From 7f28f357840fc857348b7ed42d7ee697cd221a59 Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Fri, 13 Jun 2014 12:49:59 +0300 Subject: [PATCH 1/8] ASoC: davinci-mcasp: Add dependecy to SND_DAVINCI_SOC or SND_OMAP_SOC Fixes build with SND_DAVINCI_SOC or SND_OMAP_SOC alone and adds build dependecy to SND_DAVINCI_SOC or SND_OMAP_SOC. Signed-off-by: Jyri Sarha Signed-off-by: Mark Brown --- sound/soc/davinci/Kconfig | 1 + sound/soc/davinci/davinci-mcasp.c | 8 ++++++++ 2 files changed, 9 insertions(+) diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 50a098749b9e..fdbb16fffd30 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -6,6 +6,7 @@ config SND_DAVINCI_SOC_I2S tristate config SND_DAVINCI_SOC_MCASP + depends on SND_DAVINCI_SOC || SND_OMAP_SOC tristate config SND_DAVINCI_SOC_VCIF diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 9afb14629a17..0ee4986038cc 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1223,14 +1223,22 @@ static int davinci_mcasp_probe(struct platform_device *pdev) goto err; switch (mcasp->version) { +#if IS_BUILTIN(CONFIG_SND_DAVINCI_SOC) || \ + (IS_MODULE(CONFIG_SND_DAVINCI_SOC_MCASP) && \ + IS_MODULE(CONFIG_SND_DAVINCI_SOC)) case MCASP_VERSION_1: case MCASP_VERSION_2: case MCASP_VERSION_3: ret = davinci_soc_platform_register(&pdev->dev); break; +#endif +#if IS_BUILTIN(CONFIG_SND_OMAP_SOC) || \ + (IS_MODULE(CONFIG_SND_DAVINCI_SOC_MCASP) && \ + IS_MODULE(CONFIG_SND_OMAP_SOC)) case MCASP_VERSION_4: ret = omap_pcm_platform_register(&pdev->dev); break; +#endif default: dev_err(&pdev->dev, "Invalid McASP version: %d\n", mcasp->version); From b60f363b7f226daf40025ab13972dc82e6780be3 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 10 Jun 2014 18:41:02 +0100 Subject: [PATCH 2/8] ASoC: wm5110: Power both channels for differential mono output On the wm5110 CODEC both the left and right channel must be powered when an output is being used as a mono output, although no audio is routed to the right output channel. This patch adds additional DAPM routes to link the right channel to the left in the case where an output is marked as mono. Audio must always be brought in on the left channel for mono operation. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 25 +++++++++++++++++++++++++ sound/soc/codecs/arizona.h | 1 + sound/soc/codecs/wm5110.c | 1 + 3 files changed, 27 insertions(+) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 29e198f57d4c..747c71e59c04 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -243,6 +243,31 @@ int arizona_init_spk(struct snd_soc_codec *codec) } EXPORT_SYMBOL_GPL(arizona_init_spk); +static const struct snd_soc_dapm_route arizona_mono_routes[] = { + { "OUT1R", NULL, "OUT1L" }, + { "OUT2R", NULL, "OUT2L" }, + { "OUT3R", NULL, "OUT3L" }, + { "OUT4R", NULL, "OUT4L" }, + { "OUT5R", NULL, "OUT5L" }, + { "OUT6R", NULL, "OUT6L" }, +}; + +int arizona_init_mono(struct snd_soc_codec *codec) +{ + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + int i; + + for (i = 0; i < ARIZONA_MAX_OUTPUT; ++i) { + if (arizona->pdata.out_mono[i]) + snd_soc_dapm_add_routes(&codec->dapm, + &arizona_mono_routes[i], 1); + } + + return 0; +} +EXPORT_SYMBOL_GPL(arizona_init_mono); + int arizona_init_gpio(struct snd_soc_codec *codec) { struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 05ae17f5bca3..942cfb197b6d 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -249,6 +249,7 @@ extern int arizona_set_fll(struct arizona_fll *fll, int source, extern int arizona_init_spk(struct snd_soc_codec *codec); extern int arizona_init_gpio(struct snd_soc_codec *codec); +extern int arizona_init_mono(struct snd_soc_codec *codec); extern int arizona_init_dai(struct arizona_priv *priv, int dai); diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 2e5fcb559e90..62ef54456499 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -1596,6 +1596,7 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec) arizona_init_spk(codec); arizona_init_gpio(codec); + arizona_init_mono(codec); ret = snd_soc_add_codec_controls(codec, wm_adsp2_fw_controls, 8); if (ret != 0) From e73f3de5c5eb5285a1b1d8eed111eb229702b20f Mon Sep 17 00:00:00 2001 From: Russell King Date: Thu, 26 Jun 2014 15:22:50 +0100 Subject: [PATCH 3/8] ASoC: fix debugfs directory creation bug Avoid creating duplicate directories by prefixing codecs and platforms with their separate identifiers. This avoids snd-soc-dummy (which can appear both as a dummy platform and a dummy codec on the same card) from clashing. Signed-off-by: Russell King Tested-by: Andrew Lunn Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 28 ++++++++++++++++++++++++---- 1 file changed, 24 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b87d7d882e6d..91120b8e283e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -270,12 +270,32 @@ static const struct file_operations codec_reg_fops = { .llseek = default_llseek, }; +static struct dentry *soc_debugfs_create_dir(struct dentry *parent, + const char *fmt, ...) +{ + struct dentry *de; + va_list ap; + char *s; + + va_start(ap, fmt); + s = kvasprintf(GFP_KERNEL, fmt, ap); + va_end(ap); + + if (!s) + return NULL; + + de = debugfs_create_dir(s, parent); + kfree(s); + + return de; +} + static void soc_init_codec_debugfs(struct snd_soc_codec *codec) { struct dentry *debugfs_card_root = codec->card->debugfs_card_root; - codec->debugfs_codec_root = debugfs_create_dir(codec->name, - debugfs_card_root); + codec->debugfs_codec_root = soc_debugfs_create_dir(debugfs_card_root, + "codec:%s", codec->name); if (!codec->debugfs_codec_root) { dev_warn(codec->dev, "ASoC: Failed to create codec debugfs directory\n"); @@ -306,8 +326,8 @@ static void soc_init_platform_debugfs(struct snd_soc_platform *platform) { struct dentry *debugfs_card_root = platform->card->debugfs_card_root; - platform->debugfs_platform_root = debugfs_create_dir(platform->name, - debugfs_card_root); + platform->debugfs_platform_root = soc_debugfs_create_dir(debugfs_card_root, + "platform:%s", platform->name); if (!platform->debugfs_platform_root) { dev_warn(platform->dev, "ASoC: Failed to create platform debugfs directory\n"); From 182bef863cc37a9a387ae9bc0f1b05243234bd4a Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 26 Jun 2014 08:09:24 +0300 Subject: [PATCH 4/8] ASoC: davinci-mcasp: Fix S24_LE and U24_LE support In case of S24_LE/U24_LE modes we expect 24bits on the bus while the samples are stored and transferred in memory on 32bits (lower 3 bytes of the 4 bytes). Signed-off-by: Peter Ujfalusi Tested-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 0ee4986038cc..bfcc6c3dc2fd 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -720,6 +720,10 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_U24_LE: case SNDRV_PCM_FORMAT_S24_LE: + dma_params->data_type = 4; + word_length = 24; + break; + case SNDRV_PCM_FORMAT_U32_LE: case SNDRV_PCM_FORMAT_S32_LE: dma_params->data_type = 4; From 3ad80b828b2533f37c221e2df155774efd6ed814 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Thu, 3 Jul 2014 16:51:36 +0200 Subject: [PATCH 5/8] ASoC: adau1701: fix adau1701_reg_read() Fix a long standing bug in the read register routing of adau1701. The bytes arrive in the buffer in big-endian, so the result has to be shifted before and-ing the bytes in the loop. Signed-off-by: Daniel Mack Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/adau1701.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index d71c59cf7bdd..370b742117ef 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -230,8 +230,10 @@ static int adau1701_reg_read(void *context, unsigned int reg, *value = 0; - for (i = 0; i < size; i++) - *value |= recv_buf[i] << (i * 8); + for (i = 0; i < size; i++) { + *value <<= 8; + *value |= recv_buf[i]; + } return 0; } From 0a37c6efec4a2fdc2563c5a8faa472b814deee80 Mon Sep 17 00:00:00 2001 From: Praveen Diwakar Date: Fri, 4 Jul 2014 11:17:41 +0530 Subject: [PATCH 6/8] ASoC: wm_adsp: Add missing MODULE_LICENSE Since MODULE_LICENSE is missing the module load fails, so add this for module. Signed-off-by: Praveen Diwakar Signed-off-by: Vinod Koul Reviewed-by: Charles Keepax Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm_adsp.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 060027182dcb..2537725dd53f 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1758,3 +1758,5 @@ int wm_adsp2_init(struct wm_adsp *adsp, bool dvfs) return 0; } EXPORT_SYMBOL_GPL(wm_adsp2_init); + +MODULE_LICENSE("GPL v2"); From 30443408fd7201fd1911b09daccf92fae3cc700d Mon Sep 17 00:00:00 2001 From: Scott Jiang Date: Fri, 18 Jul 2014 16:14:57 +0800 Subject: [PATCH 7/8] ASoC: blackfin: use samples to set silence The third parameter for snd_pcm_format_set_silence needs the number of samples instead of sample bytes. Signed-off-by: Scott Jiang Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/blackfin/bf5xx-i2s-pcm.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c index a3881c4381c9..bcf591373a7a 100644 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.c +++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c @@ -290,19 +290,19 @@ static int bf5xx_pcm_silence(struct snd_pcm_substream *substream, unsigned int sample_size = runtime->sample_bits / 8; void *buf = runtime->dma_area; struct bf5xx_i2s_pcm_data *dma_data; - unsigned int offset, size; + unsigned int offset, samples; dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); if (dma_data->tdm_mode) { offset = pos * 8 * sample_size; - size = count * 8 * sample_size; + samples = count * 8; } else { offset = frames_to_bytes(runtime, pos); - size = frames_to_bytes(runtime, count); + samples = count * runtime->channels; } - snd_pcm_format_set_silence(runtime->format, buf + offset, size); + snd_pcm_format_set_silence(runtime->format, buf + offset, samples); return 0; } From 95b47f8de787214f7db88b26759d7edc7c64d74a Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Thu, 17 Jul 2014 13:16:54 -0500 Subject: [PATCH 8/8] ASoC: cs42l56: Fix stereo channel register assignment for Headphone and LineOut volume mixers Stereo Headphone and LineOut volume mixers are now attached to HPA+HPB, LOA+LOB. Reported-by: Ryan Harvey Signed-off-by: Ryan Harvey Signed-off-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l56.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index fdc4bd27b0df..8e68ef5de849 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -445,9 +445,9 @@ static const struct snd_kcontrol_new cs42l56_snd_controls[] = { SOC_DOUBLE("ADC Boost Switch", CS42L56_GAIN_BIAS_CTL, 3, 2, 1, 1), SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L56_HPA_VOLUME, - CS42L56_HPA_VOLUME, 0, 0x44, 0x55, hl_tlv), + CS42L56_HPB_VOLUME, 0, 0x44, 0x55, hl_tlv), SOC_DOUBLE_R_SX_TLV("LineOut Volume", CS42L56_LOA_VOLUME, - CS42L56_LOA_VOLUME, 0, 0x44, 0x55, hl_tlv), + CS42L56_LOB_VOLUME, 0, 0x44, 0x55, hl_tlv), SOC_SINGLE_TLV("Bass Shelving Volume", CS42L56_TONE_CTL, 0, 0x00, 1, tone_tlv),