device *pdev) +{ + int ret = 0; + struct xlnx_pcm_drv_data *adata = dev_get_drvdata(&pdev->dev); + + if (adata->s2mm_presence) + ret = xlnx_formatter_pcm_reset(adata->mmio + XLNX_S2MM_OFFSET); + + /* Try MM2S reset, even if S2MM reset fails */ + if (adata->mm2s_presence) + ret = xlnx_formatter_pcm_reset(adata->mmio + XLNX_MM2S_OFFSET); + + if (ret) + dev_err(&pdev->dev, "audio formatter reset failed\n"); + + clk_disable_unprepare(adata->axi_clk); + return ret; +} + +static const struct of_device_id xlnx_formatter_pcm_of_match[] = { + { .compatible = "xlnx,audio-formatter-1.0"}, + {}, +}; +MODULE_DEVICE_TABLE(of, xlnx_formatter_pcm_of_match); + +static struct platform_driver xlnx_formatter_pcm_driver = { + .probe = xlnx_formatter_pcm_probe, + .remove = xlnx_formatter_pcm_remove, + .driver = { + .name = DRV_NAME, + .of_match_table = xlnx_formatter_pcm_of_match, + }, +}; + +module_platform_driver(xlnx_formatter_pcm_driver); +MODULE_AUTHOR("Maruthi Srinivas Bayyavarapu "); +MODULE_LICENSE("GPL v2"); From b31daa15af760747b91dbb76c80306d77d8ae05f Mon Sep 17 00:00:00 2001 From: Maruthi Srinivas Bayyavarapu Date: Fri, 21 Dec 2018 14:27:29 +0530 Subject: [PATCH 34/67] ASoC: xlnx: enable audio formatter driver build Enable audio formatter driver build. Signed-off-by: Maruthi Srinivas Bayyavarapu Signed-off-by: Mark Brown --- sound/soc/xilinx/Kconfig | 7 +++++++ sound/soc/xilinx/Makefile | 2 ++ 2 files changed, 9 insertions(+) diff --git a/sound/soc/xilinx/Kconfig b/sound/soc/xilinx/Kconfig index 723a583a8d57..ac48d6a00c36 100644 --- a/sound/soc/xilinx/Kconfig +++ b/sound/soc/xilinx/Kconfig @@ -6,3 +6,10 @@ config SND_SOC_XILINX_I2S mode, IP receives audio in AES format, extracts PCM and sends PCM data. In receiver mode, IP receives PCM audio and encapsulates PCM in AES format and sends AES data. + +config SND_SOC_XILINX_AUDIO_FORMATTER + tristate "Audio support for the the Xilinx audio formatter" + help + Select this option to enable Xilinx audio formatter + support. This provides DMA platform device support for + audio functionality. diff --git a/sound/soc/xilinx/Makefile b/sound/soc/xilinx/Makefile index 6c1209b9ee75..432693b1cc79 100644 --- a/sound/soc/xilinx/Makefile +++ b/sound/soc/xilinx/Makefile @@ -1,2 +1,4 @@ snd-soc-xlnx-i2s-objs := xlnx_i2s.o obj-$(CONFIG_SND_SOC_XILINX_I2S) += snd-soc-xlnx-i2s.o +snd-soc-xlnx-formatter-pcm-objs := xlnx_formatter_pcm.o +obj-$(CONFIG_SND_SOC_XILINX_AUDIO_FORMATTER) += snd-soc-xlnx-formatter-pcm.o From de2949fe262197298036989924d05f5de6b9815a Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 20 Dec 2018 10:45:54 +0900 Subject: [PATCH 35/67] ASoC: audio-graph-card: add 1 CPU : 1 Codec support again audio-graph-card is now supporting normal sound and DPCM sound. For DPCM sound, original sound card (= audio-graph-scu) had been supported 1 CPU : 1 Codec connection which uses hw_params_fixup() for convert-rate/channel. But, merged audio-graph-card is completely forgeting about it. This patch re-support it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/audio-graph-card.c | 44 ++++++++++++++++++++-------- 1 file changed, 32 insertions(+), 12 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index c3e80bc27e80..638333cdac66 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -408,6 +408,7 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) struct device_node *codec_ep = NULL; struct device_node *codec_port = NULL; struct device_node *codec_port_old = NULL; + struct asoc_simple_card_data adata; int rc, ret; int link_idx, dai_idx, conf_idx; int cpu; @@ -453,7 +454,13 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) dev_dbg(dev, "%pOFf <-> %pOFf\n", cpu_ep, codec_ep); - if (of_get_child_count(codec_port) > 1) { + memset(&adata, 0, sizeof(adata)); + asoc_graph_card_get_conversion(dev, codec_ep, &adata); + asoc_graph_card_get_conversion(dev, cpu_ep, &adata); + + if ((of_get_child_count(codec_port) > 1) || + adata.convert_rate || + adata.convert_channels) { /* * for DPCM sound */ @@ -495,7 +502,7 @@ static void asoc_graph_get_dais_count(struct device *dev, struct device_node *codec_ep; struct device_node *codec_port; struct device_node *codec_port_old; - struct device_node *codec_port_old2; + struct asoc_simple_card_data adata; int rc; /* @@ -534,9 +541,17 @@ static void asoc_graph_get_dais_count(struct device *dev, * => 6 links = 0xCPU-Codec + 4xCPU-dummy + 2xdummy-Codec * => 6 DAIs = 4xCPU + 2xCodec * => 2 ccnf = 2xdummy-Codec + * + * ex4) + * CPU0 --- Codec0 (convert-rate) link : 3 + * CPU1 --- Codec1 dais : 4 + * ccnf : 1 + * + * => 3 links = 1xCPU-Codec + 1xCPU-dummy + 1xdummy-Codec + * => 4 DAIs = 2xCPU + 2xCodec + * => 1 ccnf = 1xdummy-Codec */ codec_port_old = NULL; - codec_port_old2 = NULL; of_for_each_phandle(&it, rc, node, "dais", NULL, 0) { cpu_port = it.node; cpu_ep = NULL; @@ -554,17 +569,22 @@ static void asoc_graph_get_dais_count(struct device *dev, (*link_num)++; (*dais_num)++; - if (codec_port_old == codec_port) { - if (codec_port_old2 != codec_port_old) { - (*link_num)++; - (*ccnf_num)++; - } + memset(&adata, 0, sizeof(adata)); + asoc_graph_card_get_conversion(dev, codec_ep, &adata); + asoc_graph_card_get_conversion(dev, cpu_ep, &adata); - codec_port_old2 = codec_port_old; - continue; + if ((of_get_child_count(codec_port) > 1) || + adata.convert_rate || adata.convert_channels) { + + if (codec_port_old == codec_port) + continue; + + (*link_num)++; + (*ccnf_num)++; + (*dais_num)++; + } else { + (*dais_num)++; } - - (*dais_num)++; codec_port_old = codec_port; } } From 1e4771a62fd7a6bab058529c450d3d87a8bd5b1a Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 20 Dec 2018 10:45:59 +0900 Subject: [PATCH 36/67] ASoC: audio-graph-card: add link_info Current audio-graph-card is parsing DAI link for both "normal sound" and "DPCM sound". On this driver, it needs to count and parse DAIs/Links/Codec Conf from each links. Then, counting/parsing link loop are very similar, but using different implementation. Because of this background, the link loop code is very mysterious. Mystery code will be trouble in the future. To preparing cleanup code, this patch adds link_info which handles number of DAIs/Links/Codec Conf, and CPU/Codec turn. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/audio-graph-card.c | 99 ++++++++++++++-------------- 1 file changed, 51 insertions(+), 48 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 638333cdac66..cd9beb801fc1 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -39,6 +39,13 @@ struct graph_card_data { struct gpio_desc *pa_gpio; }; +struct link_info { + int dais; /* number of dai */ + int link; /* number of link */ + int conf; /* number of codec_conf */ + int cpu; /* turn for CPU / Codec */ +}; + #define graph_priv_to_card(priv) (&(priv)->snd_card) #define graph_priv_to_props(priv, i) ((priv)->dai_props + (i)) #define graph_priv_to_dev(priv) (graph_priv_to_card(priv)->dev) @@ -189,13 +196,12 @@ static int asoc_graph_card_dai_link_of_dpcm(struct device_node *top, struct device_node *cpu_ep, struct device_node *codec_ep, struct graph_card_data *priv, - int *dai_idx, int link_idx, - int *conf_idx, int is_cpu) + struct link_info *li) { struct device *dev = graph_priv_to_dev(priv); - struct snd_soc_dai_link *dai_link = graph_priv_to_link(priv, link_idx); - struct graph_dai_props *dai_props = graph_priv_to_props(priv, link_idx); - struct device_node *ep = is_cpu ? cpu_ep : codec_ep; + struct snd_soc_dai_link *dai_link = graph_priv_to_link(priv, li->link); + struct graph_dai_props *dai_props = graph_priv_to_props(priv, li->link); + struct device_node *ep = li->cpu ? cpu_ep : codec_ep; struct device_node *port = of_get_parent(ep); struct device_node *ports = of_get_parent(port); struct device_node *node = of_graph_get_port_parent(ep); @@ -203,7 +209,9 @@ static int asoc_graph_card_dai_link_of_dpcm(struct device_node *top, struct snd_soc_dai_link_component *codecs = dai_link->codecs; int ret; - dev_dbg(dev, "link_of DPCM (for %s)\n", is_cpu ? "CPU" : "Codec"); + li->link++; + + dev_dbg(dev, "link_of DPCM (%pOF)\n", ep); of_property_read_u32(top, "mclk-fs", &dai_props->mclk_fs); of_property_read_u32(ports, "mclk-fs", &dai_props->mclk_fs); @@ -215,7 +223,7 @@ static int asoc_graph_card_dai_link_of_dpcm(struct device_node *top, of_node_put(ports); of_node_put(port); - if (is_cpu) { + if (li->cpu) { /* BE is dummy */ codecs->of_node = NULL; @@ -227,7 +235,7 @@ static int asoc_graph_card_dai_link_of_dpcm(struct device_node *top, dai_link->dpcm_merged_format = 1; dai = - dai_props->cpu_dai = &priv->dais[(*dai_idx)++]; + dai_props->cpu_dai = &priv->dais[li->dais++]; ret = asoc_simple_card_parse_graph_cpu(ep, dai_link); if (ret) @@ -259,10 +267,10 @@ static int asoc_graph_card_dai_link_of_dpcm(struct device_node *top, dai_link->be_hw_params_fixup = asoc_graph_card_be_hw_params_fixup; dai = - dai_props->codec_dai = &priv->dais[(*dai_idx)++]; + dai_props->codec_dai = &priv->dais[li->dais++]; cconf = - dai_props->codec_conf = &priv->codec_conf[(*conf_idx)++]; + dai_props->codec_conf = &priv->codec_conf[li->conf++]; ret = asoc_simple_card_parse_graph_codec(ep, dai_link); if (ret < 0) @@ -314,11 +322,11 @@ static int asoc_graph_card_dai_link_of(struct device_node *top, struct device_node *cpu_ep, struct device_node *codec_ep, struct graph_card_data *priv, - int *dai_idx, int link_idx) + struct link_info *li) { struct device *dev = graph_priv_to_dev(priv); - struct snd_soc_dai_link *dai_link = graph_priv_to_link(priv, link_idx); - struct graph_dai_props *dai_props = graph_priv_to_props(priv, link_idx); + struct snd_soc_dai_link *dai_link = graph_priv_to_link(priv, li->link); + struct graph_dai_props *dai_props = graph_priv_to_props(priv, li->link); struct device_node *cpu_port = of_get_parent(cpu_ep); struct device_node *codec_port = of_get_parent(codec_ep); struct device_node *cpu_ports = of_get_parent(cpu_port); @@ -327,12 +335,14 @@ static int asoc_graph_card_dai_link_of(struct device_node *top, struct asoc_simple_dai *codec_dai; int ret; - dev_dbg(dev, "link_of\n"); + dev_dbg(dev, "link_of (%pOF)\n", cpu_ep); + + li->link++; cpu_dai = - dai_props->cpu_dai = &priv->dais[(*dai_idx)++]; + dai_props->cpu_dai = &priv->dais[li->dais++]; codec_dai = - dai_props->codec_dai = &priv->dais[(*dai_idx)++]; + dai_props->codec_dai = &priv->dais[li->dais++]; /* Factor to mclk, used in hw_params() */ of_property_read_u32(top, "mclk-fs", &dai_props->mclk_fs); @@ -409,9 +419,8 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) struct device_node *codec_port = NULL; struct device_node *codec_port_old = NULL; struct asoc_simple_card_data adata; + struct link_info li; int rc, ret; - int link_idx, dai_idx, conf_idx; - int cpu; ret = asoc_simple_card_of_parse_widgets(card, NULL); if (ret < 0) @@ -421,11 +430,9 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) if (ret < 0) return ret; - link_idx = 0; - dai_idx = 0; - conf_idx = 0; + memset(&li, 0, sizeof(li)); codec_port_old = NULL; - for (cpu = 1; cpu >= 0; cpu--) { + for (li.cpu = 1; li.cpu >= 0; li.cpu--) { /* * Detect all CPU first, and Detect all Codec 2nd. * @@ -464,22 +471,19 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) /* * for DPCM sound */ - if (!cpu) { + if (!li.cpu) { if (codec_port_old == codec_port) continue; codec_port_old = codec_port; } ret = asoc_graph_card_dai_link_of_dpcm( - top, cpu_ep, codec_ep, priv, - &dai_idx, link_idx++, - &conf_idx, cpu); - } else if (cpu) { + top, cpu_ep, codec_ep, priv, &li); + } else if (li.cpu) { /* * for Normal sound */ ret = asoc_graph_card_dai_link_of( - top, cpu_ep, codec_ep, priv, - &dai_idx, link_idx++); + top, cpu_ep, codec_ep, priv, &li); } if (ret < 0) return ret; @@ -491,9 +495,7 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) } static void asoc_graph_get_dais_count(struct device *dev, - int *link_num, - int *dais_num, - int *ccnf_num) + struct link_info *li) { struct of_phandle_iterator it; struct device_node *node = dev->of_node; @@ -566,8 +568,8 @@ static void asoc_graph_get_dais_count(struct device *dev, of_node_put(codec_ep); of_node_put(codec_port); - (*link_num)++; - (*dais_num)++; + li->link++; + li->dais++; memset(&adata, 0, sizeof(adata)); asoc_graph_card_get_conversion(dev, codec_ep, &adata); @@ -579,11 +581,11 @@ static void asoc_graph_get_dais_count(struct device *dev, if (codec_port_old == codec_port) continue; - (*link_num)++; - (*ccnf_num)++; - (*dais_num)++; + li->link++; + li->conf++; + li->dais++; } else { - (*dais_num)++; + li->dais++; } codec_port_old = codec_port; } @@ -615,7 +617,7 @@ static int asoc_graph_card_probe(struct platform_device *pdev) struct device *dev = &pdev->dev; struct snd_soc_card *card; struct snd_soc_codec_conf *cconf; - int lnum = 0, dnum = 0, cnum = 0; + struct link_info li; int ret, i; /* Allocate the private data and the DAI link array */ @@ -623,14 +625,15 @@ static int asoc_graph_card_probe(struct platform_device *pdev) if (!priv) return -ENOMEM; - asoc_graph_get_dais_count(dev, &lnum, &dnum, &cnum); - if (!lnum || !dnum) + memset(&li, 0, sizeof(li)); + asoc_graph_get_dais_count(dev, &li); + if (!li.link || !li.dais) return -EINVAL; - dai_props = devm_kcalloc(dev, lnum, sizeof(*dai_props), GFP_KERNEL); - dai_link = devm_kcalloc(dev, lnum, sizeof(*dai_link), GFP_KERNEL); - dais = devm_kcalloc(dev, dnum, sizeof(*dais), GFP_KERNEL); - cconf = devm_kcalloc(dev, cnum, sizeof(*cconf), GFP_KERNEL); + dai_props = devm_kcalloc(dev, li.link, sizeof(*dai_props), GFP_KERNEL); + dai_link = devm_kcalloc(dev, li.link, sizeof(*dai_link), GFP_KERNEL); + dais = devm_kcalloc(dev, li.dais, sizeof(*dais), GFP_KERNEL); + cconf = devm_kcalloc(dev, li.conf, sizeof(*cconf), GFP_KERNEL); if (!dai_props || !dai_link || !dais) return -ENOMEM; @@ -640,7 +643,7 @@ static int asoc_graph_card_probe(struct platform_device *pdev) * see * soc-core.c :: snd_soc_init_multicodec() */ - for (i = 0; i < lnum; i++) { + for (i = 0; i < li.link; i++) { dai_link[i].codecs = &dai_props[i].codecs; dai_link[i].num_codecs = 1; dai_link[i].platform = &dai_props[i].platform; @@ -663,12 +666,12 @@ static int asoc_graph_card_probe(struct platform_device *pdev) card->owner = THIS_MODULE; card->dev = dev; card->dai_link = dai_link; - card->num_links = lnum; + card->num_links = li.link; card->dapm_widgets = asoc_graph_card_dapm_widgets; card->num_dapm_widgets = ARRAY_SIZE(asoc_graph_card_dapm_widgets); card->probe = asoc_graph_soc_card_probe; card->codec_conf = cconf; - card->num_configs = cnum; + card->num_configs = li.conf; ret = asoc_graph_card_parse_of(priv); if (ret < 0) { From dd98fbc558a035728beed08a16c443f9fd37eb2b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 20 Dec 2018 10:46:05 +0900 Subject: [PATCH 37/67] ASoC: audio-graph-card: cleanup DAI link loop method - step1 Current audio-graph-card is parsing DAI link for both "normal sound" and "DPCM sound". On this driver, it needs to count and parse DAIs/Links/Codec Conf from each links. Then, counting/parsing link loop are very similar, but using different implementation. Because of this background, the link loop code is very mysterious. Mystery code will be trouble in the future. This patch adds/modifies counting and parsing function for "normal sound" and "DPCM sound", and call it from link loop. This is prepare for cleanup DAI link loop method. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/audio-graph-card.c | 134 ++++++++++++++++++--------- 1 file changed, 91 insertions(+), 43 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index cd9beb801fc1..fbd32129c518 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -192,23 +192,32 @@ static void asoc_graph_card_get_conversion(struct device *dev, asoc_simple_card_parse_convert(dev, ep, NULL, adata); } -static int asoc_graph_card_dai_link_of_dpcm(struct device_node *top, +static int asoc_graph_card_dai_link_of_dpcm(struct graph_card_data *priv, struct device_node *cpu_ep, struct device_node *codec_ep, - struct graph_card_data *priv, - struct link_info *li) + struct link_info *li, + int dup_codec) { struct device *dev = graph_priv_to_dev(priv); struct snd_soc_dai_link *dai_link = graph_priv_to_link(priv, li->link); struct graph_dai_props *dai_props = graph_priv_to_props(priv, li->link); + struct device_node *top = dev->of_node; struct device_node *ep = li->cpu ? cpu_ep : codec_ep; - struct device_node *port = of_get_parent(ep); - struct device_node *ports = of_get_parent(port); - struct device_node *node = of_graph_get_port_parent(ep); + struct device_node *port; + struct device_node *ports; + struct device_node *node; struct asoc_simple_dai *dai; struct snd_soc_dai_link_component *codecs = dai_link->codecs; int ret; + /* Do it all CPU endpoint, and 1st Codec endpoint */ + if (!li->cpu && dup_codec) + return 0; + + port = of_get_parent(ep); + ports = of_get_parent(port); + node = of_graph_get_port_parent(ep); + li->link++; dev_dbg(dev, "link_of DPCM (%pOF)\n", ep); @@ -222,6 +231,7 @@ static int asoc_graph_card_dai_link_of_dpcm(struct device_node *top, of_node_put(ports); of_node_put(port); + of_node_put(node); if (li->cpu) { @@ -318,23 +328,32 @@ static int asoc_graph_card_dai_link_of_dpcm(struct device_node *top, return 0; } -static int asoc_graph_card_dai_link_of(struct device_node *top, +static int asoc_graph_card_dai_link_of(struct graph_card_data *priv, struct device_node *cpu_ep, struct device_node *codec_ep, - struct graph_card_data *priv, struct link_info *li) { struct device *dev = graph_priv_to_dev(priv); struct snd_soc_dai_link *dai_link = graph_priv_to_link(priv, li->link); struct graph_dai_props *dai_props = graph_priv_to_props(priv, li->link); - struct device_node *cpu_port = of_get_parent(cpu_ep); - struct device_node *codec_port = of_get_parent(codec_ep); - struct device_node *cpu_ports = of_get_parent(cpu_port); - struct device_node *codec_ports = of_get_parent(codec_port); + struct device_node *top = dev->of_node; + struct device_node *cpu_port; + struct device_node *codec_port; + struct device_node *cpu_ports; + struct device_node *codec_ports; struct asoc_simple_dai *cpu_dai; struct asoc_simple_dai *codec_dai; int ret; + /* Do it only CPU turn */ + if (!li->cpu) + return 0; + + cpu_port = of_get_parent(cpu_ep); + cpu_ports = of_get_parent(cpu_port); + codec_port = of_get_parent(codec_ep); + codec_ports = of_get_parent(codec_port); + dev_dbg(dev, "link_of (%pOF)\n", cpu_ep); li->link++; @@ -471,22 +490,19 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) /* * for DPCM sound */ - if (!li.cpu) { - if (codec_port_old == codec_port) - continue; - codec_port_old = codec_port; - } ret = asoc_graph_card_dai_link_of_dpcm( - top, cpu_ep, codec_ep, priv, &li); + priv, cpu_ep, codec_ep, &li, + (codec_port_old == codec_port)); } else if (li.cpu) { /* * for Normal sound */ ret = asoc_graph_card_dai_link_of( - top, cpu_ep, codec_ep, priv, &li); + priv, cpu_ep, codec_ep, &li); } if (ret < 0) return ret; + codec_port_old = codec_port; } } } @@ -494,9 +510,47 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) return asoc_simple_card_parse_card_name(card, NULL); } -static void asoc_graph_get_dais_count(struct device *dev, +static int asoc_graph_card_count_noml(struct graph_card_data *priv, + struct device_node *cpu_ep, + struct device_node *codec_ep, struct link_info *li) { + struct device *dev = graph_priv_to_dev(priv); + + li->link += 1; /* 1xCPU-Codec */ + li->dais += 2; /* 1xCPU + 1xCodec */ + + dev_dbg(dev, "Count As Normal\n"); + + return 0; +} + +static int asoc_graph_card_count_dpcm(struct graph_card_data *priv, + struct device_node *cpu_ep, + struct device_node *codec_ep, + struct link_info *li, + int dup_codec) +{ + struct device *dev = graph_priv_to_dev(priv); + + li->link++; /* 1xCPU-dummy */ + li->dais++; /* 1xCPU */ + + if (!dup_codec) { + li->link++; /* 1xdummy-Codec */ + li->conf++; /* 1xdummy-Codec */ + li->dais++; /* 1xCodec */ + } + + dev_dbg(dev, "Count As DPCM\n"); + + return 0; +} + +static void asoc_graph_get_dais_count(struct graph_card_data *priv, + struct link_info *li) +{ + struct device *dev = graph_priv_to_dev(priv); struct of_phandle_iterator it; struct device_node *node = dev->of_node; struct device_node *cpu_port; @@ -568,24 +622,18 @@ static void asoc_graph_get_dais_count(struct device *dev, of_node_put(codec_ep); of_node_put(codec_port); - li->link++; - li->dais++; - memset(&adata, 0, sizeof(adata)); asoc_graph_card_get_conversion(dev, codec_ep, &adata); asoc_graph_card_get_conversion(dev, cpu_ep, &adata); if ((of_get_child_count(codec_port) > 1) || adata.convert_rate || adata.convert_channels) { - - if (codec_port_old == codec_port) - continue; - - li->link++; - li->conf++; - li->dais++; + asoc_graph_card_count_dpcm(priv, + cpu_ep, codec_ep, li, + (codec_port_old == codec_port)); } else { - li->dais++; + asoc_graph_card_count_noml(priv, + cpu_ep, codec_ep, li); } codec_port_old = codec_port; } @@ -625,8 +673,15 @@ static int asoc_graph_card_probe(struct platform_device *pdev) if (!priv) return -ENOMEM; + card = graph_priv_to_card(priv); + card->owner = THIS_MODULE; + card->dev = dev; + card->dapm_widgets = asoc_graph_card_dapm_widgets; + card->num_dapm_widgets = ARRAY_SIZE(asoc_graph_card_dapm_widgets); + card->probe = asoc_graph_soc_card_probe; + memset(&li, 0, sizeof(li)); - asoc_graph_get_dais_count(dev, &li); + asoc_graph_get_dais_count(priv, &li); if (!li.link || !li.dais) return -EINVAL; @@ -656,20 +711,13 @@ static int asoc_graph_card_probe(struct platform_device *pdev) return ret; } - priv->dai_props = dai_props; - priv->dai_link = dai_link; - priv->dais = dais; - priv->codec_conf = cconf; + priv->dai_props = dai_props; + priv->dai_link = dai_link; + priv->dais = dais; + priv->codec_conf = cconf; - /* Init snd_soc_card */ - card = graph_priv_to_card(priv); - card->owner = THIS_MODULE; - card->dev = dev; card->dai_link = dai_link; card->num_links = li.link; - card->dapm_widgets = asoc_graph_card_dapm_widgets; - card->num_dapm_widgets = ARRAY_SIZE(asoc_graph_card_dapm_widgets); - card->probe = asoc_graph_soc_card_probe; card->codec_conf = cconf; card->num_configs = li.conf; From fce9b90c1ab7e915553c57353355700c79b39c86 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 20 Dec 2018 10:46:20 +0900 Subject: [PATCH 38/67] ASoC: audio-graph-card: cleanup DAI link loop method - step2 Current audio-graph-card is parsing DAI link for both "normal sound" and "DPCM sound". On this driver, it needs to count and parse DAIs/Links/Codec Conf from each links. Then, counting/parsing link loop are very similar, but using different implementation. Because of this background, the link loop code is very mysterious. Mystery code will be trouble in the future. This patch cleanups the code by using asoc_graph_card_for_each_link() which judges normal link / DPCM link. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/audio-graph-card.c | 168 ++++++++++++--------------- 1 file changed, 77 insertions(+), 91 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index fbd32129c518..1152de37110e 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -425,22 +425,80 @@ static int asoc_graph_card_dai_link_of(struct graph_card_data *priv, return 0; } -static int asoc_graph_card_parse_of(struct graph_card_data *priv) +static int asoc_graph_card_for_each_link(struct graph_card_data *priv, + struct link_info *li, + int (*func_noml)(struct graph_card_data *priv, + struct device_node *cpu_ep, + struct device_node *codec_ep, + struct link_info *li), + int (*func_dpcm)(struct graph_card_data *priv, + struct device_node *cpu_ep, + struct device_node *codec_ep, + struct link_info *li, int dup_codec)) { struct of_phandle_iterator it; struct device *dev = graph_priv_to_dev(priv); - struct snd_soc_card *card = graph_priv_to_card(priv); - struct device_node *top = dev->of_node; - struct device_node *node = top; + struct device_node *node = dev->of_node; struct device_node *cpu_port; - struct device_node *cpu_ep = NULL; - struct device_node *codec_ep = NULL; - struct device_node *codec_port = NULL; - struct device_node *codec_port_old = NULL; + struct device_node *cpu_ep; + struct device_node *codec_ep; + struct device_node *codec_port; + struct device_node *codec_port_old = NULL; struct asoc_simple_card_data adata; - struct link_info li; int rc, ret; + /* loop for all listed CPU port */ + of_for_each_phandle(&it, rc, node, "dais", NULL, 0) { + cpu_port = it.node; + cpu_ep = NULL; + + /* loop for all CPU endpoint */ + while (1) { + cpu_ep = of_get_next_child(cpu_port, cpu_ep); + if (!cpu_ep) + break; + + /* get codec */ + codec_ep = of_graph_get_remote_endpoint(cpu_ep); + codec_port = of_get_parent(codec_ep); + + of_node_put(codec_ep); + of_node_put(codec_port); + + /* get convert-xxx property */ + memset(&adata, 0, sizeof(adata)); + asoc_graph_card_get_conversion(dev, codec_ep, &adata); + asoc_graph_card_get_conversion(dev, cpu_ep, &adata); + + /* + * It is DPCM + * if Codec port has many endpoints, + * or has convert-xxx property + */ + if ((of_get_child_count(codec_port) > 1) || + adata.convert_rate || adata.convert_channels) + ret = func_dpcm(priv, cpu_ep, codec_ep, li, + (codec_port_old == codec_port)); + /* else normal sound */ + else + ret = func_noml(priv, cpu_ep, codec_ep, li); + + if (ret < 0) + return ret; + + codec_port_old = codec_port; + } + } + + return 0; +} + +static int asoc_graph_card_parse_of(struct graph_card_data *priv) +{ + struct snd_soc_card *card = graph_priv_to_card(priv); + struct link_info li; + int ret; + ret = asoc_simple_card_of_parse_widgets(card, NULL); if (ret < 0) return ret; @@ -450,7 +508,6 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) return ret; memset(&li, 0, sizeof(li)); - codec_port_old = NULL; for (li.cpu = 1; li.cpu >= 0; li.cpu--) { /* * Detect all CPU first, and Detect all Codec 2nd. @@ -464,47 +521,11 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) * To avoid random sub-device numbering, * detect "dummy-Codec" in last; */ - of_for_each_phandle(&it, rc, node, "dais", NULL, 0) { - cpu_port = it.node; - cpu_ep = NULL; - while (1) { - cpu_ep = of_get_next_child(cpu_port, cpu_ep); - if (!cpu_ep) - break; - - codec_ep = of_graph_get_remote_endpoint(cpu_ep); - codec_port = of_get_parent(codec_ep); - - of_node_put(codec_ep); - of_node_put(codec_port); - - dev_dbg(dev, "%pOFf <-> %pOFf\n", cpu_ep, codec_ep); - - memset(&adata, 0, sizeof(adata)); - asoc_graph_card_get_conversion(dev, codec_ep, &adata); - asoc_graph_card_get_conversion(dev, cpu_ep, &adata); - - if ((of_get_child_count(codec_port) > 1) || - adata.convert_rate || - adata.convert_channels) { - /* - * for DPCM sound - */ - ret = asoc_graph_card_dai_link_of_dpcm( - priv, cpu_ep, codec_ep, &li, - (codec_port_old == codec_port)); - } else if (li.cpu) { - /* - * for Normal sound - */ - ret = asoc_graph_card_dai_link_of( - priv, cpu_ep, codec_ep, &li); - } - if (ret < 0) - return ret; - codec_port_old = codec_port; - } - } + ret = asoc_graph_card_for_each_link(priv, &li, + asoc_graph_card_dai_link_of, + asoc_graph_card_dai_link_of_dpcm); + if (ret < 0) + return ret; } return asoc_simple_card_parse_card_name(card, NULL); @@ -551,15 +572,6 @@ static void asoc_graph_get_dais_count(struct graph_card_data *priv, struct link_info *li) { struct device *dev = graph_priv_to_dev(priv); - struct of_phandle_iterator it; - struct device_node *node = dev->of_node; - struct device_node *cpu_port; - struct device_node *cpu_ep; - struct device_node *codec_ep; - struct device_node *codec_port; - struct device_node *codec_port_old; - struct asoc_simple_card_data adata; - int rc; /* * link_num : number of links. @@ -607,37 +619,11 @@ static void asoc_graph_get_dais_count(struct graph_card_data *priv, * => 4 DAIs = 2xCPU + 2xCodec * => 1 ccnf = 1xdummy-Codec */ - codec_port_old = NULL; - of_for_each_phandle(&it, rc, node, "dais", NULL, 0) { - cpu_port = it.node; - cpu_ep = NULL; - while (1) { - cpu_ep = of_get_next_child(cpu_port, cpu_ep); - if (!cpu_ep) - break; - - codec_ep = of_graph_get_remote_endpoint(cpu_ep); - codec_port = of_get_parent(codec_ep); - - of_node_put(codec_ep); - of_node_put(codec_port); - - memset(&adata, 0, sizeof(adata)); - asoc_graph_card_get_conversion(dev, codec_ep, &adata); - asoc_graph_card_get_conversion(dev, cpu_ep, &adata); - - if ((of_get_child_count(codec_port) > 1) || - adata.convert_rate || adata.convert_channels) { - asoc_graph_card_count_dpcm(priv, - cpu_ep, codec_ep, li, - (codec_port_old == codec_port)); - } else { - asoc_graph_card_count_noml(priv, - cpu_ep, codec_ep, li); - } - codec_port_old = codec_port; - } - } + asoc_graph_card_for_each_link(priv, li, + asoc_graph_card_count_noml, + asoc_graph_card_count_dpcm); + dev_dbg(dev, "link %d, dais %d, ccnf %d\n", + li->link, li->dais, li->conf); } static int asoc_graph_soc_card_probe(struct snd_soc_card *card) From 97fe6ca4146583d8dccdde51c143c52b385c2682 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 20 Dec 2018 10:46:33 +0900 Subject: [PATCH 39/67] ASoC: audio-graph-card: reduce naming prefix Current audio-graph-card is using asoc_graph_card_xxx() for function / data naming. Because of this long prefix, it is easy to be 80 character over. Let's reduce prefix from asoc_graph_card_xxx() to graph_xxx(). Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/audio-graph-card.c | 164 +++++++++++++-------------- 1 file changed, 82 insertions(+), 82 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 1152de37110e..3ec96cdc683b 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -20,7 +20,7 @@ #include #include -struct graph_card_data { +struct graph_priv { struct snd_soc_card snd_card; struct graph_dai_props { struct asoc_simple_dai *cpu_dai; @@ -53,12 +53,12 @@ struct link_info { #define PREFIX "audio-graph-card," -static int asoc_graph_card_outdrv_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, - int event) +static int graph_outdrv_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) { struct snd_soc_dapm_context *dapm = w->dapm; - struct graph_card_data *priv = snd_soc_card_get_drvdata(dapm->card); + struct graph_priv *priv = snd_soc_card_get_drvdata(dapm->card); switch (event) { case SND_SOC_DAPM_POST_PMU: @@ -74,16 +74,16 @@ static int asoc_graph_card_outdrv_event(struct snd_soc_dapm_widget *w, return 0; } -static const struct snd_soc_dapm_widget asoc_graph_card_dapm_widgets[] = { +static const struct snd_soc_dapm_widget graph_dapm_widgets[] = { SND_SOC_DAPM_OUT_DRV_E("Amplifier", SND_SOC_NOPM, - 0, 0, NULL, 0, asoc_graph_card_outdrv_event, + 0, 0, NULL, 0, graph_outdrv_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), }; -static int asoc_graph_card_startup(struct snd_pcm_substream *substream) +static int graph_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); + struct graph_priv *priv = snd_soc_card_get_drvdata(rtd->card); struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); int ret; @@ -98,10 +98,10 @@ static int asoc_graph_card_startup(struct snd_pcm_substream *substream) return ret; } -static void asoc_graph_card_shutdown(struct snd_pcm_substream *substream) +static void graph_shutdown(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); + struct graph_priv *priv = snd_soc_card_get_drvdata(rtd->card); struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); asoc_simple_card_clk_disable(dai_props->cpu_dai); @@ -109,13 +109,13 @@ static void asoc_graph_card_shutdown(struct snd_pcm_substream *substream) asoc_simple_card_clk_disable(dai_props->codec_dai); } -static int asoc_graph_card_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) +static int graph_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); + struct graph_priv *priv = snd_soc_card_get_drvdata(rtd->card); struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); unsigned int mclk, mclk_fs = 0; int ret = 0; @@ -140,15 +140,15 @@ static int asoc_graph_card_hw_params(struct snd_pcm_substream *substream, return ret; } -static const struct snd_soc_ops asoc_graph_card_ops = { - .startup = asoc_graph_card_startup, - .shutdown = asoc_graph_card_shutdown, - .hw_params = asoc_graph_card_hw_params, +static const struct snd_soc_ops graph_ops = { + .startup = graph_startup, + .shutdown = graph_shutdown, + .hw_params = graph_hw_params, }; -static int asoc_graph_card_dai_init(struct snd_soc_pcm_runtime *rtd) +static int graph_dai_init(struct snd_soc_pcm_runtime *rtd) { - struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); + struct graph_priv *priv = snd_soc_card_get_drvdata(rtd->card); struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); int ret = 0; @@ -165,10 +165,10 @@ static int asoc_graph_card_dai_init(struct snd_soc_pcm_runtime *rtd) return 0; } -static int asoc_graph_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) +static int graph_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) { - struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); + struct graph_priv *priv = snd_soc_card_get_drvdata(rtd->card); struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); asoc_simple_card_convert_fixup(&dai_props->adata, params); @@ -176,9 +176,9 @@ static int asoc_graph_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } -static void asoc_graph_card_get_conversion(struct device *dev, - struct device_node *ep, - struct asoc_simple_card_data *adata) +static void graph_get_conversion(struct device *dev, + struct device_node *ep, + struct asoc_simple_card_data *adata) { struct device_node *top = dev->of_node; struct device_node *port = of_get_parent(ep); @@ -192,11 +192,11 @@ static void asoc_graph_card_get_conversion(struct device *dev, asoc_simple_card_parse_convert(dev, ep, NULL, adata); } -static int asoc_graph_card_dai_link_of_dpcm(struct graph_card_data *priv, - struct device_node *cpu_ep, - struct device_node *codec_ep, - struct link_info *li, - int dup_codec) +static int graph_dai_link_of_dpcm(struct graph_priv *priv, + struct device_node *cpu_ep, + struct device_node *codec_ep, + struct link_info *li, + int dup_codec) { struct device *dev = graph_priv_to_dev(priv); struct snd_soc_dai_link *dai_link = graph_priv_to_link(priv, li->link); @@ -227,7 +227,7 @@ static int asoc_graph_card_dai_link_of_dpcm(struct graph_card_data *priv, of_property_read_u32(port, "mclk-fs", &dai_props->mclk_fs); of_property_read_u32(ep, "mclk-fs", &dai_props->mclk_fs); - asoc_graph_card_get_conversion(dev, ep, &dai_props->adata); + graph_get_conversion(dev, ep, &dai_props->adata); of_node_put(ports); of_node_put(port); @@ -274,7 +274,7 @@ static int asoc_graph_card_dai_link_of_dpcm(struct graph_card_data *priv, /* BE settings */ dai_link->no_pcm = 1; - dai_link->be_hw_params_fixup = asoc_graph_card_be_hw_params_fixup; + dai_link->be_hw_params_fixup = graph_be_hw_params_fixup; dai = dai_props->codec_dai = &priv->dais[li->dais++]; @@ -322,24 +322,24 @@ static int asoc_graph_card_dai_link_of_dpcm(struct graph_card_data *priv, dai_link->dpcm_playback = 1; dai_link->dpcm_capture = 1; - dai_link->ops = &asoc_graph_card_ops; - dai_link->init = asoc_graph_card_dai_init; + dai_link->ops = &graph_ops; + dai_link->init = graph_dai_init; return 0; } -static int asoc_graph_card_dai_link_of(struct graph_card_data *priv, - struct device_node *cpu_ep, - struct device_node *codec_ep, - struct link_info *li) +static int graph_dai_link_of(struct graph_priv *priv, + struct device_node *cpu_ep, + struct device_node *codec_ep, + struct link_info *li) { struct device *dev = graph_priv_to_dev(priv); struct snd_soc_dai_link *dai_link = graph_priv_to_link(priv, li->link); struct graph_dai_props *dai_props = graph_priv_to_props(priv, li->link); struct device_node *top = dev->of_node; struct device_node *cpu_port; - struct device_node *codec_port; struct device_node *cpu_ports; + struct device_node *codec_port; struct device_node *codec_ports; struct asoc_simple_dai *cpu_dai; struct asoc_simple_dai *codec_dai; @@ -416,8 +416,8 @@ static int asoc_graph_card_dai_link_of(struct graph_card_data *priv, if (ret < 0) return ret; - dai_link->ops = &asoc_graph_card_ops; - dai_link->init = asoc_graph_card_dai_init; + dai_link->ops = &graph_ops; + dai_link->init = graph_dai_init; asoc_simple_card_canonicalize_cpu(dai_link, of_graph_get_endpoint_count(dai_link->cpu_of_node) == 1); @@ -425,13 +425,13 @@ static int asoc_graph_card_dai_link_of(struct graph_card_data *priv, return 0; } -static int asoc_graph_card_for_each_link(struct graph_card_data *priv, +static int graph_for_each_link(struct graph_priv *priv, struct link_info *li, - int (*func_noml)(struct graph_card_data *priv, + int (*func_noml)(struct graph_priv *priv, struct device_node *cpu_ep, struct device_node *codec_ep, struct link_info *li), - int (*func_dpcm)(struct graph_card_data *priv, + int (*func_dpcm)(struct graph_priv *priv, struct device_node *cpu_ep, struct device_node *codec_ep, struct link_info *li, int dup_codec)) @@ -467,8 +467,8 @@ static int asoc_graph_card_for_each_link(struct graph_card_data *priv, /* get convert-xxx property */ memset(&adata, 0, sizeof(adata)); - asoc_graph_card_get_conversion(dev, codec_ep, &adata); - asoc_graph_card_get_conversion(dev, cpu_ep, &adata); + graph_get_conversion(dev, codec_ep, &adata); + graph_get_conversion(dev, cpu_ep, &adata); /* * It is DPCM @@ -493,7 +493,7 @@ static int asoc_graph_card_for_each_link(struct graph_card_data *priv, return 0; } -static int asoc_graph_card_parse_of(struct graph_card_data *priv) +static int graph_parse_of(struct graph_priv *priv) { struct snd_soc_card *card = graph_priv_to_card(priv); struct link_info li; @@ -521,9 +521,9 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) * To avoid random sub-device numbering, * detect "dummy-Codec" in last; */ - ret = asoc_graph_card_for_each_link(priv, &li, - asoc_graph_card_dai_link_of, - asoc_graph_card_dai_link_of_dpcm); + ret = graph_for_each_link(priv, &li, + graph_dai_link_of, + graph_dai_link_of_dpcm); if (ret < 0) return ret; } @@ -531,10 +531,10 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) return asoc_simple_card_parse_card_name(card, NULL); } -static int asoc_graph_card_count_noml(struct graph_card_data *priv, - struct device_node *cpu_ep, - struct device_node *codec_ep, - struct link_info *li) +static int graph_count_noml(struct graph_priv *priv, + struct device_node *cpu_ep, + struct device_node *codec_ep, + struct link_info *li) { struct device *dev = graph_priv_to_dev(priv); @@ -546,11 +546,11 @@ static int asoc_graph_card_count_noml(struct graph_card_data *priv, return 0; } -static int asoc_graph_card_count_dpcm(struct graph_card_data *priv, - struct device_node *cpu_ep, - struct device_node *codec_ep, - struct link_info *li, - int dup_codec) +static int graph_count_dpcm(struct graph_priv *priv, + struct device_node *cpu_ep, + struct device_node *codec_ep, + struct link_info *li, + int dup_codec) { struct device *dev = graph_priv_to_dev(priv); @@ -568,8 +568,8 @@ static int asoc_graph_card_count_dpcm(struct graph_card_data *priv, return 0; } -static void asoc_graph_get_dais_count(struct graph_card_data *priv, - struct link_info *li) +static void graph_get_dais_count(struct graph_priv *priv, + struct link_info *li) { struct device *dev = graph_priv_to_dev(priv); @@ -619,16 +619,16 @@ static void asoc_graph_get_dais_count(struct graph_card_data *priv, * => 4 DAIs = 2xCPU + 2xCodec * => 1 ccnf = 1xdummy-Codec */ - asoc_graph_card_for_each_link(priv, li, - asoc_graph_card_count_noml, - asoc_graph_card_count_dpcm); + graph_for_each_link(priv, li, + graph_count_noml, + graph_count_dpcm); dev_dbg(dev, "link %d, dais %d, ccnf %d\n", li->link, li->dais, li->conf); } -static int asoc_graph_soc_card_probe(struct snd_soc_card *card) +static int graph_card_probe(struct snd_soc_card *card) { - struct graph_card_data *priv = snd_soc_card_get_drvdata(card); + struct graph_priv *priv = snd_soc_card_get_drvdata(card); int ret; ret = asoc_simple_card_init_hp(card, &priv->hp_jack, NULL); @@ -642,9 +642,9 @@ static int asoc_graph_soc_card_probe(struct snd_soc_card *card) return 0; } -static int asoc_graph_card_probe(struct platform_device *pdev) +static int graph_probe(struct platform_device *pdev) { - struct graph_card_data *priv; + struct graph_priv *priv; struct snd_soc_dai_link *dai_link; struct graph_dai_props *dai_props; struct asoc_simple_dai *dais; @@ -662,12 +662,12 @@ static int asoc_graph_card_probe(struct platform_device *pdev) card = graph_priv_to_card(priv); card->owner = THIS_MODULE; card->dev = dev; - card->dapm_widgets = asoc_graph_card_dapm_widgets; - card->num_dapm_widgets = ARRAY_SIZE(asoc_graph_card_dapm_widgets); - card->probe = asoc_graph_soc_card_probe; + card->dapm_widgets = graph_dapm_widgets; + card->num_dapm_widgets = ARRAY_SIZE(graph_dapm_widgets); + card->probe = graph_card_probe; memset(&li, 0, sizeof(li)); - asoc_graph_get_dais_count(priv, &li); + graph_get_dais_count(priv, &li); if (!li.link || !li.dais) return -EINVAL; @@ -707,7 +707,7 @@ static int asoc_graph_card_probe(struct platform_device *pdev) card->codec_conf = cconf; card->num_configs = li.conf; - ret = asoc_graph_card_parse_of(priv); + ret = graph_parse_of(priv); if (ret < 0) { if (ret != -EPROBE_DEFER) dev_err(dev, "parse error %d\n", ret); @@ -727,30 +727,30 @@ static int asoc_graph_card_probe(struct platform_device *pdev) return ret; } -static int asoc_graph_card_remove(struct platform_device *pdev) +static int graph_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); return asoc_simple_card_clean_reference(card); } -static const struct of_device_id asoc_graph_of_match[] = { +static const struct of_device_id graph_of_match[] = { { .compatible = "audio-graph-card", }, { .compatible = "audio-graph-scu-card", }, {}, }; -MODULE_DEVICE_TABLE(of, asoc_graph_of_match); +MODULE_DEVICE_TABLE(of, graph_of_match); -static struct platform_driver asoc_graph_card = { +static struct platform_driver graph_card = { .driver = { .name = "asoc-audio-graph-card", .pm = &snd_soc_pm_ops, - .of_match_table = asoc_graph_of_match, + .of_match_table = graph_of_match, }, - .probe = asoc_graph_card_probe, - .remove = asoc_graph_card_remove, + .probe = graph_probe, + .remove = graph_remove, }; -module_platform_driver(asoc_graph_card); +module_platform_driver(graph_card); MODULE_ALIAS("platform:asoc-audio-graph-card"); MODULE_LICENSE("GPL v2"); From 7e5e1f8bbaa82e6877d8bc121cb1b44cb1ce7ddf Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 20 Dec 2018 10:46:42 +0900 Subject: [PATCH 40/67] ASoC: simple-card: add asoc_simple_card_get_conversion() simple-card is now supporting normal sound and DPCM sound. For DPCM sound, original sound card (= simple-scu-card) had been supported 1 CPU : 1 Codec connection which uses hw_params_fixup() for convert-rate/channel. But, merged simple-card is completely forgeting about it. To re-support 1 CPU : 1 Codec DPCM for hw_params_fixup(), it need to judge whether it is DPCM by checking convert-rate/channel. For this purpose, this patch adds asoc_simple_card_get_conversion() as preparation Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 19 ++++++++++++++++--- 1 file changed, 16 insertions(+), 3 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 37e001cf9cd1..52048069b25a 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -165,6 +165,21 @@ static int asoc_simple_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } +static void asoc_simple_card_get_conversion(struct device *dev, + struct device_node *np, + struct asoc_simple_card_data *adata) +{ + struct device_node *top = dev->of_node; + struct device_node *node = of_get_parent(np); + + asoc_simple_card_parse_convert(dev, top, PREFIX, adata); + asoc_simple_card_parse_convert(dev, node, PREFIX, adata); + asoc_simple_card_parse_convert(dev, node, NULL, adata); + asoc_simple_card_parse_convert(dev, np, NULL, adata); + + of_node_put(node); +} + static int asoc_simple_card_dai_link_of_dpcm(struct device_node *top, struct device_node *node, struct device_node *np, @@ -260,9 +275,7 @@ static int asoc_simple_card_dai_link_of_dpcm(struct device_node *top, "prefix"); } - asoc_simple_card_parse_convert(dev, top, PREFIX, &dai_props->adata); - asoc_simple_card_parse_convert(dev, node, prefix, &dai_props->adata); - asoc_simple_card_parse_convert(dev, np, NULL, &dai_props->adata); + asoc_simple_card_get_conversion(dev, np, &dai_props->adata); ret = asoc_simple_card_of_parse_tdm(np, dai); if (ret) From 7adee60ee2732f23f703ff83ee35caad561490ba Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 20 Dec 2018 10:46:47 +0900 Subject: [PATCH 41/67] ASoC: simple-card: add 1 CPU : 1 Codec support again simple-card is now supporting normal sound and DPCM sound. For DPCM sound, original sound card (= simple-scu-card) had been supported 1 CPU : 1 Codec connection which uses hw_params_fixup() for convert-rate/channel. But, merged simple-card is completely forgeting about it. This patch re-support it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 32 ++++++++++++++++++++++++++++---- 1 file changed, 28 insertions(+), 4 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 52048069b25a..b15651409c7f 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -453,6 +453,7 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) struct device_node *node; struct device_node *np; struct device_node *codec; + struct asoc_simple_card_data adata; bool is_fe; int ret, loop; int dai_idx, link_idx, conf_idx; @@ -480,8 +481,13 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) } do { + memset(&adata, 0, sizeof(adata)); + for_each_child_of_node(node, np) + asoc_simple_card_get_conversion(dev, np, &adata); + /* DPCM */ - if (of_get_child_count(node) > 2) { + if (of_get_child_count(node) > 2 || + adata.convert_rate || adata.convert_channels) { for_each_child_of_node(node, np) { codec = of_get_child_by_name(node, loop ? "codec" : @@ -495,14 +501,16 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) top, node, np, codec, priv, &dai_idx, link_idx++, &conf_idx, is_fe, !loop); + if (ret < 0) + return ret; } } else { ret = asoc_simple_card_dai_link_of( top, node, priv, &dai_idx, link_idx++, !loop); + if (ret < 0) + return ret; } - if (ret < 0) - return ret; node = of_get_next_child(top, node); } while (loop && node); @@ -523,6 +531,8 @@ static void asoc_simple_card_get_dais_count(struct device *dev, { struct device_node *top = dev->of_node; struct device_node *node; + struct device_node *np; + struct asoc_simple_card_data adata; int loop; int num; @@ -562,6 +572,15 @@ static void asoc_simple_card_get_dais_count(struct device *dev, * => 6 links = 0xCPU-Codec + 4xCPU-dummy + 2xdummy-Codec * => 6 DAIs = 4xCPU + 2xCodec * => 2 ccnf = 2xdummy-Codec + * + * ex4) + * CPU0 --- Codec0 (convert-rate) link : 3 + * CPU1 --- Codec1 dais : 4 + * ccnf : 1 + * + * => 3 links = 1xCPU-Codec + 1xCPU-dummy + 1xdummy-Codec + * => 4 DAIs = 2xCPU + 2xCodec + * => 1 ccnf = 1xdummy-Codec */ if (!top) { (*link_num) = 1; @@ -578,9 +597,14 @@ static void asoc_simple_card_get_dais_count(struct device *dev, } do { + memset(&adata, 0, sizeof(adata)); + for_each_child_of_node(node, np) + asoc_simple_card_get_conversion(dev, np, &adata); + num = of_get_child_count(node); (*dais_num) += num; - if (num > 2) { + if (num > 2 || + adata.convert_rate || adata.convert_channels) { (*link_num) += num; (*ccnf_num)++; } else { From 17029e494edc68337c9b99665e8f9b478f1d4ec5 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 20 Dec 2018 10:46:53 +0900 Subject: [PATCH 42/67] ASoC: simple-card: add link_info Current simple-card is parsing DAI link for both "normal sound" and "DPCM sound". On this driver, it needs to count and parse DAIs/Links/Codec Conf from each links. Then, counting/parsing link loop are very similar, but using different implementation. Because of this background, the link loop code is very mysterious. Mystery code will be trouble in the future. To preparing cleanup code, this patch adds link_info which handles number of DAIs/Links/Codec Conf, and CPU/Codec turn. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 94 ++++++++++++++++++--------------- 1 file changed, 50 insertions(+), 44 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index b15651409c7f..3820ad719059 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -33,6 +33,13 @@ struct simple_card_data { struct snd_soc_codec_conf *codec_conf; }; +struct link_info { + int dais; /* number of dai */ + int link; /* number of link */ + int conf; /* number of codec_conf */ + int cpu; /* turn for CPU / Codec */ +}; + #define simple_priv_to_card(priv) (&(priv)->snd_card) #define simple_priv_to_props(priv, i) ((priv)->dai_props + (i)) #define simple_priv_to_dev(priv) (simple_priv_to_card(priv)->dev) @@ -185,25 +192,27 @@ static int asoc_simple_card_dai_link_of_dpcm(struct device_node *top, struct device_node *np, struct device_node *codec, struct simple_card_data *priv, - int *dai_idx, int link_idx, - int *conf_idx, int is_fe, + struct link_info *li, bool is_top_level_node) { struct device *dev = simple_priv_to_dev(priv); - struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, link_idx); - struct simple_dai_props *dai_props = simple_priv_to_props(priv, link_idx); + struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); + struct simple_dai_props *dai_props = simple_priv_to_props(priv, li->link); struct asoc_simple_dai *dai; struct snd_soc_dai_link_component *codecs = dai_link->codecs; - char prop[128]; char *prefix = ""; int ret; + dev_dbg(dev, "link_of DPCM (%pOF)\n", np); + + li->link++; + /* For single DAI link & old style of DT node */ if (is_top_level_node) prefix = PREFIX; - if (is_fe) { + if (np != codec) { int is_single_links = 0; /* BE is dummy */ @@ -216,7 +225,7 @@ static int asoc_simple_card_dai_link_of_dpcm(struct device_node *top, dai_link->dpcm_merged_format = 1; dai = - dai_props->cpu_dai = &priv->dais[(*dai_idx)++]; + dai_props->cpu_dai = &priv->dais[li->dais++]; ret = asoc_simple_card_parse_cpu(np, dai_link, DAI, CELL, &is_single_links); @@ -247,10 +256,10 @@ static int asoc_simple_card_dai_link_of_dpcm(struct device_node *top, dai_link->be_hw_params_fixup = asoc_simple_card_be_hw_params_fixup; dai = - dai_props->codec_dai = &priv->dais[(*dai_idx)++]; + dai_props->codec_dai = &priv->dais[li->dais++]; cconf = - dai_props->codec_conf = &priv->codec_conf[(*conf_idx)++]; + dai_props->codec_conf = &priv->codec_conf[li->conf++]; ret = asoc_simple_card_parse_codec(np, dai_link, DAI, CELL); if (ret < 0) @@ -306,12 +315,12 @@ static int asoc_simple_card_dai_link_of_dpcm(struct device_node *top, static int asoc_simple_card_dai_link_of(struct device_node *top, struct device_node *node, struct simple_card_data *priv, - int *dai_idx, int link_idx, + struct link_info *li, bool is_top_level_node) { struct device *dev = simple_priv_to_dev(priv); - struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, link_idx); - struct simple_dai_props *dai_props = simple_priv_to_props(priv, link_idx); + struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); + struct simple_dai_props *dai_props = simple_priv_to_props(priv, li->link); struct asoc_simple_dai *cpu_dai; struct asoc_simple_dai *codec_dai; struct device_node *cpu = NULL; @@ -321,6 +330,10 @@ static int asoc_simple_card_dai_link_of(struct device_node *top, char *prefix = ""; int ret, single_cpu; + li->link++; + + dev_dbg(dev, "link_of (%pOF)\n", node); + /* For single DAI link & old style of DT node */ if (is_top_level_node) prefix = PREFIX; @@ -347,9 +360,9 @@ static int asoc_simple_card_dai_link_of(struct device_node *top, } cpu_dai = - dai_props->cpu_dai = &priv->dais[(*dai_idx)++]; + dai_props->cpu_dai = &priv->dais[li->dais++]; codec_dai = - dai_props->codec_dai = &priv->dais[(*dai_idx)++]; + dai_props->codec_dai = &priv->dais[li->dais++]; ret = asoc_simple_card_parse_daifmt(dev, node, codec, prefix, &dai_link->dai_fmt); @@ -454,9 +467,8 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) struct device_node *np; struct device_node *codec; struct asoc_simple_card_data adata; - bool is_fe; + struct link_info li; int ret, loop; - int dai_idx, link_idx, conf_idx; if (!top) return -EINVAL; @@ -470,10 +482,8 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) return ret; /* Single/Muti DAI link(s) & New style of DT node */ + memset(&li, 0, sizeof(li)); loop = 1; - link_idx = 0; - dai_idx = 0; - conf_idx = 0; node = of_get_child_by_name(top, PREFIX "dai-link"); if (!node) { node = dev->of_node; @@ -495,19 +505,16 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) if (!codec) return -ENODEV; - is_fe = (np != codec); - ret = asoc_simple_card_dai_link_of_dpcm( top, node, np, codec, priv, - &dai_idx, link_idx++, &conf_idx, - is_fe, !loop); + &li, !loop); if (ret < 0) return ret; } } else { ret = asoc_simple_card_dai_link_of( top, node, priv, - &dai_idx, link_idx++, !loop); + &li, !loop); if (ret < 0) return ret; } @@ -525,9 +532,7 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) } static void asoc_simple_card_get_dais_count(struct device *dev, - int *link_num, - int *dais_num, - int *ccnf_num) + struct link_info *li) { struct device_node *top = dev->of_node; struct device_node *node; @@ -583,9 +588,9 @@ static void asoc_simple_card_get_dais_count(struct device *dev, * => 1 ccnf = 1xdummy-Codec */ if (!top) { - (*link_num) = 1; - (*dais_num) = 2; - (*ccnf_num) = 0; + li->link = 1; + li->dais = 2; + li->conf = 0; return; } @@ -602,13 +607,13 @@ static void asoc_simple_card_get_dais_count(struct device *dev, asoc_simple_card_get_conversion(dev, np, &adata); num = of_get_child_count(node); - (*dais_num) += num; + li->dais += num; if (num > 2 || adata.convert_rate || adata.convert_channels) { - (*link_num) += num; - (*ccnf_num)++; + li->link += num; + li->conf++; } else { - (*link_num)++; + li->link++; } node = of_get_next_child(top, node); } while (loop && node); @@ -640,7 +645,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) struct device_node *np = dev->of_node; struct snd_soc_card *card; struct snd_soc_codec_conf *cconf; - int lnum = 0, dnum = 0, cnum = 0; + struct link_info li; int ret, i; /* Allocate the private data and the DAI link array */ @@ -648,14 +653,15 @@ static int asoc_simple_card_probe(struct platform_device *pdev) if (!priv) return -ENOMEM; - asoc_simple_card_get_dais_count(dev, &lnum, &dnum, &cnum); - if (!lnum || !dnum) + memset(&li, 0, sizeof(li)); + asoc_simple_card_get_dais_count(dev, &li); + if (!li.link || !li.dais) return -EINVAL; - dai_props = devm_kcalloc(dev, lnum, sizeof(*dai_props), GFP_KERNEL); - dai_link = devm_kcalloc(dev, lnum, sizeof(*dai_link), GFP_KERNEL); - dais = devm_kcalloc(dev, dnum, sizeof(*dais), GFP_KERNEL); - cconf = devm_kcalloc(dev, cnum, sizeof(*cconf), GFP_KERNEL); + dai_props = devm_kcalloc(dev, li.link, sizeof(*dai_props), GFP_KERNEL); + dai_link = devm_kcalloc(dev, li.link, sizeof(*dai_link), GFP_KERNEL); + dais = devm_kcalloc(dev, li.dais, sizeof(*dais), GFP_KERNEL); + cconf = devm_kcalloc(dev, li.conf, sizeof(*cconf), GFP_KERNEL); if (!dai_props || !dai_link || !dais) return -ENOMEM; @@ -665,7 +671,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) * see * soc-core.c :: snd_soc_init_multicodec() */ - for (i = 0; i < lnum; i++) { + for (i = 0; i < li.link; i++) { dai_link[i].codecs = &dai_props[i].codecs; dai_link[i].num_codecs = 1; dai_link[i].platform = &dai_props[i].platform; @@ -681,9 +687,9 @@ static int asoc_simple_card_probe(struct platform_device *pdev) card->owner = THIS_MODULE; card->dev = dev; card->dai_link = priv->dai_link; - card->num_links = lnum; + card->num_links = li.link; card->codec_conf = cconf; - card->num_configs = cnum; + card->num_configs = li.conf; card->probe = asoc_simple_soc_card_probe; if (np && of_device_is_available(np)) { From d947cdfd4be29c48c6c529c2b5ce7b1988387c67 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 20 Dec 2018 10:47:23 +0900 Subject: [PATCH 43/67] ASoC: simple-card: cleanup DAI link loop method - step1 Current simple-card is parsing DAI link for both "normal sound" and "DPCM sound". On this driver, it needs to count and parse DAIs/Links/Codec Conf from each links. Then, counting/parsing link loop are very similar, but using different implementation. Because of this background, the link loop code is very mysterious. Mystery code will be trouble in the future. This patch adds/modifies counting and parsing function for "normal sound" and "DPCM sound", and call it from link loop. This is prepare for cleanup DAI link loop method. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 208 +++++++++++++++++++++----------- 1 file changed, 136 insertions(+), 72 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 3820ad719059..4987db667165 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -187,32 +187,43 @@ static void asoc_simple_card_get_conversion(struct device *dev, of_node_put(node); } -static int asoc_simple_card_dai_link_of_dpcm(struct device_node *top, - struct device_node *node, +static int asoc_simple_card_dai_link_of_dpcm(struct simple_card_data *priv, struct device_node *np, struct device_node *codec, - struct simple_card_data *priv, struct link_info *li, - bool is_top_level_node) + bool is_top) { struct device *dev = simple_priv_to_dev(priv); struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); struct simple_dai_props *dai_props = simple_priv_to_props(priv, li->link); struct asoc_simple_dai *dai; struct snd_soc_dai_link_component *codecs = dai_link->codecs; + struct device_node *top = dev->of_node; + struct device_node *node = of_get_parent(np); char prop[128]; char *prefix = ""; int ret; + /* + * |CPU |Codec : turn + * CPU |Pass |return + * Codec |return|Pass + * np + */ + if (li->cpu == (np == codec)) + return 0; + dev_dbg(dev, "link_of DPCM (%pOF)\n", np); li->link++; + of_node_put(node); + /* For single DAI link & old style of DT node */ - if (is_top_level_node) + if (is_top) prefix = PREFIX; - if (np != codec) { + if (li->cpu) { int is_single_links = 0; /* BE is dummy */ @@ -312,53 +323,47 @@ static int asoc_simple_card_dai_link_of_dpcm(struct device_node *top, return 0; } -static int asoc_simple_card_dai_link_of(struct device_node *top, - struct device_node *node, - struct simple_card_data *priv, +static int asoc_simple_card_dai_link_of(struct simple_card_data *priv, + struct device_node *np, + struct device_node *codec, struct link_info *li, - bool is_top_level_node) + bool is_top) { struct device *dev = simple_priv_to_dev(priv); struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); struct simple_dai_props *dai_props = simple_priv_to_props(priv, li->link); struct asoc_simple_dai *cpu_dai; struct asoc_simple_dai *codec_dai; + struct device_node *top = dev->of_node; struct device_node *cpu = NULL; + struct device_node *node = NULL; struct device_node *plat = NULL; - struct device_node *codec = NULL; char prop[128]; char *prefix = ""; int ret, single_cpu; + /* + * |CPU |Codec : turn + * CPU |Pass |return + * Codec |return|return + * np + */ + if (!li->cpu || np == codec) + return 0; + + cpu = np; + node = of_get_parent(np); li->link++; dev_dbg(dev, "link_of (%pOF)\n", node); /* For single DAI link & old style of DT node */ - if (is_top_level_node) + if (is_top) prefix = PREFIX; - snprintf(prop, sizeof(prop), "%scpu", prefix); - cpu = of_get_child_by_name(node, prop); - - if (!cpu) { - ret = -EINVAL; - dev_err(dev, "%s: Can't find %s DT node\n", __func__, prop); - goto dai_link_of_err; - } - snprintf(prop, sizeof(prop), "%splat", prefix); plat = of_get_child_by_name(node, prop); - snprintf(prop, sizeof(prop), "%scodec", prefix); - codec = of_get_child_by_name(node, prop); - - if (!codec) { - ret = -EINVAL; - dev_err(dev, "%s: Can't find %s DT node\n", __func__, prop); - goto dai_link_of_err; - } - cpu_dai = dai_props->cpu_dai = &priv->dais[li->dais++]; codec_dai = @@ -421,8 +426,7 @@ static int asoc_simple_card_dai_link_of(struct device_node *top, asoc_simple_card_canonicalize_cpu(dai_link, single_cpu); dai_link_of_err: - of_node_put(cpu); - of_node_put(codec); + of_node_put(node); return ret; } @@ -464,9 +468,6 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) struct device_node *top = dev->of_node; struct snd_soc_card *card = simple_priv_to_card(priv); struct device_node *node; - struct device_node *np; - struct device_node *codec; - struct asoc_simple_card_data adata; struct link_info li; int ret, loop; @@ -483,6 +484,10 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) /* Single/Muti DAI link(s) & New style of DT node */ memset(&li, 0, sizeof(li)); + + /* FIXME */ + li.cpu = 1; +parse_loop: loop = 1; node = of_get_child_by_name(top, PREFIX "dai-link"); if (!node) { @@ -491,37 +496,59 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) } do { + struct asoc_simple_card_data adata; + struct device_node *codec; + struct device_node *np; + int num = of_get_child_count(node); + int ret; + + codec = of_get_child_by_name(node, !loop ? + PREFIX "codec" : "codec"); + if (!codec) + return -ENODEV; + + of_node_put(codec); + memset(&adata, 0, sizeof(adata)); for_each_child_of_node(node, np) asoc_simple_card_get_conversion(dev, np, &adata); - /* DPCM */ - if (of_get_child_count(node) > 2 || - adata.convert_rate || adata.convert_channels) { - for_each_child_of_node(node, np) { - codec = of_get_child_by_name(node, - loop ? "codec" : - PREFIX "codec"); - if (!codec) - return -ENODEV; + /* + * Detect all CPU first, and Detect all Codec 2nd. + * + * In Normal sound case, all DAIs are detected + * as "CPU-Codec". + * + * In DPCM sound case, + * all CPUs are detected as "CPU-dummy", and + * all Codecs are detected as "dummy-Codec". + * To avoid random sub-device numbering, + * detect "dummy-Codec" in last; + */ + /* loop for all CPU/Codec node */ + for_each_child_of_node(node, np) { + if (num > 2 || + adata.convert_rate || adata.convert_channels) { ret = asoc_simple_card_dai_link_of_dpcm( - top, node, np, codec, priv, - &li, !loop); + priv, np, codec, &li, !loop); + if (ret < 0) + return ret; + } else { + ret = asoc_simple_card_dai_link_of( + priv, np, codec, &li, !loop); if (ret < 0) return ret; } - } else { - ret = asoc_simple_card_dai_link_of( - top, node, priv, - &li, !loop); - if (ret < 0) - return ret; } - node = of_get_next_child(top, node); } while (loop && node); + /* FIXME */ + li.cpu--; + if (li.cpu >= 0) + goto parse_loop; + ret = asoc_simple_card_parse_card_name(card, PREFIX); if (ret < 0) return ret; @@ -531,12 +558,39 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) return ret; } -static void asoc_simple_card_get_dais_count(struct device *dev, +static int asoc_simple_card_count_noml(struct simple_card_data *priv, + struct device_node *np, + struct device_node *codec, + struct link_info *li, bool is_top) +{ + li->dais++; /* CPU or Codec */ + if (np != codec) + li->link++; /* CPU-Codec */ + + return 0; +} + +static int asoc_simple_card_count_dpcm(struct simple_card_data *priv, + struct device_node *np, + struct device_node *codec, + struct link_info *li, bool is_top) +{ + li->dais++; /* CPU or Codec */ + li->link++; /* CPU-dummy or dummy-Codec */ + if (np == codec) + li->conf++; + + return 0; +} + +static void asoc_simple_card_get_dais_count(struct simple_card_data *priv, struct link_info *li) { + struct device *dev = simple_priv_to_dev(priv); struct device_node *top = dev->of_node; struct device_node *node; struct device_node *np; + struct device_node *codec; struct asoc_simple_card_data adata; int loop; int num; @@ -602,18 +656,28 @@ static void asoc_simple_card_get_dais_count(struct device *dev, } do { + num = of_get_child_count(node); + + codec = of_get_child_by_name(node, !loop ? + PREFIX "codec" : "codec"); + if (!codec) + return; + + of_node_put(codec); + memset(&adata, 0, sizeof(adata)); for_each_child_of_node(node, np) asoc_simple_card_get_conversion(dev, np, &adata); - num = of_get_child_count(node); - li->dais += num; - if (num > 2 || - adata.convert_rate || adata.convert_channels) { - li->link += num; - li->conf++; - } else { - li->link++; + for_each_child_of_node(node, np) { + if (num > 2 || + adata.convert_rate || adata.convert_channels) { + asoc_simple_card_count_dpcm(priv, np, codec, + li, !loop); + } else { + asoc_simple_card_count_noml(priv, np, codec, + li, !loop); + } } node = of_get_next_child(top, node); } while (loop && node); @@ -653,8 +717,13 @@ static int asoc_simple_card_probe(struct platform_device *pdev) if (!priv) return -ENOMEM; + card = simple_priv_to_card(priv); + card->owner = THIS_MODULE; + card->dev = dev; + card->probe = asoc_simple_soc_card_probe; + memset(&li, 0, sizeof(li)); - asoc_simple_card_get_dais_count(dev, &li); + asoc_simple_card_get_dais_count(priv, &li); if (!li.link || !li.dais) return -EINVAL; @@ -677,20 +746,15 @@ static int asoc_simple_card_probe(struct platform_device *pdev) dai_link[i].platform = &dai_props[i].platform; } - priv->dai_props = dai_props; - priv->dai_link = dai_link; - priv->dais = dais; - priv->codec_conf = cconf; + priv->dai_props = dai_props; + priv->dai_link = dai_link; + priv->dais = dais; + priv->codec_conf = cconf; - /* Init snd_soc_card */ - card = simple_priv_to_card(priv); - card->owner = THIS_MODULE; - card->dev = dev; card->dai_link = priv->dai_link; card->num_links = li.link; card->codec_conf = cconf; card->num_configs = li.conf; - card->probe = asoc_simple_soc_card_probe; if (np && of_device_is_available(np)) { From c39291a76444e3177f7a89d603eae7f83fbdb9f9 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 20 Dec 2018 10:47:28 +0900 Subject: [PATCH 44/67] ASoC: simple-card: cleanup DAI link loop method - step2 Current simple-card is parsing DAI link for both "normal sound" and "DPCM sound". On this driver, it needs to count and parse DAIs/Links/Codec Conf from each links. Then, counting/parsing link loop are very similar, but using different implementation. Because of this background, the link loop code is very mysterious. Mystery code will be trouble in the future. This patch cleanups the code by using asoc_simple_card_for_each_link() which judges normal link / DPCM link. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 174 +++++++++++++++----------------- 1 file changed, 81 insertions(+), 93 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 4987db667165..e796b1516b40 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -431,6 +431,74 @@ static int asoc_simple_card_dai_link_of(struct simple_card_data *priv, return ret; } +static int asoc_simple_card_for_each_link(struct simple_card_data *priv, + struct link_info *li, + int (*func_noml)(struct simple_card_data *priv, + struct device_node *np, + struct device_node *codec, + struct link_info *li, bool is_top), + int (*func_dpcm)(struct simple_card_data *priv, + struct device_node *np, + struct device_node *codec, + struct link_info *li, bool is_top)) +{ + struct device *dev = simple_priv_to_dev(priv); + struct device_node *top = dev->of_node; + struct device_node *node; + bool is_top = 0; + + /* Check if it has dai-link */ + node = of_get_child_by_name(top, PREFIX "dai-link"); + if (!node) { + node = top; + is_top = 1; + } + + /* loop for all dai-link */ + do { + struct asoc_simple_card_data adata; + struct device_node *codec; + struct device_node *np; + int num = of_get_child_count(node); + int ret; + + /* get codec */ + codec = of_get_child_by_name(node, is_top ? + PREFIX "codec" : "codec"); + if (!codec) + return -ENODEV; + + of_node_put(codec); + + /* get convert-xxx property */ + memset(&adata, 0, sizeof(adata)); + for_each_child_of_node(node, np) + asoc_simple_card_get_conversion(dev, np, &adata); + + /* loop for all CPU/Codec node */ + for_each_child_of_node(node, np) { + /* + * It is DPCM + * if it has many CPUs, + * or has convert-xxx property + */ + if (num > 2 || + adata.convert_rate || adata.convert_channels) + ret = func_dpcm(priv, np, codec, li, is_top); + /* else normal sound */ + else + ret = func_noml(priv, np, codec, li, is_top); + + if (ret < 0) + return ret; + } + + node = of_get_next_child(top, node); + } while (!is_top && node); + + return 0; +} + static int asoc_simple_card_parse_aux_devs(struct device_node *node, struct simple_card_data *priv) { @@ -467,9 +535,8 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) struct device *dev = simple_priv_to_dev(priv); struct device_node *top = dev->of_node; struct snd_soc_card *card = simple_priv_to_card(priv); - struct device_node *node; struct link_info li; - int ret, loop; + int ret; if (!top) return -EINVAL; @@ -484,35 +551,7 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) /* Single/Muti DAI link(s) & New style of DT node */ memset(&li, 0, sizeof(li)); - - /* FIXME */ - li.cpu = 1; -parse_loop: - loop = 1; - node = of_get_child_by_name(top, PREFIX "dai-link"); - if (!node) { - node = dev->of_node; - loop = 0; - } - - do { - struct asoc_simple_card_data adata; - struct device_node *codec; - struct device_node *np; - int num = of_get_child_count(node); - int ret; - - codec = of_get_child_by_name(node, !loop ? - PREFIX "codec" : "codec"); - if (!codec) - return -ENODEV; - - of_node_put(codec); - - memset(&adata, 0, sizeof(adata)); - for_each_child_of_node(node, np) - asoc_simple_card_get_conversion(dev, np, &adata); - + for (li.cpu = 1; li.cpu >= 0; li.cpu--) { /* * Detect all CPU first, and Detect all Codec 2nd. * @@ -525,29 +564,12 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) * To avoid random sub-device numbering, * detect "dummy-Codec" in last; */ - - /* loop for all CPU/Codec node */ - for_each_child_of_node(node, np) { - if (num > 2 || - adata.convert_rate || adata.convert_channels) { - ret = asoc_simple_card_dai_link_of_dpcm( - priv, np, codec, &li, !loop); - if (ret < 0) - return ret; - } else { - ret = asoc_simple_card_dai_link_of( - priv, np, codec, &li, !loop); - if (ret < 0) - return ret; - } - } - node = of_get_next_child(top, node); - } while (loop && node); - - /* FIXME */ - li.cpu--; - if (li.cpu >= 0) - goto parse_loop; + ret = asoc_simple_card_for_each_link(priv, &li, + asoc_simple_card_dai_link_of, + asoc_simple_card_dai_link_of_dpcm); + if (ret < 0) + return ret; + } ret = asoc_simple_card_parse_card_name(card, PREFIX); if (ret < 0) @@ -588,12 +610,6 @@ static void asoc_simple_card_get_dais_count(struct simple_card_data *priv, { struct device *dev = simple_priv_to_dev(priv); struct device_node *top = dev->of_node; - struct device_node *node; - struct device_node *np; - struct device_node *codec; - struct asoc_simple_card_data adata; - int loop; - int num; /* * link_num : number of links. @@ -648,39 +664,11 @@ static void asoc_simple_card_get_dais_count(struct simple_card_data *priv, return; } - loop = 1; - node = of_get_child_by_name(top, PREFIX "dai-link"); - if (!node) { - node = top; - loop = 0; - } - - do { - num = of_get_child_count(node); - - codec = of_get_child_by_name(node, !loop ? - PREFIX "codec" : "codec"); - if (!codec) - return; - - of_node_put(codec); - - memset(&adata, 0, sizeof(adata)); - for_each_child_of_node(node, np) - asoc_simple_card_get_conversion(dev, np, &adata); - - for_each_child_of_node(node, np) { - if (num > 2 || - adata.convert_rate || adata.convert_channels) { - asoc_simple_card_count_dpcm(priv, np, codec, - li, !loop); - } else { - asoc_simple_card_count_noml(priv, np, codec, - li, !loop); - } - } - node = of_get_next_child(top, node); - } while (loop && node); + asoc_simple_card_for_each_link(priv, li, + asoc_simple_card_count_noml, + asoc_simple_card_count_dpcm); + dev_dbg(dev, "link %d, dais %d, ccnf %d\n", + li->link, li->dais, li->conf); } static int asoc_simple_soc_card_probe(struct snd_soc_card *card) From 2d01a84605a55cf07ea9c6886049cc85c5e98454 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 20 Dec 2018 10:47:34 +0900 Subject: [PATCH 45/67] ASoC: simple-card: reduce naming prefix Current simple-card is using asoc_simple_card_xxx() for function / data naming. Because of this long prefix, it is easy to be 80 character over. Let's reduce prefix from asoc_simple_card_xxx() to simple_xxx(). Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 157 ++++++++++++++++---------------- 1 file changed, 79 insertions(+), 78 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index e796b1516b40..479de236e694 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -15,7 +15,7 @@ #include #include -struct simple_card_data { +struct simple_priv { struct snd_soc_card snd_card; struct simple_dai_props { struct asoc_simple_dai *cpu_dai; @@ -49,10 +49,10 @@ struct link_info { #define CELL "#sound-dai-cells" #define PREFIX "simple-audio-card," -static int asoc_simple_card_startup(struct snd_pcm_substream *substream) +static int simple_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); + struct simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); int ret; @@ -68,10 +68,10 @@ static int asoc_simple_card_startup(struct snd_pcm_substream *substream) return ret; } -static void asoc_simple_card_shutdown(struct snd_pcm_substream *substream) +static void simple_shutdown(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); + struct simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); @@ -80,8 +80,8 @@ static void asoc_simple_card_shutdown(struct snd_pcm_substream *substream) asoc_simple_card_clk_disable(dai_props->codec_dai); } -static int asoc_simple_set_clk_rate(struct asoc_simple_dai *simple_dai, - unsigned long rate) +static int simple_set_clk_rate(struct asoc_simple_dai *simple_dai, + unsigned long rate) { if (!simple_dai) return 0; @@ -95,13 +95,13 @@ static int asoc_simple_set_clk_rate(struct asoc_simple_dai *simple_dai, return clk_set_rate(simple_dai->clk, rate); } -static int asoc_simple_card_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) +static int simple_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); + struct simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); unsigned int mclk, mclk_fs = 0; @@ -113,11 +113,11 @@ static int asoc_simple_card_hw_params(struct snd_pcm_substream *substream, if (mclk_fs) { mclk = params_rate(params) * mclk_fs; - ret = asoc_simple_set_clk_rate(dai_props->codec_dai, mclk); + ret = simple_set_clk_rate(dai_props->codec_dai, mclk); if (ret < 0) return ret; - ret = asoc_simple_set_clk_rate(dai_props->cpu_dai, mclk); + ret = simple_set_clk_rate(dai_props->cpu_dai, mclk); if (ret < 0) return ret; @@ -136,15 +136,15 @@ static int asoc_simple_card_hw_params(struct snd_pcm_substream *substream, return ret; } -static const struct snd_soc_ops asoc_simple_card_ops = { - .startup = asoc_simple_card_startup, - .shutdown = asoc_simple_card_shutdown, - .hw_params = asoc_simple_card_hw_params, +static const struct snd_soc_ops simple_ops = { + .startup = simple_startup, + .shutdown = simple_shutdown, + .hw_params = simple_hw_params, }; -static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) +static int simple_dai_init(struct snd_soc_pcm_runtime *rtd) { - struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); + struct simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); int ret; @@ -161,10 +161,10 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) return 0; } -static int asoc_simple_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) +static int simple_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) { - struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); + struct simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); asoc_simple_card_convert_fixup(&dai_props->adata, params); @@ -172,9 +172,9 @@ static int asoc_simple_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } -static void asoc_simple_card_get_conversion(struct device *dev, - struct device_node *np, - struct asoc_simple_card_data *adata) +static void simple_get_conversion(struct device *dev, + struct device_node *np, + struct asoc_simple_card_data *adata) { struct device_node *top = dev->of_node; struct device_node *node = of_get_parent(np); @@ -187,11 +187,11 @@ static void asoc_simple_card_get_conversion(struct device *dev, of_node_put(node); } -static int asoc_simple_card_dai_link_of_dpcm(struct simple_card_data *priv, - struct device_node *np, - struct device_node *codec, - struct link_info *li, - bool is_top) +static int simple_dai_link_of_dpcm(struct simple_priv *priv, + struct device_node *np, + struct device_node *codec, + struct link_info *li, + bool is_top) { struct device *dev = simple_priv_to_dev(priv); struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); @@ -264,7 +264,7 @@ static int asoc_simple_card_dai_link_of_dpcm(struct simple_card_data *priv, /* BE settings */ dai_link->no_pcm = 1; - dai_link->be_hw_params_fixup = asoc_simple_card_be_hw_params_fixup; + dai_link->be_hw_params_fixup = simple_be_hw_params_fixup; dai = dai_props->codec_dai = &priv->dais[li->dais++]; @@ -295,7 +295,7 @@ static int asoc_simple_card_dai_link_of_dpcm(struct simple_card_data *priv, "prefix"); } - asoc_simple_card_get_conversion(dev, np, &dai_props->adata); + simple_get_conversion(dev, np, &dai_props->adata); ret = asoc_simple_card_of_parse_tdm(np, dai); if (ret) @@ -317,17 +317,17 @@ static int asoc_simple_card_dai_link_of_dpcm(struct simple_card_data *priv, dai_link->dpcm_playback = 1; dai_link->dpcm_capture = 1; - dai_link->ops = &asoc_simple_card_ops; - dai_link->init = asoc_simple_card_dai_init; + dai_link->ops = &simple_ops; + dai_link->init = simple_dai_init; return 0; } -static int asoc_simple_card_dai_link_of(struct simple_card_data *priv, - struct device_node *np, - struct device_node *codec, - struct link_info *li, - bool is_top) +static int simple_dai_link_of(struct simple_priv *priv, + struct device_node *np, + struct device_node *codec, + struct link_info *li, + bool is_top) { struct device *dev = simple_priv_to_dev(priv); struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); @@ -420,8 +420,8 @@ static int asoc_simple_card_dai_link_of(struct simple_card_data *priv, if (ret < 0) goto dai_link_of_err; - dai_link->ops = &asoc_simple_card_ops; - dai_link->init = asoc_simple_card_dai_init; + dai_link->ops = &simple_ops; + dai_link->init = simple_dai_init; asoc_simple_card_canonicalize_cpu(dai_link, single_cpu); @@ -431,13 +431,13 @@ static int asoc_simple_card_dai_link_of(struct simple_card_data *priv, return ret; } -static int asoc_simple_card_for_each_link(struct simple_card_data *priv, +static int simple_for_each_link(struct simple_priv *priv, struct link_info *li, - int (*func_noml)(struct simple_card_data *priv, + int (*func_noml)(struct simple_priv *priv, struct device_node *np, struct device_node *codec, struct link_info *li, bool is_top), - int (*func_dpcm)(struct simple_card_data *priv, + int (*func_dpcm)(struct simple_priv *priv, struct device_node *np, struct device_node *codec, struct link_info *li, bool is_top)) @@ -473,7 +473,7 @@ static int asoc_simple_card_for_each_link(struct simple_card_data *priv, /* get convert-xxx property */ memset(&adata, 0, sizeof(adata)); for_each_child_of_node(node, np) - asoc_simple_card_get_conversion(dev, np, &adata); + simple_get_conversion(dev, np, &adata); /* loop for all CPU/Codec node */ for_each_child_of_node(node, np) { @@ -499,8 +499,8 @@ static int asoc_simple_card_for_each_link(struct simple_card_data *priv, return 0; } -static int asoc_simple_card_parse_aux_devs(struct device_node *node, - struct simple_card_data *priv) +static int simple_parse_aux_devs(struct device_node *node, + struct simple_priv *priv) { struct device *dev = simple_priv_to_dev(priv); struct device_node *aux_node; @@ -530,7 +530,7 @@ static int asoc_simple_card_parse_aux_devs(struct device_node *node, return 0; } -static int asoc_simple_card_parse_of(struct simple_card_data *priv) +static int simple_parse_of(struct simple_priv *priv) { struct device *dev = simple_priv_to_dev(priv); struct device_node *top = dev->of_node; @@ -564,9 +564,9 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) * To avoid random sub-device numbering, * detect "dummy-Codec" in last; */ - ret = asoc_simple_card_for_each_link(priv, &li, - asoc_simple_card_dai_link_of, - asoc_simple_card_dai_link_of_dpcm); + ret = simple_for_each_link(priv, &li, + simple_dai_link_of, + simple_dai_link_of_dpcm); if (ret < 0) return ret; } @@ -575,15 +575,15 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) if (ret < 0) return ret; - ret = asoc_simple_card_parse_aux_devs(top, priv); + ret = simple_parse_aux_devs(top, priv); return ret; } -static int asoc_simple_card_count_noml(struct simple_card_data *priv, - struct device_node *np, - struct device_node *codec, - struct link_info *li, bool is_top) +static int simple_count_noml(struct simple_priv *priv, + struct device_node *np, + struct device_node *codec, + struct link_info *li, bool is_top) { li->dais++; /* CPU or Codec */ if (np != codec) @@ -592,10 +592,10 @@ static int asoc_simple_card_count_noml(struct simple_card_data *priv, return 0; } -static int asoc_simple_card_count_dpcm(struct simple_card_data *priv, - struct device_node *np, - struct device_node *codec, - struct link_info *li, bool is_top) +static int simple_count_dpcm(struct simple_priv *priv, + struct device_node *np, + struct device_node *codec, + struct link_info *li, bool is_top) { li->dais++; /* CPU or Codec */ li->link++; /* CPU-dummy or dummy-Codec */ @@ -605,8 +605,8 @@ static int asoc_simple_card_count_dpcm(struct simple_card_data *priv, return 0; } -static void asoc_simple_card_get_dais_count(struct simple_card_data *priv, - struct link_info *li) +static void simple_get_dais_count(struct simple_priv *priv, + struct link_info *li) { struct device *dev = simple_priv_to_dev(priv); struct device_node *top = dev->of_node; @@ -664,16 +664,17 @@ static void asoc_simple_card_get_dais_count(struct simple_card_data *priv, return; } - asoc_simple_card_for_each_link(priv, li, - asoc_simple_card_count_noml, - asoc_simple_card_count_dpcm); + simple_for_each_link(priv, li, + simple_count_noml, + simple_count_dpcm); + dev_dbg(dev, "link %d, dais %d, ccnf %d\n", li->link, li->dais, li->conf); } -static int asoc_simple_soc_card_probe(struct snd_soc_card *card) +static int simple_soc_probe(struct snd_soc_card *card) { - struct simple_card_data *priv = snd_soc_card_get_drvdata(card); + struct simple_priv *priv = snd_soc_card_get_drvdata(card); int ret; ret = asoc_simple_card_init_hp(card, &priv->hp_jack, PREFIX); @@ -687,9 +688,9 @@ static int asoc_simple_soc_card_probe(struct snd_soc_card *card) return 0; } -static int asoc_simple_card_probe(struct platform_device *pdev) +static int simple_probe(struct platform_device *pdev) { - struct simple_card_data *priv; + struct simple_priv *priv; struct snd_soc_dai_link *dai_link; struct simple_dai_props *dai_props; struct asoc_simple_dai *dais; @@ -708,10 +709,10 @@ static int asoc_simple_card_probe(struct platform_device *pdev) card = simple_priv_to_card(priv); card->owner = THIS_MODULE; card->dev = dev; - card->probe = asoc_simple_soc_card_probe; + card->probe = simple_soc_probe; memset(&li, 0, sizeof(li)); - asoc_simple_card_get_dais_count(priv, &li); + simple_get_dais_count(priv, &li); if (!li.link || !li.dais) return -EINVAL; @@ -746,7 +747,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) if (np && of_device_is_available(np)) { - ret = asoc_simple_card_parse_of(priv); + ret = simple_parse_of(priv); if (ret < 0) { if (ret != -EPROBE_DEFER) dev_err(dev, "parse error %d\n", ret); @@ -789,7 +790,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) dai_link->stream_name = cinfo->name; dai_link->cpu_dai_name = cinfo->cpu_dai.name; dai_link->dai_fmt = cinfo->daifmt; - dai_link->init = asoc_simple_card_dai_init; + dai_link->init = simple_dai_init; memcpy(priv->dai_props->cpu_dai, &cinfo->cpu_dai, sizeof(*priv->dai_props->cpu_dai)); memcpy(priv->dai_props->codec_dai, &cinfo->codec_dai, @@ -809,28 +810,28 @@ static int asoc_simple_card_probe(struct platform_device *pdev) return ret; } -static int asoc_simple_card_remove(struct platform_device *pdev) +static int simple_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); return asoc_simple_card_clean_reference(card); } -static const struct of_device_id asoc_simple_of_match[] = { +static const struct of_device_id simple_of_match[] = { { .compatible = "simple-audio-card", }, { .compatible = "simple-scu-audio-card", }, {}, }; -MODULE_DEVICE_TABLE(of, asoc_simple_of_match); +MODULE_DEVICE_TABLE(of, simple_of_match); static struct platform_driver asoc_simple_card = { .driver = { .name = "asoc-simple-card", .pm = &snd_soc_pm_ops, - .of_match_table = asoc_simple_of_match, + .of_match_table = simple_of_match, }, - .probe = asoc_simple_card_probe, - .remove = asoc_simple_card_remove, + .probe = simple_probe, + .remove = simple_remove, }; module_platform_driver(asoc_simple_card); From c32759035ad246d3e4c65d23a07f9e6ba32caeaf Mon Sep 17 00:00:00 2001 From: Katsuhiro Suzuki Date: Fri, 21 Dec 2018 00:36:35 +0900 Subject: [PATCH 46/67] ASoC: rockchip: support ACODEC for rk3328 This patch adds support for audio CODEC core of rk3328. Rockchip does not publish detail specification of this core but driver source code is opened on their GitHub repository. https://github.com/rockchip-linux/kernel So I ported this code to linux-next and added some trivial fixes. Signed-off-by: Katsuhiro Suzuki Signed-off-by: Mark Brown --- .../bindings/sound/rockchip,rk3328-codec.txt | 23 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rk3328_codec.c | 517 ++++++++++++++++++ sound/soc/codecs/rk3328_codec.h | 210 +++++++ 5 files changed, 757 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt create mode 100644 sound/soc/codecs/rk3328_codec.c create mode 100644 sound/soc/codecs/rk3328_codec.h diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt b/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt new file mode 100644 index 000000000000..2469588c7ccb --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt @@ -0,0 +1,23 @@ +* Rockchip Rk3328 internal codec + +Required properties: + +- compatible: "rockchip,rk3328-codec" +- reg: physical base address of the controller and length of memory mapped + region. +- rockchip,grf: the phandle of the syscon node for GRF register. +- clocks: a list of phandle + clock-specifer pairs, one for each entry in clock-names. +- clock-names: should be "pclk". +- spk-depop-time-ms: speak depop time msec. + +Example for rk3328 internal codec: + +codec: codec@ff410000 { + compatible = "rockchip,rk3328-codec"; + reg = <0x0 0xff410000 0x0 0x1000>; + rockchip,grf = <&grf>; + clocks = <&cru PCLK_ACODEC>; + clock-names = "pclk"; + spk-depop-time-ms = 100; + status = "disabled"; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index d46de3e04ff6..87cb9c51e6f5 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -130,6 +130,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_PCM5102A select SND_SOC_PCM512x_I2C if I2C select SND_SOC_PCM512x_SPI if SPI_MASTER + select SND_SOC_RK3328 select SND_SOC_RT274 if I2C select SND_SOC_RT286 if I2C select SND_SOC_RT298 if I2C @@ -806,6 +807,10 @@ config SND_SOC_PCM512x_SPI select SND_SOC_PCM512x select REGMAP_SPI +config SND_SOC_RK3328 + tristate "Rockchip RK3328 audio CODEC" + select REGMAP_MMIO + config SND_SOC_RL6231 tristate default y if SND_SOC_RT5514=y diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index fbe36e6177b0..9bb3346fab2f 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -133,6 +133,7 @@ snd-soc-pcm5102a-objs := pcm5102a.o snd-soc-pcm512x-objs := pcm512x.o snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o snd-soc-pcm512x-spi-objs := pcm512x-spi.o +snd-soc-rk3328-objs := rk3328_codec.o snd-soc-rl6231-objs := rl6231.o snd-soc-rl6347a-objs := rl6347a.o snd-soc-rt1305-objs := rt1305.o @@ -400,6 +401,7 @@ obj-$(CONFIG_SND_SOC_PCM5102A) += snd-soc-pcm5102a.o obj-$(CONFIG_SND_SOC_PCM512x) += snd-soc-pcm512x.o obj-$(CONFIG_SND_SOC_PCM512x_I2C) += snd-soc-pcm512x-i2c.o obj-$(CONFIG_SND_SOC_PCM512x_SPI) += snd-soc-pcm512x-spi.o +obj-$(CONFIG_SND_SOC_RK3328) += snd-soc-rk3328.o obj-$(CONFIG_SND_SOC_RL6231) += snd-soc-rl6231.o obj-$(CONFIG_SND_SOC_RL6347A) += snd-soc-rl6347a.o obj-$(CONFIG_SND_SOC_RT1305) += snd-soc-rt1305.o diff --git a/sound/soc/codecs/rk3328_codec.c b/sound/soc/codecs/rk3328_codec.c new file mode 100644 index 000000000000..71f3fc2d970c --- /dev/null +++ b/sound/soc/codecs/rk3328_codec.c @@ -0,0 +1,517 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// rk3328 ALSA SoC Audio driver +// +// Copyright (c) 2017, Fuzhou Rockchip Electronics Co., Ltd All rights reserved. + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "rk3328_codec.h" + +/* + * volume setting + * 0: -39dB + * 26: 0dB + * 31: 6dB + * Step: 1.5dB + */ +#define OUT_VOLUME (0x18) +#define RK3328_GRF_SOC_CON2 (0x0408) +#define RK3328_GRF_SOC_CON10 (0x0428) +#define INITIAL_FREQ (11289600) + +struct rk3328_codec_priv { + struct regmap *regmap; + struct regmap *grf; + struct clk *mclk; + struct clk *pclk; + unsigned int sclk; + int spk_depop_time; /* msec */ +}; + +static const struct reg_default rk3328_codec_reg_defaults[] = { + { CODEC_RESET, 0x03 }, + { DAC_INIT_CTRL1, 0x00 }, + { DAC_INIT_CTRL2, 0x50 }, + { DAC_INIT_CTRL3, 0x0e }, + { DAC_PRECHARGE_CTRL, 0x01 }, + { DAC_PWR_CTRL, 0x00 }, + { DAC_CLK_CTRL, 0x00 }, + { HPMIX_CTRL, 0x00 }, + { HPOUT_CTRL, 0x00 }, + { HPOUTL_GAIN_CTRL, 0x00 }, + { HPOUTR_GAIN_CTRL, 0x00 }, + { HPOUT_POP_CTRL, 0x11 }, +}; + +static int rk3328_codec_reset(struct rk3328_codec_priv *rk3328) +{ + regmap_write(rk3328->regmap, CODEC_RESET, 0x00); + mdelay(10); + regmap_write(rk3328->regmap, CODEC_RESET, 0x03); + + return 0; +} + +static int rk3328_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct rk3328_codec_priv *rk3328 = + snd_soc_component_get_drvdata(dai->component); + unsigned int val; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + val = PIN_DIRECTION_IN | DAC_I2S_MODE_SLAVE; + break; + case SND_SOC_DAIFMT_CBM_CFM: + val = PIN_DIRECTION_OUT | DAC_I2S_MODE_MASTER; + break; + default: + return -EINVAL; + } + + regmap_update_bits(rk3328->regmap, DAC_INIT_CTRL1, + PIN_DIRECTION_MASK | DAC_I2S_MODE_MASK, val); + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + val = DAC_MODE_PCM; + break; + case SND_SOC_DAIFMT_I2S: + val = DAC_MODE_I2S; + break; + case SND_SOC_DAIFMT_RIGHT_J: + val = DAC_MODE_RJM; + break; + case SND_SOC_DAIFMT_LEFT_J: + val = DAC_MODE_LJM; + break; + default: + return -EINVAL; + } + + regmap_update_bits(rk3328->regmap, DAC_INIT_CTRL2, + DAC_MODE_MASK, val); + + return 0; +} + +static void rk3328_analog_output(struct rk3328_codec_priv *rk3328, int mute) +{ + unsigned int val = BIT(17); + + if (mute) + val |= BIT(1); + + regmap_write(rk3328->grf, RK3328_GRF_SOC_CON10, val); +} + +static int rk3328_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct rk3328_codec_priv *rk3328 = + snd_soc_component_get_drvdata(dai->component); + unsigned int val; + + if (mute) + val = HPOUTL_MUTE | HPOUTR_MUTE; + else + val = HPOUTL_UNMUTE | HPOUTR_UNMUTE; + + regmap_update_bits(rk3328->regmap, HPOUT_CTRL, + HPOUTL_MUTE_MASK | HPOUTR_MUTE_MASK, val); + + return 0; +} + +static int rk3328_codec_power_on(struct rk3328_codec_priv *rk3328, int wait_ms) +{ + regmap_update_bits(rk3328->regmap, DAC_PRECHARGE_CTRL, + DAC_CHARGE_XCHARGE_MASK, DAC_CHARGE_PRECHARGE); + mdelay(10); + regmap_update_bits(rk3328->regmap, DAC_PRECHARGE_CTRL, + DAC_CHARGE_CURRENT_ALL_MASK, + DAC_CHARGE_CURRENT_ALL_ON); + mdelay(wait_ms); + + return 0; +} + +static int rk3328_codec_power_off(struct rk3328_codec_priv *rk3328, int wait_ms) +{ + regmap_update_bits(rk3328->regmap, DAC_PRECHARGE_CTRL, + DAC_CHARGE_XCHARGE_MASK, DAC_CHARGE_DISCHARGE); + mdelay(10); + regmap_update_bits(rk3328->regmap, DAC_PRECHARGE_CTRL, + DAC_CHARGE_CURRENT_ALL_MASK, + DAC_CHARGE_CURRENT_ALL_ON); + mdelay(wait_ms); + + return 0; +} + +static const struct rk3328_reg_msk_val playback_open_list[] = { + { DAC_PWR_CTRL, DAC_PWR_MASK, DAC_PWR_ON }, + { DAC_PWR_CTRL, DACL_PATH_REFV_MASK | DACR_PATH_REFV_MASK, + DACL_PATH_REFV_ON | DACR_PATH_REFV_ON }, + { DAC_PWR_CTRL, HPOUTL_ZERO_CROSSING_MASK | HPOUTR_ZERO_CROSSING_MASK, + HPOUTL_ZERO_CROSSING_ON | HPOUTR_ZERO_CROSSING_ON }, + { HPOUT_POP_CTRL, HPOUTR_POP_MASK | HPOUTL_POP_MASK, + HPOUTR_POP_WORK | HPOUTL_POP_WORK }, + { HPMIX_CTRL, HPMIXL_MASK | HPMIXR_MASK, HPMIXL_EN | HPMIXR_EN }, + { HPMIX_CTRL, HPMIXL_INIT_MASK | HPMIXR_INIT_MASK, + HPMIXL_INIT_EN | HPMIXR_INIT_EN }, + { HPOUT_CTRL, HPOUTL_MASK | HPOUTR_MASK, HPOUTL_EN | HPOUTR_EN }, + { HPOUT_CTRL, HPOUTL_INIT_MASK | HPOUTR_INIT_MASK, + HPOUTL_INIT_EN | HPOUTR_INIT_EN }, + { DAC_CLK_CTRL, DACL_REFV_MASK | DACR_REFV_MASK, + DACL_REFV_ON | DACR_REFV_ON }, + { DAC_CLK_CTRL, DACL_CLK_MASK | DACR_CLK_MASK, + DACL_CLK_ON | DACR_CLK_ON }, + { DAC_CLK_CTRL, DACL_MASK | DACR_MASK, DACL_ON | DACR_ON }, + { DAC_CLK_CTRL, DACL_INIT_MASK | DACR_INIT_MASK, + DACL_INIT_ON | DACR_INIT_ON }, + { DAC_SELECT, DACL_SELECT_MASK | DACR_SELECT_MASK, + DACL_SELECT | DACR_SELECT }, + { HPMIX_CTRL, HPMIXL_INIT2_MASK | HPMIXR_INIT2_MASK, + HPMIXL_INIT2_EN | HPMIXR_INIT2_EN }, + { HPOUT_CTRL, HPOUTL_MUTE_MASK | HPOUTR_MUTE_MASK, + HPOUTL_UNMUTE | HPOUTR_UNMUTE }, +}; + +static int rk3328_codec_open_playback(struct rk3328_codec_priv *rk3328) +{ + int i; + + regmap_update_bits(rk3328->regmap, DAC_PRECHARGE_CTRL, + DAC_CHARGE_CURRENT_ALL_MASK, + DAC_CHARGE_CURRENT_I); + + for (i = 0; i < ARRAY_SIZE(playback_open_list); i++) { + regmap_update_bits(rk3328->regmap, + playback_open_list[i].reg, + playback_open_list[i].msk, + playback_open_list[i].val); + mdelay(1); + } + + msleep(rk3328->spk_depop_time); + rk3328_analog_output(rk3328, 1); + + regmap_update_bits(rk3328->regmap, HPOUTL_GAIN_CTRL, + HPOUTL_GAIN_MASK, OUT_VOLUME); + regmap_update_bits(rk3328->regmap, HPOUTR_GAIN_CTRL, + HPOUTR_GAIN_MASK, OUT_VOLUME); + + return 0; +} + +static const struct rk3328_reg_msk_val playback_close_list[] = { + { HPMIX_CTRL, HPMIXL_INIT2_MASK | HPMIXR_INIT2_MASK, + HPMIXL_INIT2_DIS | HPMIXR_INIT2_DIS }, + { DAC_SELECT, DACL_SELECT_MASK | DACR_SELECT_MASK, + DACL_UNSELECT | DACR_UNSELECT }, + { HPOUT_CTRL, HPOUTL_MUTE_MASK | HPOUTR_MUTE_MASK, + HPOUTL_MUTE | HPOUTR_MUTE }, + { HPOUT_CTRL, HPOUTL_INIT_MASK | HPOUTR_INIT_MASK, + HPOUTL_INIT_DIS | HPOUTR_INIT_DIS }, + { HPOUT_CTRL, HPOUTL_MASK | HPOUTR_MASK, HPOUTL_DIS | HPOUTR_DIS }, + { HPMIX_CTRL, HPMIXL_MASK | HPMIXR_MASK, HPMIXL_DIS | HPMIXR_DIS }, + { DAC_CLK_CTRL, DACL_MASK | DACR_MASK, DACL_OFF | DACR_OFF }, + { DAC_CLK_CTRL, DACL_CLK_MASK | DACR_CLK_MASK, + DACL_CLK_OFF | DACR_CLK_OFF }, + { DAC_CLK_CTRL, DACL_REFV_MASK | DACR_REFV_MASK, + DACL_REFV_OFF | DACR_REFV_OFF }, + { HPOUT_POP_CTRL, HPOUTR_POP_MASK | HPOUTL_POP_MASK, + HPOUTR_POP_XCHARGE | HPOUTL_POP_XCHARGE }, + { DAC_PWR_CTRL, DACL_PATH_REFV_MASK | DACR_PATH_REFV_MASK, + DACL_PATH_REFV_OFF | DACR_PATH_REFV_OFF }, + { DAC_PWR_CTRL, DAC_PWR_MASK, DAC_PWR_OFF }, + { HPMIX_CTRL, HPMIXL_INIT_MASK | HPMIXR_INIT_MASK, + HPMIXL_INIT_DIS | HPMIXR_INIT_DIS }, + { DAC_CLK_CTRL, DACL_INIT_MASK | DACR_INIT_MASK, + DACL_INIT_OFF | DACR_INIT_OFF }, +}; + +static int rk3328_codec_close_playback(struct rk3328_codec_priv *rk3328) +{ + size_t i; + + rk3328_analog_output(rk3328, 0); + + regmap_update_bits(rk3328->regmap, HPOUTL_GAIN_CTRL, + HPOUTL_GAIN_MASK, 0); + regmap_update_bits(rk3328->regmap, HPOUTR_GAIN_CTRL, + HPOUTR_GAIN_MASK, 0); + + for (i = 0; i < ARRAY_SIZE(playback_close_list); i++) { + regmap_update_bits(rk3328->regmap, + playback_close_list[i].reg, + playback_close_list[i].msk, + playback_close_list[i].val); + mdelay(1); + } + + regmap_update_bits(rk3328->regmap, DAC_PRECHARGE_CTRL, + DAC_CHARGE_CURRENT_ALL_MASK, + DAC_CHARGE_CURRENT_I); + + return 0; +} + +static int rk3328_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct rk3328_codec_priv *rk3328 = + snd_soc_component_get_drvdata(dai->component); + unsigned int val = 0; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + val = DAC_VDL_16BITS; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val = DAC_VDL_20BITS; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val = DAC_VDL_24BITS; + break; + case SNDRV_PCM_FORMAT_S32_LE: + val = DAC_VDL_32BITS; + break; + default: + return -EINVAL; + } + regmap_update_bits(rk3328->regmap, DAC_INIT_CTRL2, DAC_VDL_MASK, val); + + val = DAC_WL_32BITS | DAC_RST_DIS; + regmap_update_bits(rk3328->regmap, DAC_INIT_CTRL3, + DAC_WL_MASK | DAC_RST_MASK, val); + + return 0; +} + +static int rk3328_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct rk3328_codec_priv *rk3328 = + snd_soc_component_get_drvdata(dai->component); + + return rk3328_codec_open_playback(rk3328); +} + +static void rk3328_pcm_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct rk3328_codec_priv *rk3328 = + snd_soc_component_get_drvdata(dai->component); + + rk3328_codec_close_playback(rk3328); +} + +static const struct snd_soc_dai_ops rk3328_dai_ops = { + .hw_params = rk3328_hw_params, + .set_fmt = rk3328_set_dai_fmt, + .digital_mute = rk3328_digital_mute, + .startup = rk3328_pcm_startup, + .shutdown = rk3328_pcm_shutdown, +}; + +static struct snd_soc_dai_driver rk3328_dai[] = { + { + .name = "rk3328-hifi", + .id = RK3328_HIFI, + .playback = { + .stream_name = "HIFI Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = (SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S20_3LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE), + }, + .capture = { + .stream_name = "HIFI Capture", + .channels_min = 2, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = (SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S20_3LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE), + }, + .ops = &rk3328_dai_ops, + }, +}; + +static int rk3328_codec_probe(struct snd_soc_component *component) +{ + struct rk3328_codec_priv *rk3328 = + snd_soc_component_get_drvdata(component); + + rk3328_codec_reset(rk3328); + rk3328_codec_power_on(rk3328, 0); + + return 0; +} + +static void rk3328_codec_remove(struct snd_soc_component *component) +{ + struct rk3328_codec_priv *rk3328 = + snd_soc_component_get_drvdata(component); + + rk3328_codec_close_playback(rk3328); + rk3328_codec_power_off(rk3328, 0); +} + +static const struct snd_soc_component_driver soc_codec_rk3328 = { + .probe = rk3328_codec_probe, + .remove = rk3328_codec_remove, +}; + +static bool rk3328_codec_write_read_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CODEC_RESET: + case DAC_INIT_CTRL1: + case DAC_INIT_CTRL2: + case DAC_INIT_CTRL3: + case DAC_PRECHARGE_CTRL: + case DAC_PWR_CTRL: + case DAC_CLK_CTRL: + case HPMIX_CTRL: + case DAC_SELECT: + case HPOUT_CTRL: + case HPOUTL_GAIN_CTRL: + case HPOUTR_GAIN_CTRL: + case HPOUT_POP_CTRL: + return true; + default: + return false; + } +} + +static bool rk3328_codec_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CODEC_RESET: + return true; + default: + return false; + } +} + +static const struct regmap_config rk3328_codec_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = HPOUT_POP_CTRL, + .writeable_reg = rk3328_codec_write_read_reg, + .readable_reg = rk3328_codec_write_read_reg, + .volatile_reg = rk3328_codec_volatile_reg, + .reg_defaults = rk3328_codec_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(rk3328_codec_reg_defaults), + .cache_type = REGCACHE_FLAT, +}; + +static int rk3328_platform_probe(struct platform_device *pdev) +{ + struct device_node *rk3328_np = pdev->dev.of_node; + struct rk3328_codec_priv *rk3328; + struct resource *res; + struct regmap *grf; + void __iomem *base; + int ret = 0; + + rk3328 = devm_kzalloc(&pdev->dev, sizeof(*rk3328), GFP_KERNEL); + if (!rk3328) + return -ENOMEM; + + grf = syscon_regmap_lookup_by_phandle(rk3328_np, + "rockchip,grf"); + if (IS_ERR(grf)) { + dev_err(&pdev->dev, "missing 'rockchip,grf'\n"); + return PTR_ERR(grf); + } + rk3328->grf = grf; + /* enable i2s_acodec_en */ + regmap_write(grf, RK3328_GRF_SOC_CON2, + (BIT(14) << 16 | BIT(14))); + + ret = of_property_read_u32(rk3328_np, "spk-depop-time-ms", + &rk3328->spk_depop_time); + if (ret < 0) { + dev_info(&pdev->dev, "spk_depop_time use default value.\n"); + rk3328->spk_depop_time = 200; + } + + rk3328_analog_output(rk3328, 0); + + rk3328->mclk = devm_clk_get(&pdev->dev, "mclk"); + if (IS_ERR(rk3328->mclk)) + return PTR_ERR(rk3328->mclk); + + ret = clk_prepare_enable(rk3328->mclk); + if (ret) + return ret; + clk_set_rate(rk3328->mclk, INITIAL_FREQ); + + rk3328->pclk = devm_clk_get(&pdev->dev, "pclk"); + if (IS_ERR(rk3328->pclk)) { + dev_err(&pdev->dev, "can't get acodec pclk\n"); + return PTR_ERR(rk3328->pclk); + } + + ret = clk_prepare_enable(rk3328->pclk); + if (ret < 0) { + dev_err(&pdev->dev, "failed to enable acodec pclk\n"); + return ret; + } + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) + return PTR_ERR(base); + + rk3328->regmap = devm_regmap_init_mmio(&pdev->dev, base, + &rk3328_codec_regmap_config); + if (IS_ERR(rk3328->regmap)) + return PTR_ERR(rk3328->regmap); + + platform_set_drvdata(pdev, rk3328); + + return devm_snd_soc_register_component(&pdev->dev, &soc_codec_rk3328, + rk3328_dai, + ARRAY_SIZE(rk3328_dai)); +} + +static const struct of_device_id rk3328_codec_of_match[] = { + { .compatible = "rockchip,rk3328-codec", }, + {}, +}; +MODULE_DEVICE_TABLE(of, rk3328_codec_of_match); + +static struct platform_driver rk3328_codec_driver = { + .driver = { + .name = "rk3328-codec", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(rk3328_codec_of_match), + }, + .probe = rk3328_platform_probe, +}; +module_platform_driver(rk3328_codec_driver); + +MODULE_AUTHOR("Sugar Zhang "); +MODULE_DESCRIPTION("ASoC rk3328 codec driver"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rk3328_codec.h b/sound/soc/codecs/rk3328_codec.h new file mode 100644 index 000000000000..655103586241 --- /dev/null +++ b/sound/soc/codecs/rk3328_codec.h @@ -0,0 +1,210 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * rk3328 ALSA SoC Audio driver + * + * Copyright (c) 2017, Fuzhou Rockchip Electronics Co., Ltd All rights reserved. + */ + +#ifndef _RK3328_CODEC_H +#define _RK3328_CODEC_H + +#include + +/* codec register */ +#define CODEC_RESET (0x00 << 2) +#define DAC_INIT_CTRL1 (0x03 << 2) +#define DAC_INIT_CTRL2 (0x04 << 2) +#define DAC_INIT_CTRL3 (0x05 << 2) +#define DAC_PRECHARGE_CTRL (0x22 << 2) +#define DAC_PWR_CTRL (0x23 << 2) +#define DAC_CLK_CTRL (0x24 << 2) +#define HPMIX_CTRL (0x25 << 2) +#define DAC_SELECT (0x26 << 2) +#define HPOUT_CTRL (0x27 << 2) +#define HPOUTL_GAIN_CTRL (0x28 << 2) +#define HPOUTR_GAIN_CTRL (0x29 << 2) +#define HPOUT_POP_CTRL (0x2a << 2) + +/* REG00: CODEC_RESET */ +#define PWR_RST_BYPASS_DIS (0x0 << 6) +#define PWR_RST_BYPASS_EN (0x1 << 6) +#define DIG_CORE_RST (0x0 << 1) +#define DIG_CORE_WORK (0x1 << 1) +#define SYS_RST (0x0 << 0) +#define SYS_WORK (0x1 << 0) + +/* REG03: DAC_INIT_CTRL1 */ +#define PIN_DIRECTION_MASK BIT(5) +#define PIN_DIRECTION_IN (0x0 << 5) +#define PIN_DIRECTION_OUT (0x1 << 5) +#define DAC_I2S_MODE_MASK BIT(4) +#define DAC_I2S_MODE_SLAVE (0x0 << 4) +#define DAC_I2S_MODE_MASTER (0x1 << 4) + +/* REG04: DAC_INIT_CTRL2 */ +#define DAC_I2S_LRP_MASK BIT(7) +#define DAC_I2S_LRP_NORMAL (0x0 << 7) +#define DAC_I2S_LRP_REVERSAL (0x1 << 7) +#define DAC_VDL_MASK GENMASK(6, 5) +#define DAC_VDL_16BITS (0x0 << 5) +#define DAC_VDL_20BITS (0x1 << 5) +#define DAC_VDL_24BITS (0x2 << 5) +#define DAC_VDL_32BITS (0x3 << 5) +#define DAC_MODE_MASK GENMASK(4, 3) +#define DAC_MODE_RJM (0x0 << 3) +#define DAC_MODE_LJM (0x1 << 3) +#define DAC_MODE_I2S (0x2 << 3) +#define DAC_MODE_PCM (0x3 << 3) +#define DAC_LR_SWAP_MASK BIT(2) +#define DAC_LR_SWAP_DIS (0x0 << 2) +#define DAC_LR_SWAP_EN (0x1 << 2) + +/* REG05: DAC_INIT_CTRL3 */ +#define DAC_WL_MASK GENMASK(3, 2) +#define DAC_WL_16BITS (0x0 << 2) +#define DAC_WL_20BITS (0x1 << 2) +#define DAC_WL_24BITS (0x2 << 2) +#define DAC_WL_32BITS (0x3 << 2) +#define DAC_RST_MASK BIT(1) +#define DAC_RST_EN (0x0 << 1) +#define DAC_RST_DIS (0x1 << 1) +#define DAC_BCP_MASK BIT(0) +#define DAC_BCP_NORMAL (0x0 << 0) +#define DAC_BCP_REVERSAL (0x1 << 0) + +/* REG22: DAC_PRECHARGE_CTRL */ +#define DAC_CHARGE_XCHARGE_MASK BIT(7) +#define DAC_CHARGE_DISCHARGE (0x0 << 7) +#define DAC_CHARGE_PRECHARGE (0x1 << 7) +#define DAC_CHARGE_CURRENT_64I_MASK BIT(6) +#define DAC_CHARGE_CURRENT_64I (0x1 << 6) +#define DAC_CHARGE_CURRENT_32I_MASK BIT(5) +#define DAC_CHARGE_CURRENT_32I (0x1 << 5) +#define DAC_CHARGE_CURRENT_16I_MASK BIT(4) +#define DAC_CHARGE_CURRENT_16I (0x1 << 4) +#define DAC_CHARGE_CURRENT_08I_MASK BIT(3) +#define DAC_CHARGE_CURRENT_08I (0x1 << 3) +#define DAC_CHARGE_CURRENT_04I_MASK BIT(2) +#define DAC_CHARGE_CURRENT_04I (0x1 << 2) +#define DAC_CHARGE_CURRENT_02I_MASK BIT(1) +#define DAC_CHARGE_CURRENT_02I (0x1 << 1) +#define DAC_CHARGE_CURRENT_I_MASK BIT(0) +#define DAC_CHARGE_CURRENT_I (0x1 << 0) +#define DAC_CHARGE_CURRENT_ALL_MASK GENMASK(6, 0) +#define DAC_CHARGE_CURRENT_ALL_OFF 0x00 +#define DAC_CHARGE_CURRENT_ALL_ON 0x7f + +/* REG23: DAC_PWR_CTRL */ +#define DAC_PWR_MASK BIT(6) +#define DAC_PWR_OFF (0x0 << 6) +#define DAC_PWR_ON (0x1 << 6) +#define DACL_PATH_REFV_MASK BIT(5) +#define DACL_PATH_REFV_OFF (0x0 << 5) +#define DACL_PATH_REFV_ON (0x1 << 5) +#define HPOUTL_ZERO_CROSSING_MASK BIT(4) +#define HPOUTL_ZERO_CROSSING_OFF (0x0 << 4) +#define HPOUTL_ZERO_CROSSING_ON (0x1 << 4) +#define DACR_PATH_REFV_MASK BIT(1) +#define DACR_PATH_REFV_OFF (0x0 << 1) +#define DACR_PATH_REFV_ON (0x1 << 1) +#define HPOUTR_ZERO_CROSSING_MASK BIT(0) +#define HPOUTR_ZERO_CROSSING_OFF (0x0 << 0) +#define HPOUTR_ZERO_CROSSING_ON (0x1 << 0) + +/* REG24: DAC_CLK_CTRL */ +#define DACL_REFV_MASK BIT(7) +#define DACL_REFV_OFF (0x0 << 7) +#define DACL_REFV_ON (0x1 << 7) +#define DACL_CLK_MASK BIT(6) +#define DACL_CLK_OFF (0x0 << 6) +#define DACL_CLK_ON (0x1 << 6) +#define DACL_MASK BIT(5) +#define DACL_OFF (0x0 << 5) +#define DACL_ON (0x1 << 5) +#define DACL_INIT_MASK BIT(4) +#define DACL_INIT_OFF (0x0 << 4) +#define DACL_INIT_ON (0x1 << 4) +#define DACR_REFV_MASK BIT(3) +#define DACR_REFV_OFF (0x0 << 3) +#define DACR_REFV_ON (0x1 << 3) +#define DACR_CLK_MASK BIT(2) +#define DACR_CLK_OFF (0x0 << 2) +#define DACR_CLK_ON (0x1 << 2) +#define DACR_MASK BIT(1) +#define DACR_OFF (0x0 << 1) +#define DACR_ON (0x1 << 1) +#define DACR_INIT_MASK BIT(0) +#define DACR_INIT_OFF (0x0 << 0) +#define DACR_INIT_ON (0x1 << 0) + +/* REG25: HPMIX_CTRL*/ +#define HPMIXL_MASK BIT(6) +#define HPMIXL_DIS (0x0 << 6) +#define HPMIXL_EN (0x1 << 6) +#define HPMIXL_INIT_MASK BIT(5) +#define HPMIXL_INIT_DIS (0x0 << 5) +#define HPMIXL_INIT_EN (0x1 << 5) +#define HPMIXL_INIT2_MASK BIT(4) +#define HPMIXL_INIT2_DIS (0x0 << 4) +#define HPMIXL_INIT2_EN (0x1 << 4) +#define HPMIXR_MASK BIT(2) +#define HPMIXR_DIS (0x0 << 2) +#define HPMIXR_EN (0x1 << 2) +#define HPMIXR_INIT_MASK BIT(1) +#define HPMIXR_INIT_DIS (0x0 << 1) +#define HPMIXR_INIT_EN (0x1 << 1) +#define HPMIXR_INIT2_MASK BIT(0) +#define HPMIXR_INIT2_DIS (0x0 << 0) +#define HPMIXR_INIT2_EN (0x1 << 0) + +/* REG26: DAC_SELECT */ +#define DACL_SELECT_MASK BIT(4) +#define DACL_UNSELECT (0x0 << 4) +#define DACL_SELECT (0x1 << 4) +#define DACR_SELECT_MASK BIT(0) +#define DACR_UNSELECT (0x0 << 0) +#define DACR_SELECT (0x1 << 0) + +/* REG27: HPOUT_CTRL */ +#define HPOUTL_MASK BIT(7) +#define HPOUTL_DIS (0x0 << 7) +#define HPOUTL_EN (0x1 << 7) +#define HPOUTL_INIT_MASK BIT(6) +#define HPOUTL_INIT_DIS (0x0 << 6) +#define HPOUTL_INIT_EN (0x1 << 6) +#define HPOUTL_MUTE_MASK BIT(5) +#define HPOUTL_MUTE (0x0 << 5) +#define HPOUTL_UNMUTE (0x1 << 5) +#define HPOUTR_MASK BIT(4) +#define HPOUTR_DIS (0x0 << 4) +#define HPOUTR_EN (0x1 << 4) +#define HPOUTR_INIT_MASK BIT(3) +#define HPOUTR_INIT_DIS (0x0 << 3) +#define HPOUTR_INIT_EN (0x1 << 3) +#define HPOUTR_MUTE_MASK BIT(2) +#define HPOUTR_MUTE (0x0 << 2) +#define HPOUTR_UNMUTE (0x1 << 2) + +/* REG28: HPOUTL_GAIN_CTRL */ +#define HPOUTL_GAIN_MASK GENMASK(4, 0) + +/* REG29: HPOUTR_GAIN_CTRL */ +#define HPOUTR_GAIN_MASK GENMASK(4, 0) + +/* REG2a: HPOUT_POP_CTRL */ +#define HPOUTR_POP_MASK GENMASK(5, 4) +#define HPOUTR_POP_XCHARGE (0x1 << 4) +#define HPOUTR_POP_WORK (0x2 << 4) +#define HPOUTL_POP_MASK GENMASK(1, 0) +#define HPOUTL_POP_XCHARGE (0x1 << 0) +#define HPOUTL_POP_WORK (0x2 << 0) + +#define RK3328_HIFI 0 + +struct rk3328_reg_msk_val { + unsigned int reg; + unsigned int msk; + unsigned int val; +}; + +#endif From f5758544d98c8bec7793aeea28928f5e8bd45d47 Mon Sep 17 00:00:00 2001 From: Katsuhiro Suzuki Date: Fri, 21 Dec 2018 00:36:36 +0900 Subject: [PATCH 47/67] ASoC: rockchip: add workaround for silence of rk3288 ACODEC This patch adds reset and precharge in shutdown of PCM device. ACODEC goes to silence if we change Fs to 44.1kHz from 48kHz. This workaround seems to work but I don't know this workaround is correct sequence or not for ACODEC. Signed-off-by: Katsuhiro Suzuki Signed-off-by: Mark Brown --- sound/soc/codecs/rk3328_codec.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rk3328_codec.c b/sound/soc/codecs/rk3328_codec.c index 71f3fc2d970c..f3442a2283ea 100644 --- a/sound/soc/codecs/rk3328_codec.c +++ b/sound/soc/codecs/rk3328_codec.c @@ -261,9 +261,12 @@ static int rk3328_codec_close_playback(struct rk3328_codec_priv *rk3328) mdelay(1); } + /* Workaround for silence when changed Fs 48 -> 44.1kHz */ + rk3328_codec_reset(rk3328); + regmap_update_bits(rk3328->regmap, DAC_PRECHARGE_CTRL, DAC_CHARGE_CURRENT_ALL_MASK, - DAC_CHARGE_CURRENT_I); + DAC_CHARGE_CURRENT_ALL_ON); return 0; } From 1d38b4e903d577f05393eb0ac6727f40d90dd6c6 Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Wed, 26 Dec 2018 15:11:06 -0600 Subject: [PATCH 48/67] ASoC: xlnx: fix error handling in xlnx_formatter_pcm_probe Currently, if platform_get_irq_byname() fails, the returned error turns into a huge value, once it is being store into a variable of type unsigned int, hence never actually reporting any error and causing unexpected behavior when using the values stored in aud_drv_data->s2mm_irq and aud_drv_data->mm2s_irq. Fix this by changing the type of variables s2mm_irq and mm2s_irq in structure xlnx_pcm_drv_data from unsigned int to int. Addresses-Coverity-ID: 1476096 ("Unsigned compared against 0") Fixes: 796175a94a7f ("ASoC: xlnx: add pcm formatter platform driver") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/xilinx/xlnx_formatter_pcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/xilinx/xlnx_formatter_pcm.c b/sound/soc/xilinx/xlnx_formatter_pcm.c index f7235f7664d7..d2194da928e7 100644 --- a/sound/soc/xilinx/xlnx_formatter_pcm.c +++ b/sound/soc/xilinx/xlnx_formatter_pcm.c @@ -76,8 +76,8 @@ struct xlnx_pcm_drv_data { void __iomem *mmio; bool s2mm_presence; bool mm2s_presence; - unsigned int s2mm_irq; - unsigned int mm2s_irq; + int s2mm_irq; + int mm2s_irq; struct snd_pcm_substream *play_stream; struct snd_pcm_substream *capture_stream; struct clk *axi_clk; From e1de3d237b5083fcd0da6fcf600848a4cef9cc67 Mon Sep 17 00:00:00 2001 From: YueHaibing Date: Tue, 25 Dec 2018 02:20:36 +0000 Subject: [PATCH 49/67] ASoC: rockchip: fix platform_no_drv_owner.cocci warnings Remove .owner field if calls are used which set it automatically Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci Signed-off-by: YueHaibing Signed-off-by: Mark Brown --- sound/soc/codecs/rk3328_codec.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rk3328_codec.c b/sound/soc/codecs/rk3328_codec.c index f3442a2283ea..24f8f86d58e9 100644 --- a/sound/soc/codecs/rk3328_codec.c +++ b/sound/soc/codecs/rk3328_codec.c @@ -508,7 +508,6 @@ MODULE_DEVICE_TABLE(of, rk3328_codec_of_match); static struct platform_driver rk3328_codec_driver = { .driver = { .name = "rk3328-codec", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(rk3328_codec_of_match), }, .probe = rk3328_platform_probe, From 8c3590de0a378c2449fc1aec127cc693632458e4 Mon Sep 17 00:00:00 2001 From: Yizhuo Date: Thu, 3 Jan 2019 13:59:12 -0800 Subject: [PATCH 50/67] ASoC: Variable "val" in function rt274_i2c_probe() could be uninitialized Inside function rt274_i2c_probe(), if regmap_read() function returns -EINVAL, then local variable "val" leaves uninitialized but used in if statement. This is potentially unsafe. Signed-off-by: Yizhuo Signed-off-by: Mark Brown --- sound/soc/codecs/rt274.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt274.c b/sound/soc/codecs/rt274.c index 0ef966d56bac..e2855ab9a2c6 100644 --- a/sound/soc/codecs/rt274.c +++ b/sound/soc/codecs/rt274.c @@ -1128,8 +1128,11 @@ static int rt274_i2c_probe(struct i2c_client *i2c, return ret; } - regmap_read(rt274->regmap, + ret = regmap_read(rt274->regmap, RT274_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &val); + if (ret) + return ret; + if (val != RT274_VENDOR_ID) { dev_err(&i2c->dev, "Device with ID register %#x is not rt274\n", val); From 822257661031faa437336058d8a32bf1844ad9c6 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Thu, 3 Jan 2019 14:45:26 +0100 Subject: [PATCH 51/67] ASoC: es8316: Add jack-detect support Adding jack-detect support may seem weird for a codec with only a single output, but it is necessary. The ES8316 appnote showing the intended usage uses a jack-receptacle which physically disconnects the speakers from the output when a jack is plugged in. But all 3 devices using the es8316 which I have (2 Cherry Trail devices and one Bay Trail CR device), use an analog mux to disconnect the speakers, driven by a GPIO. In order to enable/disable the speakers at the right time, we need jack-detect. The same goes for the microphone where we must correctly set the mux for the single ADC to either the internal or the headset microphone. All devices I have support the es8316's builtin jack-detect functionality. Signed-off-by: Hans de Goede Signed-off-by: Mark Brown --- sound/soc/codecs/es8316.c | 195 +++++++++++++++++++++++++++++++++++++- sound/soc/codecs/es8316.h | 7 ++ 2 files changed, 198 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index e97d12d578b0..26413851e434 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -15,12 +15,14 @@ #include #include #include +#include #include #include #include #include #include #include +#include #include "es8316.h" /* In slave mode at single speed, the codec is documented as accepting 5 @@ -33,6 +35,11 @@ static const unsigned int supported_mclk_lrck_ratios[] = { }; struct es8316_priv { + struct mutex lock; + struct regmap *regmap; + struct snd_soc_component *component; + struct snd_soc_jack *jack; + int irq; unsigned int sysclk; unsigned int allowed_rates[NR_SUPPORTED_MCLK_LRCK_RATIOS]; struct snd_pcm_hw_constraint_list sysclk_constraints; @@ -529,8 +536,162 @@ static struct snd_soc_dai_driver es8316_dai = { .symmetric_rates = 1, }; +static void es8316_enable_micbias_for_mic_gnd_short_detect( + struct snd_soc_component *component) +{ + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + + snd_soc_dapm_mutex_lock(dapm); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "Bias"); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "Analog power"); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "Mic Bias"); + snd_soc_dapm_sync_unlocked(dapm); + snd_soc_dapm_mutex_unlock(dapm); + + msleep(20); +} + +static void es8316_disable_micbias_for_mic_gnd_short_detect( + struct snd_soc_component *component) +{ + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + + snd_soc_dapm_mutex_lock(dapm); + snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Bias"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Analog power"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Bias"); + snd_soc_dapm_sync_unlocked(dapm); + snd_soc_dapm_mutex_unlock(dapm); +} + +static irqreturn_t es8316_irq(int irq, void *data) +{ + struct es8316_priv *es8316 = data; + struct snd_soc_component *comp = es8316->component; + unsigned int flags; + + mutex_lock(&es8316->lock); + + regmap_read(es8316->regmap, ES8316_GPIO_FLAG, &flags); + if (flags == 0x00) + goto out; /* Powered-down / reset */ + + /* Catch spurious IRQ before set_jack is called */ + if (!es8316->jack) + goto out; + + dev_dbg(comp->dev, "gpio flags %#04x\n", flags); + if (flags & ES8316_GPIO_FLAG_HP_NOT_INSERTED) { + /* Jack removed, or spurious IRQ? */ + if (es8316->jack->status & SND_JACK_MICROPHONE) + es8316_disable_micbias_for_mic_gnd_short_detect(comp); + + if (es8316->jack->status & SND_JACK_HEADPHONE) { + snd_soc_jack_report(es8316->jack, 0, + SND_JACK_HEADSET | SND_JACK_BTN_0); + dev_dbg(comp->dev, "jack unplugged\n"); + } + } else if (!(es8316->jack->status & SND_JACK_HEADPHONE)) { + /* Jack inserted, determine type */ + es8316_enable_micbias_for_mic_gnd_short_detect(comp); + regmap_read(es8316->regmap, ES8316_GPIO_FLAG, &flags); + dev_dbg(comp->dev, "gpio flags %#04x\n", flags); + if (flags & ES8316_GPIO_FLAG_HP_NOT_INSERTED) { + /* Jack unplugged underneath us */ + es8316_disable_micbias_for_mic_gnd_short_detect(comp); + } else if (flags & ES8316_GPIO_FLAG_GM_NOT_SHORTED) { + /* Open, headset */ + snd_soc_jack_report(es8316->jack, + SND_JACK_HEADSET, + SND_JACK_HEADSET); + /* Keep mic-gnd-short detection on for button press */ + } else { + /* Shorted, headphones */ + snd_soc_jack_report(es8316->jack, + SND_JACK_HEADPHONE, + SND_JACK_HEADSET); + /* No longer need mic-gnd-short detection */ + es8316_disable_micbias_for_mic_gnd_short_detect(comp); + } + } else if (es8316->jack->status & SND_JACK_MICROPHONE) { + /* Interrupt while jack inserted, report button state */ + if (flags & ES8316_GPIO_FLAG_GM_NOT_SHORTED) { + /* Open, button release */ + snd_soc_jack_report(es8316->jack, 0, SND_JACK_BTN_0); + } else { + /* Short, button press */ + snd_soc_jack_report(es8316->jack, + SND_JACK_BTN_0, + SND_JACK_BTN_0); + } + } + +out: + mutex_unlock(&es8316->lock); + return IRQ_HANDLED; +} + +static void es8316_enable_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *jack) +{ + struct es8316_priv *es8316 = snd_soc_component_get_drvdata(component); + + mutex_lock(&es8316->lock); + + es8316->jack = jack; + + if (es8316->jack->status & SND_JACK_MICROPHONE) + es8316_enable_micbias_for_mic_gnd_short_detect(component); + + snd_soc_component_update_bits(component, ES8316_GPIO_DEBOUNCE, + ES8316_GPIO_ENABLE_INTERRUPT, + ES8316_GPIO_ENABLE_INTERRUPT); + + mutex_unlock(&es8316->lock); + + /* Enable irq and sync initial jack state */ + enable_irq(es8316->irq); + es8316_irq(es8316->irq, es8316); +} + +static void es8316_disable_jack_detect(struct snd_soc_component *component) +{ + struct es8316_priv *es8316 = snd_soc_component_get_drvdata(component); + + disable_irq(es8316->irq); + + mutex_lock(&es8316->lock); + + snd_soc_component_update_bits(component, ES8316_GPIO_DEBOUNCE, + ES8316_GPIO_ENABLE_INTERRUPT, 0); + + if (es8316->jack->status & SND_JACK_MICROPHONE) { + es8316_disable_micbias_for_mic_gnd_short_detect(component); + snd_soc_jack_report(es8316->jack, 0, SND_JACK_BTN_0); + } + + es8316->jack = NULL; + + mutex_unlock(&es8316->lock); +} + +static int es8316_set_jack(struct snd_soc_component *component, + struct snd_soc_jack *jack, void *data) +{ + if (jack) + es8316_enable_jack_detect(component, jack); + else + es8316_disable_jack_detect(component); + + return 0; +} + static int es8316_probe(struct snd_soc_component *component) { + struct es8316_priv *es8316 = snd_soc_component_get_drvdata(component); + + es8316->component = component; + /* Reset codec and enable current state machine */ snd_soc_component_write(component, ES8316_RESET, 0x3f); usleep_range(5000, 5500); @@ -555,6 +716,7 @@ static int es8316_probe(struct snd_soc_component *component) static const struct snd_soc_component_driver soc_component_dev_es8316 = { .probe = es8316_probe, + .set_jack = es8316_set_jack, .controls = es8316_snd_controls, .num_controls = ARRAY_SIZE(es8316_snd_controls), .dapm_widgets = es8316_dapm_widgets, @@ -566,18 +728,29 @@ static const struct snd_soc_component_driver soc_component_dev_es8316 = { .non_legacy_dai_naming = 1, }; +static const struct regmap_range es8316_volatile_ranges[] = { + regmap_reg_range(ES8316_GPIO_FLAG, ES8316_GPIO_FLAG), +}; + +static const struct regmap_access_table es8316_volatile_table = { + .yes_ranges = es8316_volatile_ranges, + .n_yes_ranges = ARRAY_SIZE(es8316_volatile_ranges), +}; + static const struct regmap_config es8316_regmap = { .reg_bits = 8, .val_bits = 8, .max_register = 0x53, + .volatile_table = &es8316_volatile_table, .cache_type = REGCACHE_RBTREE, }; static int es8316_i2c_probe(struct i2c_client *i2c_client, const struct i2c_device_id *id) { + struct device *dev = &i2c_client->dev; struct es8316_priv *es8316; - struct regmap *regmap; + int ret; es8316 = devm_kzalloc(&i2c_client->dev, sizeof(struct es8316_priv), GFP_KERNEL); @@ -586,9 +759,23 @@ static int es8316_i2c_probe(struct i2c_client *i2c_client, i2c_set_clientdata(i2c_client, es8316); - regmap = devm_regmap_init_i2c(i2c_client, &es8316_regmap); - if (IS_ERR(regmap)) - return PTR_ERR(regmap); + es8316->regmap = devm_regmap_init_i2c(i2c_client, &es8316_regmap); + if (IS_ERR(es8316->regmap)) + return PTR_ERR(es8316->regmap); + + es8316->irq = i2c_client->irq; + mutex_init(&es8316->lock); + + ret = devm_request_threaded_irq(dev, es8316->irq, NULL, es8316_irq, + IRQF_TRIGGER_HIGH | IRQF_ONESHOT, + "es8316", es8316); + if (ret == 0) { + /* Gets re-enabled by es8316_set_jack() */ + disable_irq(es8316->irq); + } else { + dev_warn(dev, "Failed to get IRQ %d: %d\n", es8316->irq, ret); + es8316->irq = -ENXIO; + } return devm_snd_soc_register_component(&i2c_client->dev, &soc_component_dev_es8316, diff --git a/sound/soc/codecs/es8316.h b/sound/soc/codecs/es8316.h index 6bcdd63ea459..439a0130cbb7 100644 --- a/sound/soc/codecs/es8316.h +++ b/sound/soc/codecs/es8316.h @@ -126,4 +126,11 @@ #define ES8316_SERDATA2_LEN_16 0x0c #define ES8316_SERDATA2_LEN_32 0x10 +/* ES8316_GPIO_DEBOUNCE */ +#define ES8316_GPIO_ENABLE_INTERRUPT 0x02 + +/* ES8316_GPIO_FLAG */ +#define ES8316_GPIO_FLAG_GM_NOT_SHORTED 0x02 +#define ES8316_GPIO_FLAG_HP_NOT_INSERTED 0x04 + #endif From 24b53f17a3f24967b8b523243f9f7fc361427119 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Thu, 3 Jan 2019 14:45:27 +0100 Subject: [PATCH 52/67] ASoC: es8316: Add DAC mono mix switch mixer control Export the DAC functionality to mix left + right together and then output the same (mixed) signal on both outputs. Various (x86) tablets with an ES8316 codec use a single speaker connected between the headhpone LOUT and ROUT pins, expecting the output to be in a mono differential mode. Presumably this is done to use the power of both the left and right outputs to allow the speaker to be louder. The ES8316 codec does not have a differential output mode, but we can emulate this by making both channels output the same through the mono mix switch, combined with setting the Playback Polarity control to "R Invert", which applias a 180 degrees phase inversion to the right channel. Signed-off-by: Hans de Goede Signed-off-by: Mark Brown --- sound/soc/codecs/es8316.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index 26413851e434..98464ba1046c 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -101,6 +101,7 @@ static const struct snd_kcontrol_new es8316_snd_controls[] = { SOC_SINGLE("DAC Notch Filter Switch", ES8316_DAC_SET2, 6, 1, 0), SOC_SINGLE("DAC Double Fs Switch", ES8316_DAC_SET2, 7, 1, 0), SOC_SINGLE("DAC Stereo Enhancement", ES8316_DAC_SET3, 0, 7, 0), + SOC_SINGLE("DAC Mono Mix Switch", ES8316_DAC_SET3, 3, 1, 0), SOC_ENUM("Capture Polarity", adcpol), SOC_SINGLE("Mic Boost Switch", ES8316_ADC_D2SEPGA, 0, 1, 0), From 6ca382c4363d6c636200ccdd9ac95f44b1a498ea Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Thu, 3 Jan 2019 14:45:28 +0100 Subject: [PATCH 53/67] ASoC: Intel: bytcht_es8316: Sort includes alphabetically For lack of a better (non-random) way of sorting includes more and more files in the kernel are moving over to sorting the includes alphabetically. Move the bytcht_es8316 driver over to this sorting before we add a bunch of more includes. Signed-off-by: Hans de Goede Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcht_es8316.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index adc26dfc7d65..5d8ecc100766 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -19,13 +19,13 @@ * * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ */ +#include +#include #include #include #include -#include #include #include -#include #include #include #include From 86909c8f77c5eda17a9b5dc954849e25df1ffe0f Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Thu, 3 Jan 2019 14:45:29 +0100 Subject: [PATCH 54/67] ASoC: Intel: bytcht_es8316: Minor refactoring Some minor refactoring: 1) Group the code setting the card dev and prive pointers together with registering the card 2) Properly put the comment about registering the card at the place where we actually register the card and add a new comment for getting the clk 3) Add a struct device *dev helper variable (this will be used more in follow up commits) 4) Reword error message to have the same "foo failed: %d" wording as others Signed-off-by: Hans de Goede Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcht_es8316.c | 26 +++++++++++++------------- 1 file changed, 13 insertions(+), 13 deletions(-) diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 5d8ecc100766..e29f00560b00 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -237,17 +237,18 @@ static char codec_name[SND_ACPI_I2C_ID_LEN]; static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) { struct byt_cht_es8316_private *priv; + struct device *dev = &pdev->dev; struct snd_soc_acpi_mach *mach; const char *i2c_name = NULL; int dai_index = 0; int i; int ret = 0; - priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); if (!priv) return -ENOMEM; - mach = (&pdev->dev)->platform_data; + mach = dev->platform_data; /* fix index of codec dai */ for (i = 0; i < ARRAY_SIZE(byt_cht_es8316_dais); i++) { if (!strcmp(byt_cht_es8316_dais[i].codec_name, @@ -265,26 +266,25 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) byt_cht_es8316_dais[dai_index].codec_name = codec_name; } - /* register the soc card */ - byt_cht_es8316_card.dev = &pdev->dev; - snd_soc_card_set_drvdata(&byt_cht_es8316_card, priv); - - priv->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3"); + /* get the clock */ + priv->mclk = devm_clk_get(dev, "pmc_plt_clk_3"); if (IS_ERR(priv->mclk)) { ret = PTR_ERR(priv->mclk); - dev_err(&pdev->dev, - "Failed to get MCLK from pmc_plt_clk_3: %d\n", - ret); + dev_err(dev, "clk_get pmc_plt_clk_3 failed: %d\n", ret); return ret; } - ret = devm_snd_soc_register_card(&pdev->dev, &byt_cht_es8316_card); + /* register the soc card */ + byt_cht_es8316_card.dev = dev; + snd_soc_card_set_drvdata(&byt_cht_es8316_card, priv); + + ret = devm_snd_soc_register_card(dev, &byt_cht_es8316_card); if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card failed %d\n", ret); + dev_err(dev, "snd_soc_register_card failed: %d\n", ret); return ret; } platform_set_drvdata(pdev, &byt_cht_es8316_card); - return ret; + return 0; } static struct platform_driver snd_byt_cht_es8316_mc_driver = { From 349e13862c9975c613aac9dc7fa953e70cff9d06 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Thu, 3 Jan 2019 14:45:30 +0100 Subject: [PATCH 55/67] ASoC: Intel: bytcht_es8316: Add support for SSP0 (BYTCR) Add support for having the codec connected to SSP0 instead of SSP2. This is controlled through a new quirk parameter, similar to how this is done in the bytcr_rt5640 and bytcr_rt5651 machine drivers. Bay Trail CR (cost reduced) SoCs do not have an SSP2, so we default to SSP0 there. Note the SPP0 quirk gets BIT(16) because bits 0-15 are reserved for non boolean quirks like the input-map added in a later commit in this series. Signed-off-by: Hans de Goede Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcht_es8316.c | 76 ++++++++++++++++++++++++-- 1 file changed, 72 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index e29f00560b00..3358d82499a3 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -25,6 +25,8 @@ #include #include #include +#include +#include #include #include #include @@ -37,6 +39,20 @@ struct byt_cht_es8316_private { struct clk *mclk; }; +#define BYT_CHT_ES8316_SSP0 BIT(16) + +static int quirk; + +static int quirk_override = -1; +module_param_named(quirk, quirk_override, int, 0444); +MODULE_PARM_DESC(quirk, "Board-specific quirk override"); + +static void log_quirks(struct device *dev) +{ + if (quirk & BYT_CHT_ES8316_SSP0) + dev_info(dev, "quirk SSP0 enabled"); +} + static const struct snd_soc_dapm_widget byt_cht_es8316_widgets[] = { SND_SOC_DAPM_HP("Headphone", NULL), @@ -55,7 +71,16 @@ static const struct snd_soc_dapm_route byt_cht_es8316_audio_map[] = { {"Headphone", NULL, "HPOL"}, {"Headphone", NULL, "HPOR"}, +}; +static const struct snd_soc_dapm_route byt_cht_es8316_ssp0_map[] = { + {"Playback", NULL, "ssp0 Tx"}, + {"ssp0 Tx", NULL, "modem_out"}, + {"modem_in", NULL, "ssp0 Rx"}, + {"ssp0 Rx", NULL, "Capture"}, +}; + +static const struct snd_soc_dapm_route byt_cht_es8316_ssp2_map[] = { {"Playback", NULL, "ssp2 Tx"}, {"ssp2 Tx", NULL, "codec_out0"}, {"ssp2 Tx", NULL, "codec_out1"}, @@ -74,10 +99,23 @@ static int byt_cht_es8316_init(struct snd_soc_pcm_runtime *runtime) { struct snd_soc_card *card = runtime->card; struct byt_cht_es8316_private *priv = snd_soc_card_get_drvdata(card); + const struct snd_soc_dapm_route *custom_map; + int num_routes; int ret; card->dapm.idle_bias_off = true; + if (quirk & BYT_CHT_ES8316_SSP0) { + custom_map = byt_cht_es8316_ssp0_map; + num_routes = ARRAY_SIZE(byt_cht_es8316_ssp0_map); + } else { + custom_map = byt_cht_es8316_ssp2_map; + num_routes = ARRAY_SIZE(byt_cht_es8316_ssp2_map); + } + ret = snd_soc_dapm_add_routes(&card->dapm, custom_map, num_routes); + if (ret) + return ret; + /* * The firmware might enable the clock at boot (this information * may or may not be reflected in the enable clock register). @@ -123,14 +161,21 @@ static int byt_cht_es8316_codec_fixup(struct snd_soc_pcm_runtime *rtd, SNDRV_PCM_HW_PARAM_RATE); struct snd_interval *channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); - int ret; + int ret, bits; /* The DSP will covert the FE rate to 48k, stereo */ rate->min = rate->max = 48000; channels->min = channels->max = 2; - /* set SSP2 to 24-bit */ - params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); + if (quirk & BYT_CHT_ES8316_SSP0) { + /* set SSP0 to 16-bit */ + params_set_format(params, SNDRV_PCM_FORMAT_S16_LE); + bits = 16; + } else { + /* set SSP2 to 24-bit */ + params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); + bits = 24; + } /* * Default mode for SSP configuration is TDM 4 slot, override config @@ -147,7 +192,7 @@ static int byt_cht_es8316_codec_fixup(struct snd_soc_pcm_runtime *rtd, return ret; } - ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, 24); + ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, bits); if (ret < 0) { dev_err(rtd->dev, "can't set I2S config, err %d\n", ret); return ret; @@ -232,6 +277,11 @@ static struct snd_soc_card byt_cht_es8316_card = { .fully_routed = true, }; +static const struct x86_cpu_id baytrail_cpu_ids[] = { + { X86_VENDOR_INTEL, 6, INTEL_FAM6_ATOM_SILVERMONT }, /* Valleyview */ + {} +}; + static char codec_name[SND_ACPI_I2C_ID_LEN]; static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) @@ -266,6 +316,24 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) byt_cht_es8316_dais[dai_index].codec_name = codec_name; } + /* Check for BYTCR or other platform and setup quirks */ + if (x86_match_cpu(baytrail_cpu_ids) && + mach->mach_params.acpi_ipc_irq_index == 0) { + /* On BYTCR default to SSP0 */ + quirk = BYT_CHT_ES8316_SSP0; + } else { + quirk = 0; + } + if (quirk_override != -1) { + dev_info(dev, "Overriding quirk 0x%x => 0x%x\n", quirk, + quirk_override); + quirk = quirk_override; + } + log_quirks(dev); + + if (quirk & BYT_CHT_ES8316_SSP0) + byt_cht_es8316_dais[dai_index].cpu_dai_name = "ssp0-port"; + /* get the clock */ priv->mclk = devm_clk_get(dev, "pmc_plt_clk_3"); if (IS_ERR(priv->mclk)) { From 4bf538b42933253296daf86aab7ede56b5fb97bf Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Thu, 3 Jan 2019 14:45:31 +0100 Subject: [PATCH 56/67] ASoC: Intel: bytcht_es8316: Add jack-detect support Hookup the jack-detect support added to the codec driver. Signed-off-by: Hans de Goede Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcht_es8316.c | 67 +++++++++++++++++++++++++- 1 file changed, 65 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 3358d82499a3..905dd6904710 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -22,12 +22,14 @@ #include #include #include +#include #include #include #include #include #include #include +#include #include #include #include @@ -37,6 +39,7 @@ struct byt_cht_es8316_private { struct clk *mclk; + struct snd_soc_jack jack; }; #define BYT_CHT_ES8316_SSP0 BIT(16) @@ -55,6 +58,7 @@ static void log_quirks(struct device *dev) static const struct snd_soc_dapm_widget byt_cht_es8316_widgets[] = { SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), /* * The codec supports two analog microphone inputs. I have only @@ -68,6 +72,7 @@ static const struct snd_soc_dapm_widget byt_cht_es8316_widgets[] = { static const struct snd_soc_dapm_route byt_cht_es8316_audio_map[] = { {"MIC1", NULL, "Microphone 1"}, {"MIC2", NULL, "Microphone 2"}, + {"MIC1", NULL, "Headset Mic"}, {"Headphone", NULL, "HPOL"}, {"Headphone", NULL, "HPOR"}, @@ -91,12 +96,25 @@ static const struct snd_soc_dapm_route byt_cht_es8316_ssp2_map[] = { static const struct snd_kcontrol_new byt_cht_es8316_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), SOC_DAPM_PIN_SWITCH("Microphone 1"), SOC_DAPM_PIN_SWITCH("Microphone 2"), }; +static struct snd_soc_jack_pin byt_cht_es8316_jack_pins[] = { + { + .pin = "Headphone", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, +}; + static int byt_cht_es8316_init(struct snd_soc_pcm_runtime *runtime) { + struct snd_soc_component *codec = runtime->codec_dai->component; struct snd_soc_card *card = runtime->card; struct byt_cht_es8316_private *priv = snd_soc_card_get_drvdata(card); const struct snd_soc_dapm_route *custom_map; @@ -143,6 +161,18 @@ static int byt_cht_es8316_init(struct snd_soc_pcm_runtime *runtime) return ret; } + ret = snd_soc_card_jack_new(card, "Headset", + SND_JACK_HEADSET | SND_JACK_BTN_0, + &priv->jack, byt_cht_es8316_jack_pins, + ARRAY_SIZE(byt_cht_es8316_jack_pins)); + if (ret) { + dev_err(card->dev, "jack creation failed %d\n", ret); + return ret; + } + + snd_jack_set_key(priv->jack.jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_soc_component_set_jack(codec, &priv->jack, NULL); + return 0; } @@ -263,6 +293,39 @@ static struct snd_soc_dai_link byt_cht_es8316_dais[] = { /* SoC card */ +static char codec_name[SND_ACPI_I2C_ID_LEN]; + +static int byt_cht_es8316_suspend(struct snd_soc_card *card) +{ + struct snd_soc_component *component; + + for_each_card_components(card, component) { + if (!strcmp(component->name, codec_name)) { + dev_dbg(component->dev, "disabling jack detect before suspend\n"); + snd_soc_component_set_jack(component, NULL, NULL); + break; + } + } + + return 0; +} + +static int byt_cht_es8316_resume(struct snd_soc_card *card) +{ + struct byt_cht_es8316_private *priv = snd_soc_card_get_drvdata(card); + struct snd_soc_component *component; + + for_each_card_components(card, component) { + if (!strcmp(component->name, codec_name)) { + dev_dbg(component->dev, "re-enabling jack detect after resume\n"); + snd_soc_component_set_jack(component, &priv->jack, NULL); + break; + } + } + + return 0; +} + static struct snd_soc_card byt_cht_es8316_card = { .name = "bytcht-es8316", .owner = THIS_MODULE, @@ -275,6 +338,8 @@ static struct snd_soc_card byt_cht_es8316_card = { .controls = byt_cht_es8316_controls, .num_controls = ARRAY_SIZE(byt_cht_es8316_controls), .fully_routed = true, + .suspend_pre = byt_cht_es8316_suspend, + .resume_post = byt_cht_es8316_resume, }; static const struct x86_cpu_id baytrail_cpu_ids[] = { @@ -282,8 +347,6 @@ static const struct x86_cpu_id baytrail_cpu_ids[] = { {} }; -static char codec_name[SND_ACPI_I2C_ID_LEN]; - static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) { struct byt_cht_es8316_private *priv; From 0d3e91da0750835cfd5c16487ffb3cdd752ea53a Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Thu, 3 Jan 2019 14:45:32 +0100 Subject: [PATCH 57/67] ASoC: Intel: bytcht_es8316: Add external speaker mux support The ES8316 only has a single (amplified) output. The ES8316 appnote showing the intended usage uses a jack-receptacle which physically disconnects the speakers from the output when a jack is plugged in. But all 3 devices using the es8316 which I have (2 Cherry Trail devices and one Bay Trail CR device), use an analog mux to disconnect the speakers, driven by a GPIO. This commit adds support for this, modelling this as a separate speaker widget / dapm pin-switch which sets the mux to drive the speakers when selected. The intend is for userspace to use the recently added jack-detect support and then automatically select either the Headphone or Speaker output based on that. Note this commit includes a workaround for an ACPI table bug which is present on 2 of the 3 devices I have, see the added comment in the code. Signed-off-by: Hans de Goede Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcht_es8316.c | 98 ++++++++++++++++++++++++++ 1 file changed, 98 insertions(+) diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 905dd6904710..8e504fca4624 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -19,8 +19,11 @@ * * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ */ +#include #include #include +#include +#include #include #include #include @@ -40,6 +43,8 @@ struct byt_cht_es8316_private { struct clk *mclk; struct snd_soc_jack jack; + struct gpio_desc *speaker_en_gpio; + bool speaker_en; }; #define BYT_CHT_ES8316_SSP0 BIT(16) @@ -56,7 +61,24 @@ static void log_quirks(struct device *dev) dev_info(dev, "quirk SSP0 enabled"); } +static int byt_cht_es8316_speaker_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_card *card = w->dapm->card; + struct byt_cht_es8316_private *priv = snd_soc_card_get_drvdata(card); + + if (SND_SOC_DAPM_EVENT_ON(event)) + priv->speaker_en = true; + else + priv->speaker_en = false; + + gpiod_set_value_cansleep(priv->speaker_en_gpio, priv->speaker_en); + + return 0; +} + static const struct snd_soc_dapm_widget byt_cht_es8316_widgets[] = { + SND_SOC_DAPM_SPK("Speaker", NULL), SND_SOC_DAPM_HP("Headphone", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), @@ -67,6 +89,10 @@ static const struct snd_soc_dapm_widget byt_cht_es8316_widgets[] = { */ SND_SOC_DAPM_MIC("Microphone 1", NULL), SND_SOC_DAPM_MIC("Microphone 2", NULL), + + SND_SOC_DAPM_SUPPLY("Speaker Power", SND_SOC_NOPM, 0, 0, + byt_cht_es8316_speaker_power_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), }; static const struct snd_soc_dapm_route byt_cht_es8316_audio_map[] = { @@ -76,6 +102,14 @@ static const struct snd_soc_dapm_route byt_cht_es8316_audio_map[] = { {"Headphone", NULL, "HPOL"}, {"Headphone", NULL, "HPOR"}, + + /* + * There is no separate speaker output instead the speakers are muxed to + * the HP outputs. The mux is controlled by the "Speaker Power" supply. + */ + {"Speaker", NULL, "HPOL"}, + {"Speaker", NULL, "HPOR"}, + {"Speaker", NULL, "Speaker Power"}, }; static const struct snd_soc_dapm_route byt_cht_es8316_ssp0_map[] = { @@ -95,6 +129,7 @@ static const struct snd_soc_dapm_route byt_cht_es8316_ssp2_map[] = { }; static const struct snd_kcontrol_new byt_cht_es8316_controls[] = { + SOC_DAPM_PIN_SWITCH("Speaker"), SOC_DAPM_PIN_SWITCH("Headphone"), SOC_DAPM_PIN_SWITCH("Headset Mic"), SOC_DAPM_PIN_SWITCH("Microphone 1"), @@ -323,6 +358,25 @@ static int byt_cht_es8316_resume(struct snd_soc_card *card) } } + /* + * Some Cherry Trail boards with an ES8316 codec have a bug in their + * ACPI tables where the MSSL1680 touchscreen's _PS0 and _PS3 methods + * wrongly also set the speaker-enable GPIO to 1/0. Testing has shown + * that this really is a bug and the GPIO has no influence on the + * touchscreen at all. + * + * The silead.c touchscreen driver does not support runtime suspend, so + * the GPIO can only be changed underneath us during a system suspend. + * This resume() function runs from a pm complete() callback, and thus + * is guaranteed to run after the touchscreen driver/ACPI-subsys has + * brought the touchscreen back up again (and thus changed the GPIO). + * + * So to work around this we pass GPIOD_FLAGS_BIT_NONEXCLUSIVE when + * requesting the GPIO and we set its value here to undo any changes + * done by the touchscreen's broken _PS0 ACPI method. + */ + gpiod_set_value_cansleep(priv->speaker_en_gpio, priv->speaker_en); + return 0; } @@ -347,12 +401,20 @@ static const struct x86_cpu_id baytrail_cpu_ids[] = { {} }; +static const struct acpi_gpio_params first_gpio = { 0, 0, false }; + +static const struct acpi_gpio_mapping byt_cht_es8316_gpios[] = { + { "speaker-enable-gpios", &first_gpio, 1 }, + { }, +}; + static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) { struct byt_cht_es8316_private *priv; struct device *dev = &pdev->dev; struct snd_soc_acpi_mach *mach; const char *i2c_name = NULL; + struct device *codec_dev; int dai_index = 0; int i; int ret = 0; @@ -405,12 +467,39 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) return ret; } + /* get speaker enable GPIO */ + codec_dev = bus_find_device_by_name(&i2c_bus_type, NULL, codec_name); + if (!codec_dev) + return -EPROBE_DEFER; + + devm_acpi_dev_add_driver_gpios(codec_dev, byt_cht_es8316_gpios); + priv->speaker_en_gpio = + gpiod_get_index(codec_dev, "speaker-enable", 0, + /* see comment in byt_cht_es8316_resume */ + GPIOD_OUT_LOW | GPIOD_FLAGS_BIT_NONEXCLUSIVE); + put_device(codec_dev); + + if (IS_ERR(priv->speaker_en_gpio)) { + ret = PTR_ERR(priv->speaker_en_gpio); + switch (ret) { + case -ENOENT: + priv->speaker_en_gpio = NULL; + break; + default: + dev_err(dev, "get speaker GPIO failed: %d\n", ret); + /* fall through */ + case -EPROBE_DEFER: + return ret; + } + } + /* register the soc card */ byt_cht_es8316_card.dev = dev; snd_soc_card_set_drvdata(&byt_cht_es8316_card, priv); ret = devm_snd_soc_register_card(dev, &byt_cht_es8316_card); if (ret) { + gpiod_put(priv->speaker_en_gpio); dev_err(dev, "snd_soc_register_card failed: %d\n", ret); return ret; } @@ -418,11 +507,20 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) return 0; } +static int snd_byt_cht_es8316_mc_remove(struct platform_device *pdev) +{ + struct byt_cht_es8316_private *priv = platform_get_drvdata(pdev); + + gpiod_put(priv->speaker_en_gpio); + return 0; +} + static struct platform_driver snd_byt_cht_es8316_mc_driver = { .driver = { .name = "bytcht_es8316", }, .probe = snd_byt_cht_es8316_mc_probe, + .remove = snd_byt_cht_es8316_mc_remove, }; module_platform_driver(snd_byt_cht_es8316_mc_driver); From 730501a91d94b652275e049e101ed44cdbfdf31b Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Thu, 3 Jan 2019 14:45:33 +0100 Subject: [PATCH 58/67] ASoC: Intel: bytcht_es8316: Add input-map support After adding jack-detect support we have 3 microphone input switches: "Microphone 1", "Microphone 2" and "Headset Mic". But the ES8316 has only 2 microphone inputs. In the app-note explaining how to use the codec and on the 3 boards I have one input is used for an internal microphone and one for the headset microphone. On the 2 CHT boards I have the internal mic is on on MIC1 and the headset mic is on MIC2, on the BYTCR board I have it is the other way around. This commit replaces the 2 "Microphone 1" and "Microphone 2" input switches with a single "Internal Mic" switch and adds support for selecting either possible input mapping. Signed-off-by: Hans de Goede Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcht_es8316.c | 58 ++++++++++++++++++-------- 1 file changed, 41 insertions(+), 17 deletions(-) diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 8e504fca4624..941a66f94660 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -47,6 +47,12 @@ struct byt_cht_es8316_private { bool speaker_en; }; +enum { + BYT_CHT_ES8316_INTMIC_IN1_MAP, + BYT_CHT_ES8316_INTMIC_IN2_MAP, +}; + +#define BYT_CHT_ES8316_MAP(quirk) ((quirk) & GENMASK(3, 0)) #define BYT_CHT_ES8316_SSP0 BIT(16) static int quirk; @@ -57,6 +63,10 @@ MODULE_PARM_DESC(quirk, "Board-specific quirk override"); static void log_quirks(struct device *dev) { + if (BYT_CHT_ES8316_MAP(quirk) == BYT_CHT_ES8316_INTMIC_IN1_MAP) + dev_info(dev, "quirk IN1_MAP enabled"); + if (BYT_CHT_ES8316_MAP(quirk) == BYT_CHT_ES8316_INTMIC_IN2_MAP) + dev_info(dev, "quirk IN2_MAP enabled"); if (quirk & BYT_CHT_ES8316_SSP0) dev_info(dev, "quirk SSP0 enabled"); } @@ -81,14 +91,7 @@ static const struct snd_soc_dapm_widget byt_cht_es8316_widgets[] = { SND_SOC_DAPM_SPK("Speaker", NULL), SND_SOC_DAPM_HP("Headphone", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), - - /* - * The codec supports two analog microphone inputs. I have only - * tested MIC1. A DMIC route could also potentially be added - * if such functionality is found on another platform. - */ - SND_SOC_DAPM_MIC("Microphone 1", NULL), - SND_SOC_DAPM_MIC("Microphone 2", NULL), + SND_SOC_DAPM_MIC("Internal Mic", NULL), SND_SOC_DAPM_SUPPLY("Speaker Power", SND_SOC_NOPM, 0, 0, byt_cht_es8316_speaker_power_event, @@ -96,10 +99,6 @@ static const struct snd_soc_dapm_widget byt_cht_es8316_widgets[] = { }; static const struct snd_soc_dapm_route byt_cht_es8316_audio_map[] = { - {"MIC1", NULL, "Microphone 1"}, - {"MIC2", NULL, "Microphone 2"}, - {"MIC1", NULL, "Headset Mic"}, - {"Headphone", NULL, "HPOL"}, {"Headphone", NULL, "HPOR"}, @@ -112,6 +111,16 @@ static const struct snd_soc_dapm_route byt_cht_es8316_audio_map[] = { {"Speaker", NULL, "Speaker Power"}, }; +static const struct snd_soc_dapm_route byt_cht_es8316_intmic_in1_map[] = { + {"MIC1", NULL, "Internal Mic"}, + {"MIC2", NULL, "Headset Mic"}, +}; + +static const struct snd_soc_dapm_route byt_cht_es8316_intmic_in2_map[] = { + {"MIC2", NULL, "Internal Mic"}, + {"MIC1", NULL, "Headset Mic"}, +}; + static const struct snd_soc_dapm_route byt_cht_es8316_ssp0_map[] = { {"Playback", NULL, "ssp0 Tx"}, {"ssp0 Tx", NULL, "modem_out"}, @@ -132,8 +141,7 @@ static const struct snd_kcontrol_new byt_cht_es8316_controls[] = { SOC_DAPM_PIN_SWITCH("Speaker"), SOC_DAPM_PIN_SWITCH("Headphone"), SOC_DAPM_PIN_SWITCH("Headset Mic"), - SOC_DAPM_PIN_SWITCH("Microphone 1"), - SOC_DAPM_PIN_SWITCH("Microphone 2"), + SOC_DAPM_PIN_SWITCH("Internal Mic"), }; static struct snd_soc_jack_pin byt_cht_es8316_jack_pins[] = { @@ -158,6 +166,21 @@ static int byt_cht_es8316_init(struct snd_soc_pcm_runtime *runtime) card->dapm.idle_bias_off = true; + switch (BYT_CHT_ES8316_MAP(quirk)) { + case BYT_CHT_ES8316_INTMIC_IN1_MAP: + default: + custom_map = byt_cht_es8316_intmic_in1_map; + num_routes = ARRAY_SIZE(byt_cht_es8316_intmic_in1_map); + break; + case BYT_CHT_ES8316_INTMIC_IN2_MAP: + custom_map = byt_cht_es8316_intmic_in2_map; + num_routes = ARRAY_SIZE(byt_cht_es8316_intmic_in2_map); + break; + } + ret = snd_soc_dapm_add_routes(&card->dapm, custom_map, num_routes); + if (ret) + return ret; + if (quirk & BYT_CHT_ES8316_SSP0) { custom_map = byt_cht_es8316_ssp0_map; num_routes = ARRAY_SIZE(byt_cht_es8316_ssp0_map); @@ -444,10 +467,11 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) /* Check for BYTCR or other platform and setup quirks */ if (x86_match_cpu(baytrail_cpu_ids) && mach->mach_params.acpi_ipc_irq_index == 0) { - /* On BYTCR default to SSP0 */ - quirk = BYT_CHT_ES8316_SSP0; + /* On BYTCR default to SSP0, internal-mic-in2-map */ + quirk = BYT_CHT_ES8316_SSP0 | BYT_CHT_ES8316_INTMIC_IN2_MAP; } else { - quirk = 0; + /* Others default to internal-mic-in1-map */ + quirk = BYT_CHT_ES8316_INTMIC_IN1_MAP; } if (quirk_override != -1) { dev_info(dev, "Overriding quirk 0x%x => 0x%x\n", quirk, From 249d2fc9e55c324dda968252ea3ad0ac21c72b8f Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Thu, 3 Jan 2019 14:45:34 +0100 Subject: [PATCH 59/67] ASoC: Intel: bytcht_es8316: Set card long_name based on quirks Depending on the input-map and on if 1 or 2 speakers are connected, userspace needs to use a different UCM profile. Since we already deal with quirks in the kernel driver and set the input-map from the kernel, add a quirk for devices with a single / mono speaker and set the card's long_name based on the input and speaker quirks, so that userspace can use the long_name to pick the right UCM profile. This change, including how the long_name is build-up mirrors how we do this in the bytcr_rt5640 and bytcr_rt5651 machine drivers. Note since all devices I have access to use a mono speaker setup I've chosen to default the speaker setting to mono. Signed-off-by: Hans de Goede Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcht_es8316.c | 19 +++++++++++++++---- 1 file changed, 15 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 941a66f94660..cdf2061e7613 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -54,6 +54,7 @@ enum { #define BYT_CHT_ES8316_MAP(quirk) ((quirk) & GENMASK(3, 0)) #define BYT_CHT_ES8316_SSP0 BIT(16) +#define BYT_CHT_ES8316_MONO_SPEAKER BIT(17) static int quirk; @@ -69,6 +70,8 @@ static void log_quirks(struct device *dev) dev_info(dev, "quirk IN2_MAP enabled"); if (quirk & BYT_CHT_ES8316_SSP0) dev_info(dev, "quirk SSP0 enabled"); + if (quirk & BYT_CHT_ES8316_MONO_SPEAKER) + dev_info(dev, "quirk MONO_SPEAKER enabled\n"); } static int byt_cht_es8316_speaker_power_event(struct snd_soc_dapm_widget *w, @@ -352,6 +355,7 @@ static struct snd_soc_dai_link byt_cht_es8316_dais[] = { /* SoC card */ static char codec_name[SND_ACPI_I2C_ID_LEN]; +static char long_name[50]; /* = "bytcht-es8316-*-spk-*-mic" */ static int byt_cht_es8316_suspend(struct snd_soc_card *card) { @@ -433,6 +437,7 @@ static const struct acpi_gpio_mapping byt_cht_es8316_gpios[] = { static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) { + const char * const mic_name[] = { "in1", "in2" }; struct byt_cht_es8316_private *priv; struct device *dev = &pdev->dev; struct snd_soc_acpi_mach *mach; @@ -467,11 +472,13 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) /* Check for BYTCR or other platform and setup quirks */ if (x86_match_cpu(baytrail_cpu_ids) && mach->mach_params.acpi_ipc_irq_index == 0) { - /* On BYTCR default to SSP0, internal-mic-in2-map */ - quirk = BYT_CHT_ES8316_SSP0 | BYT_CHT_ES8316_INTMIC_IN2_MAP; + /* On BYTCR default to SSP0, internal-mic-in2-map, mono-spk */ + quirk = BYT_CHT_ES8316_SSP0 | BYT_CHT_ES8316_INTMIC_IN2_MAP | + BYT_CHT_ES8316_MONO_SPEAKER; } else { - /* Others default to internal-mic-in1-map */ - quirk = BYT_CHT_ES8316_INTMIC_IN1_MAP; + /* Others default to internal-mic-in1-map, mono-speaker */ + quirk = BYT_CHT_ES8316_INTMIC_IN1_MAP | + BYT_CHT_ES8316_MONO_SPEAKER; } if (quirk_override != -1) { dev_info(dev, "Overriding quirk 0x%x => 0x%x\n", quirk, @@ -518,6 +525,10 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) } /* register the soc card */ + snprintf(long_name, sizeof(long_name), "bytcht-es8316-%s-spk-%s-mic", + (quirk & BYT_CHT_ES8316_MONO_SPEAKER) ? "mono" : "stereo", + mic_name[BYT_CHT_ES8316_MAP(quirk)]); + byt_cht_es8316_card.long_name = long_name; byt_cht_es8316_card.dev = dev; snd_soc_card_set_drvdata(&byt_cht_es8316_card, priv); From 5198baf8817d7e6e0fe2f3e74ea2ead714b74d9c Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Thu, 3 Jan 2019 14:45:35 +0100 Subject: [PATCH 60/67] ASoC: Intel: Add ACPI match table entry for ES8316 codec on BYTCR platform Some BYTCR devices use an ES8316 codec, add an ACPI match table entry for this. Signed-off-by: Hans de Goede Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/common/soc-acpi-intel-byt-match.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-byt-match.c b/sound/soc/intel/common/soc-acpi-intel-byt-match.c index 097dc06377ba..47a90909b956 100644 --- a/sound/soc/intel/common/soc-acpi-intel-byt-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-byt-match.c @@ -154,6 +154,15 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[] = { .sof_tplg_filename = "intel/sof-byt-da7213.tplg", .asoc_plat_name = "sst-mfld-platform", }, + { + .id = "ESSX8316", + .drv_name = "bytcht_es8316", + .fw_filename = "intel/fw_sst_0f28.bin", + .board = "bytcht_es8316", + .sof_fw_filename = "intel/sof-byt.ri", + .sof_tplg_filename = "intel/sof-byt-es8316.tplg", + .asoc_plat_name = "sst-mfld-platform", + }, /* some Baytrail platforms rely on RT5645, use CHT machine driver */ { .id = "10EC5645", From b97205ef95efddee018061dfee14c995be08dde3 Mon Sep 17 00:00:00 2001 From: Stephan Gerhold Date: Wed, 2 Jan 2019 20:39:03 +0100 Subject: [PATCH 61/67] ASoC: Intel: sst: Simplify is_byt_cr() is_byt_cr() and its usage can be simplified by returning the bool directly, instead of through a pointer. This works because the return value is just treated as bytcr = false and is not used otherwise. This patch also removes the extra check of IS_ENABLED(CONFIG_IOSF_MBI) in favor of checking iosf_mbi_available() directly. The header already takes care of returning false if the config option is not enabled. No functional change. Signed-off-by: Stephan Gerhold Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/atom/sst/sst_acpi.c | 33 ++++++++++++----------------- 1 file changed, 14 insertions(+), 19 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index 3a95ebbfc45d..9eaac450f864 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -255,18 +255,15 @@ static int is_byt(void) return status; } -static int is_byt_cr(struct device *dev, bool *bytcr) +static bool is_byt_cr(struct device *dev) { int status = 0; - if (IS_ENABLED(CONFIG_IOSF_MBI)) { + if (!is_byt()) + return false; + + if (iosf_mbi_available()) { u32 bios_status; - - if (!is_byt() || !iosf_mbi_available()) { - /* bail silently */ - return status; - } - status = iosf_mbi_read(BT_MBI_UNIT_PMC, /* 0x04 PUNIT */ MBI_REG_READ, /* 0x10 */ 0x006, /* BIOS_CONFIG */ @@ -278,15 +275,17 @@ static int is_byt_cr(struct device *dev, bool *bytcr) /* bits 26:27 mirror PMIC options */ bios_status = (bios_status >> 26) & 3; - if ((bios_status == 1) || (bios_status == 3)) - *bytcr = true; - else - dev_info(dev, "BYT-CR not detected\n"); + if (bios_status == 1 || bios_status == 3) { + dev_info(dev, "Detected Baytrail-CR platform\n"); + return true; + } + + dev_info(dev, "BYT-CR not detected\n"); } } else { - dev_info(dev, "IOSF_MBI not enabled, no BYT-CR detection\n"); + dev_info(dev, "IOSF_MBI not available, no BYT-CR detection\n"); } - return status; + return false; } @@ -301,7 +300,6 @@ static int sst_acpi_probe(struct platform_device *pdev) struct platform_device *plat_dev; struct sst_platform_info *pdata; unsigned int dev_id; - bool bytcr = false; id = acpi_match_device(dev->driver->acpi_match_table, dev); if (!id) @@ -333,10 +331,7 @@ static int sst_acpi_probe(struct platform_device *pdev) if (ret < 0) return ret; - ret = is_byt_cr(dev, &bytcr); - if (!(ret < 0 || !bytcr)) { - dev_info(dev, "Detected Baytrail-CR platform\n"); - + if (is_byt_cr(dev)) { /* override resource info */ byt_rvp_platform_data.res_info = &bytcr_res_info; } From fee15714552dbf420264da6f88dd813b8502592b Mon Sep 17 00:00:00 2001 From: Stephan Gerhold Date: Wed, 2 Jan 2019 20:39:06 +0100 Subject: [PATCH 62/67] ASoC: Intel: sst: Fallback to BYT-CR if IRQ 5 is missing Some devices detected as BYT-T by the PMIC-type based detection have only a single IRQ listed in the 80860F28 ACPI device. This causes -ENXIO later when attempting to get the IRQ at index 5. It turns out these devices behave more like BYT-CR devices, and using the IRQ at index 0 makes sound work correctly. This patch adds a fallback for these devices to is_byt_cr(): If there is no IRQ resource at index 5, treating the device as BYT-T is guaranteed to fail later, so we can safely treat these devices as BYT-CR without breaking any working device. Link: http://mailman.alsa-project.org/pipermail/alsa-devel/2018-December/143176.html Signed-off-by: Stephan Gerhold Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/atom/sst/sst_acpi.c | 16 ++++++++++++++-- 1 file changed, 14 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index 9eaac450f864..ae17ce4677a5 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -255,8 +255,9 @@ static int is_byt(void) return status; } -static bool is_byt_cr(struct device *dev) +static bool is_byt_cr(struct platform_device *pdev) { + struct device *dev = &pdev->dev; int status = 0; if (!is_byt()) @@ -285,6 +286,17 @@ static bool is_byt_cr(struct device *dev) } else { dev_info(dev, "IOSF_MBI not available, no BYT-CR detection\n"); } + + if (platform_get_resource(pdev, IORESOURCE_IRQ, 5) == NULL) { + /* + * Some devices detected as BYT-T have only a single IRQ listed, + * causing platform_get_irq with index 5 to return -ENXIO. + * The correct IRQ in this case is at index 0, as on BYT-CR. + */ + dev_info(dev, "Falling back to Baytrail-CR platform\n"); + return true; + } + return false; } @@ -331,7 +343,7 @@ static int sst_acpi_probe(struct platform_device *pdev) if (ret < 0) return ret; - if (is_byt_cr(dev)) { + if (is_byt_cr(pdev)) { /* override resource info */ byt_rvp_platform_data.res_info = &bytcr_res_info; } From 51a13e401a83ef37aa98c049c2c30bba885676c2 Mon Sep 17 00:00:00 2001 From: Stephan Gerhold Date: Wed, 2 Jan 2019 20:39:08 +0100 Subject: [PATCH 63/67] ASoC: Intel: bytcr_rt5640: Add quirks for ASUS MeMO Pad 7 (ME176C) Add quirks to select the correct input map, jack-detect options and channel map to make sound work on the ASUS MeMO Pad 7 (ME176C). Note: Although sound works out of the box, jack detection currently requires overriding the ACPI DSDT table. This is necessary because the rt5640 ACPI device (10EC5640) has the wrong GPIO listed as interrupt (one of the Bluetooth GPIOs). The correct GPIO is GPO2 0x0004 (listed as the first GPIO in the Intel(R) Audio Machine Driver - AMCR0F28 device). Signed-off-by: Stephan Gerhold Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5640.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index a22366ce33c4..ca8b4d5ff70f 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -428,6 +428,18 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "ME176C"), + }, + .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), From 2130f15d6cd9898a68dc7244084985353030514f Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 3 Jan 2019 14:53:45 +0200 Subject: [PATCH 64/67] ASoC: ti: davinci-mcasp: No need for IS_MODULE/BUILTIN check for pcm driver Since the platform drivers are selected by the DAI drivers (including McASP) there is no longer a need to check whether the modules are built-in or module. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/ti/davinci-mcasp.c | 16 ---------------- 1 file changed, 16 deletions(-) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index eeda6d5565bc..ee9ab58d4d0c 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -2149,26 +2149,10 @@ static int davinci_mcasp_probe(struct platform_device *pdev) ret = davinci_mcasp_get_dma_type(mcasp); switch (ret) { case PCM_EDMA: -#if IS_BUILTIN(CONFIG_SND_SOC_TI_EDMA_PCM) || \ - (IS_MODULE(CONFIG_SND_SOC_DAVINCI_MCASP) && \ - IS_MODULE(CONFIG_SND_SOC_TI_EDMA_PCM)) ret = edma_pcm_platform_register(&pdev->dev); -#else - dev_err(&pdev->dev, "Missing SND_EDMA_SOC\n"); - ret = -EINVAL; - goto err; -#endif break; case PCM_SDMA: -#if IS_BUILTIN(CONFIG_SND_SOC_TI_SDMA_PCM) || \ - (IS_MODULE(CONFIG_SND_SOC_DAVINCI_MCASP) && \ - IS_MODULE(CONFIG_SND_SOC_TI_SDMA_PCM)) ret = sdma_pcm_platform_register(&pdev->dev, NULL, NULL); -#else - dev_err(&pdev->dev, "Missing SND_SDMA_SOC\n"); - ret = -EINVAL; - goto err; -#endif break; default: dev_err(&pdev->dev, "No DMA controller found (%d)\n", ret); From 6175471755075d256c1c654151fc1cad183c1e33 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 3 Jan 2019 16:05:50 +0200 Subject: [PATCH 65/67] ASoC: ti: davinci-mcasp: Move context save/restore to runtime_pm callbacks McASP can loose it's context when runtime_pm is disabled. Save and restore the context when suspending and resuming the device. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/ti/davinci-mcasp.c | 136 +++++++++++++++++------------------ 1 file changed, 64 insertions(+), 72 deletions(-) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index eeda6d5565bc..a10fcb5963c6 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -108,7 +108,7 @@ struct davinci_mcasp { /* Used for comstraint setting on the second stream */ u32 channels; -#ifdef CONFIG_PM_SLEEP +#ifdef CONFIG_PM struct davinci_mcasp_context context; #endif @@ -1486,74 +1486,6 @@ static int davinci_mcasp_dai_probe(struct snd_soc_dai *dai) return 0; } -#ifdef CONFIG_PM_SLEEP -static int davinci_mcasp_suspend(struct snd_soc_dai *dai) -{ - struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); - struct davinci_mcasp_context *context = &mcasp->context; - u32 reg; - int i; - - context->pm_state = pm_runtime_active(mcasp->dev); - if (!context->pm_state) - pm_runtime_get_sync(mcasp->dev); - - for (i = 0; i < ARRAY_SIZE(context_regs); i++) - context->config_regs[i] = mcasp_get_reg(mcasp, context_regs[i]); - - if (mcasp->txnumevt) { - reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET; - context->afifo_regs[0] = mcasp_get_reg(mcasp, reg); - } - if (mcasp->rxnumevt) { - reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET; - context->afifo_regs[1] = mcasp_get_reg(mcasp, reg); - } - - for (i = 0; i < mcasp->num_serializer; i++) - context->xrsr_regs[i] = mcasp_get_reg(mcasp, - DAVINCI_MCASP_XRSRCTL_REG(i)); - - pm_runtime_put_sync(mcasp->dev); - - return 0; -} - -static int davinci_mcasp_resume(struct snd_soc_dai *dai) -{ - struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); - struct davinci_mcasp_context *context = &mcasp->context; - u32 reg; - int i; - - pm_runtime_get_sync(mcasp->dev); - - for (i = 0; i < ARRAY_SIZE(context_regs); i++) - mcasp_set_reg(mcasp, context_regs[i], context->config_regs[i]); - - if (mcasp->txnumevt) { - reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET; - mcasp_set_reg(mcasp, reg, context->afifo_regs[0]); - } - if (mcasp->rxnumevt) { - reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET; - mcasp_set_reg(mcasp, reg, context->afifo_regs[1]); - } - - for (i = 0; i < mcasp->num_serializer; i++) - mcasp_set_reg(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i), - context->xrsr_regs[i]); - - if (!context->pm_state) - pm_runtime_put_sync(mcasp->dev); - - return 0; -} -#else -#define davinci_mcasp_suspend NULL -#define davinci_mcasp_resume NULL -#endif - #define DAVINCI_MCASP_RATES SNDRV_PCM_RATE_8000_192000 #define DAVINCI_MCASP_PCM_FMTS (SNDRV_PCM_FMTBIT_S8 | \ @@ -1571,8 +1503,6 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = { { .name = "davinci-mcasp.0", .probe = davinci_mcasp_dai_probe, - .suspend = davinci_mcasp_suspend, - .resume = davinci_mcasp_resume, .playback = { .channels_min = 1, .channels_max = 32 * 16, @@ -1976,7 +1906,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) } mcasp->num_serializer = pdata->num_serializer; -#ifdef CONFIG_PM_SLEEP +#ifdef CONFIG_PM mcasp->context.xrsr_regs = devm_kcalloc(&pdev->dev, mcasp->num_serializer, sizeof(u32), GFP_KERNEL); @@ -2196,11 +2126,73 @@ static int davinci_mcasp_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_PM +static int davinci_mcasp_runtime_suspend(struct device *dev) +{ + struct davinci_mcasp *mcasp = dev_get_drvdata(dev); + struct davinci_mcasp_context *context = &mcasp->context; + u32 reg; + int i; + + for (i = 0; i < ARRAY_SIZE(context_regs); i++) + context->config_regs[i] = mcasp_get_reg(mcasp, context_regs[i]); + + if (mcasp->txnumevt) { + reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET; + context->afifo_regs[0] = mcasp_get_reg(mcasp, reg); + } + if (mcasp->rxnumevt) { + reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET; + context->afifo_regs[1] = mcasp_get_reg(mcasp, reg); + } + + for (i = 0; i < mcasp->num_serializer; i++) + context->xrsr_regs[i] = mcasp_get_reg(mcasp, + DAVINCI_MCASP_XRSRCTL_REG(i)); + + return 0; +} + +static int davinci_mcasp_runtime_resume(struct device *dev) +{ + struct davinci_mcasp *mcasp = dev_get_drvdata(dev); + struct davinci_mcasp_context *context = &mcasp->context; + u32 reg; + int i; + + for (i = 0; i < ARRAY_SIZE(context_regs); i++) + mcasp_set_reg(mcasp, context_regs[i], context->config_regs[i]); + + if (mcasp->txnumevt) { + reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET; + mcasp_set_reg(mcasp, reg, context->afifo_regs[0]); + } + if (mcasp->rxnumevt) { + reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET; + mcasp_set_reg(mcasp, reg, context->afifo_regs[1]); + } + + for (i = 0; i < mcasp->num_serializer; i++) + mcasp_set_reg(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i), + context->xrsr_regs[i]); + + return 0; +} + +#endif + +static const struct dev_pm_ops davinci_mcasp_pm_ops = { + SET_RUNTIME_PM_OPS(davinci_mcasp_runtime_suspend, + davinci_mcasp_runtime_resume, + NULL) +}; + static struct platform_driver davinci_mcasp_driver = { .probe = davinci_mcasp_probe, .remove = davinci_mcasp_remove, .driver = { .name = "davinci-mcasp", + .pm = &davinci_mcasp_pm_ops, .of_match_table = mcasp_dt_ids, }, }; From 4664b94c98b4f9fdd3845da41d5c65288e59c66c Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 3 Jan 2019 16:05:51 +0200 Subject: [PATCH 66/67] ASoC: davinci-mcasp: Document GPIO support McASP pins can be used as GPIO, add optional section to enable GPIO support for McASP. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- .../bindings/sound/davinci-mcasp-audio.txt | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt index b279b6072bd5..a58f79f5345c 100644 --- a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt +++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt @@ -45,6 +45,23 @@ Optional properties: - fck_parent : Should contain a valid clock name which will be used as parent for the McASP fck +Optional GPIO support: +If any McASP pin need to be used as GPIO then the McASP node must have: +... + gpio-controller + #gpio-cells = <2>; +... + +When requesting a GPIO, the first parameter is the PIN index in McASP_P* +registers. +For example to request the AXR2 pin of mcasp8: +function-gpios = <&mcasp8 2 0>; + +Or to request the ACLKR pin of mcasp8: +function-gpios = <&mcasp8 29 0>; + +For generic gpio information, please refer to bindings/gpio/gpio.txt + Example: mcasp0: mcasp0@1d00000 { From 540f1ba7b3a5596827a3bfeaae9c5e754347c933 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 3 Jan 2019 16:05:52 +0200 Subject: [PATCH 67/67] ASoC: ti: davinci-mcasp: Add support for GPIO mode of the pins All McASP pin can be configured as GPIO. Add gpiochip support for McASP and only enable it when the gpio-controller is present in the DT node. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/ti/davinci-mcasp.c | 159 ++++++++++++++++++++++++++++++++++- 1 file changed, 156 insertions(+), 3 deletions(-) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index a6a470a76900..a3a67a8f0f54 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -29,6 +29,7 @@ #include #include #include +#include #include #include @@ -54,6 +55,7 @@ static u32 context_regs[] = { DAVINCI_MCASP_AHCLKXCTL_REG, DAVINCI_MCASP_AHCLKRCTL_REG, DAVINCI_MCASP_PDIR_REG, + DAVINCI_MCASP_PFUNC_REG, DAVINCI_MCASP_RXMASK_REG, DAVINCI_MCASP_TXMASK_REG, DAVINCI_MCASP_RXTDM_REG, @@ -108,6 +110,10 @@ struct davinci_mcasp { /* Used for comstraint setting on the second stream */ u32 channels; +#ifdef CONFIG_GPIOLIB + struct gpio_chip gpio_chip; +#endif + #ifdef CONFIG_PM struct davinci_mcasp_context context; #endif @@ -818,9 +824,6 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream, if (mcasp->version < MCASP_VERSION_3) mcasp_set_bits(mcasp, DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT); - /* All PINS as McASP */ - mcasp_set_reg(mcasp, DAVINCI_MCASP_PFUNC_REG, 0x00000000); - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { mcasp_set_reg(mcasp, DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF); mcasp_clr_bits(mcasp, DAVINCI_MCASP_XEVTCTL_REG, TXDATADMADIS); @@ -1845,6 +1848,147 @@ static u32 davinci_mcasp_rxdma_offset(struct davinci_mcasp_pdata *pdata) return offset; } +#ifdef CONFIG_GPIOLIB +static int davinci_mcasp_gpio_request(struct gpio_chip *chip, unsigned offset) +{ + struct davinci_mcasp *mcasp = gpiochip_get_data(chip); + + if (mcasp->num_serializer && offset < mcasp->num_serializer && + mcasp->serial_dir[offset] != INACTIVE_MODE) { + dev_err(mcasp->dev, "AXR%u pin is used for audio\n", offset); + return -EBUSY; + } + + /* Do not change the PIN yet */ + + return pm_runtime_get_sync(mcasp->dev); +} + +static void davinci_mcasp_gpio_free(struct gpio_chip *chip, unsigned offset) +{ + struct davinci_mcasp *mcasp = gpiochip_get_data(chip); + + /* Set the direction to input */ + mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG, BIT(offset)); + + /* Set the pin as McASP pin */ + mcasp_clr_bits(mcasp, DAVINCI_MCASP_PFUNC_REG, BIT(offset)); + + pm_runtime_put_sync(mcasp->dev); +} + +static int davinci_mcasp_gpio_direction_out(struct gpio_chip *chip, + unsigned offset, int value) +{ + struct davinci_mcasp *mcasp = gpiochip_get_data(chip); + u32 val; + + if (value) + mcasp_set_bits(mcasp, DAVINCI_MCASP_PDOUT_REG, BIT(offset)); + else + mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDOUT_REG, BIT(offset)); + + val = mcasp_get_reg(mcasp, DAVINCI_MCASP_PFUNC_REG); + if (!(val & BIT(offset))) { + /* Set the pin as GPIO pin */ + mcasp_set_bits(mcasp, DAVINCI_MCASP_PFUNC_REG, BIT(offset)); + + /* Set the direction to output */ + mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, BIT(offset)); + } + + return 0; +} + +static void davinci_mcasp_gpio_set(struct gpio_chip *chip, unsigned offset, + int value) +{ + struct davinci_mcasp *mcasp = gpiochip_get_data(chip); + + if (value) + mcasp_set_bits(mcasp, DAVINCI_MCASP_PDOUT_REG, BIT(offset)); + else + mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDOUT_REG, BIT(offset)); +} + +static int davinci_mcasp_gpio_direction_in(struct gpio_chip *chip, + unsigned offset) +{ + struct davinci_mcasp *mcasp = gpiochip_get_data(chip); + u32 val; + + val = mcasp_get_reg(mcasp, DAVINCI_MCASP_PFUNC_REG); + if (!(val & BIT(offset))) { + /* Set the direction to input */ + mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG, BIT(offset)); + + /* Set the pin as GPIO pin */ + mcasp_set_bits(mcasp, DAVINCI_MCASP_PFUNC_REG, BIT(offset)); + } + + return 0; +} + +static int davinci_mcasp_gpio_get(struct gpio_chip *chip, unsigned offset) +{ + struct davinci_mcasp *mcasp = gpiochip_get_data(chip); + u32 val; + + val = mcasp_get_reg(mcasp, DAVINCI_MCASP_PDSET_REG); + if (val & BIT(offset)) + return 1; + + return 0; +} + +static int davinci_mcasp_gpio_get_direction(struct gpio_chip *chip, + unsigned offset) +{ + struct davinci_mcasp *mcasp = gpiochip_get_data(chip); + u32 val; + + val = mcasp_get_reg(mcasp, DAVINCI_MCASP_PDIR_REG); + if (val & BIT(offset)) + return 0; + + return 1; +} + +static const struct gpio_chip davinci_mcasp_template_chip = { + .owner = THIS_MODULE, + .request = davinci_mcasp_gpio_request, + .free = davinci_mcasp_gpio_free, + .direction_output = davinci_mcasp_gpio_direction_out, + .set = davinci_mcasp_gpio_set, + .direction_input = davinci_mcasp_gpio_direction_in, + .get = davinci_mcasp_gpio_get, + .get_direction = davinci_mcasp_gpio_get_direction, + .base = -1, + .ngpio = 32, +}; + +static int davinci_mcasp_init_gpiochip(struct davinci_mcasp *mcasp) +{ + if (!of_property_read_bool(mcasp->dev->of_node, "gpio-controller")) + return 0; + + mcasp->gpio_chip = davinci_mcasp_template_chip; + mcasp->gpio_chip.label = dev_name(mcasp->dev); + mcasp->gpio_chip.parent = mcasp->dev; +#ifdef CONFIG_OF_GPIO + mcasp->gpio_chip.of_node = mcasp->dev->of_node; +#endif + + return devm_gpiochip_add_data(mcasp->dev, &mcasp->gpio_chip, mcasp); +} + +#else /* CONFIG_GPIOLIB */ +static inline int davinci_mcasp_init_gpiochip(struct davinci_mcasp *mcasp) +{ + return 0; +} +#endif /* CONFIG_GPIOLIB */ + static int davinci_mcasp_probe(struct platform_device *pdev) { struct snd_dmaengine_dai_dma_data *dma_data; @@ -2069,6 +2213,15 @@ static int davinci_mcasp_probe(struct platform_device *pdev) mcasp_reparent_fck(pdev); + /* All PINS as McASP */ + pm_runtime_get_sync(mcasp->dev); + mcasp_set_reg(mcasp, DAVINCI_MCASP_PFUNC_REG, 0x00000000); + pm_runtime_put(mcasp->dev); + + ret = davinci_mcasp_init_gpiochip(mcasp); + if (ret) + goto err; + ret = devm_snd_soc_register_component(&pdev->dev, &davinci_mcasp_component, &davinci_mcasp_dai[pdata->op_mode], 1);