From 64eae986fc1e3a281b00f04b7c9c00b145ec8a57 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 18 Jun 2014 17:54:43 +0900 Subject: [PATCH 01/27] ASoC: rsnd: fixup rsnd_gen_dma_addr() for Gen1 ad32d0c7b0e993433df152ae747652647eb65a27 (ASoC: rsnd: add rsnd_gen_dma_addr() for DMAC addr) added rsnd_gen_dma_addr() to calculate DMA addr, but, it is necessary only for Gen2. This patch ignores Gen1 case. Kernel will be panic without this patch. Special thanks to Simon Reported-by: Simon Horman Signed-off-by: Kuninori Morimoto Tested-by: Simon Horman Signed-off-by: Mark Brown --- sound/soc/sh/rcar/gen.c | 33 +++++++++++++++++++++------------ 1 file changed, 21 insertions(+), 12 deletions(-) diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 1dd2b7d38c2c..0280a11c0899 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -184,7 +184,7 @@ static int rsnd_gen_regmap_init(struct rsnd_priv *priv, #define RDMA_CMD_O_N(addr, i) (addr ##_reg - 0x004f8000 + (0x400 * i)) #define RDMA_CMD_O_P(addr, i) (addr ##_reg - 0x001f8000 + (0x400 * i)) -void rsnd_gen_dma_addr(struct rsnd_priv *priv, +static void rsnd_gen2_dma_addr(struct rsnd_priv *priv, struct rsnd_dma *dma, struct dma_slave_config *cfg, int is_play, int slave_id) @@ -226,17 +226,6 @@ void rsnd_gen_dma_addr(struct rsnd_priv *priv, } }; - cfg->slave_id = slave_id; - cfg->src_addr = 0; - cfg->dst_addr = 0; - cfg->direction = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM; - - /* - * gen1 uses default DMA addr - */ - if (rsnd_is_gen1(priv)) - return; - /* it shouldn't happen */ if (use_dvc & !use_src) { dev_err(dev, "DVC is selected without SRC\n"); @@ -250,6 +239,26 @@ void rsnd_gen_dma_addr(struct rsnd_priv *priv, id, cfg->src_addr, cfg->dst_addr); } +void rsnd_gen_dma_addr(struct rsnd_priv *priv, + struct rsnd_dma *dma, + struct dma_slave_config *cfg, + int is_play, int slave_id) +{ + cfg->slave_id = slave_id; + cfg->src_addr = 0; + cfg->dst_addr = 0; + cfg->direction = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM; + + /* + * gen1 uses default DMA addr + */ + if (rsnd_is_gen1(priv)) + return; + + rsnd_gen2_dma_addr(priv, dma, cfg, is_play, slave_id); +} + + /* * Gen2 */ From c08c3b088053cec1465051258844e7934d3e3e37 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 18 Jun 2014 17:55:09 +0900 Subject: [PATCH 02/27] ASoC: rsnd: fixup loop exit timing of dma name search Current dma name search loop didn't care about SSI index This patch fixes it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 4e86265f625c..ed76901f8202 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -297,7 +297,6 @@ static void rsnd_dma_of_name(struct rsnd_dma *dma, for (i = 1; i < MOD_MAX; i++) { if (!src) { mod[i] = ssi; - break; } else if (!dvc) { mod[i] = src; src = NULL; @@ -308,6 +307,9 @@ static void rsnd_dma_of_name(struct rsnd_dma *dma, if (mod[i] == this) index = i; + + if (mod[i] == ssi) + break; } if (is_play) { From 4adeb0ccf86a5af1825bbfe290dee9e60a5ab870 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Thu, 19 Jun 2014 09:32:05 +0300 Subject: [PATCH 03/27] ASoC: max98090: Fix missing free_irq max98090.c doesn't free the threaded interrupt it requests. This causes an oops when doing "cat /proc/interrupts" after snd-soc-max98090.ko is unloaded. Fix this by requesting the interrupt by using devm_request_threaded_irq(). Signed-off-by: Jarkko Nikula Cc: Stable # 3.10+ Signed-off-by: Mark Brown --- sound/soc/codecs/max98090.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index f5fccc7a8e89..d97f1ce7ff7d 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2284,7 +2284,7 @@ static int max98090_probe(struct snd_soc_codec *codec) /* Register for interrupts */ dev_dbg(codec->dev, "irq = %d\n", max98090->irq); - ret = request_threaded_irq(max98090->irq, NULL, + ret = devm_request_threaded_irq(codec->dev, max98090->irq, NULL, max98090_interrupt, IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "max98090_interrupt", codec); if (ret < 0) { From 7f28f357840fc857348b7ed42d7ee697cd221a59 Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Fri, 13 Jun 2014 12:49:59 +0300 Subject: [PATCH 04/27] 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 c7099eb1c19e60251e6725d6302354dfabae5303 Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Fri, 13 Jun 2014 13:04:36 +0300 Subject: [PATCH 05/27] ASoC: simple-card: Make u32 DT parameter handling 64-bit proof Passing unsigned int pointers as u32 ponters may be dangerous on 64-bit system. Signed-off-by: Jyri Sarha Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 03a7fdcdf114..159e517fa09a 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -116,6 +116,7 @@ asoc_simple_card_sub_parse_of(struct device_node *np, { struct device_node *node; struct clk *clk; + u32 val; int ret; /* @@ -151,10 +152,8 @@ asoc_simple_card_sub_parse_of(struct device_node *np, } dai->sysclk = clk_get_rate(clk); - } else if (of_property_read_bool(np, "system-clock-frequency")) { - of_property_read_u32(np, - "system-clock-frequency", - &dai->sysclk); + } else if (!of_property_read_u32(np, "system-clock-frequency", &val)) { + dai->sysclk = val; } else { clk = of_clk_get(node, 0); if (!IS_ERR(clk)) @@ -303,6 +302,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, { struct snd_soc_dai_link *dai_link = priv->snd_card.dai_link; struct simple_dai_props *dai_props = priv->dai_props; + u32 val; int ret; /* parsing the card name from DT */ @@ -325,8 +325,9 @@ static int asoc_simple_card_parse_of(struct device_node *node, } /* Factor to mclk, used in hw_params() */ - of_property_read_u32(node, "simple-audio-card,mclk-fs", - &priv->mclk_fs); + ret = of_property_read_u32(node, "simple-audio-card,mclk-fs", &val); + if (ret == 0) + priv->mclk_fs = val; dev_dbg(dev, "New simple-card: %s\n", priv->snd_card.name ? priv->snd_card.name : ""); From 7ed9de76ff342cbd717a9cf897044b99272cb8f8 Mon Sep 17 00:00:00 2001 From: Qiao Zhou Date: Wed, 4 Jun 2014 19:42:06 +0800 Subject: [PATCH 06/27] ASoC: pcm: fix dpcm_path_put in dpcm runtime update we need to release dapm widget list after dpcm_path_get in soc_dpcm_runtime_update. otherwise, there will be potential memory leak. add dpcm_path_put to fix it. Signed-off-by: Qiao Zhou Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/soc-pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 54d18f22a33e..4ea656770d65 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2069,6 +2069,7 @@ int soc_dpcm_runtime_update(struct snd_soc_card *card) dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK); } + dpcm_path_put(&list); capture: /* skip if FE doesn't have capture capability */ if (!fe->cpu_dai->driver->capture.channels_min) From b60f363b7f226daf40025ab13972dc82e6780be3 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 10 Jun 2014 18:41:02 +0100 Subject: [PATCH 07/27] 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 a5b37bf36f541dd06d876f9a3743abc224a25fd3 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Mon, 23 Jun 2014 16:29:37 +0300 Subject: [PATCH 08/27] ASoC: Intel: byt-max98090: Move MICBIAS widget to supply of Headset Mic Move "MICBIAS" as a supply widget to "Headset Mic" instead of keeping it between input pin "IN34" and "Headset Mic". Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/intel/byt-max98090.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/intel/byt-max98090.c b/sound/soc/intel/byt-max98090.c index 5fc98c64a3f4..a3c5f02a60fc 100644 --- a/sound/soc/intel/byt-max98090.c +++ b/sound/soc/intel/byt-max98090.c @@ -39,8 +39,7 @@ static const struct snd_soc_dapm_widget byt_max98090_widgets[] = { static const struct snd_soc_dapm_route byt_max98090_audio_map[] = { {"IN34", NULL, "Headset Mic"}, - {"IN34", NULL, "MICBIAS"}, - {"MICBIAS", NULL, "Headset Mic"}, + {"Headset Mic", NULL, "MICBIAS"}, {"DMICL", NULL, "Int Mic"}, {"Headphone", NULL, "HPL"}, {"Headphone", NULL, "HPR"}, From 725a6dfd014d02b4ad1d00df364a6d4a15ff2ebb Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Mon, 23 Jun 2014 16:29:38 +0300 Subject: [PATCH 09/27] ASoC: Intel: byt-max98090: Fix mic detect GPIO polarity Mic detect GPIO is active low when headset microphone is detected. Found both by debugging and checking the schematics. Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/intel/byt-max98090.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/byt-max98090.c b/sound/soc/intel/byt-max98090.c index a3c5f02a60fc..3d06440278ec 100644 --- a/sound/soc/intel/byt-max98090.c +++ b/sound/soc/intel/byt-max98090.c @@ -83,6 +83,7 @@ static struct snd_soc_jack_gpio hs_jack_gpios[] = { { .name = "mic-gpio", .idx = 1, + .invert = 1, .report = SND_JACK_MICROPHONE | SND_JACK_LINEIN, .debounce_time = 200, }, From 6a0cdccad87a7032440f204c04a599e4bcc8656b Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Mon, 23 Jun 2014 16:29:39 +0300 Subject: [PATCH 10/27] ASoC: Intel: byt-max98090: Do not report SND_JACK_LINEIN Headset jack has only mono microphone input. Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/intel/byt-max98090.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/byt-max98090.c b/sound/soc/intel/byt-max98090.c index 3d06440278ec..165ab6d6c4a3 100644 --- a/sound/soc/intel/byt-max98090.c +++ b/sound/soc/intel/byt-max98090.c @@ -84,7 +84,7 @@ static struct snd_soc_jack_gpio hs_jack_gpios[] = { .name = "mic-gpio", .idx = 1, .invert = 1, - .report = SND_JACK_MICROPHONE | SND_JACK_LINEIN, + .report = SND_JACK_MICROPHONE, .debounce_time = 200, }, }; From 24988992932612ba1c069ad597125dbce3e19621 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Mon, 23 Jun 2014 16:29:40 +0300 Subject: [PATCH 11/27] ASoC: Intel: byt-max98090: Fix jack type in order to report correctly Pass actual jack type bitmask to snd_soc_jack_new() in order to report also microphone detections and not only headphone. While at it change also jack name and pass also SND_JACK_LINEOUT type. Reported-by: Jin Yao Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/intel/byt-max98090.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/byt-max98090.c b/sound/soc/intel/byt-max98090.c index 165ab6d6c4a3..3e2e7152ae66 100644 --- a/sound/soc/intel/byt-max98090.c +++ b/sound/soc/intel/byt-max98090.c @@ -108,7 +108,8 @@ static int byt_max98090_init(struct snd_soc_pcm_runtime *runtime) } /* Enable jack detection */ - ret = snd_soc_jack_new(codec, "Headphone", SND_JACK_HEADPHONE, jack); + ret = snd_soc_jack_new(codec, "Headset", + SND_JACK_LINEOUT | SND_JACK_HEADSET, jack); if (ret) return ret; From ab6f7d0d932313cedfb1696e0536bdbe661cd9c9 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Mon, 23 Jun 2014 16:29:42 +0300 Subject: [PATCH 12/27] ASoC: Intel: byt-max98090: Do not enable MAX98090 microphone detection It turned out there is no need to enable microphone detection in MAX98090 codec. Headset microphone is anyway detected by a GPIO signal from another chip and headset button presses cannot be detected either because a signal needed for it is not connected. Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/intel/byt-max98090.c | 10 +++------- 1 file changed, 3 insertions(+), 7 deletions(-) diff --git a/sound/soc/intel/byt-max98090.c b/sound/soc/intel/byt-max98090.c index 3e2e7152ae66..5cfb41ec3fab 100644 --- a/sound/soc/intel/byt-max98090.c +++ b/sound/soc/intel/byt-max98090.c @@ -118,13 +118,9 @@ static int byt_max98090_init(struct snd_soc_pcm_runtime *runtime) if (ret) return ret; - ret = snd_soc_jack_add_gpiods(card->dev->parent, jack, - ARRAY_SIZE(hs_jack_gpios), - hs_jack_gpios); - if (ret) - return ret; - - return max98090_mic_detect(codec, jack); + return snd_soc_jack_add_gpiods(card->dev->parent, jack, + ARRAY_SIZE(hs_jack_gpios), + hs_jack_gpios); } static struct snd_soc_dai_link byt_max98090_dais[] = { From e73f3de5c5eb5285a1b1d8eed111eb229702b20f Mon Sep 17 00:00:00 2001 From: Russell King Date: Thu, 26 Jun 2014 15:22:50 +0100 Subject: [PATCH 13/27] 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 25ccb22ed55cf4ed1b94e2627b80f9ef44637558 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 26 Jun 2014 08:06:55 +0300 Subject: [PATCH 14/27] ASoC: tlv320aic3x: Correct S24_3LE support Correct the hw_params callback to configure the codec correctly in case of S24_3LE format since in case of S24_3LE the codec has been configured to 16bit format mode. S24_LE is not defined as supported format for the codec. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index e12fafbb1e09..5360772bc1ad 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -879,7 +879,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S20_3LE: data |= (0x01 << 4); break; - case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S24_3LE: data |= (0x02 << 4); break; case SNDRV_PCM_FORMAT_S32_LE: From 182bef863cc37a9a387ae9bc0f1b05243234bd4a Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 26 Jun 2014 08:09:24 +0300 Subject: [PATCH 15/27] 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 16/27] 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 17/27] 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 001eaa25ca8cf4bfdb76fa8f2287b8d34287b9b6 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 30 Jun 2014 10:01:37 +0200 Subject: [PATCH 18/27] ASoC: s6000-i2s: Fix s6000_i2s_remove() return type The platform_driver remove callback return type is int not void. Fixes the following warning: sound/soc/s6000/s6000-i2s.c:604:19: warning: incorrect type in initializer (different base types) sound/soc/s6000/s6000-i2s.c:604:19: expected int ( *remove )( ... ) sound/soc/s6000/s6000-i2s.c:604:19: got void ( static [toplevel] *)( ... ) Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/s6000/s6000-i2s.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/s6000/s6000-i2s.c b/sound/soc/s6000/s6000-i2s.c index 7eba7979b9af..1c8d01166e5b 100644 --- a/sound/soc/s6000/s6000-i2s.c +++ b/sound/soc/s6000/s6000-i2s.c @@ -570,7 +570,7 @@ static int s6000_i2s_probe(struct platform_device *pdev) return ret; } -static void s6000_i2s_remove(struct platform_device *pdev) +static int s6000_i2s_remove(struct platform_device *pdev) { struct s6000_i2s_dev *dev = dev_get_drvdata(&pdev->dev); struct resource *region; @@ -597,6 +597,8 @@ static void s6000_i2s_remove(struct platform_device *pdev) iounmap(mmio); region = platform_get_resource(pdev, IORESOURCE_IO, 0); release_mem_region(region->start, resource_size(region)); + + return 0; } static struct platform_driver s6000_i2s_driver = { From d3d4e5247b013008a39e4d5f69ce4c60ed57f997 Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki Date: Fri, 4 Jul 2014 16:05:45 +0200 Subject: [PATCH 19/27] ASoC: samsung: Correct I2S DAI suspend/resume ops We should save/restore relevant I2S registers regardless of the dai->active flag, otherwise some settings are being lost after system suspend/resume cycle. E.g. I2S slave mode set only during dai initialization is not preserved and the device ends up in master mode after system resume. Signed-off-by: Sylwester Nawrocki Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/samsung/i2s.c | 16 ++++++---------- 1 file changed, 6 insertions(+), 10 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 2ac76fa3e742..5f9b255a8b38 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -920,11 +920,9 @@ static int i2s_suspend(struct snd_soc_dai *dai) { struct i2s_dai *i2s = to_info(dai); - if (dai->active) { - i2s->suspend_i2smod = readl(i2s->addr + I2SMOD); - i2s->suspend_i2scon = readl(i2s->addr + I2SCON); - i2s->suspend_i2spsr = readl(i2s->addr + I2SPSR); - } + i2s->suspend_i2smod = readl(i2s->addr + I2SMOD); + i2s->suspend_i2scon = readl(i2s->addr + I2SCON); + i2s->suspend_i2spsr = readl(i2s->addr + I2SPSR); return 0; } @@ -933,11 +931,9 @@ static int i2s_resume(struct snd_soc_dai *dai) { struct i2s_dai *i2s = to_info(dai); - if (dai->active) { - writel(i2s->suspend_i2scon, i2s->addr + I2SCON); - writel(i2s->suspend_i2smod, i2s->addr + I2SMOD); - writel(i2s->suspend_i2spsr, i2s->addr + I2SPSR); - } + writel(i2s->suspend_i2scon, i2s->addr + I2SCON); + writel(i2s->suspend_i2smod, i2s->addr + I2SMOD); + writel(i2s->suspend_i2spsr, i2s->addr + I2SPSR); return 0; } From e42be7e1420389349400e990a3b13d23b9c60d39 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Mon, 7 Jul 2014 14:26:48 -0300 Subject: [PATCH 20/27] ASoC: sgtl5000: Fix driver unbound Using the sgtl5000 codec driver as a module and trying to remove it causes the followig kernel oops: root@freescale /$ rmmod snd-soc-imx-sgtl5000 [ 117.122920] ------------[ cut here ]------------ [ 117.127609] WARNING: CPU: 0 PID: 631 at drivers/regulator/core.c:3604 regula) [ 117.137046] Modules linked in: snd_soc_imx_sgtl5000(-) snd_soc_sgtl5000 evbug [ 117.144315] CPU: 0 PID: 631 Comm: rmmod Not tainted 3.16.0-rc3-next-201407043 [ 117.153366] Backtrace: [ 117.155865] [<80011e5c>] (dump_backtrace) from [<80011ff8>] (show_stack+0x18) [ 117.163484] r6:802fcc48 r5:00000000 r4:00000000 r3:00000000 [ 117.169228] [<80011fe0>] (show_stack) from [<80668cc0>] (dump_stack+0x88/0xa) [ 117.176508] [<80668c38>] (dump_stack) from [<80029a38>] (warn_slowpath_commo) [ 117.184696] r5:00000009 r4:00000000 [ 117.188322] [<800299c8>] (warn_slowpath_common) from [<80029a80>] (warn_slow) [ 117.197150] r8:dd60d600 r7:ddfa6d00 r6:dd5d9064 r5:dd5e0f90 r4:dd5d9400 [ 117.203983] [<80029a5c>] (warn_slowpath_null) from [<802fcc48>] (regulator_u) [ 117.212828] [<802fcb74>] (regulator_unregister) from [<7f0047c4>] (ldo_regul) [ 117.223475] r4:dd59e300 r3:dd5e0f90 [ 117.227100] [<7f00479c>] (ldo_regulator_remove [snd_soc_sgtl5000]) from [<7f) [ 117.238959] r4:dd5d8000 r3:ddd51420 [ 117.242623] [<7f0047dc>] (sgtl5000_remove [snd_soc_sgtl5000]) from [<804e5b1) [ 117.252489] r5:00000000 r4:dd5d8000 [ 117.256111] [<804e5af8>] (soc_remove_codec) from [<804e5ed4>] (soc_remove_da) [ 117.264933] r4:ddfb640c r3:00000000 [ 117.268555] [<804e5c10>] (soc_remove_dai_links) from [<804e5fbc>] (snd_soc_u) [ 117.277810] r10:80359e48 r9:dd684000 r8:dd5ca800 r7:dd685e60 r6:00000001 r5c [ 117.285761] r4:dd5d9064 [ 117.288329] [<804e5f28>] (snd_soc_unregister_card) from [<804f29f0>] (devm_c) [ 117.297324] r6:00000004 r5:ddd0fc10 r4:dd5a7980 r3:804f29dc [ 117.303095] [<804f29dc>] (devm_card_release) from [<8035a418>] (release_node) [ 117.311369] [<8035a2a8>] (release_nodes) from [<8035aab4>] (devres_release_a) [ 117.319583] r10:00000000 r9:dd684000 r8:8000ed64 r7:00000081 r6:ddd0fc44 r54 [ 117.327543] r4:ddd0fc10 [ 117.330108] [<8035aa7c>] (devres_release_all) from [<803571c8>] (__device_re) [ 117.339214] r4:ddd0fc10 r3:dd5d9010 [ 117.342989] [<80357148>] (__device_release_driver) from [<80357a48>] (driver) [ 117.351607] r5:7f00c9e4 r4:ddd0fc10 [ 117.355285] [<8035798c>] (driver_detach) from [<80357030>] (bus_remove_drive) [ 117.363454] r6:00000880 r5:00000000 r4:7f00c9e4 r3:dd6c3a80 [ 117.369195] [<80356fdc>] (bus_remove_driver) from [<803580b8>] (driver_unreg) [ 117.377683] r4:7f00c9e4 r3:dd60d480 [ 117.381305] [<80358088>] (driver_unregister) from [<80358bb4>] (platform_dri) [ 117.390642] r4:7f00ca28 r3:7f00c35c [ 117.394309] [<80358ba0>] (platform_driver_unregister) from [<7f00c370>] (imx) [ 117.406188] [<7f00c35c>] (imx_sgtl5000_driver_exit [snd_soc_imx_sgtl5000]) f) [ 117.417452] [<8008c2b8>] (SyS_delete_module) from [<8000eba0>] (ret_fast_sys) [ 117.425753] r6:5f636f73 r5:5f646e73 r4:00016f40 [ 117.430428] ---[ end trace 8fd8a5cb39e46d0e ]--- This problem is well explained by Russell King: "The sgtl5000 uses a two-stage initialisation process. The first stage is when the platform driver is probed, where some resources are found and initialised. The second stage is via the codec driver's probe function, where regulators are found and initialised using the managed resource support. The problem here is that this works fine until the codec driver is unbound. When this occurs, sgtl5000_remove() is called which is a no-op as far as the managed resource code is concerned. The regulators remain allocated, and their pointers in sgtl5000_priv remain valid. If the codec is now re-probed, it will again try and find the regulators, which will now be busy. This will fail. That's not the only problem - if using the LDO regulator, the regulator is unregistered from the regulator core code, but it still has a user. When the user is cleaned up (eg, by removing the module) it hits the free'd regulator, and this can oops the kernel. This bug was originally introduced by 63e54cd9caa3ce ("ASoC: sgtl5000: Use devm_regulator_bulk_get()")." This reverts commit 63e54cd9caa3ce. Tested on a imx53-qsb board. Reported-by: Russell King Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 3d39f0b5b4a8..8f4c73d17c87 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1277,7 +1277,7 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) return ret; } - ret = devm_regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies), + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); if (ret) goto err_ldo_remove; @@ -1285,13 +1285,16 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) ret = regulator_bulk_enable(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); if (ret) - goto err_ldo_remove; + goto err_regulator_free; /* wait for all power rails bring up */ udelay(10); return 0; +err_regulator_free: + regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); err_ldo_remove: if (!external_vddd) ldo_regulator_remove(codec); @@ -1361,6 +1364,8 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) err: regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); + regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); ldo_regulator_remove(codec); return ret; @@ -1374,6 +1379,8 @@ static int sgtl5000_remove(struct snd_soc_codec *codec) regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); + regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); ldo_regulator_remove(codec); return 0; From b97c60abf9a561f86ae71bd741add02673cc1a08 Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki Date: Thu, 10 Jul 2014 18:11:13 +0200 Subject: [PATCH 21/27] ASoC: samsung-i2s: Maintain CDCLK settings across i2s_{shutdown/startup} Currently configuration of the CDCLK pad is being overwritten in the i2s_shutdown() callback in order to gate the SoC output clock. However if an ASoC machine driver doesn't restore that clock settings each time after opening the sound device this results in the CDCLK pin being permanently configured into input mode. I.e. the output clock will always stay disabled. Fix that by saving the CDCLKCON bit state in i2s_shutdown() and and restoring it in the i2s_startup() callback. Signed-off-by: Chen Zhen Signed-off-by: Sylwester Nawrocki Signed-off-by: Mark Brown --- sound/soc/samsung/i2s.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 5f9b255a8b38..d2533dbc8399 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -68,6 +68,8 @@ struct i2s_dai { #define DAI_OPENED (1 << 0) /* Dai is opened */ #define DAI_MANAGER (1 << 1) /* Dai is the manager */ unsigned mode; + /* CDCLK pin direction: 0 - input, 1 - output */ + unsigned int cdclk_out:1; /* Driver for this DAI */ struct snd_soc_dai_driver i2s_dai_drv; /* DMA parameters */ @@ -737,6 +739,9 @@ static int i2s_startup(struct snd_pcm_substream *substream, spin_unlock_irqrestore(&lock, flags); + if (!is_opened(other) && i2s->cdclk_out) + i2s_set_sysclk(dai, SAMSUNG_I2S_CDCLK, + 0, SND_SOC_CLOCK_OUT); return 0; } @@ -752,9 +757,13 @@ static void i2s_shutdown(struct snd_pcm_substream *substream, i2s->mode &= ~DAI_OPENED; i2s->mode &= ~DAI_MANAGER; - if (is_opened(other)) + if (is_opened(other)) { other->mode |= DAI_MANAGER; - + } else { + u32 mod = readl(i2s->addr + I2SMOD); + i2s->cdclk_out = !(mod & MOD_CDCLKCON); + other->cdclk_out = i2s->cdclk_out; + } /* Reset any constraint on RFS and BFS */ i2s->rfs = 0; i2s->bfs = 0; From 8e89761876611f06ef4be865b4780b4361caf4af Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Mon, 14 Jul 2014 17:37:36 +0800 Subject: [PATCH 22/27] ASoC: Intel: Cleanup HSW pcm format support This change removes unsupported formats from System, Capture and Loopback FE DAIs. Also it fixes S24_LE support on all DAIs. While at this fix 24 bit flag for BYT as well. Signed-off-by: Jie Yang Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/intel/sst-baytrail-pcm.c | 2 +- sound/soc/intel/sst-haswell-pcm.c | 27 ++++++++++++++++++--------- 2 files changed, 19 insertions(+), 10 deletions(-) diff --git a/sound/soc/intel/sst-baytrail-pcm.c b/sound/soc/intel/sst-baytrail-pcm.c index 8eab97368ea7..599401c0c655 100644 --- a/sound/soc/intel/sst-baytrail-pcm.c +++ b/sound/soc/intel/sst-baytrail-pcm.c @@ -32,7 +32,7 @@ static const struct snd_pcm_hardware sst_byt_pcm_hardware = { SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, .formats = SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FORMAT_S24_LE, + SNDRV_PCM_FMTBIT_S24_LE, .period_bytes_min = 384, .period_bytes_max = 48000, .periods_min = 2, diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 058efb17c568..61bf6da4bb02 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -80,7 +80,7 @@ static const struct snd_pcm_hardware hsw_pcm_hardware = { SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_NO_PERIOD_WAKEUP, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE | + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE, .period_bytes_min = PAGE_SIZE, .period_bytes_max = (HSW_PCM_PERIODS_MAX / HSW_PCM_PERIODS_MIN) * PAGE_SIZE, @@ -400,7 +400,15 @@ static int hsw_pcm_hw_params(struct snd_pcm_substream *substream, sst_hsw_stream_set_valid(hsw, pcm_data->stream, 16); break; case SNDRV_PCM_FORMAT_S24_LE: - bits = SST_HSW_DEPTH_24BIT; + bits = SST_HSW_DEPTH_32BIT; + sst_hsw_stream_set_valid(hsw, pcm_data->stream, 24); + break; + case SNDRV_PCM_FORMAT_S8: + bits = SST_HSW_DEPTH_8BIT; + sst_hsw_stream_set_valid(hsw, pcm_data->stream, 8); + break; + case SNDRV_PCM_FORMAT_S32_LE: + bits = SST_HSW_DEPTH_32BIT; sst_hsw_stream_set_valid(hsw, pcm_data->stream, 32); break; default: @@ -685,8 +693,9 @@ static int hsw_pcm_new(struct snd_soc_pcm_runtime *rtd) } #define HSW_FORMATS \ - (SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S16_LE |\ - SNDRV_PCM_FMTBIT_S32_LE) + (SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S8) static struct snd_soc_dai_driver hsw_dais[] = { { @@ -696,7 +705,7 @@ static struct snd_soc_dai_driver hsw_dais[] = { .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, + .formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE, }, }, { @@ -727,8 +736,8 @@ static struct snd_soc_dai_driver hsw_dais[] = { .stream_name = "Loopback Capture", .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000, - .formats = HSW_FORMATS, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE, }, }, { @@ -737,8 +746,8 @@ static struct snd_soc_dai_driver hsw_dais[] = { .stream_name = "Analog Capture", .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000, - .formats = HSW_FORMATS, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE, }, }, }; From c761b58306b26ea78fa91488cbb04c61b35d583d Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Mon, 14 Jul 2014 17:11:11 +0800 Subject: [PATCH 23/27] ASoC: Intel: Add dummy read for SRAM block enable Add dummy read after each block enable, to workaround SRAM write missing bytes issue. Signed-off-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-dsp.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c index 535f517629fd..a33b931181dc 100644 --- a/sound/soc/intel/sst-haswell-dsp.c +++ b/sound/soc/intel/sst-haswell-dsp.c @@ -359,6 +359,17 @@ static u32 hsw_block_get_bit(struct sst_mem_block *block) return bit; } +/*dummy read a SRAM block.*/ +static void sst_mem_block_dummy_read(struct sst_mem_block *block) +{ + u32 size; + u8 tmp_buf[4]; + struct sst_dsp *sst = block->dsp; + + size = block->size > 4 ? 4 : block->size; + memcpy_fromio(tmp_buf, sst->addr.lpe + block->offset, size); +} + /* enable 32kB memory block - locks held by caller */ static int hsw_block_enable(struct sst_mem_block *block) { @@ -378,6 +389,8 @@ static int hsw_block_enable(struct sst_mem_block *block) /* wait 18 DSP clock ticks */ udelay(10); + /*add a dummy read before the SRAM block is written, otherwise the writing may miss bytes sometimes.*/ + sst_mem_block_dummy_read(block); return 0; } From 30443408fd7201fd1911b09daccf92fae3cc700d Mon Sep 17 00:00:00 2001 From: Scott Jiang Date: Fri, 18 Jul 2014 16:14:57 +0800 Subject: [PATCH 24/27] 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 eff952b733d4c1ff3a6b35accce940b223372978 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Thu, 17 Jul 2014 21:21:37 +0800 Subject: [PATCH 25/27] ASoC: fsl_sai: Reset FIFOs after disabling TE/RE SAI will not clear their FIFOs after disabling TE/RE. Therfore, the driver should take care the task so as not to let useless data remain in the FIFO. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index c5a0e8af8226..b10dbd835d5c 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -371,10 +371,13 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, /* Check if the opposite FRDE is also disabled */ if (!(tx ? rcsr & FSL_SAI_CSR_FRDE : tcsr & FSL_SAI_CSR_FRDE)) { + /* Disable both directions and reset their FIFOs */ regmap_update_bits(sai->regmap, FSL_SAI_TCSR, - FSL_SAI_CSR_TERE, 0); + FSL_SAI_CSR_TERE | FSL_SAI_CSR_FR, + FSL_SAI_CSR_FR); regmap_update_bits(sai->regmap, FSL_SAI_RCSR, - FSL_SAI_CSR_TERE, 0); + FSL_SAI_CSR_TERE | FSL_SAI_CSR_FR, + FSL_SAI_CSR_FR); } break; default: From 4800f88b615f194ae3c1577038a7ccd871c907c9 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Thu, 17 Jul 2014 21:21:38 +0800 Subject: [PATCH 26/27] ASoC: fsl_sai: Fix incorrect register writing in fsl_sai_isr() In the rx irq handling part, we should clear the flags in RCSR not TCSR. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index b10dbd835d5c..1b6ee2ce849f 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -106,7 +106,7 @@ static irqreturn_t fsl_sai_isr(int irq, void *devid) xcsr &= ~FSL_SAI_CSR_xF_MASK; if (flags) - regmap_write(sai->regmap, FSL_SAI_TCSR, flags | xcsr); + regmap_write(sai->regmap, FSL_SAI_RCSR, flags | xcsr); out: if (irq_none) From 95b47f8de787214f7db88b26759d7edc7c64d74a Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Thu, 17 Jul 2014 13:16:54 -0500 Subject: [PATCH 27/27] 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),