October 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [asoc:topic/arizona 4/5] sound/soc/codecs/cs47l24.c:1326:24: error: 'arizona' undeclared
by kbuild test robot 26 Oct '16

26 Oct '16

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/arizona head: 0809492e894f18cb19f8811f202e95d08e83b7af commit: 31833ead95c2c0a374f35a8ae8148c00459a0d49 [4/5] ASoC: arizona: Move request of speaker IRQs into bus probe config: i386-allmodconfig (attached as .config) compiler: gcc-6 (Debian 6.2.0-3) 6.2.0 20160901 reproduce: git checkout 31833ead95c2c0a374f35a8ae8148c00459a0d49 # save the attached .config to linux build tree make ARCH=i386 All errors (new ones prefixed by >>): sound/soc/codecs/cs47l24.c: In function 'cs47l24_remove': >> sound/soc/codecs/cs47l24.c:1326:24: error: 'arizona' undeclared (first use in this function) arizona_free_spk_irqs(arizona); ^~~~~~~ sound/soc/codecs/cs47l24.c:1326:24: note: each undeclared identifier is reported only once for each function it appears in vim +/arizona +1326 sound/soc/codecs/cs47l24.c 1320 snd_soc_unregister_codec(&pdev->dev); 1321 pm_runtime_disable(&pdev->dev); 1322 1323 wm_adsp2_remove(&cs47l24->core.adsp[1]); 1324 wm_adsp2_remove(&cs47l24->core.adsp[2]); 1325 > 1326 arizona_free_spk_irqs(arizona); 1327 1328 return 0; 1329 } --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH] Codec to codec dai link description
by anish kumar 26 Oct '16

26 Oct '16

Signed-off-by: anish kumar <yesanishhere(a)gmail.com> --- Documentation/sound/alsa/soc/codec_to_codec.txt | 114 ++++++++++++++++++++++++ 1 file changed, 114 insertions(+) create mode 100644 Documentation/sound/alsa/soc/codec_to_codec.txt diff --git a/Documentation/sound/alsa/soc/codec_to_codec.txt b/Documentation/sound/alsa/soc/codec_to_codec.txt new file mode 100644 index 0000000..b0f221d --- /dev/null +++ b/Documentation/sound/alsa/soc/codec_to_codec.txt @@ -0,0 +1,114 @@ +Creating codec to codec dai link for ALSA dapm +=================================================== + +Mostly the flow of audio is always from CPU to codec so your system +will look as below: + + ---------- --------- +| | dai | | + CPU -------> codec +| | | | + --------- --------- + +In case your system looks as below: + --------- + | | + codec-2 + | | + --------- + | + dai-2 + | + ---------- --------- +| | dai-1 | | + CPU -------> codec-1 +| | | | + ---------- --------- + | + dai-3 + | + --------- + | | + codec-3 + | | + --------- + +Suppose codec-2 is a bluetooth chip and codec-3 is connected to +a speaker and you have a below scenario: +codec-2 will receive the audio data and the user wants to play that +audio through codec-3 without involving the CPU.This +aforementioned case is the ideal case when codec to codec +connection should be used. + +Your dai_link should appear as below in your machine +file: + +static const struct snd_soc_pcm_stream dummy_params = { + .formats = SNDRV_PCM_FMTBIT_S24_LE, + .rate_min = 48000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, +}; + +{ + .name = "your_name", + .stream_name = "your_stream_name", + .cpu_dai_name = "snd-soc-dummy-dai", + .codec_name = "codec-2, + .codec_dai_name = "codec-2-dai_name", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .ignore_suspend = 1, + .params = &dummy_params, +}, +{ + .name = "your_name", + .stream_name = "your_stream_name", + .cpu_dai_name = "snd-soc-dummy-dai", + .codec_name = "codec-3, + .codec_dai_name = "codec-3-dai_name", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .ignore_suspend = 1, + .params = &dummy_params, +}, + +Note the "params" callback which lets the dapm know that this +dai_link is a codec to codec connection. +Also, in above code cpu_dai should be replaced with your actual +cpu dai but in case you don't have a actual cpu dai then dummy will +do. + +You can browse the speyside.c for an actual example code in mainline. + +Note that in current device tree there is no way to mark a dai_link +as codec to codec. However, it may change in future. + +In dapm core a route is created between cpu_dai playback widget +and codec_dai capture widget for playback path and vice-versa is +true for capture path. In order for this aforementioned route to get +triggered, DAPM needs to find a valid endpoint which could be either +a sink or source widget corresponding to playback and capture path +respectively. + +Below is what you can use it to trigger the widgets provided you have +stream name ending with "Playback" and "Capture" for cpu and +codec dai's. + +static const struct snd_soc_dapm_widget aif_dapm_widgets[] = { + SND_SOC_DAPM_SPK("dummyspk", NULL), + SND_SOC_DAPM_MIC("dummymic", NULL), +}; + +static const struct snd_soc_dapm_route audio_i2s_map[] = { + {"dummyspk", NULL, "Playback"}, + {"Capture", NULL, "dummymic"}, +}; + +Above code is good for quick testing but in order to mainline it +you are expected to create a thin codec driver for the speaker +amp rather than doing this sort of thing, as that at least +sets appropriate constraints for the device even if it needs +no control. For an example of such a driver you can see: +sound/soc/codecs/wm8727.c -- 2.8.4 (Apple Git-73)

3 5

[alsa-devel] [PATCH 0/7] ASoC: sti audio fixes and legacy cleaning
by Arnaud Pouliquen 26 Oct '16

26 Oct '16

Audio fixes and cleaning of code associated to stih416 platform. Arnaud Pouliquen (7): ASoC: sti: fix errors management ASoC: sti: fix channel status update after playback start ASoC: sti: reset refactoring ASoC: sti: clean unused include ASoC: codec: clean legacy in sti-sas ASoC: sti: sti-sas: add missing return in messages strings ASoc: codec: sti enable fast io for regmap sound/soc/codecs/sti-sas.c | 181 +++++----------------------------------- sound/soc/sti/sti_uniperif.c | 43 ++++++++-- sound/soc/sti/uniperif.h | 2 + sound/soc/sti/uniperif_player.c | 97 +++++++++------------ sound/soc/sti/uniperif_reader.c | 41 +++------ 5 files changed, 111 insertions(+), 253 deletions(-) -- 1.9.1

2 15

[alsa-devel] [PATCH 1/2] ASoC: arizona: Move request of speaker IRQs into bus probe
by Charles Keepax 26 Oct '16

26 Oct '16

It is more idiomatic to request all resources in the bus level probe, this patch moves the request of the speaker thermal event IRQs from the ASoC level probe into the bus level probe. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 17 +++++++++++------ sound/soc/codecs/arizona.h | 3 ++- sound/soc/codecs/cs47l24.c | 19 +++++++++++++++---- sound/soc/codecs/wm5102.c | 20 ++++++++++++++++---- sound/soc/codecs/wm5110.c | 20 ++++++++++++++++---- sound/soc/codecs/wm8997.c | 26 +++++++++++++++++++++----- sound/soc/codecs/wm8998.c | 28 +++++++++++++++++++++++----- 7 files changed, 104 insertions(+), 29 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 7f7d4b3..99ce6a0e 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -191,6 +191,14 @@ int arizona_init_spk(struct snd_soc_codec *codec) break; } + return 0; +} +EXPORT_SYMBOL_GPL(arizona_init_spk); + +int arizona_init_spk_irqs(struct arizona *arizona) +{ + int ret; + ret = arizona_request_irq(arizona, ARIZONA_IRQ_SPK_OVERHEAT_WARN, "Thermal warning", arizona_thermal_warn, arizona); @@ -209,19 +217,16 @@ int arizona_init_spk(struct snd_soc_codec *codec) return 0; } -EXPORT_SYMBOL_GPL(arizona_init_spk); +EXPORT_SYMBOL_GPL(arizona_init_spk_irqs); -int arizona_free_spk(struct snd_soc_codec *codec) +int arizona_free_spk_irqs(struct arizona *arizona) { - struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); - struct arizona *arizona = priv->arizona; - arizona_free_irq(arizona, ARIZONA_IRQ_SPK_OVERHEAT_WARN, arizona); arizona_free_irq(arizona, ARIZONA_IRQ_SPK_OVERHEAT, arizona); return 0; } -EXPORT_SYMBOL_GPL(arizona_free_spk); +EXPORT_SYMBOL_GPL(arizona_free_spk_irqs); static const struct snd_soc_dapm_route arizona_mono_routes[] = { { "OUT1R", NULL, "OUT1L" }, diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 9d0997b..34d65d1 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -317,7 +317,8 @@ extern int arizona_init_gpio(struct snd_soc_codec *codec); extern int arizona_init_mono(struct snd_soc_codec *codec); extern int arizona_init_notifiers(struct snd_soc_codec *codec); -extern int arizona_free_spk(struct snd_soc_codec *codec); +extern int arizona_init_spk_irqs(struct arizona *arizona); +extern int arizona_free_spk_irqs(struct arizona *arizona); extern int arizona_init_dai(struct arizona_priv *priv, int dai); diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c index 7df6a67..17e52c1 100644 --- a/sound/soc/codecs/cs47l24.c +++ b/sound/soc/codecs/cs47l24.c @@ -1170,8 +1170,6 @@ static int cs47l24_codec_remove(struct snd_soc_codec *codec) arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, priv); - arizona_free_spk(codec); - return 0; } @@ -1287,20 +1285,31 @@ static int cs47l24_probe(struct platform_device *pdev) pm_runtime_enable(&pdev->dev); pm_runtime_idle(&pdev->dev); + ret = arizona_init_spk_irqs(arizona); + if (ret < 0) + return ret; + ret = snd_soc_register_platform(&pdev->dev, &cs47l24_compr_platform); if (ret < 0) { dev_err(&pdev->dev, "Failed to register platform: %d\n", ret); - return ret; + goto err_spk_irqs; } ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_cs47l24, cs47l24_dai, ARRAY_SIZE(cs47l24_dai)); if (ret < 0) { dev_err(&pdev->dev, "Failed to register codec: %d\n", ret); - snd_soc_unregister_platform(&pdev->dev); + goto err_platform; } return ret; + +err_platform: + snd_soc_unregister_platform(&pdev->dev); +err_spk_irqs: + arizona_free_spk_irqs(arizona); + + return ret; } static int cs47l24_remove(struct platform_device *pdev) @@ -1314,6 +1323,8 @@ static int cs47l24_remove(struct platform_device *pdev) wm_adsp2_remove(&cs47l24->core.adsp[1]); wm_adsp2_remove(&cs47l24->core.adsp[2]); + arizona_free_spk_irqs(arizona); + return 0; } diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index bb3de5b..6f40ee6 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1978,8 +1978,6 @@ static int wm5102_codec_remove(struct snd_soc_codec *codec) arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, priv); - arizona_free_spk(codec); - return 0; } @@ -2097,25 +2095,37 @@ static int wm5102_probe(struct platform_device *pdev) pm_runtime_enable(&pdev->dev); pm_runtime_idle(&pdev->dev); + ret = arizona_init_spk_irqs(arizona); + if (ret < 0) + return ret; + ret = snd_soc_register_platform(&pdev->dev, &wm5102_compr_platform); if (ret < 0) { dev_err(&pdev->dev, "Failed to register platform: %d\n", ret); - return ret; + goto err_spk_irqs; } ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm5102, wm5102_dai, ARRAY_SIZE(wm5102_dai)); if (ret < 0) { dev_err(&pdev->dev, "Failed to register codec: %d\n", ret); - snd_soc_unregister_platform(&pdev->dev); + goto err_platform; } return ret; + +err_platform: + snd_soc_unregister_platform(&pdev->dev); +err_spk_irqs: + arizona_free_spk_irqs(arizona); + + return ret; } static int wm5102_remove(struct platform_device *pdev) { struct wm5102_priv *wm5102 = platform_get_drvdata(pdev); + struct arizona *arizona = wm5102->core.arizona; snd_soc_unregister_platform(&pdev->dev); snd_soc_unregister_codec(&pdev->dev); @@ -2123,6 +2133,8 @@ static int wm5102_remove(struct platform_device *pdev) wm_adsp2_remove(&wm5102->core.adsp[0]); + arizona_free_spk_irqs(arizona); + return 0; } diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 407dc4a..667fc25 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -2330,8 +2330,6 @@ static int wm5110_codec_remove(struct snd_soc_codec *codec) arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, priv); - arizona_free_spk(codec); - return 0; } @@ -2453,25 +2451,37 @@ static int wm5110_probe(struct platform_device *pdev) pm_runtime_enable(&pdev->dev); pm_runtime_idle(&pdev->dev); + ret = arizona_init_spk_irqs(arizona); + if (ret < 0) + return ret; + ret = snd_soc_register_platform(&pdev->dev, &wm5110_compr_platform); if (ret < 0) { dev_err(&pdev->dev, "Failed to register platform: %d\n", ret); - return ret; + goto err_spk_irqs; } ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm5110, wm5110_dai, ARRAY_SIZE(wm5110_dai)); if (ret < 0) { dev_err(&pdev->dev, "Failed to register codec: %d\n", ret); - snd_soc_unregister_platform(&pdev->dev); + goto err_platform; } return ret; + +err_platform: + snd_soc_unregister_platform(&pdev->dev); +err_spk_irqs: + arizona_free_spk_irqs(arizona); + + return ret; } static int wm5110_remove(struct platform_device *pdev) { struct wm5110_priv *wm5110 = platform_get_drvdata(pdev); + struct arizona *arizona = wm5110->core.arizona; int i; snd_soc_unregister_platform(&pdev->dev); @@ -2481,6 +2491,8 @@ static int wm5110_remove(struct platform_device *pdev) for (i = 0; i < WM5110_NUM_ADSP; i++) wm_adsp2_remove(&wm5110->core.adsp[i]); + arizona_free_spk_irqs(arizona); + return 0; } diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index f5f5d43..600595c 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -1077,8 +1077,6 @@ static int wm8997_codec_remove(struct snd_soc_codec *codec) priv->core.arizona->dapm = NULL; - arizona_free_spk(codec); - return 0; } @@ -1124,7 +1122,7 @@ static int wm8997_probe(struct platform_device *pdev) { struct arizona *arizona = dev_get_drvdata(pdev->dev.parent); struct wm8997_priv *wm8997; - int i; + int i, ret; wm8997 = devm_kzalloc(&pdev->dev, sizeof(struct wm8997_priv), GFP_KERNEL); @@ -1164,15 +1162,33 @@ static int wm8997_probe(struct platform_device *pdev) pm_runtime_enable(&pdev->dev); pm_runtime_idle(&pdev->dev); - return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm8997, - wm8997_dai, ARRAY_SIZE(wm8997_dai)); + ret = arizona_init_spk_irqs(arizona); + if (ret < 0) + return ret; + + ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm8997, + wm8997_dai, ARRAY_SIZE(wm8997_dai)); + if (ret < 0) { + dev_err(&pdev->dev, "Failed to register codec: %d\n", ret); + goto err_spk_irqs; + } + +err_spk_irqs: + arizona_free_spk_irqs(arizona); + + return ret; } static int wm8997_remove(struct platform_device *pdev) { + struct wm8997_priv *wm8997 = platform_get_drvdata(pdev); + struct arizona *arizona = wm8997->core.arizona; + snd_soc_unregister_codec(&pdev->dev); pm_runtime_disable(&pdev->dev); + arizona_free_spk_irqs(arizona); + return 0; } diff --git a/sound/soc/codecs/wm8998.c b/sound/soc/codecs/wm8998.c index 3eaa361..4cccaae 100644 --- a/sound/soc/codecs/wm8998.c +++ b/sound/soc/codecs/wm8998.c @@ -1337,8 +1337,6 @@ static int wm8998_codec_remove(struct snd_soc_codec *codec) priv->core.arizona->dapm = NULL; - arizona_free_spk(codec); - return 0; } @@ -1387,7 +1385,7 @@ static int wm8998_probe(struct platform_device *pdev) { struct arizona *arizona = dev_get_drvdata(pdev->dev.parent); struct wm8998_priv *wm8998; - int i; + int i, ret; wm8998 = devm_kzalloc(&pdev->dev, sizeof(struct wm8998_priv), GFP_KERNEL); @@ -1419,15 +1417,35 @@ static int wm8998_probe(struct platform_device *pdev) pm_runtime_enable(&pdev->dev); pm_runtime_idle(&pdev->dev); - return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm8998, - wm8998_dai, ARRAY_SIZE(wm8998_dai)); + ret = arizona_init_spk_irqs(arizona); + if (ret < 0) + return ret; + + ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm8998, + wm8998_dai, ARRAY_SIZE(wm8998_dai)); + if (ret < 0) { + dev_err(&pdev->dev, "Failed to register codec: %d\n", ret); + goto err_spk_irqs; + } + + return ret; + +err_spk_irqs: + arizona_free_spk_irqs(arizona); + + return ret; } static int wm8998_remove(struct platform_device *pdev) { + struct wm8998_priv *wm8998 = platform_get_drvdata(pdev); + struct arizona *arizona = wm8998->core.arizona; + snd_soc_unregister_codec(&pdev->dev); pm_runtime_disable(&pdev->dev); + arizona_free_spk_irqs(arizona); + return 0; } -- 2.1.4

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[alsa-devel] [patch] ASoC: Intel: Skylake: Fix a shift wrapping bug
by Dan Carpenter 26 Oct '16

26 Oct '16

"*val" is a u64. It definitely looks like we intend to use the high 32 bits as well. Fixes: 700a9a63f9c1 ("ASoC: Intel: Skylake: Add module instance id generation APIs") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/intel/skylake/skl-sst-utils.c b/sound/soc/intel/skylake/skl-sst-utils.c index 8dc0303..ea162fb 100644 --- a/sound/soc/intel/skylake/skl-sst-utils.c +++ b/sound/soc/intel/skylake/skl-sst-utils.c @@ -179,7 +179,7 @@ static inline int skl_getid_32(struct uuid_module *module, u64 *val, index = ffz(mask_val); pvt_id = index + word1_mask + word2_mask; if (pvt_id <= (max_inst - 1)) { - *val |= 1 << (index + word1_mask); + *val |= 1ULL << (index + word1_mask); return pvt_id; } }

3 2

[alsa-devel] [PATCH] SoC: mxs-saif: check validity of ids in mxs_saif_probe()
by Alexey Khoroshilov 26 Oct '16

26 Oct '16

There is a check for validity of one of ids in mxs_saif_probe(), while array dereferece is made by the other id. The patch adds the check for the second saif id. Found by Linux Driver Verification project (linuxtesting.org). Signed-off-by: Alexey Khoroshilov <khoroshilov(a)ispras.ru> --- sound/soc/mxs/mxs-saif.c | 13 +++++++++---- 1 file changed, 9 insertions(+), 4 deletions(-) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 13631003cb7c..a002ab892772 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -735,6 +735,11 @@ static int mxs_saif_probe(struct platform_device *pdev) else saif->id = ret; + if (saif->id >= ARRAY_SIZE(mxs_saif)) { + dev_err(&pdev->dev, "get wrong saif id\n"); + return -EINVAL; + } + /* * If there is no "fsl,saif-master" phandle, it's a saif * master. Otherwise, it's a slave and its phandle points @@ -749,11 +754,11 @@ static int mxs_saif_probe(struct platform_device *pdev) return ret; else saif->master_id = ret; - } - if (saif->master_id >= ARRAY_SIZE(mxs_saif)) { - dev_err(&pdev->dev, "get wrong master id\n"); - return -EINVAL; + if (saif->master_id >= ARRAY_SIZE(mxs_saif)) { + dev_err(&pdev->dev, "get wrong master id\n"); + return -EINVAL; + } } mxs_saif[saif->id] = saif; -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: cs35l34: Remove CS35L34_CHIP_ID from cs35l34_readable_register
by Axel Lin 25 Oct '16

25 Oct '16

CS35L34_CHIP_ID is not a register address, it's the value read from CS35L34_DEVID_AB/CD/E registers. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs35l34.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/cs35l34.c b/sound/soc/codecs/cs35l34.c index 4045383..e0f672a 100644 --- a/sound/soc/codecs/cs35l34.c +++ b/sound/soc/codecs/cs35l34.c @@ -145,7 +145,6 @@ static bool cs35l34_volatile_register(struct device *dev, unsigned int reg) static bool cs35l34_readable_register(struct device *dev, unsigned int reg) { switch (reg) { - case CS35L34_CHIP_ID: case CS35L34_DEVID_AB: case CS35L34_DEVID_CD: case CS35L34_DEVID_E: -- 2.7.4

2 1

[alsa-devel] [PATCH 1/3] ASoC: Drop SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME flag
by Sylwester Nawrocki 25 Oct '16

25 Oct '16

Since commit 194c7dea00c68c1b1f8ff26304fa937a006f66dd "ASoC: dmaengine: add custom DMA config to snd_dmaengine_pcm_config" custom DMA channels can be also specified in chan_names[] field of struct snd_dmaengine_pcm_config. This patch drops chan_name field of struct snd_dmaengine_dai_dma_data and makes soc-generic-dmaengine-pcm module using config->chan_names instead. Remaining users of struct snd_dmaengine_dai_dma_data::chan_name are updated to not require that field. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- include/sound/dmaengine_pcm.h | 6 ------ sound/soc/samsung/dmaengine.c | 8 ++------ sound/soc/samsung/i2s.c | 3 --- sound/soc/samsung/s3c2412-i2s.c | 2 -- sound/soc/samsung/s3c24xx-i2s.c | 2 -- sound/soc/soc-generic-dmaengine-pcm.c | 12 +++--------- 6 files changed, 5 insertions(+), 28 deletions(-) diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index 67be244..1c8f9e1 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -71,7 +71,6 @@ struct dma_chan *snd_dmaengine_pcm_request_channel(dma_filter_fn filter_fn, * @slave_id: Slave requester id for the DMA channel. * @filter_data: Custom DMA channel filter data, this will usually be used when * requesting the DMA channel. - * @chan_name: Custom channel name to use when requesting DMA channel. * @fifo_size: FIFO size of the DAI controller in bytes * @flags: PCM_DAI flags, only SND_DMAENGINE_PCM_DAI_FLAG_PACK for now */ @@ -81,7 +80,6 @@ struct snd_dmaengine_dai_dma_data { u32 maxburst; unsigned int slave_id; void *filter_data; - const char *chan_name; unsigned int fifo_size; unsigned int flags; }; @@ -107,10 +105,6 @@ void snd_dmaengine_pcm_set_config_from_dai_data( * playback. */ #define SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX BIT(3) -/* - * The PCM streams have custom channel names specified. - */ -#define SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME BIT(4) /** * struct snd_dmaengine_pcm_config - Configuration data for dmaengine based PCM diff --git a/sound/soc/samsung/dmaengine.c b/sound/soc/samsung/dmaengine.c index 9104c98..cda656e 100644 --- a/sound/soc/samsung/dmaengine.c +++ b/sound/soc/samsung/dmaengine.c @@ -37,12 +37,8 @@ int samsung_asoc_dma_platform_register(struct device *dev, dma_filter_fn filter, pcm_conf->prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config; pcm_conf->compat_filter_fn = filter; - if (dev->of_node) { - pcm_conf->chan_names[SNDRV_PCM_STREAM_PLAYBACK] = tx; - pcm_conf->chan_names[SNDRV_PCM_STREAM_CAPTURE] = rx; - } else { - flags |= SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME; - } + pcm_conf->chan_names[SNDRV_PCM_STREAM_PLAYBACK] = tx; + pcm_conf->chan_names[SNDRV_PCM_STREAM_CAPTURE] = rx; return devm_snd_dmaengine_pcm_register(dev, pcm_conf, flags); } diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 588aa01..e06aa31 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1305,8 +1305,6 @@ static int samsung_i2s_probe(struct platform_device *pdev) } pri_dai->dma_playback.addr = regs_base + I2STXD; pri_dai->dma_capture.addr = regs_base + I2SRXD; - pri_dai->dma_playback.chan_name = "tx"; - pri_dai->dma_capture.chan_name = "rx"; pri_dai->dma_playback.addr_width = 4; pri_dai->dma_capture.addr_width = 4; pri_dai->quirks = quirks; @@ -1326,7 +1324,6 @@ static int samsung_i2s_probe(struct platform_device *pdev) sec_dai->lock = &pri_dai->spinlock; sec_dai->variant_regs = pri_dai->variant_regs; sec_dai->dma_playback.addr = regs_base + I2STXDS; - sec_dai->dma_playback.chan_name = "tx-sec"; if (!np) { sec_dai->dma_playback.filter_data = i2s_pdata->dma_play_sec; diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index 3e89fbc..40331a0 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -35,12 +35,10 @@ #include <linux/platform_data/asoc-s3c.h> static struct snd_dmaengine_dai_dma_data s3c2412_i2s_pcm_stereo_out = { - .chan_name = "tx", .addr_width = 4, }; static struct snd_dmaengine_dai_dma_data s3c2412_i2s_pcm_stereo_in = { - .chan_name = "rx", .addr_width = 4, }; diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index c78a936..d59a47d 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -33,12 +33,10 @@ #include <linux/platform_data/asoc-s3c.h> static struct snd_dmaengine_dai_dma_data s3c24xx_i2s_pcm_stereo_out = { - .chan_name = "tx", .addr_width = 2, }; static struct snd_dmaengine_dai_dma_data s3c24xx_i2s_pcm_stereo_in = { - .chan_name = "rx", .addr_width = 2, }; diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 6cef397..f2d372d 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -263,7 +263,6 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform); const struct snd_dmaengine_pcm_config *config = pcm->config; struct device *dev = rtd->platform->dev; - struct snd_dmaengine_dai_dma_data *dma_data; struct snd_pcm_substream *substream; size_t prealloc_buffer_size; size_t max_buffer_size; @@ -284,12 +283,9 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) if (!substream) continue; - dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - - if (!pcm->chan[i] && - (pcm->flags & SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME)) + if (!pcm->chan[i] && config->chan_names[i]) pcm->chan[i] = dma_request_slave_channel(dev, - dma_data->chan_name); + config->chan_names[i]); if (!pcm->chan[i] && (pcm->flags & SND_DMAENGINE_PCM_FLAG_COMPAT)) { pcm->chan[i] = dmaengine_pcm_compat_request_channel(rtd, @@ -359,9 +355,7 @@ static int dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, const char *name; struct dma_chan *chan; - if ((pcm->flags & (SND_DMAENGINE_PCM_FLAG_NO_DT | - SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME)) || - !dev->of_node) + if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_NO_DT) || !dev->of_node) return 0; if (config && config->dma_dev) { -- 1.9.1

3 6

[alsa-devel] [PATCH] ASoC: tlv320aic31xx: add explicit support for tlv320dac31xx
by Nikita Yushchenko 25 Oct '16

25 Oct '16

tlv320dac31xx is a subset of tlv320aic31xx: - it does not have MIC inputs and ADC, thus capture is not supported, - it has analog inputs AIN1/AIN2 that can be mixed into output. Although tlv320dac31xx does work with tlv320aic31xx driver, this setup does register non-existent widgets and non-existent capture stream. Thus userspace lists non-existent objects in user interfaces, an can access these, causing operations with device registers that are declared as "reserved" in tlv320dac31xx datasheet. This patch fixes this situation by separating controls/widgets/routes into common, aic31xx-specific, and dac31xx-specific parts. Only parts that match actual hardware (as declared in "compatible" device tree property) are registered. Signed-off-by: Nikita Yushchenko <nikita.yoush(a)cogentembedded.com> --- sound/soc/codecs/tlv320aic31xx.c | 212 ++++++++++++++++++++++++++++----------- sound/soc/codecs/tlv320aic31xx.h | 2 + 2 files changed, 158 insertions(+), 56 deletions(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 3c5e1df..fb7d648 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -273,10 +273,20 @@ static const DECLARE_TLV_DB_SCALE(sp_vol_tlv, -6350, 50, 0); /* * controls to be exported to the user space */ -static const struct snd_kcontrol_new aic31xx_snd_controls[] = { +static const struct snd_kcontrol_new common31xx_snd_controls[] = { SOC_DOUBLE_R_S_TLV("DAC Playback Volume", AIC31XX_LDACVOL, AIC31XX_RDACVOL, 0, -127, 48, 7, 0, dac_vol_tlv), + SOC_DOUBLE_R("HP Driver Playback Switch", AIC31XX_HPLGAIN, + AIC31XX_HPRGAIN, 2, 1, 0), + SOC_DOUBLE_R_TLV("HP Driver Playback Volume", AIC31XX_HPLGAIN, + AIC31XX_HPRGAIN, 3, 0x09, 0, hp_drv_tlv), + + SOC_DOUBLE_R_TLV("HP Analog Playback Volume", AIC31XX_LANALOGHPL, + AIC31XX_RANALOGHPR, 0, 0x7F, 1, hp_vol_tlv), +}; + +static const struct snd_kcontrol_new aic31xx_snd_controls[] = { SOC_SINGLE_TLV("ADC Fine Capture Volume", AIC31XX_ADCFGA, 4, 4, 1, adc_fgain_tlv), @@ -286,14 +296,6 @@ static const struct snd_kcontrol_new aic31xx_snd_controls[] = { SOC_SINGLE_TLV("Mic PGA Capture Volume", AIC31XX_MICPGA, 0, 119, 0, mic_pga_tlv), - - SOC_DOUBLE_R("HP Driver Playback Switch", AIC31XX_HPLGAIN, - AIC31XX_HPRGAIN, 2, 1, 0), - SOC_DOUBLE_R_TLV("HP Driver Playback Volume", AIC31XX_HPLGAIN, - AIC31XX_HPRGAIN, 3, 0x09, 0, hp_drv_tlv), - - SOC_DOUBLE_R_TLV("HP Analog Playback Volume", AIC31XX_LANALOGHPL, - AIC31XX_RANALOGHPR, 0, 0x7F, 1, hp_vol_tlv), }; static const struct snd_kcontrol_new aic311x_snd_controls[] = { @@ -397,17 +399,28 @@ static int aic31xx_dapm_power_event(struct snd_soc_dapm_widget *w, return 0; } -static const struct snd_kcontrol_new left_output_switches[] = { +static const struct snd_kcontrol_new aic31xx_left_output_switches[] = { SOC_DAPM_SINGLE("From Left DAC", AIC31XX_DACMIXERROUTE, 6, 1, 0), SOC_DAPM_SINGLE("From MIC1LP", AIC31XX_DACMIXERROUTE, 5, 1, 0), SOC_DAPM_SINGLE("From MIC1RP", AIC31XX_DACMIXERROUTE, 4, 1, 0), }; -static const struct snd_kcontrol_new right_output_switches[] = { +static const struct snd_kcontrol_new aic31xx_right_output_switches[] = { SOC_DAPM_SINGLE("From Right DAC", AIC31XX_DACMIXERROUTE, 2, 1, 0), SOC_DAPM_SINGLE("From MIC1RP", AIC31XX_DACMIXERROUTE, 1, 1, 0), }; +static const struct snd_kcontrol_new dac31xx_left_output_switches[] = { + SOC_DAPM_SINGLE("From Left DAC", AIC31XX_DACMIXERROUTE, 6, 1, 0), + SOC_DAPM_SINGLE("From AIN1", AIC31XX_DACMIXERROUTE, 5, 1, 0), + SOC_DAPM_SINGLE("From AIN2", AIC31XX_DACMIXERROUTE, 4, 1, 0), +}; + +static const struct snd_kcontrol_new dac31xx_right_output_switches[] = { + SOC_DAPM_SINGLE("From Right DAC", AIC31XX_DACMIXERROUTE, 2, 1, 0), + SOC_DAPM_SINGLE("From AIN2", AIC31XX_DACMIXERROUTE, 1, 1, 0), +}; + static const struct snd_kcontrol_new p_term_mic1lp = SOC_DAPM_ENUM("MIC1LP P-Terminal", mic1lp_p_enum); @@ -457,7 +470,7 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w, return 0; } -static const struct snd_soc_dapm_widget aic31xx_dapm_widgets[] = { +static const struct snd_soc_dapm_widget common31xx_dapm_widgets[] = { SND_SOC_DAPM_AIF_IN("DAC IN", "DAC Playback", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_MUX("DAC Left Input", @@ -473,14 +486,7 @@ static const struct snd_soc_dapm_widget aic31xx_dapm_widgets[] = { AIC31XX_DACSETUP, 6, 0, aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), - /* Output Mixers */ - SND_SOC_DAPM_MIXER("Output Left", SND_SOC_NOPM, 0, 0, - left_output_switches, - ARRAY_SIZE(left_output_switches)), - SND_SOC_DAPM_MIXER("Output Right", SND_SOC_NOPM, 0, 0, - right_output_switches, - ARRAY_SIZE(right_output_switches)), - + /* HP */ SND_SOC_DAPM_SWITCH("HP Left", SND_SOC_NOPM, 0, 0, &aic31xx_dapm_hpl_switch), SND_SOC_DAPM_SWITCH("HP Right", SND_SOC_NOPM, 0, 0, @@ -494,10 +500,34 @@ static const struct snd_soc_dapm_widget aic31xx_dapm_widgets[] = { NULL, 0, aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_POST_PMU), - /* ADC */ - SND_SOC_DAPM_ADC_E("ADC", "Capture", AIC31XX_ADCSETUP, 7, 0, - aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU | - SND_SOC_DAPM_POST_PMD), + /* Mic Bias */ + SND_SOC_DAPM_SUPPLY("MICBIAS", SND_SOC_NOPM, 0, 0, mic_bias_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), +}; + +static const struct snd_soc_dapm_widget dac31xx_dapm_widgets[] = { + /* Inputs */ + SND_SOC_DAPM_INPUT("AIN1"), + SND_SOC_DAPM_INPUT("AIN2"), + + /* Output Mixers */ + SND_SOC_DAPM_MIXER("Output Left", SND_SOC_NOPM, 0, 0, + dac31xx_left_output_switches, + ARRAY_SIZE(dac31xx_left_output_switches)), + SND_SOC_DAPM_MIXER("Output Right", SND_SOC_NOPM, 0, 0, + dac31xx_right_output_switches, + ARRAY_SIZE(dac31xx_right_output_switches)), +}; + +static const struct snd_soc_dapm_widget aic31xx_dapm_widgets[] = { + /* Inputs */ + SND_SOC_DAPM_INPUT("MIC1LP"), + SND_SOC_DAPM_INPUT("MIC1RP"), + SND_SOC_DAPM_INPUT("MIC1LM"), /* Input Selection to MIC_PGA */ SND_SOC_DAPM_MUX("MIC1LP P-Terminal", SND_SOC_NOPM, 0, 0, @@ -507,24 +537,25 @@ static const struct snd_soc_dapm_widget aic31xx_dapm_widgets[] = { SND_SOC_DAPM_MUX("MIC1LM P-Terminal", SND_SOC_NOPM, 0, 0, &p_term_mic1lm), + /* ADC */ + SND_SOC_DAPM_ADC_E("ADC", "Capture", AIC31XX_ADCSETUP, 7, 0, + aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX("MIC1LM M-Terminal", SND_SOC_NOPM, 0, 0, &m_term_mic1lm), + /* Enabling & Disabling MIC Gain Ctl */ SND_SOC_DAPM_PGA("MIC_GAIN_CTL", AIC31XX_MICPGA, 7, 1, NULL, 0), - /* Mic Bias */ - SND_SOC_DAPM_SUPPLY("MICBIAS", SND_SOC_NOPM, 0, 0, mic_bias_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), - - /* Outputs */ - SND_SOC_DAPM_OUTPUT("HPL"), - SND_SOC_DAPM_OUTPUT("HPR"), - - /* Inputs */ - SND_SOC_DAPM_INPUT("MIC1LP"), - SND_SOC_DAPM_INPUT("MIC1RP"), - SND_SOC_DAPM_INPUT("MIC1LM"), + /* Output Mixers */ + SND_SOC_DAPM_MIXER("Output Left", SND_SOC_NOPM, 0, 0, + aic31xx_left_output_switches, + ARRAY_SIZE(aic31xx_left_output_switches)), + SND_SOC_DAPM_MIXER("Output Right", SND_SOC_NOPM, 0, 0, + aic31xx_right_output_switches, + ARRAY_SIZE(aic31xx_right_output_switches)), }; static const struct snd_soc_dapm_widget aic311x_dapm_widgets[] = { @@ -554,7 +585,7 @@ static const struct snd_soc_dapm_widget aic310x_dapm_widgets[] = { }; static const struct snd_soc_dapm_route -aic31xx_audio_map[] = { +common31xx_audio_map[] = { /* DAC Input Routing */ {"DAC Left Input", "Left Data", "DAC IN"}, {"DAC Left Input", "Right Data", "DAC IN"}, @@ -565,6 +596,31 @@ aic31xx_audio_map[] = { {"DAC Left", NULL, "DAC Left Input"}, {"DAC Right", NULL, "DAC Right Input"}, + /* HPL path */ + {"HP Left", "Switch", "Output Left"}, + {"HPL Driver", NULL, "HP Left"}, + {"HPL", NULL, "HPL Driver"}, + + /* HPR path */ + {"HP Right", "Switch", "Output Right"}, + {"HPR Driver", NULL, "HP Right"}, + {"HPR", NULL, "HPR Driver"}, +}; + +static const struct snd_soc_dapm_route +dac31xx_audio_map[] = { + /* Left Output */ + {"Output Left", "From Left DAC", "DAC Left"}, + {"Output Left", "From AIN1", "AIN1"}, + {"Output Left", "From AIN2", "AIN2"}, + + /* Right Output */ + {"Output Right", "From Right DAC", "DAC Right"}, + {"Output Right", "From AIN2", "AIN2"}, +}; + +static const struct snd_soc_dapm_route +aic31xx_audio_map[] = { /* Mic input */ {"MIC1LP P-Terminal", "FFR 10 Ohm", "MIC1LP"}, {"MIC1LP P-Terminal", "FFR 20 Ohm", "MIC1LP"}, @@ -595,16 +651,6 @@ aic31xx_audio_map[] = { /* Right Output */ {"Output Right", "From Right DAC", "DAC Right"}, {"Output Right", "From MIC1RP", "MIC1RP"}, - - /* HPL path */ - {"HP Left", "Switch", "Output Left"}, - {"HPL Driver", NULL, "HP Left"}, - {"HPL", NULL, "HPL Driver"}, - - /* HPR path */ - {"HP Right", "Switch", "Output Right"}, - {"HPR Driver", NULL, "HP Right"}, - {"HPR", NULL, "HPR Driver"}, }; static const struct snd_soc_dapm_route @@ -633,6 +679,13 @@ static int aic31xx_add_controls(struct snd_soc_codec *codec) int ret = 0; struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + if (!(aic31xx->pdata.codec_type & DAC31XX_BIT)) + ret = snd_soc_add_codec_controls( + codec, aic31xx_snd_controls, + ARRAY_SIZE(aic31xx_snd_controls)); + if (ret) + return ret; + if (aic31xx->pdata.codec_type & AIC31XX_STEREO_CLASS_D_BIT) ret = snd_soc_add_codec_controls( codec, aic311x_snd_controls, @@ -651,6 +704,30 @@ static int aic31xx_add_widgets(struct snd_soc_codec *codec) struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); int ret = 0; + if (aic31xx->pdata.codec_type & DAC31XX_BIT) { + ret = snd_soc_dapm_new_controls( + dapm, dac31xx_dapm_widgets, + ARRAY_SIZE(dac31xx_dapm_widgets)); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(dapm, dac31xx_audio_map, + ARRAY_SIZE(dac31xx_audio_map)); + if (ret) + return ret; + } else { + ret = snd_soc_dapm_new_controls( + dapm, aic31xx_dapm_widgets, + ARRAY_SIZE(aic31xx_dapm_widgets)); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(dapm, aic31xx_audio_map, + ARRAY_SIZE(aic31xx_audio_map)); + if (ret) + return ret; + } + if (aic31xx->pdata.codec_type & AIC31XX_STEREO_CLASS_D_BIT) { ret = snd_soc_dapm_new_controls( dapm, aic311x_dapm_widgets, @@ -1114,12 +1191,12 @@ static struct snd_soc_codec_driver soc_codec_driver_aic31xx = { .set_bias_level = aic31xx_set_bias_level, .suspend_bias_off = true, - .controls = aic31xx_snd_controls, - .num_controls = ARRAY_SIZE(aic31xx_snd_controls), - .dapm_widgets = aic31xx_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(aic31xx_dapm_widgets), - .dapm_routes = aic31xx_audio_map, - .num_dapm_routes = ARRAY_SIZE(aic31xx_audio_map), + .controls = common31xx_snd_controls, + .num_controls = ARRAY_SIZE(common31xx_snd_controls), + .dapm_widgets = common31xx_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(common31xx_dapm_widgets), + .dapm_routes = common31xx_audio_map, + .num_dapm_routes = ARRAY_SIZE(common31xx_audio_map), }; static const struct snd_soc_dai_ops aic31xx_dai_ops = { @@ -1129,6 +1206,21 @@ static const struct snd_soc_dai_ops aic31xx_dai_ops = { .digital_mute = aic31xx_dac_mute, }; +static struct snd_soc_dai_driver dac31xx_dai_driver[] = { + { + .name = "tlv32dac31xx-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = AIC31XX_RATES, + .formats = AIC31XX_FORMATS, + }, + .ops = &aic31xx_dai_ops, + .symmetric_rates = 1, + } +}; + static struct snd_soc_dai_driver aic31xx_dai_driver[] = { { .name = "tlv320aic31xx-hifi", @@ -1259,9 +1351,16 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, if (ret) return ret; - return snd_soc_register_codec(&i2c->dev, &soc_codec_driver_aic31xx, - aic31xx_dai_driver, - ARRAY_SIZE(aic31xx_dai_driver)); + if (aic31xx->pdata.codec_type & DAC31XX_BIT) + return snd_soc_register_codec(&i2c->dev, + &soc_codec_driver_aic31xx, + dac31xx_dai_driver, + ARRAY_SIZE(dac31xx_dai_driver)); + else + return snd_soc_register_codec(&i2c->dev, + &soc_codec_driver_aic31xx, + aic31xx_dai_driver, + ARRAY_SIZE(aic31xx_dai_driver)); } static int aic31xx_i2c_remove(struct i2c_client *i2c) @@ -1277,6 +1376,7 @@ static const struct i2c_device_id aic31xx_i2c_id[] = { { "tlv320aic3110", AIC3110 }, { "tlv320aic3120", AIC3120 }, { "tlv320aic3111", AIC3111 }, + { "tlv320dac3100", DAC3100 }, { } }; MODULE_DEVICE_TABLE(i2c, aic31xx_i2c_id); diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h index ac9b146..5acd5b6 100644 --- a/sound/soc/codecs/tlv320aic31xx.h +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -24,12 +24,14 @@ #define AIC31XX_STEREO_CLASS_D_BIT 0x1 #define AIC31XX_MINIDSP_BIT 0x2 +#define DAC31XX_BIT 0x4 enum aic31xx_type { AIC3100 = 0, AIC3110 = AIC31XX_STEREO_CLASS_D_BIT, AIC3120 = AIC31XX_MINIDSP_BIT, AIC3111 = (AIC31XX_STEREO_CLASS_D_BIT | AIC31XX_MINIDSP_BIT), + DAC3100 = DAC31XX_BIT, }; struct aic31xx_pdata { -- 2.1.4

5 6

[alsa-devel] [PATCH 0/8] ASoC: s3c24xx_uda134x card fixes and cleanups
by Sylwester Nawrocki 25 Oct '16

25 Oct '16

This series might have a bit of an archaeological aspect, nevertheless my goal was to have sound working on one of the older S3C24XX SoC based boards for testing any patches touching the older Samsung SoC IP block related code. Together with patches: ASoC: samsung: Fix clock handling in S3C24XX_UDA134X card ARM: S3C24XX: Add dma_mask assignments for DMA devices ARM: S3C24XX: Add missing DMA device for Mini2440 board this series fixes sound on Micro2440SDK/mini2440 board broken since the 3.12 kernel release. I'm not sure how it worked before but currently the uda134x codec always fails to probe due to uninitialized platform_data, which is supposed to be set by the ASoC machine driver. This series addresses the issue by moving all required codec platform_data to the board file and exposing optional GPIO bitbang L3 bus ops which are selected by the codec. ASoC dependency patches: ASoC: samsung: Fix clock handling in S3C24XX_UDA134X card The patch "ARM: S3C24XX: Specify audio codec platform_data for mini2440 board" is for arm-soc but it would be good to merge whole series through one tree to avoid compilation breaks due to unmet dependencies. Sylwester Nawrocki (8): ASoC: L3 bus: Add default gpio ops ARM: S3C24XX: Specify audio codec platform_data for mini2440 board ASoC: uda134x: Optionally initialize L3 ops to default GPIO ops ASoC: s3c24xx_uda134x: Remove unused power() callback ASoC: s3c24xx_uda134x: Drop initialization of codec's platform data ASoC: samsung: Convert s3c24xx_uda134x to use devm_snd_soc_register_card() ASoC: samsung: s3c24xx_uda134x: debug/error trace cleanup ASoC: s3c244_uda134x: Allocate private data dynamically arch/arm/mach-s3c24xx/mach-mini2440.c | 19 ++++ include/sound/l3.h | 15 ++- include/sound/s3c24xx_uda134x.h | 1 - sound/soc/codecs/l3.c | 71 ++++++++++-- sound/soc/codecs/uda134x.c | 7 ++ sound/soc/samsung/s3c24xx_uda134x.c | 205 ++++++++++------------------------ 6 files changed, 158 insertions(+), 160 deletions(-) -- 1.9.1

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