From ce9594c6b332fd6fe464e22a83b0e6e0a287aac6 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 27 Feb 2015 14:13:02 +0100 Subject: ASoC: ak4671: Fix control-less DAPM routes Routes without a control must use NULL for the control name. The ak4671 driver uses "NULL" instead in a few places. Previous to commit 5fe5b767dc6f ("ASoC: dapm: Do not pretend to support controls for non mixer/mux widgets") the DAPM core silently ignored non-NULL controls on non-mixer and non-mux routes. But starting with that commit it will complain and not add the route breaking the ak4671 driver in the process. This patch replaces the incorrect "NULL" control name with NULL to fix the issue. Fixes: 5fe5b767dc6f ("ASoC: dapm: Do not pretend to support controls for non mixer/mux widgets") Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/ak4671.c | 44 ++++++++++++++++++++++---------------------- 1 file changed, 22 insertions(+), 22 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index 632e89f..2a58b1d 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -343,25 +343,25 @@ static const struct snd_soc_dapm_widget ak4671_dapm_widgets[] = { }; static const struct snd_soc_dapm_route ak4671_intercon[] = { - {"DAC Left", "NULL", "PMPLL"}, - {"DAC Right", "NULL", "PMPLL"}, - {"ADC Left", "NULL", "PMPLL"}, - {"ADC Right", "NULL", "PMPLL"}, + {"DAC Left", NULL, "PMPLL"}, + {"DAC Right", NULL, "PMPLL"}, + {"ADC Left", NULL, "PMPLL"}, + {"ADC Right", NULL, "PMPLL"}, /* Outputs */ - {"LOUT1", "NULL", "LOUT1 Mixer"}, - {"ROUT1", "NULL", "ROUT1 Mixer"}, - {"LOUT2", "NULL", "LOUT2 Mix Amp"}, - {"ROUT2", "NULL", "ROUT2 Mix Amp"}, - {"LOUT3", "NULL", "LOUT3 Mixer"}, - {"ROUT3", "NULL", "ROUT3 Mixer"}, + {"LOUT1", NULL, "LOUT1 Mixer"}, + {"ROUT1", NULL, "ROUT1 Mixer"}, + {"LOUT2", NULL, "LOUT2 Mix Amp"}, + {"ROUT2", NULL, "ROUT2 Mix Amp"}, + {"LOUT3", NULL, "LOUT3 Mixer"}, + {"ROUT3", NULL, "ROUT3 Mixer"}, {"LOUT1 Mixer", "DACL", "DAC Left"}, {"ROUT1 Mixer", "DACR", "DAC Right"}, {"LOUT2 Mixer", "DACHL", "DAC Left"}, {"ROUT2 Mixer", "DACHR", "DAC Right"}, - {"LOUT2 Mix Amp", "NULL", "LOUT2 Mixer"}, - {"ROUT2 Mix Amp", "NULL", "ROUT2 Mixer"}, + {"LOUT2 Mix Amp", NULL, "LOUT2 Mixer"}, + {"ROUT2 Mix Amp", NULL, "ROUT2 Mixer"}, {"LOUT3 Mixer", "DACSL", "DAC Left"}, {"ROUT3 Mixer", "DACSR", "DAC Right"}, @@ -381,18 +381,18 @@ static const struct snd_soc_dapm_route ak4671_intercon[] = { {"LIN2", NULL, "Mic Bias"}, {"RIN2", NULL, "Mic Bias"}, - {"ADC Left", "NULL", "LIN MUX"}, - {"ADC Right", "NULL", "RIN MUX"}, + {"ADC Left", NULL, "LIN MUX"}, + {"ADC Right", NULL, "RIN MUX"}, /* Analog Loops */ - {"LIN1 Mixing Circuit", "NULL", "LIN1"}, - {"RIN1 Mixing Circuit", "NULL", "RIN1"}, - {"LIN2 Mixing Circuit", "NULL", "LIN2"}, - {"RIN2 Mixing Circuit", "NULL", "RIN2"}, - {"LIN3 Mixing Circuit", "NULL", "LIN3"}, - {"RIN3 Mixing Circuit", "NULL", "RIN3"}, - {"LIN4 Mixing Circuit", "NULL", "LIN4"}, - {"RIN4 Mixing Circuit", "NULL", "RIN4"}, + {"LIN1 Mixing Circuit", NULL, "LIN1"}, + {"RIN1 Mixing Circuit", NULL, "RIN1"}, + {"LIN2 Mixing Circuit", NULL, "LIN2"}, + {"RIN2 Mixing Circuit", NULL, "RIN2"}, + {"LIN3 Mixing Circuit", NULL, "LIN3"}, + {"RIN3 Mixing Circuit", NULL, "RIN3"}, + {"LIN4 Mixing Circuit", NULL, "LIN4"}, + {"RIN4 Mixing Circuit", NULL, "RIN4"}, {"LOUT1 Mixer", "LINL1", "LIN1 Mixing Circuit"}, {"ROUT1 Mixer", "RINR1", "RIN1 Mixing Circuit"}, -- cgit v1.1 From 8e6a75c102f8e232b599a06e06731d8c5d5f2c5d Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 27 Feb 2015 14:13:03 +0100 Subject: ASoC: da732x: Fix control-less DAPM routes Routes without a control must use NULL for the control name. The da732x driver uses "NULL" instead in a few places. Previous to commit 5fe5b767dc6f ("ASoC: dapm: Do not pretend to support controls for non mixer/mux widgets") the DAPM core silently ignored non-NULL controls on non-mixer and non-mux routes. But starting with that commit it will complain and not add the route breaking the da732x driver in the process. This patch replaces the incorrect "NULL" control name with NULL to fix the issue. Fixes: 5fe5b767dc6f ("ASoC: dapm: Do not pretend to support controls for non mixer/mux widgets") Signed-off-by: Lars-Peter Clausen Acked-by: Adam Thomson Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/da732x.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index ffe9617..911c26c 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -876,11 +876,11 @@ static const struct snd_soc_dapm_widget da732x_dapm_widgets[] = { static const struct snd_soc_dapm_route da732x_dapm_routes[] = { /* Inputs */ - {"AUX1L PGA", "NULL", "AUX1L"}, - {"AUX1R PGA", "NULL", "AUX1R"}, + {"AUX1L PGA", NULL, "AUX1L"}, + {"AUX1R PGA", NULL, "AUX1R"}, {"MIC1 PGA", NULL, "MIC1"}, - {"MIC2 PGA", "NULL", "MIC2"}, - {"MIC3 PGA", "NULL", "MIC3"}, + {"MIC2 PGA", NULL, "MIC2"}, + {"MIC3 PGA", NULL, "MIC3"}, /* Capture Path */ {"ADC1 Left MUX", "MIC1", "MIC1 PGA"}, -- cgit v1.1 From cdd3d2a93f08823a0b9802147dc28c99029dfdfd Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 27 Feb 2015 14:13:04 +0100 Subject: ASoC: sn95031: Fix control-less DAPM routes Routes without a control must use NULL for the control name. The sn95031 driver uses "NULL" instead in a few places. Previous to commit 5fe5b767dc6f ("ASoC: dapm: Do not pretend to support controls for non mixer/mux widgets") the DAPM core silently ignored non-NULL controls on non-mixer and non-mux routes. But starting with that commit it will complain and not add the route breaking the sn95031 driver in the process. This patch replaces the incorrect "NULL" control name with NULL to fix the issue. Fixes: 5fe5b767dc6f ("ASoC: dapm: Do not pretend to support controls for non mixer/mux widgets") Signed-off-by: Lars-Peter Clausen Acked-by: Vinod Koul Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/sn95031.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 47b257e..82095d6c 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -538,8 +538,8 @@ static const struct snd_soc_dapm_route sn95031_audio_map[] = { /* speaker map */ { "IHFOUTL", NULL, "Speaker Rail"}, { "IHFOUTR", NULL, "Speaker Rail"}, - { "IHFOUTL", "NULL", "Speaker Left Playback"}, - { "IHFOUTR", "NULL", "Speaker Right Playback"}, + { "IHFOUTL", NULL, "Speaker Left Playback"}, + { "IHFOUTR", NULL, "Speaker Right Playback"}, { "Speaker Left Playback", NULL, "Speaker Left Filter"}, { "Speaker Right Playback", NULL, "Speaker Right Filter"}, { "Speaker Left Filter", NULL, "IHFDAC Left"}, -- cgit v1.1 From 90aff15b3e0858eaefdcd390e64849542845d489 Mon Sep 17 00:00:00 2001 From: "Maciej S. Szmigiero" Date: Wed, 4 Mar 2015 22:48:30 +0100 Subject: fsl_ssi: fix of_property_read_u32_array return value check of_property_read_u32_array returns 0 on success, so the return value shouldn't be inverted twice, first on assignment then in condition expression. Signed-off-by: Maciej Szmigiero Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index d7365c5..134388f 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1227,7 +1227,7 @@ static int fsl_ssi_imx_probe(struct platform_device *pdev, ssi_private->dma_params_tx.addr = ssi_private->ssi_phys + CCSR_SSI_STX0; ssi_private->dma_params_rx.addr = ssi_private->ssi_phys + CCSR_SSI_SRX0; - ret = !of_property_read_u32_array(np, "dmas", dmas, 4); + ret = of_property_read_u32_array(np, "dmas", dmas, 4); if (ssi_private->use_dma && !ret && dmas[2] == IMX_DMATYPE_SSI_DUAL) { ssi_private->use_dual_fifo = true; /* When using dual fifo mode, we need to keep watermark -- cgit v1.1 From 6c8ca30eec7b6f8eb09c957e8dcced89e5f100c7 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Wed, 4 Mar 2015 21:05:04 -0800 Subject: ASoC: fsl_ssi: Don't try to round-up for PM divisor calculation According to i.MX6 Series Reference Manual, the formula to calculate the sys clock is sysclk rate = bclk rate * (div2 + 1) * (7 * psr + 1) * (pm + 1) * 2 Commit aafa85e71a75 ("ASoC: fsl_ssi: Add DAI master mode support for SSI on i.MX series") added the divisor calculation which relies on the clk_round_rate(). However, at that time, clk_round_rate() didn't provide closest clock rates for some cases because it might not use a correct rounding policy. So using the original formula (pm + 1) for PM divisor was not able to give us a desired clock rate. And then we used (pm + 2) to do the trick. However, the clk-divider driver has been refined a lot since commit b11d282dbea2 ("clk: divider: fix rate calculation for fractional rates") Now using (pm + 2) trick would result an incorrect clock rate. So this patch fixes the problem by removing the useless trick. Reported-by: Stephane Cerveau Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 134388f..7eebc08 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -603,7 +603,7 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, factor = (div2 + 1) * (7 * psr + 1) * 2; for (i = 0; i < 255; i++) { - tmprate = freq * factor * (i + 2); + tmprate = freq * factor * (i + 1); if (baudclk_is_used) clkrate = clk_get_rate(ssi_private->baudclk); -- cgit v1.1 From c7d910b87d3c8e9fcf4077089ca4327c12eee099 Mon Sep 17 00:00:00 2001 From: Eric Nelson Date: Fri, 27 Feb 2015 08:06:45 -0700 Subject: ASoC: sgtl5000: remove useless register write clearing CHRGPUMP_POWERUP The SGTL5000_CHIP_ANA_POWER register is cached. Update the cached value instead of writing it directly. Patch inspired by Russell King's more colorful remarks in this patch: https://github.com/SolidRun/linux-imx6-3.14/commit/dd4bf6a Signed-off-by: Eric Nelson Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/sgtl5000.c | 8 +------- 1 file changed, 1 insertion(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index e182e65..3593a14 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1151,13 +1151,7 @@ static int sgtl5000_set_power_regs(struct snd_soc_codec *codec) /* Enable VDDC charge pump */ ana_pwr |= SGTL5000_VDDC_CHRGPMP_POWERUP; } else if (vddio >= 3100 && vdda >= 3100) { - /* - * if vddio and vddd > 3.1v, - * charge pump should be clean before set ana_pwr - */ - snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, - SGTL5000_VDDC_CHRGPMP_POWERUP, 0); - + ana_pwr &= ~SGTL5000_VDDC_CHRGPMP_POWERUP; /* VDDC use VDDIO rail */ lreg_ctrl |= SGTL5000_VDDC_ASSN_OVRD; lreg_ctrl |= SGTL5000_VDDC_MAN_ASSN_VDDIO << -- cgit v1.1 From a4ee556137a5bb4b542c5023e6fead4b7cf33495 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Fri, 6 Mar 2015 10:12:58 +0800 Subject: ASoC: rt286: Change the DMI mapping for Dino The board ID will be changed between revisions. So, it is better to map it by project name. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt286.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index f374840..9b541e5 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -1198,7 +1198,7 @@ static struct dmi_system_id dmi_dell_dino[] = { .ident = "Dell Dino", .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc."), - DMI_MATCH(DMI_BOARD_NAME, "0144P8") + DMI_MATCH(DMI_PRODUCT_NAME, "XPS 13 9343") } }, { } -- cgit v1.1 From 3fe0607a04ed7deea7c048052fd63b8670e7a176 Mon Sep 17 00:00:00 2001 From: "Lu, Han" Date: Wed, 25 Feb 2015 08:26:21 +0800 Subject: ASoC: Intel: remove conflicts when load/unload multiple firmware images Details: 1. Unload all modules on fw_list of dsp when suspend, and reload all modules on fw_list when resume. 2. A DSP expects only one scratch, but hsw_parse_fw_image() allocates scratch blocks for each firmware image it parses. Move the allocate function sst_block_alloc_scratch() out of hsw_parse_fw_image() to make sure a scratch be allocated only after all firmware images be parsed. Signed-off-by: Lu, Han Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-dsp.c | 3 --- sound/soc/intel/sst-haswell-ipc.c | 32 ++++++++++++++++++++++++-------- 2 files changed, 24 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c index 57039b0..f6e1e6b 100644 --- a/sound/soc/intel/sst-haswell-dsp.c +++ b/sound/soc/intel/sst-haswell-dsp.c @@ -207,9 +207,6 @@ static int hsw_parse_fw_image(struct sst_fw *sst_fw) module = (void *)module + sizeof(*module) + module->mod_size; } - /* allocate scratch mem regions */ - sst_block_alloc_scratch(dsp); - return 0; } diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 8156cc1..6c7052a 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -1870,6 +1870,7 @@ static void sst_hsw_drop_all(struct sst_hsw *hsw) int sst_hsw_dsp_load(struct sst_hsw *hsw) { struct sst_dsp *dsp = hsw->dsp; + struct sst_fw *sst_fw, *t; int ret; dev_dbg(hsw->dev, "loading audio DSP...."); @@ -1886,12 +1887,17 @@ int sst_hsw_dsp_load(struct sst_hsw *hsw) return ret; } - ret = sst_fw_reload(hsw->sst_fw); - if (ret < 0) { - dev_err(hsw->dev, "error: SST FW reload failed\n"); - sst_dsp_dma_put_channel(dsp); - return -ENOMEM; + list_for_each_entry_safe_reverse(sst_fw, t, &dsp->fw_list, list) { + ret = sst_fw_reload(sst_fw); + if (ret < 0) { + dev_err(hsw->dev, "error: SST FW reload failed\n"); + sst_dsp_dma_put_channel(dsp); + return -ENOMEM; + } } + ret = sst_block_alloc_scratch(hsw->dsp); + if (ret < 0) + return -EINVAL; sst_dsp_dma_put_channel(dsp); return 0; @@ -1947,12 +1953,17 @@ int sst_hsw_dsp_runtime_suspend(struct sst_hsw *hsw) int sst_hsw_dsp_runtime_sleep(struct sst_hsw *hsw) { - sst_fw_unload(hsw->sst_fw); - sst_block_free_scratch(hsw->dsp); + struct sst_fw *sst_fw, *t; + struct sst_dsp *dsp = hsw->dsp; + + list_for_each_entry_safe(sst_fw, t, &dsp->fw_list, list) { + sst_fw_unload(sst_fw); + } + sst_block_free_scratch(dsp); hsw->boot_complete = false; - sst_dsp_sleep(hsw->dsp); + sst_dsp_sleep(dsp); return 0; } @@ -2081,6 +2092,11 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) goto fw_err; } + /* allocate scratch mem regions */ + ret = sst_block_alloc_scratch(hsw->dsp); + if (ret < 0) + goto boot_err; + /* wait for DSP boot completion */ sst_dsp_boot(hsw->dsp); ret = wait_event_timeout(hsw->boot_wait, hsw->boot_complete, -- cgit v1.1 From 34e81ab4556f3b1371763861e74e3600818924b5 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 7 Mar 2015 19:34:03 +0100 Subject: ASoC: Fix component lists locking Any access to the component_list, codec_list and platform_list needs to be properly locked by the client_mutex. Otherwise undefined behavior can occur if the list is modified in one thread and concurrently accessed from another thread. This patch adds the missing locking to the debugfs file handlers that display the registered components, as well as the various components unregister functions. Furthermore the client_lock is now held for the whole snd_soc_instantiate_card() sequence to make sure that component removal does not race against the card registration. Reported-by: Takashi Iwai Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 41 ++++++++++++++++++++++++++++++----------- 1 file changed, 30 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 30579ca..e5c9908 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -347,6 +347,8 @@ static ssize_t codec_list_read_file(struct file *file, char __user *user_buf, if (!buf) return -ENOMEM; + mutex_lock(&client_mutex); + list_for_each_entry(codec, &codec_list, list) { len = snprintf(buf + ret, PAGE_SIZE - ret, "%s\n", codec->component.name); @@ -358,6 +360,8 @@ static ssize_t codec_list_read_file(struct file *file, char __user *user_buf, } } + mutex_unlock(&client_mutex); + if (ret >= 0) ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret); @@ -382,6 +386,8 @@ static ssize_t dai_list_read_file(struct file *file, char __user *user_buf, if (!buf) return -ENOMEM; + mutex_lock(&client_mutex); + list_for_each_entry(component, &component_list, list) { list_for_each_entry(dai, &component->dai_list, list) { len = snprintf(buf + ret, PAGE_SIZE - ret, "%s\n", @@ -395,6 +401,8 @@ static ssize_t dai_list_read_file(struct file *file, char __user *user_buf, } } + mutex_unlock(&client_mutex); + ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret); kfree(buf); @@ -418,6 +426,8 @@ static ssize_t platform_list_read_file(struct file *file, if (!buf) return -ENOMEM; + mutex_lock(&client_mutex); + list_for_each_entry(platform, &platform_list, list) { len = snprintf(buf + ret, PAGE_SIZE - ret, "%s\n", platform->component.name); @@ -429,6 +439,8 @@ static ssize_t platform_list_read_file(struct file *file, } } + mutex_unlock(&client_mutex); + ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret); kfree(buf); @@ -836,6 +848,8 @@ static struct snd_soc_component *soc_find_component( { struct snd_soc_component *component; + lockdep_assert_held(&client_mutex); + list_for_each_entry(component, &component_list, list) { if (of_node) { if (component->dev->of_node == of_node) @@ -854,6 +868,8 @@ static struct snd_soc_dai *snd_soc_find_dai( struct snd_soc_component *component; struct snd_soc_dai *dai; + lockdep_assert_held(&client_mutex); + /* Find CPU DAI from registered DAIs*/ list_for_each_entry(component, &component_list, list) { if (dlc->of_node && component->dev->of_node != dlc->of_node) @@ -1508,6 +1524,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) struct snd_soc_codec *codec; int ret, i, order; + mutex_lock(&client_mutex); mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_INIT); /* bind DAIs */ @@ -1662,6 +1679,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) card->instantiated = 1; snd_soc_dapm_sync(&card->dapm); mutex_unlock(&card->mutex); + mutex_unlock(&client_mutex); return 0; @@ -1680,6 +1698,7 @@ card_probe_error: base_error: mutex_unlock(&card->mutex); + mutex_unlock(&client_mutex); return ret; } @@ -2713,13 +2732,6 @@ static void snd_soc_component_del_unlocked(struct snd_soc_component *component) list_del(&component->list); } -static void snd_soc_component_del(struct snd_soc_component *component) -{ - mutex_lock(&client_mutex); - snd_soc_component_del_unlocked(component); - mutex_unlock(&client_mutex); -} - int snd_soc_register_component(struct device *dev, const struct snd_soc_component_driver *cmpnt_drv, struct snd_soc_dai_driver *dai_drv, @@ -2767,14 +2779,17 @@ void snd_soc_unregister_component(struct device *dev) { struct snd_soc_component *cmpnt; + mutex_lock(&client_mutex); list_for_each_entry(cmpnt, &component_list, list) { if (dev == cmpnt->dev && cmpnt->registered_as_component) goto found; } + mutex_unlock(&client_mutex); return; found: - snd_soc_component_del(cmpnt); + snd_soc_component_del_unlocked(cmpnt); + mutex_unlock(&client_mutex); snd_soc_component_cleanup(cmpnt); kfree(cmpnt); } @@ -2882,10 +2897,14 @@ struct snd_soc_platform *snd_soc_lookup_platform(struct device *dev) { struct snd_soc_platform *platform; + mutex_lock(&client_mutex); list_for_each_entry(platform, &platform_list, list) { - if (dev == platform->dev) + if (dev == platform->dev) { + mutex_unlock(&client_mutex); return platform; + } } + mutex_unlock(&client_mutex); return NULL; } @@ -3090,15 +3109,15 @@ void snd_soc_unregister_codec(struct device *dev) { struct snd_soc_codec *codec; + mutex_lock(&client_mutex); list_for_each_entry(codec, &codec_list, list) { if (dev == codec->dev) goto found; } + mutex_unlock(&client_mutex); return; found: - - mutex_lock(&client_mutex); list_del(&codec->list); snd_soc_component_del_unlocked(&codec->component); mutex_unlock(&client_mutex); -- cgit v1.1 From a1f3f1ca66bd12c339b17a0c2ef93a093f90a277 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 8 Mar 2015 18:29:50 +0100 Subject: ALSA: hda - Fix regression of HD-audio controller fallback modes The commit [63e51fd708f5: ALSA: hda - Don't take unresponsive D3 transition too serious] introduced a conditional fallback behavior to the HD-audio controller depending on the flag set. However, it introduced a silly bug, too, that the flag was evaluated in a reverse way. This resulted in a regression of HD-audio controller driver where it can't go to the fallback mode at communication errors. Unfortunately (or fortunately?) this didn't come up until recently because the affected code path is an error handling that happens only on an unstable hardware chip. Most of recent chips work stably, thus they didn't hit this problem. Now, we've got a regression report with a VIA chip, and this seems indeed requiring the fallback to the polling mode, and finally the bug was revealed. The fix is a oneliner to remove the wrong logical NOT in the check. (Lesson learned - be careful about double negation.) The bug should be backported to stable, but the patch won't be applicable to 3.13 or earlier because of the code splits. The stable fix patches for earlier kernels will be posted later manually. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=94021 Fixes: 63e51fd708f5 ('ALSA: hda - Don't take unresponsive D3 transition too serious') Cc: # v3.14+ Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_controller.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index a2ce773..17c2637 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1164,7 +1164,7 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus, } } - if (!bus->no_response_fallback) + if (bus->no_response_fallback) return -1; if (!chip->polling_mode && chip->poll_count < 2) { -- cgit v1.1 From 5b1274efe2a24eb5a85a00cc48c334b1cdfc75aa Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Tue, 10 Mar 2015 21:58:48 +0900 Subject: Revert "ALSA: dice: fix wrong offsets for Dice interface" This reverts commit 8cdebf71098c07168ef6335e2f1f35d85dbe3049. The reverted commit breaks out-stream functionality of Dice driver. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/firewire/dice/dice-interface.h | 18 +++++++++--------- sound/firewire/dice/dice-proc.c | 4 ++-- 2 files changed, 11 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/firewire/dice/dice-interface.h b/sound/firewire/dice/dice-interface.h index de7602b..27b044f 100644 --- a/sound/firewire/dice/dice-interface.h +++ b/sound/firewire/dice/dice-interface.h @@ -299,23 +299,23 @@ #define RX_ISOCHRONOUS 0x008 /* + * Index of first quadlet to be interpreted; read/write. If > 0, that many + * quadlets at the beginning of each data block will be ignored, and all the + * audio and MIDI quadlets will follow. + */ +#define RX_SEQ_START 0x00c + +/* * The number of audio channels; read-only. There will be one quadlet per * channel. */ -#define RX_NUMBER_AUDIO 0x00c +#define RX_NUMBER_AUDIO 0x010 /* * The number of MIDI ports, 0-8; read-only. If > 0, there will be one * additional quadlet in each data block, following the audio quadlets. */ -#define RX_NUMBER_MIDI 0x010 - -/* - * Index of first quadlet to be interpreted; read/write. If > 0, that many - * quadlets at the beginning of each data block will be ignored, and all the - * audio and MIDI quadlets will follow. - */ -#define RX_SEQ_START 0x014 +#define RX_NUMBER_MIDI 0x014 /* * Names of all audio channels; read-only. Quadlets are byte-swapped. Names diff --git a/sound/firewire/dice/dice-proc.c b/sound/firewire/dice/dice-proc.c index ecfe20f..f5c1d1b 100644 --- a/sound/firewire/dice/dice-proc.c +++ b/sound/firewire/dice/dice-proc.c @@ -99,9 +99,9 @@ static void dice_proc_read(struct snd_info_entry *entry, } tx; struct { u32 iso; + u32 seq_start; u32 number_audio; u32 number_midi; - u32 seq_start; char names[RX_NAMES_SIZE]; u32 ac3_caps; u32 ac3_enable; @@ -204,10 +204,10 @@ static void dice_proc_read(struct snd_info_entry *entry, break; snd_iprintf(buffer, "rx %u:\n", stream); snd_iprintf(buffer, " iso channel: %d\n", (int)buf.rx.iso); + snd_iprintf(buffer, " sequence start: %u\n", buf.rx.seq_start); snd_iprintf(buffer, " audio channels: %u\n", buf.rx.number_audio); snd_iprintf(buffer, " midi ports: %u\n", buf.rx.number_midi); - snd_iprintf(buffer, " sequence start: %u\n", buf.rx.seq_start); if (quadlets >= 68) { dice_proc_fixup_string(buf.rx.names, RX_NAMES_SIZE); snd_iprintf(buffer, " names: %s\n", buf.rx.names); -- cgit v1.1 From 59294a01d7037f63fb8bf994af10ce63c618770a Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Tue, 10 Mar 2015 21:54:35 +0900 Subject: ALSA: firewire-lib: leave unit reference counting completely With previous commit, this module managed to leave the counting to each drivers, but the isochronous resources functionality still increment/decrement the count. This commit purge such codes to leave the responsibility to each drivers. Fix: c6f224dc20ad ('ALSA: firewire-lib: remove reference counting') Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/firewire/iso-resources.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/firewire/iso-resources.c b/sound/firewire/iso-resources.c index 5f17b77..f0e4d50 100644 --- a/sound/firewire/iso-resources.c +++ b/sound/firewire/iso-resources.c @@ -26,7 +26,7 @@ int fw_iso_resources_init(struct fw_iso_resources *r, struct fw_unit *unit) { r->channels_mask = ~0uLL; - r->unit = fw_unit_get(unit); + r->unit = unit; mutex_init(&r->mutex); r->allocated = false; @@ -42,7 +42,6 @@ void fw_iso_resources_destroy(struct fw_iso_resources *r) { WARN_ON(r->allocated); mutex_destroy(&r->mutex); - fw_unit_put(r->unit); } EXPORT_SYMBOL(fw_iso_resources_destroy); -- cgit v1.1 From 2bf4c1d483d911cda5dd385527194d23e5cea73d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Mar 2015 12:39:03 +0100 Subject: ASoC: adav80x: Fix wrong value references for boolean kctl The correct values referred by a boolean control are value.integer.value[], not value.enumerated.item[]. The former is long while the latter is int, so it's even incompatible on 64bit architectures. Signed-off-by: Takashi Iwai Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown Cc: --- sound/soc/codecs/adav80x.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index b67480f..4373ada 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -317,7 +317,7 @@ static int adav80x_put_deemph(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); - unsigned int deemph = ucontrol->value.enumerated.item[0]; + unsigned int deemph = ucontrol->value.integer.value[0]; if (deemph > 1) return -EINVAL; @@ -333,7 +333,7 @@ static int adav80x_get_deemph(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); - ucontrol->value.enumerated.item[0] = adav80x->deemph; + ucontrol->value.integer.value[0] = adav80x->deemph; return 0; }; -- cgit v1.1 From 08641d9b7bf915144a57a736b42642e13eb1167f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Mar 2015 12:39:04 +0100 Subject: ASoC: ak4641: Fix wrong value references for boolean kctl The correct values referred by a boolean control are value.integer.value[], not value.enumerated.item[]. The former is long while the latter is int, so it's even incompatible on 64bit architectures. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown Cc: --- sound/soc/codecs/ak4641.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index 70861c7..81b54a2 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -76,7 +76,7 @@ static int ak4641_put_deemph(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec); - int deemph = ucontrol->value.enumerated.item[0]; + int deemph = ucontrol->value.integer.value[0]; if (deemph > 1) return -EINVAL; @@ -92,7 +92,7 @@ static int ak4641_get_deemph(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec); - ucontrol->value.enumerated.item[0] = ak4641->deemph; + ucontrol->value.integer.value[0] = ak4641->deemph; return 0; }; -- cgit v1.1 From e8371aa0fecb73fb8a4b2e0296b025b11e7d6229 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Mar 2015 12:39:05 +0100 Subject: ASoC: cs4271: Fix wrong value references for boolean kctl The correct values referred by a boolean control are value.integer.value[], not value.enumerated.item[]. The former is long while the latter is int, so it's even incompatible on 64bit architectures. Signed-off-by: Takashi Iwai Acked-by: Paul Handrigan Signed-off-by: Mark Brown Cc: --- sound/soc/codecs/cs4271.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 79a4efc..7d3a6ac 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -286,7 +286,7 @@ static int cs4271_get_deemph(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); - ucontrol->value.enumerated.item[0] = cs4271->deemph; + ucontrol->value.integer.value[0] = cs4271->deemph; return 0; } @@ -296,7 +296,7 @@ static int cs4271_put_deemph(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); - cs4271->deemph = ucontrol->value.enumerated.item[0]; + cs4271->deemph = ucontrol->value.integer.value[0]; return cs4271_set_deemph(codec); } -- cgit v1.1 From d223b0e7fcfecc23380e7de45eb6a0e7b328c17c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Mar 2015 12:39:06 +0100 Subject: ASoC: es8238: Fix wrong value references for boolean kctl The correct values referred by a boolean control are value.integer.value[], not value.enumerated.item[]. The former is long while the latter is int, so it's even incompatible on 64bit architectures. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown Cc: --- sound/soc/codecs/es8328.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index f273251..c5f35a0 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -120,7 +120,7 @@ static int es8328_get_deemph(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec); - ucontrol->value.enumerated.item[0] = es8328->deemph; + ucontrol->value.integer.value[0] = es8328->deemph; return 0; } @@ -129,7 +129,7 @@ static int es8328_put_deemph(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec); - int deemph = ucontrol->value.enumerated.item[0]; + int deemph = ucontrol->value.integer.value[0]; int ret; if (deemph > 1) -- cgit v1.1 From d7f58db49d9ad92bdb12d21fdc2308b76bc2ed38 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Mar 2015 12:39:07 +0100 Subject: ASoC: pcm1681: Fix wrong value references for boolean kctl The correct values referred by a boolean control are value.integer.value[], not value.enumerated.item[]. The former is long while the latter is int, so it's even incompatible on 64bit architectures. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown Cc: --- sound/soc/codecs/pcm1681.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c index a722a02..477e13d 100644 --- a/sound/soc/codecs/pcm1681.c +++ b/sound/soc/codecs/pcm1681.c @@ -118,7 +118,7 @@ static int pcm1681_get_deemph(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); - ucontrol->value.enumerated.item[0] = priv->deemph; + ucontrol->value.integer.value[0] = priv->deemph; return 0; } @@ -129,7 +129,7 @@ static int pcm1681_put_deemph(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); - priv->deemph = ucontrol->value.enumerated.item[0]; + priv->deemph = ucontrol->value.integer.value[0]; return pcm1681_set_deemph(codec); } -- cgit v1.1 From 4c523ef61160b7d478371ddc9f48c8ce0a00d675 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Mar 2015 12:39:08 +0100 Subject: ASoC: tas5086: Fix wrong value references for boolean kctl The correct values referred by a boolean control are value.integer.value[], not value.enumerated.item[]. The former is long while the latter is int, so it's even incompatible on 64bit architectures. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown Cc: --- sound/soc/codecs/tas5086.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index 249ef5c..32942be 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -281,7 +281,7 @@ static int tas5086_get_deemph(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); - ucontrol->value.enumerated.item[0] = priv->deemph; + ucontrol->value.integer.value[0] = priv->deemph; return 0; } @@ -292,7 +292,7 @@ static int tas5086_put_deemph(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); - priv->deemph = ucontrol->value.enumerated.item[0]; + priv->deemph = ucontrol->value.integer.value[0]; return tas5086_set_deemph(codec); } -- cgit v1.1 From 00a14c2968e3d55817e0fa35c78106ca840537bf Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Mar 2015 12:39:09 +0100 Subject: ASoC: wm2000: Fix wrong value references for boolean kctl The correct values referred by a boolean control are value.integer.value[], not value.enumerated.item[]. The former is long while the latter is int, so it's even incompatible on 64bit architectures. Signed-off-by: Takashi Iwai Acked-by: Charles Keepax Signed-off-by: Mark Brown Cc: --- sound/soc/codecs/wm2000.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 8d9de49..21d5402 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -610,7 +610,7 @@ static int wm2000_anc_mode_get(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev); - ucontrol->value.enumerated.item[0] = wm2000->anc_active; + ucontrol->value.integer.value[0] = wm2000->anc_active; return 0; } @@ -620,7 +620,7 @@ static int wm2000_anc_mode_put(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev); - int anc_active = ucontrol->value.enumerated.item[0]; + int anc_active = ucontrol->value.integer.value[0]; int ret; if (anc_active > 1) @@ -643,7 +643,7 @@ static int wm2000_speaker_get(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev); - ucontrol->value.enumerated.item[0] = wm2000->spk_ena; + ucontrol->value.integer.value[0] = wm2000->spk_ena; return 0; } @@ -653,7 +653,7 @@ static int wm2000_speaker_put(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev); - int val = ucontrol->value.enumerated.item[0]; + int val = ucontrol->value.integer.value[0]; int ret; if (val > 1) -- cgit v1.1 From bd14016fbf31aa199026f1e2358eab695f374eb1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Mar 2015 12:39:10 +0100 Subject: ASoC: wm8731: Fix wrong value references for boolean kctl The correct values referred by a boolean control are value.integer.value[], not value.enumerated.item[]. The former is long while the latter is int, so it's even incompatible on 64bit architectures. Signed-off-by: Takashi Iwai Acked-by: Charles Keepax Signed-off-by: Mark Brown Cc: --- sound/soc/codecs/wm8731.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 098c143..c6d1053 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -125,7 +125,7 @@ static int wm8731_get_deemph(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec); - ucontrol->value.enumerated.item[0] = wm8731->deemph; + ucontrol->value.integer.value[0] = wm8731->deemph; return 0; } @@ -135,7 +135,7 @@ static int wm8731_put_deemph(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec); - int deemph = ucontrol->value.enumerated.item[0]; + int deemph = ucontrol->value.integer.value[0]; int ret = 0; if (deemph > 1) -- cgit v1.1 From 24cc883c1fd16df34211ae41624aa6d3cd906693 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Mar 2015 12:39:11 +0100 Subject: ASoC: wm8903: Fix wrong value references for boolean kctl The correct values referred by a boolean control are value.integer.value[], not value.enumerated.item[]. The former is long while the latter is int, so it's even incompatible on 64bit architectures. Signed-off-by: Takashi Iwai Acked-by: Charles Keepax Signed-off-by: Mark Brown Cc: --- sound/soc/codecs/wm8903.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index dde462c..04b04f8 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -442,7 +442,7 @@ static int wm8903_get_deemph(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); - ucontrol->value.enumerated.item[0] = wm8903->deemph; + ucontrol->value.integer.value[0] = wm8903->deemph; return 0; } @@ -452,7 +452,7 @@ static int wm8903_put_deemph(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); - int deemph = ucontrol->value.enumerated.item[0]; + int deemph = ucontrol->value.integer.value[0]; int ret = 0; if (deemph > 1) -- cgit v1.1 From eaddf6fd959074f6a6e71deffe079c71eef35da6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Mar 2015 12:39:12 +0100 Subject: ASoC: wm8904: Fix wrong value references for boolean kctl The correct values referred by a boolean control are value.integer.value[], not value.enumerated.item[]. The former is long while the latter is int, so it's even incompatible on 64bit architectures. Signed-off-by: Takashi Iwai Acked-by: Charles Keepax Signed-off-by: Mark Brown Cc: --- sound/soc/codecs/wm8904.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index d3b3f57..215e93c 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -525,7 +525,7 @@ static int wm8904_get_deemph(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec); - ucontrol->value.enumerated.item[0] = wm8904->deemph; + ucontrol->value.integer.value[0] = wm8904->deemph; return 0; } @@ -534,7 +534,7 @@ static int wm8904_put_deemph(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec); - int deemph = ucontrol->value.enumerated.item[0]; + int deemph = ucontrol->value.integer.value[0]; if (deemph > 1) return -EINVAL; -- cgit v1.1 From 07892b10356f17717abdc578acbef72db86c880e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Mar 2015 12:39:13 +0100 Subject: ASoC: wm8955: Fix wrong value references for boolean kctl The correct values referred by a boolean control are value.integer.value[], not value.enumerated.item[]. The former is long while the latter is int, so it's even incompatible on 64bit architectures. Signed-off-by: Takashi Iwai Acked-by: Charles Keepax Signed-off-by: Mark Brown Cc: --- sound/soc/codecs/wm8955.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index 1ab2d46..00bec91 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -393,7 +393,7 @@ static int wm8955_get_deemph(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm8955_priv *wm8955 = snd_soc_codec_get_drvdata(codec); - ucontrol->value.enumerated.item[0] = wm8955->deemph; + ucontrol->value.integer.value[0] = wm8955->deemph; return 0; } @@ -402,7 +402,7 @@ static int wm8955_put_deemph(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm8955_priv *wm8955 = snd_soc_codec_get_drvdata(codec); - int deemph = ucontrol->value.enumerated.item[0]; + int deemph = ucontrol->value.integer.value[0]; if (deemph > 1) return -EINVAL; -- cgit v1.1 From b4a18c8b1af15ebfa9054a3d2aef7b0a7e6f2a05 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Mar 2015 12:39:14 +0100 Subject: ASoC: wm8960: Fix wrong value references for boolean kctl The correct values referred by a boolean control are value.integer.value[], not value.enumerated.item[]. The former is long while the latter is int, so it's even incompatible on 64bit architectures. Signed-off-by: Takashi Iwai Acked-by: Charles Keepax Signed-off-by: Mark Brown Cc: --- sound/soc/codecs/wm8960.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index cf8fecf..3035d98 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -184,7 +184,7 @@ static int wm8960_get_deemph(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); - ucontrol->value.enumerated.item[0] = wm8960->deemph; + ucontrol->value.integer.value[0] = wm8960->deemph; return 0; } @@ -193,7 +193,7 @@ static int wm8960_put_deemph(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); - int deemph = ucontrol->value.enumerated.item[0]; + int deemph = ucontrol->value.integer.value[0]; if (deemph > 1) return -EINVAL; -- cgit v1.1 From 4b0b669b86a963f71feaa1a694e881832fdf4f86 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Mar 2015 12:39:15 +0100 Subject: ASoC: wm9712: Fix wrong value references for boolean kctl The correct values referred by a boolean control are value.integer.value[], not value.enumerated.item[]. The former is long while the latter is int, so it's even incompatible on 64bit architectures. Signed-off-by: Takashi Iwai Acked-by: Charles Keepax Signed-off-by: Mark Brown Cc: --- sound/soc/codecs/wm9712.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 9517571..98c9525 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -180,7 +180,7 @@ static int wm9712_hp_mixer_put(struct snd_kcontrol *kcontrol, struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol); struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec); - unsigned int val = ucontrol->value.enumerated.item[0]; + unsigned int val = ucontrol->value.integer.value[0]; struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; unsigned int mixer, mask, shift, old; @@ -193,7 +193,7 @@ static int wm9712_hp_mixer_put(struct snd_kcontrol *kcontrol, mutex_lock(&wm9712->lock); old = wm9712->hp_mixer[mixer]; - if (ucontrol->value.enumerated.item[0]) + if (ucontrol->value.integer.value[0]) wm9712->hp_mixer[mixer] |= mask; else wm9712->hp_mixer[mixer] &= ~mask; @@ -231,7 +231,7 @@ static int wm9712_hp_mixer_get(struct snd_kcontrol *kcontrol, mixer = mc->shift >> 8; shift = mc->shift & 0xff; - ucontrol->value.enumerated.item[0] = + ucontrol->value.integer.value[0] = (wm9712->hp_mixer[mixer] >> shift) & 1; return 0; -- cgit v1.1 From 87a8b286e2f63c048a586dc677140d4a5b5808aa Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Mar 2015 12:39:16 +0100 Subject: ASoC: wm9713: Fix wrong value references for boolean kctl The correct values referred by a boolean control are value.integer.value[], not value.enumerated.item[]. The former is long while the latter is int, so it's even incompatible on 64bit architectures. Signed-off-by: Takashi Iwai Acked-by: Charles Keepax Signed-off-by: Mark Brown Cc: --- sound/soc/codecs/wm9713.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 6822291..7955295 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -255,7 +255,7 @@ static int wm9713_hp_mixer_put(struct snd_kcontrol *kcontrol, struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol); struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec); - unsigned int val = ucontrol->value.enumerated.item[0]; + unsigned int val = ucontrol->value.integer.value[0]; struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; unsigned int mixer, mask, shift, old; @@ -268,7 +268,7 @@ static int wm9713_hp_mixer_put(struct snd_kcontrol *kcontrol, mutex_lock(&wm9713->lock); old = wm9713->hp_mixer[mixer]; - if (ucontrol->value.enumerated.item[0]) + if (ucontrol->value.integer.value[0]) wm9713->hp_mixer[mixer] |= mask; else wm9713->hp_mixer[mixer] &= ~mask; @@ -306,7 +306,7 @@ static int wm9713_hp_mixer_get(struct snd_kcontrol *kcontrol, mixer = mc->shift >> 8; shift = mc->shift & 0xff; - ucontrol->value.enumerated.item[0] = + ucontrol->value.integer.value[0] = (wm9713->hp_mixer[mixer] >> shift) & 1; return 0; -- cgit v1.1 From ddb6ca75b5671b8fbf1909bc588c449ee74b34f9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 11 Mar 2015 16:05:19 +0100 Subject: ALSA: hda - Fix built-in mic on Compaq Presario CQ60 Compaq Presario CQ60 laptop with CX20561 gives a wrong pin for the built-in mic NID 0x17 instead of NID 0x1d, and it results in the non-working mic. This patch just remaps the pin correctly via fixup. Bugzilla: https://bugzilla.opensuse.org/show_bug.cgi?id=920604 Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 11 +++++++++++ 1 file changed, 11 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index fd3ed18..da67ea8 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -223,6 +223,7 @@ enum { CXT_PINCFG_LENOVO_TP410, CXT_PINCFG_LEMOTE_A1004, CXT_PINCFG_LEMOTE_A1205, + CXT_PINCFG_COMPAQ_CQ60, CXT_FIXUP_STEREO_DMIC, CXT_FIXUP_INC_MIC_BOOST, CXT_FIXUP_HEADPHONE_MIC_PIN, @@ -660,6 +661,15 @@ static const struct hda_fixup cxt_fixups[] = { .type = HDA_FIXUP_PINS, .v.pins = cxt_pincfg_lemote, }, + [CXT_PINCFG_COMPAQ_CQ60] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + /* 0x17 was falsely set up as a mic, it should 0x1d */ + { 0x17, 0x400001f0 }, + { 0x1d, 0x97a70120 }, + { } + } + }, [CXT_FIXUP_STEREO_DMIC] = { .type = HDA_FIXUP_FUNC, .v.func = cxt_fixup_stereo_dmic, @@ -769,6 +779,7 @@ static const struct hda_model_fixup cxt5047_fixup_models[] = { }; static const struct snd_pci_quirk cxt5051_fixups[] = { + SND_PCI_QUIRK(0x103c, 0x360b, "Compaq CQ60", CXT_PINCFG_COMPAQ_CQ60), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT_PINCFG_LENOVO_X200), {} }; -- cgit v1.1 From 81efec851477957f964f9978921d5ae36d521d45 Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Wed, 25 Feb 2015 22:53:37 +0800 Subject: ASoC: fsl_spdif: fix struct clk pointer comparing Since commit 035a61c314eb ("clk: Make clk API return per-user struct clk instances"), clk API users can no longer check if two struct clk pointers are pointing to the same hardware clock, i.e. struct clk_hw, by simply comparing two pointers. That's because with the per-user clk change, a brand new struct clk is created whenever clients try to look up the clock by calling clk_get() or sister functions like clk_get_sys() and of_clk_get(). This changes the original behavior where the struct clk is only created for once when clock driver registers the clock to CCF in the first place. The net change here is before commit 035a61c314eb the struct clk pointer is unique for given hardware clock, while after the commit the pointers returned by clk lookup calls become different for the same hardware clock. That said, the struct clk pointer comparing in the code doesn't work any more. Call helper function clk_is_match() instead to fix the problem. Signed-off-by: Shawn Guo Signed-off-by: Michael Turquette Signed-off-by: Stephen Boyd --- sound/soc/fsl/fsl_spdif.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 75870c0..91eb3ae 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1049,7 +1049,7 @@ static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv, enum spdif_txrate index, bool round) { const u32 rate[] = { 32000, 44100, 48000, 96000, 192000 }; - bool is_sysclk = clk == spdif_priv->sysclk; + bool is_sysclk = clk_is_match(clk, spdif_priv->sysclk); u64 rate_ideal, rate_actual, sub; u32 sysclk_dfmin, sysclk_dfmax; u32 txclk_df, sysclk_df, arate; @@ -1143,7 +1143,7 @@ static int fsl_spdif_probe_txclk(struct fsl_spdif_priv *spdif_priv, spdif_priv->txclk_src[index], rate[index]); dev_dbg(&pdev->dev, "use txclk df %d for %dHz sample rate\n", spdif_priv->txclk_df[index], rate[index]); - if (spdif_priv->txclk[index] == spdif_priv->sysclk) + if (clk_is_match(spdif_priv->txclk[index], spdif_priv->sysclk)) dev_dbg(&pdev->dev, "use sysclk df %d for %dHz sample rate\n", spdif_priv->sysclk_df[index], rate[index]); dev_dbg(&pdev->dev, "the best rate for %dHz sample rate is %dHz\n", -- cgit v1.1 From aaa6d06282a749d0df8e5e22e73f8a3372f96853 Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Wed, 25 Feb 2015 22:53:38 +0800 Subject: ASoC: kirkwood: fix struct clk pointer comparing Since commit 035a61c314eb ("clk: Make clk API return per-user struct clk instances"), clk API users can no longer check if two struct clk pointers are pointing to the same hardware clock, i.e. struct clk_hw, by simply comparing two pointers. That's because with the per-user clk change, a brand new struct clk is created whenever clients try to look up the clock by calling clk_get() or sister functions like clk_get_sys() and of_clk_get(). This changes the original behavior where the struct clk is only created for once when clock driver registers the clock to CCF in the first place. The net change here is before commit 035a61c314eb the struct clk pointer is unique for given hardware clock, while after the commit the pointers returned by clk lookup calls become different for the same hardware clock. That said, the struct clk pointer comparing in the code doesn't work any more. Call helper function clk_is_match() instead to fix the problem. Signed-off-by: Shawn Guo Signed-off-by: Michael Turquette Signed-off-by: Stephen Boyd --- sound/soc/kirkwood/kirkwood-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index def7d82..d194830 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -579,7 +579,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) if (PTR_ERR(priv->extclk) == -EPROBE_DEFER) return -EPROBE_DEFER; } else { - if (priv->extclk == priv->clk) { + if (clk_is_match(priv->extclk, priv->clk)) { devm_clk_put(&pdev->dev, priv->extclk); priv->extclk = ERR_PTR(-EINVAL); } else { -- cgit v1.1 From be3bb8236db2d0fcd705062ae2e2a9d75131222f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 11 Mar 2015 18:12:49 +0100 Subject: ALSA: control: Add sanity checks for user ctl id name string There was no check about the id string of user control elements, so we accepted even a control element with an empty string, which is obviously bogus. This patch adds more sanity checks of id strings. Cc: Signed-off-by: Takashi Iwai --- sound/core/control.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/core/control.c b/sound/core/control.c index 35324a8..eeb691d 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1170,6 +1170,10 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, if (info->count < 1) return -EINVAL; + if (!*info->id.name) + return -EINVAL; + if (strnlen(info->id.name, sizeof(info->id.name)) >= sizeof(info->id.name)) + return -EINVAL; access = info->access == 0 ? SNDRV_CTL_ELEM_ACCESS_READWRITE : (info->access & (SNDRV_CTL_ELEM_ACCESS_READWRITE| SNDRV_CTL_ELEM_ACCESS_INACTIVE| -- cgit v1.1 From fcdcd1dec6d2c7b718385ec743ae5a9a233edad4 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Thu, 12 Mar 2015 09:41:32 +0100 Subject: ALSA: snd-usb: add quirks for Roland UA-22 MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The device complies to the UAC1 standard but hides that fact with proprietary descriptors. The autodetect quirk for Roland devices catches the audio interface but misses the MIDI part, so a specific quirk is needed. Signed-off-by: Daniel Mack Reported-by: Rafa Lafuente Tested-by: Raphaël Doursenaud Cc: Signed-off-by: Takashi Iwai --- sound/usb/quirks-table.h | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) (limited to 'sound') diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 67d4765..07f984d 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1773,6 +1773,36 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + USB_DEVICE(0x0582, 0x0159), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "Roland", */ + /* .product_name = "UA-22", */ + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0001, + .in_cables = 0x0001 + } + }, + { + .ifnum = -1 + } + } + } +}, /* this catches most recent vendor-specific Roland devices */ { .match_flags = USB_DEVICE_ID_MATCH_VENDOR | -- cgit v1.1 From bad994f5b4ab57eec8d56c180edca00505c3eeb2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 Mar 2015 20:28:04 +0100 Subject: ALSA: hda - Set single_adc_amp flag for CS420x codecs CS420x codecs seem to deal only the single amps of ADC nodes even though the nodes receive multiple inputs. This leads to the inconsistent amp value after S3/S4 resume, for example. The fix is just to set codec->single_adc_amp flag. Then the driver handles these ADC amps as if single connections. Reported-and-tested-by: Vasil Zlatanov Cc: # 3.9+ Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 1589c9b..ab687ff 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -584,6 +584,7 @@ static int patch_cs420x(struct hda_codec *codec) return -ENOMEM; spec->gen.automute_hook = cs_automute; + codec->single_adc_amp = 1; snd_hda_pick_fixup(codec, cs420x_models, cs420x_fixup_tbl, cs420x_fixups); -- cgit v1.1 From 2ddee91abe9cc34ddb6294ee14702b46ae07d460 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 Mar 2015 20:47:15 +0100 Subject: ALSA: hda - Add workaround for MacBook Air 5,2 built-in mic MacBook Air 5,2 has the same problem as MacBook Pro 8,1 where the built-in mic records only the right channel. Apply the same workaround as MBP8,1 to spread the mono channel via a Cirrus codec vendor-specific COEF setup. Reported-and-tested-by: Vasil Zlatanov Cc: # 3.9+ Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index ab687ff..dd2b3d9 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -393,6 +393,7 @@ static const struct snd_pci_quirk cs420x_fixup_tbl[] = { SND_PCI_QUIRK(0x106b, 0x1c00, "MacBookPro 8,1", CS420X_MBP81), SND_PCI_QUIRK(0x106b, 0x2000, "iMac 12,2", CS420X_IMAC27_122), SND_PCI_QUIRK(0x106b, 0x2800, "MacBookPro 10,1", CS420X_MBP101), + SND_PCI_QUIRK(0x106b, 0x5600, "MacBookAir 5,2", CS420X_MBP81), SND_PCI_QUIRK(0x106b, 0x5b00, "MacBookAir 4,2", CS420X_MBA42), SND_PCI_QUIRK_VENDOR(0x106b, "Apple", CS420X_APPLE), {} /* terminator */ -- cgit v1.1 From ef403edb75580a3ec5d155f5de82155f0419c621 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 Mar 2015 08:30:11 +0100 Subject: ALSA: hda - Don't access stereo amps for mono channel widgets The current HDA generic parser initializes / modifies the amp values always in stereo, but this seems causing the problem on ALC3229 codec that has a few mono channel widgets: namely, these mono widgets react to actions for both channels equally. In the driver code, we do care the mono channel and create a control only for the left channel (as defined in HD-audio spec) for such a node. When the control is updated, only the left channel value is changed. However, in the resume, the right channel value is also restored from the initial value we took as stereo, and this overwrites the left channel value. This ends up being the silent output as the right channel has been never touched and remains muted. This patch covers the places where unconditional stereo amp accesses are done and converts to the conditional accesses. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=94581 Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_generic.c | 30 ++++++++++++++++++++++-------- 1 file changed, 22 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index b680b4e..fe18071 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -692,7 +692,23 @@ static void init_amp(struct hda_codec *codec, hda_nid_t nid, int dir, int idx) { unsigned int caps = query_amp_caps(codec, nid, dir); int val = get_amp_val_to_activate(codec, nid, dir, caps, false); - snd_hda_codec_amp_init_stereo(codec, nid, dir, idx, 0xff, val); + + if (get_wcaps(codec, nid) & AC_WCAP_STEREO) + snd_hda_codec_amp_init_stereo(codec, nid, dir, idx, 0xff, val); + else + snd_hda_codec_amp_init(codec, nid, 0, dir, idx, 0xff, val); +} + +/* update the amp, doing in stereo or mono depending on NID */ +static int update_amp(struct hda_codec *codec, hda_nid_t nid, int dir, int idx, + unsigned int mask, unsigned int val) +{ + if (get_wcaps(codec, nid) & AC_WCAP_STEREO) + return snd_hda_codec_amp_stereo(codec, nid, dir, idx, + mask, val); + else + return snd_hda_codec_amp_update(codec, nid, 0, dir, idx, + mask, val); } /* calculate amp value mask we can modify; @@ -732,7 +748,7 @@ static void activate_amp(struct hda_codec *codec, hda_nid_t nid, int dir, return; val &= mask; - snd_hda_codec_amp_stereo(codec, nid, dir, idx, mask, val); + update_amp(codec, nid, dir, idx, mask, val); } static void activate_amp_out(struct hda_codec *codec, struct nid_path *path, @@ -4424,13 +4440,11 @@ static void mute_all_mixer_nid(struct hda_codec *codec, hda_nid_t mix) has_amp = nid_has_mute(codec, mix, HDA_INPUT); for (i = 0; i < nums; i++) { if (has_amp) - snd_hda_codec_amp_stereo(codec, mix, - HDA_INPUT, i, - 0xff, HDA_AMP_MUTE); + update_amp(codec, mix, HDA_INPUT, i, + 0xff, HDA_AMP_MUTE); else if (nid_has_volume(codec, conn[i], HDA_OUTPUT)) - snd_hda_codec_amp_stereo(codec, conn[i], - HDA_OUTPUT, 0, - 0xff, HDA_AMP_MUTE); + update_amp(codec, conn[i], HDA_OUTPUT, 0, + 0xff, HDA_AMP_MUTE); } } -- cgit v1.1 From cc261738add93947d138d2fabad9f4dbed4e5c00 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 16 Mar 2015 10:18:08 +0100 Subject: ALSA: hda - Treat stereo-to-mono mix properly The commit [ef403edb7558: ALSA: hda - Don't access stereo amps for mono channel widgets] fixed the handling of mono widgets in general, but it still misses an exceptional case: namely, a mono mixer widget taking a single stereo input. In this case, it has stereo volumes although it's a mono widget, and thus we have to take care of both left and right input channels, as stated in HD-audio spec ("7.1.3 Widget Interconnection Rules"). This patch covers this missing piece by adding proper checks of stereo amps in both the generic parser and the proc output codes. Reported-by: Raymond Yau Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_generic.c | 21 +++++++++++++++++++-- sound/pci/hda/hda_proc.c | 38 ++++++++++++++++++++++++++++++-------- 2 files changed, 49 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index fe18071..8ec5289 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -687,13 +687,30 @@ static int get_amp_val_to_activate(struct hda_codec *codec, hda_nid_t nid, return val; } +/* is this a stereo widget or a stereo-to-mono mix? */ +static bool is_stereo_amps(struct hda_codec *codec, hda_nid_t nid, int dir) +{ + unsigned int wcaps = get_wcaps(codec, nid); + hda_nid_t conn; + + if (wcaps & AC_WCAP_STEREO) + return true; + if (dir != HDA_INPUT || get_wcaps_type(wcaps) != AC_WID_AUD_MIX) + return false; + if (snd_hda_get_num_conns(codec, nid) != 1) + return false; + if (snd_hda_get_connections(codec, nid, &conn, 1) < 0) + return false; + return !!(get_wcaps(codec, conn) & AC_WCAP_STEREO); +} + /* initialize the amp value (only at the first time) */ static void init_amp(struct hda_codec *codec, hda_nid_t nid, int dir, int idx) { unsigned int caps = query_amp_caps(codec, nid, dir); int val = get_amp_val_to_activate(codec, nid, dir, caps, false); - if (get_wcaps(codec, nid) & AC_WCAP_STEREO) + if (is_stereo_amps(codec, nid, dir)) snd_hda_codec_amp_init_stereo(codec, nid, dir, idx, 0xff, val); else snd_hda_codec_amp_init(codec, nid, 0, dir, idx, 0xff, val); @@ -703,7 +720,7 @@ static void init_amp(struct hda_codec *codec, hda_nid_t nid, int dir, int idx) static int update_amp(struct hda_codec *codec, hda_nid_t nid, int dir, int idx, unsigned int mask, unsigned int val) { - if (get_wcaps(codec, nid) & AC_WCAP_STEREO) + if (is_stereo_amps(codec, nid, dir)) return snd_hda_codec_amp_stereo(codec, nid, dir, idx, mask, val); else diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index ce5a6da..05e19f7 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -134,13 +134,38 @@ static void print_amp_caps(struct snd_info_buffer *buffer, (caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT); } +/* is this a stereo widget or a stereo-to-mono mix? */ +static bool is_stereo_amps(struct hda_codec *codec, hda_nid_t nid, + int dir, unsigned int wcaps, int indices) +{ + hda_nid_t conn; + + if (wcaps & AC_WCAP_STEREO) + return true; + /* check for a stereo-to-mono mix; it must be: + * only a single connection, only for input, and only a mixer widget + */ + if (indices != 1 || dir != HDA_INPUT || + get_wcaps_type(wcaps) != AC_WID_AUD_MIX) + return false; + + if (snd_hda_get_raw_connections(codec, nid, &conn, 1) < 0) + return false; + /* the connection source is a stereo? */ + wcaps = snd_hda_param_read(codec, conn, AC_PAR_AUDIO_WIDGET_CAP); + return !!(wcaps & AC_WCAP_STEREO); +} + static void print_amp_vals(struct snd_info_buffer *buffer, struct hda_codec *codec, hda_nid_t nid, - int dir, int stereo, int indices) + int dir, unsigned int wcaps, int indices) { unsigned int val; + bool stereo; int i; + stereo = is_stereo_amps(codec, nid, dir, wcaps, indices); + dir = dir == HDA_OUTPUT ? AC_AMP_GET_OUTPUT : AC_AMP_GET_INPUT; for (i = 0; i < indices; i++) { snd_iprintf(buffer, " ["); @@ -757,12 +782,10 @@ static void print_codec_info(struct snd_info_entry *entry, (codec->single_adc_amp && wid_type == AC_WID_AUD_IN)) print_amp_vals(buffer, codec, nid, HDA_INPUT, - wid_caps & AC_WCAP_STEREO, - 1); + wid_caps, 1); else print_amp_vals(buffer, codec, nid, HDA_INPUT, - wid_caps & AC_WCAP_STEREO, - conn_len); + wid_caps, conn_len); } if (wid_caps & AC_WCAP_OUT_AMP) { snd_iprintf(buffer, " Amp-Out caps: "); @@ -771,11 +794,10 @@ static void print_codec_info(struct snd_info_entry *entry, if (wid_type == AC_WID_PIN && codec->pin_amp_workaround) print_amp_vals(buffer, codec, nid, HDA_OUTPUT, - wid_caps & AC_WCAP_STEREO, - conn_len); + wid_caps, conn_len); else print_amp_vals(buffer, codec, nid, HDA_OUTPUT, - wid_caps & AC_WCAP_STEREO, 1); + wid_caps, 1); } switch (wid_type) { -- cgit v1.1