January 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 01/29] ASoC: arizona: Replace w->codec snd_soc_dapm_to_codec(w->dapm)
by Lars-Peter Clausen 14 Jan '15

14 Jan '15

The codec field of the snd_soc_widget struct is eventually going to be removed, use snd_soc_dapm_to_codec(w->dapm) instead. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/arizona.c | 18 ++++++++++-------- 1 file changed, 10 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 9550d74..65977bc 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -84,7 +84,7 @@ static int arizona_spk_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct arizona *arizona = dev_get_drvdata(codec->dev->parent); struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); bool manual_ena = false; @@ -692,7 +692,8 @@ static void arizona_in_set_vu(struct snd_soc_codec *codec, int ena) int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct arizona_priv *priv = snd_soc_codec_get_drvdata(w->codec); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); unsigned int reg; if (w->shift % 2) @@ -705,25 +706,25 @@ int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, priv->in_pending++; break; case SND_SOC_DAPM_POST_PMU: - snd_soc_update_bits(w->codec, reg, ARIZONA_IN1L_MUTE, 0); + snd_soc_update_bits(codec, reg, ARIZONA_IN1L_MUTE, 0); /* If this is the last input pending then allow VU */ priv->in_pending--; if (priv->in_pending == 0) { msleep(1); - arizona_in_set_vu(w->codec, 1); + arizona_in_set_vu(codec, 1); } break; case SND_SOC_DAPM_PRE_PMD: - snd_soc_update_bits(w->codec, reg, + snd_soc_update_bits(codec, reg, ARIZONA_IN1L_MUTE | ARIZONA_IN_VU, ARIZONA_IN1L_MUTE | ARIZONA_IN_VU); break; case SND_SOC_DAPM_POST_PMD: /* Disable volume updates if no inputs are enabled */ - reg = snd_soc_read(w->codec, ARIZONA_INPUT_ENABLES); + reg = snd_soc_read(codec, ARIZONA_INPUT_ENABLES); if (reg == 0) - arizona_in_set_vu(w->codec, 0); + arizona_in_set_vu(codec, 0); } return 0; @@ -760,7 +761,8 @@ int arizona_hp_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct arizona_priv *priv = snd_soc_codec_get_drvdata(w->codec); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); struct arizona *arizona = priv->arizona; unsigned int mask = 1 << w->shift; unsigned int val; -- 1.8.0

3 30

[alsa-devel] [PATCH 1/5] ASoC: 88pm860x: Replace w->codec snd_soc_dapm_to_codec(w->dapm)
by Lars-Peter Clausen 14 Jan '15

14 Jan '15

The codec field of the snd_soc_widget struct is eventually going to be removed, use snd_soc_dapm_to_codec(w->dapm) instead. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/88pm860x-codec.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index a2bf27f..a0f2653 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -386,7 +386,7 @@ static int snd_soc_put_volsw_2r_out(struct snd_kcontrol *kcontrol, static int pm860x_rsync_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); /* * In order to avoid current on the load, mute power-on and power-off @@ -403,7 +403,7 @@ static int pm860x_rsync_event(struct snd_soc_dapm_widget *w, static int pm860x_dac_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); unsigned int dac = 0; int data; -- 1.8.0

4 7

[alsa-devel] [PATCH 0/4] ASoC: Make set_tdm_slot() semantics consistent
by Lars-Peter Clausen 14 Jan '15

14 Jan '15

Changes since v1: * Don't modify the fsl-esai driver as it already uses the non-inverted semantics Original cover letter below: This is something we shortly discussed during the ALSA mini conference in Duesseldorf last year. We want to do more useful stuff with the snd_soc_set_tdm_slot() API and build generic infrastructure on top of it. But there are currently two different incompatible semantics of how the tx_mask and rx_mask parameters of snd_soc_dai_set_tdm_slot() are used. If we want to be able to use the API in a generic way we need to fix this inconsistency. Both interpretations agree that the slot masks should be represented as bitmasks which bit 0 representing the first slot, bit 1 the second slot and so on. But where they differ is in the meaning of value of the bit. Some implementations assume that if a bit is set the channel is enabled and if the bit is cleared the channel is disabled, others have inverse semantics, they assume that if a bit is cleared the channel is enabled and if the bit is set the channel is disabled. The former is used by most drivers and later is only used by some the Freescale DAI drivers and the MC13783 CODEC driver. This appears to be a result of Freescale DAI drivers directly passing the masks through to the registers which expect the inverted semantics. And the MC13783 appears to have picked up the same meaning since it is used on a board which has a Freescale DAI on the CPU side of the DAI link. This series updates the MC13783 and Freescale DAI drivers to use the same semantics as other drivers and also updates the documentation of the snd_soc_dai_set_tdm_slot() function to be more specific on the intended semantics. - Lars Lars-Peter Clausen (4): ASoC: mc13783: Update set_tdm_slot() semantics ASoC: fsl: Update set_tdm_slot() semantics ASoC: fsl: Remove fsl_asoc_xlate_tdm_slot_mask() ASoC: Update snd_soc_dai_set_tdm_slot() documentation sound/soc/codecs/mc13783.c | 10 +++++----- sound/soc/fsl/eukrea-tlv320.c | 2 +- sound/soc/fsl/fsl_ssi.c | 4 ++-- sound/soc/fsl/fsl_utils.c | 27 --------------------------- sound/soc/fsl/fsl_utils.h | 3 --- sound/soc/fsl/imx-mc13783.c | 5 ++--- sound/soc/fsl/imx-ssi.c | 5 ++--- sound/soc/fsl/wm1133-ev1.c | 4 ++-- sound/soc/soc-core.c | 20 ++++++++++++++++---- 9 files changed, 30 insertions(+), 50 deletions(-) -- 1.8.0

3 6

[alsa-devel] [PATCH v3] ASoC: wm8904: fix runtime warning
by Bo Shen 14 Jan '15

14 Jan '15

Correct the path name for mux to get rid of the following warning: --->8--- wm8904 1-001a: Control not supported for path ADCL -> [Left] -> AIFOUTL wm8904 1-001a: ASoC: no dapm match for ADCL --> Left --> AIFOUTL wm8904 1-001a: ASoC: Failed to add route ADCL -> Left -> AIFOUTL wm8904 1-001a: Control not supported for path ADCR -> [Right] -> AIFOUTL wm8904 1-001a: ASoC: no dapm match for ADCR --> Right --> AIFOUTL wm8904 1-001a: ASoC: Failed to add route ADCR -> Right -> AIFOUTL wm8904 1-001a: Control not supported for path ADCL -> [Left] -> AIFOUTR wm8904 1-001a: ASoC: no dapm match for ADCL --> Left --> AIFOUTR wm8904 1-001a: ASoC: Failed to add route ADCL -> Left -> AIFOUTR wm8904 1-001a: Control not supported for path ADCR -> [Right] -> AIFOUTR wm8904 1-001a: ASoC: no dapm match for ADCR --> Right --> AIFOUTR wm8904 1-001a: ASoC: Failed to add route ADCR -> Right -> AIFOUTR wm8904 1-001a: Control not supported for path AIFINR -> [Right] -> DACL wm8904 1-001a: ASoC: no dapm match for AIFINR --> Right --> DACL wm8904 1-001a: ASoC: Failed to add route AIFINR -> Right -> DACL wm8904 1-001a: Control not supported for path AIFINL -> [Left] -> DACL wm8904 1-001a: ASoC: no dapm match for AIFINL --> Left --> DACL wm8904 1-001a: ASoC: Failed to add route AIFINL -> Left -> DACL wm8904 1-001a: Control not supported for path AIFINR -> [Right] -> DACR wm8904 1-001a: ASoC: no dapm match for AIFINR --> Right --> DACR wm8904 1-001a: ASoC: Failed to add route AIFINR -> Right -> DACR wm8904 1-001a: Control not supported for path AIFINL -> [Left] -> DACR wm8904 1-001a: ASoC: no dapm match for AIFINL --> Left --> DACR wm8904 1-001a: ASoC: Failed to add route AIFINL -> Left -> DACR ---8<--- Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- Changes in v3: - Correct the DAC related route path Changes in v2: - Correct the mux path name. - Route the mux to the pins. sound/soc/codecs/wm8904.c | 23 +++++++++++++++-------- 1 file changed, 15 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 4d2d2b1..75b87c5 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1076,10 +1076,13 @@ static const struct snd_soc_dapm_route adc_intercon[] = { { "Right Capture PGA", NULL, "Right Capture Mux" }, { "Right Capture PGA", NULL, "Right Capture Inverting Mux" }, - { "AIFOUTL", "Left", "ADCL" }, - { "AIFOUTL", "Right", "ADCR" }, - { "AIFOUTR", "Left", "ADCL" }, - { "AIFOUTR", "Right", "ADCR" }, + { "AIFOUTL Mux", "Left", "ADCL" }, + { "AIFOUTL Mux", "Right", "ADCR" }, + { "AIFOUTR Mux", "Left", "ADCL" }, + { "AIFOUTR Mux", "Right", "ADCR" }, + + { "AIFOUTL", NULL, "AIFOUTL Mux" }, + { "AIFOUTR", NULL, "AIFOUTR Mux" }, { "ADCL", NULL, "CLK_DSP" }, { "ADCL", NULL, "Left Capture PGA" }, @@ -1089,12 +1092,16 @@ static const struct snd_soc_dapm_route adc_intercon[] = { }; static const struct snd_soc_dapm_route dac_intercon[] = { - { "DACL", "Right", "AIFINR" }, - { "DACL", "Left", "AIFINL" }, + { "DACL Mux", "Left", "AIFINL" }, + { "DACL Mux", "Right", "AIFINR" }, + + { "DACR Mux", "Left", "AIFINL" }, + { "DACR Mux", "Right", "AIFINR" }, + + { "DACL", NULL, "DACL Mux" }, { "DACL", NULL, "CLK_DSP" }, - { "DACR", "Right", "AIFINR" }, - { "DACR", "Left", "AIFINL" }, + { "DACR", NULL, "DACR Mux" }, { "DACR", NULL, "CLK_DSP" }, { "Charge pump", NULL, "SYSCLK" }, -- 2.1.0.24.g4109c28

3 2

[alsa-devel] [PATCH] Asoc: sam9x5_wm8731: depend on ARCH_AT91 instead of SOC_AT91SAM9X5
by Alexandre Belloni 14 Jan '15

14 Jan '15

The SOC_AT91SAM9X5 option is going to be removed, only depend on ARCH_AT91 like for the other drivers. Signed-off-by: Alexandre Belloni <alexandre.belloni(a)free-electrons.com> --- sound/soc/atmel/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index fb3878312bf8..1579e994acf8 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -45,7 +45,7 @@ config SND_ATMEL_SOC_WM8904 config SND_AT91_SOC_SAM9X5_WM8731 tristate "SoC Audio support for WM8731-based at91sam9x5 board" - depends on ATMEL_SSC && SND_ATMEL_SOC && SOC_AT91SAM9X5 + depends on ARCH_AT91 && ATMEL_SSC && SND_ATMEL_SOC select SND_ATMEL_SOC_SSC select SND_ATMEL_SOC_DMA select SND_SOC_WM8731 -- 2.1.0

4 3

[alsa-devel] [PATCH] ASoC: fsl: imx-spdif: Set the card owner field
by Fabio Estevam 14 Jan '15

14 Jan '15

From: Fabio Estevam <fabio.estevam(a)freescale.com> Set the card owner field to avoid getting a kernel crash when the imx-spdif is unloaded while the playback is active. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Build-tested only. sound/soc/fsl/imx-spdif.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c index e94704f..33da26a 100644 --- a/sound/soc/fsl/imx-spdif.c +++ b/sound/soc/fsl/imx-spdif.c @@ -60,6 +60,7 @@ static int imx_spdif_audio_probe(struct platform_device *pdev) data->card.dev = &pdev->dev; data->card.dai_link = &data->dai; data->card.num_links = 1; + data->card.owner = THIS_MODULE; ret = snd_soc_of_parse_card_name(&data->card, "model"); if (ret) -- 1.9.1

2 1

[alsa-devel] [PATCH] conf/ucm: GoogleNyan: Add configuration
by Tomeu Vizoso 14 Jan '15

14 Jan '15

Taken from the ChromeOS sources, this configuration should apply to all Nyan boards from Google, so far HP Chromebook 14 (nyan-blaze) and Acer Chromebook 13 (nyan-big). Signed-off-by: Tomeu Vizoso <tomeu.vizoso(a)collabora.com> --- configure.ac | 1 + src/conf/ucm/GoogleNyan/GoogleNyan.conf | 5 ++ src/conf/ucm/GoogleNyan/HiFi.conf | 83 +++++++++++++++++++++++++++++++++ src/conf/ucm/GoogleNyan/Makefile.am | 4 ++ src/conf/ucm/Makefile.am | 2 +- 5 files changed, 94 insertions(+), 1 deletion(-) create mode 100644 src/conf/ucm/GoogleNyan/GoogleNyan.conf create mode 100644 src/conf/ucm/GoogleNyan/HiFi.conf create mode 100644 src/conf/ucm/GoogleNyan/Makefile.am diff --git a/configure.ac b/configure.ac index 734163c..916280b 100644 --- a/configure.ac +++ b/configure.ac @@ -654,6 +654,7 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \ src/conf/ucm/SDP4430/Makefile \ src/conf/ucm/tegraalc5632/Makefile \ src/conf/ucm/PAZ00/Makefile \ + src/conf/ucm/GoogleNyan/Makefile \ modules/Makefile modules/mixer/Makefile modules/mixer/simple/Makefile \ alsalisp/Makefile aserver/Makefile \ test/Makefile test/lsb/Makefile \ diff --git a/src/conf/ucm/GoogleNyan/GoogleNyan.conf b/src/conf/ucm/GoogleNyan/GoogleNyan.conf new file mode 100644 index 0000000..ebd0dd9 --- /dev/null +++ b/src/conf/ucm/GoogleNyan/GoogleNyan.conf @@ -0,0 +1,5 @@ + Comment "Nyan internal card" + SectionUseCase."HiFi" { + File "HiFi.conf" + Comment "Default" + } diff --git a/src/conf/ucm/GoogleNyan/HiFi.conf b/src/conf/ucm/GoogleNyan/HiFi.conf new file mode 100644 index 0000000..cb624e7 --- /dev/null +++ b/src/conf/ucm/GoogleNyan/HiFi.conf @@ -0,0 +1,83 @@ + SectionVerb { + Value { + OutputDspName "speaker_eq" + } + EnableSequence [ + cdev "hw:GoogleNyan" + cset "name='Left Speaker Mixer Left DAC Switch' on" + cset "name='Right Speaker Mixer Right DAC Switch' on" + cset "name='Headphone Left Switch' on" + cset "name='Headphone Right Switch' on" + cset "name='Digital EQ 3 Band Switch' off" + cset "name='Digital EQ 5 Band Switch' off" + cset "name='Digital EQ 7 Band Switch' off" + cset "name='Biquad Switch' off" + cset "name='Filter Mode' Music" + cset "name='ADC Oversampling Rate' 0" + cset "name='DMIC Mux' DMIC" + cset "name='MIC2 Mux' IN34" + cset "name='Right ADC Mixer MIC2 Switch' on" + cset "name='Left ADC Mixer MIC2 Switch' on" + cset "name='MIC2 Volume' 10" + cset "name='MIC2 Boost Volume' 0" + cset "name='Mic Jack Switch' off" + cset "name='Int Mic Switch' on" + cset "name='ADCR Boost Volume' 4" + cset "name='ADCL Boost Volume' 4" + cset "name='ADCR Volume' 11" + cset "name='ADCL Volume' 11" + cset "name='Left Speaker Mixer Left DAC Switch' on" + cset "name='Right Speaker Mixer Right DAC Switch' on" + cset "name='Speaker Left Mixer Volume' 2" + cset "name='Speaker Right Mixer Volume' 2" + cset "name='Record Path DC Blocking' on" + cset "name='Playback Path DC Blocking' on" + cset "name='Headphone Left Switch' on" + cset "name='Headphone Right Switch' on" + cset "name='HP Left Out Switch' off" + cset "name='HP Right Out Switch' off" + cset "name='Speaker Left Switch' on" + cset "name='Speaker Right Switch' on" + cset "name='Speakers Switch' on" + ] + DisableSequence [ + ] + } + SectionDevice."Headphone".0 { + Value { + JackName "Google Nyan Headphone Jack" + OutputDspName "" + } + EnableSequence [ + cdev "hw:GoogleNyan" + cset "name='Speakers Switch' off" + cset "name='HP Left Out Switch' on" + cset "name='HP Right Out Switch' on" + ] + DisableSequence [ + cdev "hw:GoogleNyan" + cset "name='HP Left Out Switch' off" + cset "name='HP Right Out Switch' off" + cset "name='Speakers Switch' on" + ] + } + SectionDevice."Mic".0 { + Value { + JackName "Google Nyan Mic Jack" + CaptureControl "MIC2" + } + EnableSequence [ + cdev "hw:GoogleNyan" + cset "name='Int Mic Switch' off" + cset "name='DMIC Mux' ADC" + cset "name='Mic Jack Switch' on" + cset "name='Record Path DC Blocking' on" + ] + DisableSequence [ + cdev "hw:GoogleNyan" + cset "name='Mic Jack Switch' off" + cset "name='DMIC Mux' DMIC" + cset "name='Int Mic Switch' on" + cset "name='Record Path DC Blocking' off" + ] + } diff --git a/src/conf/ucm/GoogleNyan/Makefile.am b/src/conf/ucm/GoogleNyan/Makefile.am new file mode 100644 index 0000000..47c7fc9 --- /dev/null +++ b/src/conf/ucm/GoogleNyan/Makefile.am @@ -0,0 +1,4 @@ +alsaconfigdir = @ALSA_CONFIG_DIR@ +ucmdir = $(alsaconfigdir)/ucm/GoogleNyan +ucm_DATA = GoogleNyan.conf HiFi.conf +EXTRA_DIST = $(ucm_DATA) diff --git a/src/conf/ucm/Makefile.am b/src/conf/ucm/Makefile.am index bde89df..14fc7ae 100644 --- a/src/conf/ucm/Makefile.am +++ b/src/conf/ucm/Makefile.am @@ -1 +1 @@ -SUBDIRS=DAISY-I2S PandaBoard PandaBoardES SDP4430 tegraalc5632 PAZ00 +SUBDIRS=DAISY-I2S PandaBoard PandaBoardES SDP4430 tegraalc5632 PAZ00 GoogleNyan -- 1.9.3

3 3

[alsa-devel] [PATCH 1/2] ASoC: rt5677: Use the regmap functions instead of the snd_soc functions
by Oder Chiou 14 Jan '15

14 Jan '15

The patch uses the regmap functions instead of the snd_soc functions in some cases. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 13 ++++++++++--- 1 file changed, 10 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 32c368c..24fc592 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -925,6 +925,8 @@ static int is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, static int is_using_asrc(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { + struct snd_soc_codec *codec = source->codec; + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); unsigned int reg, shift, val; if (source->reg == RT5677_ASRC_1) { @@ -991,7 +993,9 @@ static int is_using_asrc(struct snd_soc_dapm_widget *source, } } - val = (snd_soc_read(source->codec, reg) >> shift) & 0xf; + regmap_read(rt5677->regmap, reg, &val); + val = (val >> shift) & 0xf; + switch (val) { case 1 ... 6: return 1; @@ -4098,6 +4102,7 @@ static int rt5677_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { struct snd_soc_codec *codec = dai->codec; + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); unsigned int val = 0; if (rx_mask || tx_mask) @@ -4135,10 +4140,12 @@ static int rt5677_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, switch (dai->id) { case RT5677_AIF1: - snd_soc_update_bits(codec, RT5677_TDM1_CTRL1, 0x1f00, val); + regmap_update_bits(rt5677->regmap, RT5677_TDM1_CTRL1, 0x1f00, + val); break; case RT5677_AIF2: - snd_soc_update_bits(codec, RT5677_TDM2_CTRL1, 0x1f00, val); + regmap_update_bits(rt5677->regmap, RT5677_TDM2_CTRL1, 0x1f00, + val); break; default: break; -- 1.8.1.1.439.g50a6b54

2 2

[alsa-devel] [PATCH] ASoC: simple-card: Fix crash in asoc_simple_card_unref()
by Geert Uytterhoeven 14 Jan '15

14 Jan '15

If asoc_simple_card_probe() fails, asoc_simple_card_unref() may be called before dev_set_drvdata(), causing a NULL pointer dereference in asoc_simple_card_unref(): Unable to handle kernel NULL pointer dereference at virtual address 000000d4 ... PC is at asoc_simple_card_unref+0x14/0x48 LR is at asoc_simple_card_probe+0x3d4/0x40c This typically happens because asoc_simple_card_parse_of() returns -EPROBE_DEFER, but other failure modes are possible. devm_snd_soc_register_card()/snd_soc_register_card() may fail either before or after dev_set_drvdata(). Pass a snd_soc_card pointer instead of a platform_device pointer to asoc_simple_card_unref() to fix this. Note that if CONFIG_OF_DYNAMIC=n, of_node_put() is a dummy, and gcc may optimize away the loop over card->dai_link, never actually dereferencing card, and thus avoiding the crash... Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> Fixes: e512e001dafa54e5 ("ASoC: simple-card: Fix the reference count of device nodes") --- Seen on sh73a0/kzm9g-multiplatform and r8a7740/armadillo-multiplatform, first with CONFIG_OF_DYNAMIC=y, later with extra debug code that caused the loop not to be optimized away. It took me a while to understand what was really happening... I think this should be applied to -stable. In v3.15 and v3.16, only 8xx, pseries, and wsp selected OF_DYNAMIC, and I believe they don't use aSoC? In v3.17 and later, OF_DYNAMIC is selected by OF_SELFTEST/OF_UNITTEST. --- sound/soc/generic/simple-card.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index fb9240fdc9b70095..7fe3009b1c43c63c 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -452,9 +452,8 @@ static int asoc_simple_card_parse_of(struct device_node *node, } /* Decrease the reference count of the device nodes */ -static int asoc_simple_card_unref(struct platform_device *pdev) +static int asoc_simple_card_unref(struct snd_soc_card *card) { - struct snd_soc_card *card = platform_get_drvdata(pdev); struct snd_soc_dai_link *dai_link; int num_links; @@ -556,7 +555,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) return ret; err: - asoc_simple_card_unref(pdev); + asoc_simple_card_unref(&priv->snd_card); return ret; } @@ -572,7 +571,7 @@ static int asoc_simple_card_remove(struct platform_device *pdev) snd_soc_jack_free_gpios(&simple_card_mic_jack, 1, &simple_card_mic_jack_gpio); - return asoc_simple_card_unref(pdev); + return asoc_simple_card_unref(card); } static const struct of_device_id asoc_simple_of_match[] = { -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: fsl_ssi: Change irq type to 'int'
by Fabio Estevam 14 Jan '15

14 Jan '15

From: Fabio Estevam <fabio.estevam(a)freescale.com> Since commit 2ffa531078037a0 ("ASoC: fsl_ssi: Fix module unbound") the irq number is retrieved via platform_get_irq(), which may fail and return a negative number, so adapt its type to 'int'. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 65400be..46549de 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -160,7 +160,7 @@ struct fsl_ssi_soc_data { */ struct fsl_ssi_private { struct regmap *regs; - unsigned int irq; + int irq; struct snd_soc_dai_driver cpu_dai_drv; unsigned int dai_fmt; -- 1.9.1

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