November 2010 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 0/8] ASoC: fix resource leak in error paths
by Axel Lin 25 Nov '10

25 Nov '10

This serial of patches fix resource leak in error paths. Axel Lin (8): ASoC: efika-audio-fabric: fix resource leak in efika_fabric_init error path ASoC: pcm030-audio-fabric: fix resource leak in pcm030_fabric_init error path ASoC: snd-soc-afeb9260: remove unneeded platform_device_del in error path ASoC: sam9g20_wm8731: fix resource leak in at91sam9g20ek_init error path ASoC: smdk_wm9713: fix resource leak in smdk_init error path ASoC: simone: fix resource leak in simone_init error path ASoC: imx-ssi: fix resource leak ASoC: phycore-ac97: fix resource leak sound/soc/atmel/sam9g20_wm8731.c | 5 ++++- sound/soc/atmel/snd-soc-afeb9260.c | 1 - sound/soc/ep93xx/simone.c | 18 ++++++++++-------- sound/soc/fsl/efika-audio-fabric.c | 1 + sound/soc/fsl/pcm030-audio-fabric.c | 1 + sound/soc/imx/imx-ssi.c | 15 +++++++++++---- sound/soc/imx/phycore-ac97.c | 28 +++++++++++++++++++++------- sound/soc/samsung/smdk_wm9713.c | 17 ++++++++++------- 8 files changed, 58 insertions(+), 28 deletions(-) -- 1.7.2

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[alsa-devel] [PATCH] ASoC: fix SND_PXA2XX_LIB Kconfig warning
by Dmitry Artamonow 25 Nov '10

25 Nov '10

Fix following warning observed when SND_PXA2XX_SOC is set and SND_ARM isn't: warning: (SND_PXA2XX_AC97 && SOUND && !M68K && SND && SND_ARM && ARCH_PXA || SND_PXA2XX_SOC && SOUND && !M68K && SND && SND_SOC && ARCH_PXA) selects SND_PXA2XX_LIB which has unmet direct dependencies (SOUND && !M68K && SND && SND_ARM) Signed-off-by: Dmitry Artamonow <mad_soft(a)inbox.ru> --- This is fix for 2.6.37-rc3 sound/soc/pxa/Kconfig | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 37f191b..580f485 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -1,6 +1,7 @@ config SND_PXA2XX_SOC tristate "SoC Audio for the Intel PXA2xx chip" depends on ARCH_PXA + select SND_ARM select SND_PXA2XX_LIB help Say Y or M if you want to add support for codecs attached to -- 1.7.0.4

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[alsa-devel] [PATCH] ASoC: davinci-vcif - fix a memory leak
by Axel Lin 25 Nov '10

25 Nov '10

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/davinci/davinci-vcif.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c index fb4cc1e..9d2afcc 100644 --- a/sound/soc/davinci/davinci-vcif.c +++ b/sound/soc/davinci/davinci-vcif.c @@ -247,7 +247,10 @@ fail: static int davinci_vcif_remove(struct platform_device *pdev) { + struct davinci_vcif_dev *davinci_vcif_dev = dev_get_drvdata(&pdev->dev); + snd_soc_unregister_dai(&pdev->dev); + kfree(davinci_vcif_dev); return 0; } -- 1.7.2

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[alsa-devel] [PATCH] ASoC: Call snd_soc_unregister_dais instead of snd_soc_unregister_dai in sh4_soc_dai_remove
by Axel Lin 25 Nov '10

25 Nov '10

We call snd_soc_register_dais() in sh4_soc_dai_probe(), thus we should call snd_soc_unregister_dais() in sh4_soc_dai_remove(). Otherwise, we got "too many arguments to function 'snd_soc_unregister_dai'" error message. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/sh/ssi.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c index 40bbdf1..05192d9 100644 --- a/sound/soc/sh/ssi.c +++ b/sound/soc/sh/ssi.c @@ -387,7 +387,7 @@ static int __devinit sh4_soc_dai_probe(struct platform_device *pdev) static int __devexit sh4_soc_dai_remove(struct platform_device *pdev) { - snd_soc_unregister_dai(&pdev->dev, ARRAY_SIZE(sh4_ssi_dai)); + snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(sh4_ssi_dai)); return 0; } -- 1.7.2

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[alsa-devel] [PATCH] ASoC: Update MAINTAINERS for Samsung driver move
by Mark Brown 25 Nov '10

25 Nov '10

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- MAINTAINERS | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 33a4091..e585c6a 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -5133,7 +5133,7 @@ SAMSUNG AUDIO (ASoC) DRIVERS M: Jassi Brar <jassi.brar(a)samsung.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Supported -F: sound/soc/s3c24xx +F: sound/soc/samsung TLG2300 VIDEO4LINUX-2 DRIVER M: Huang Shijie <shijie8(a)gmail.com> -- 1.7.1

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[alsa-devel] [PATCH] ASoC: Make SMDK WM8580 driver unloadable
by Mark Brown 25 Nov '10

25 Nov '10

Supply a module exit function so that the driver can be unloaded. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/smdk_wm8580.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/soc/samsung/smdk_wm8580.c b/sound/soc/samsung/smdk_wm8580.c index 0ae3d57..e2d56b9 100644 --- a/sound/soc/samsung/smdk_wm8580.c +++ b/sound/soc/samsung/smdk_wm8580.c @@ -280,6 +280,12 @@ static int __init smdk_audio_init(void) } module_init(smdk_audio_init); +static void __exit smdk_audio_exit(void) +{ + platform_device_unregister(smdk_snd_device); +} +module_exit(smdk_audio_exit); + MODULE_AUTHOR("Jaswinder Singh, jassi.brar(a)samsung.com"); MODULE_DESCRIPTION("ALSA SoC SMDK WM8580"); MODULE_LICENSE("GPL"); -- 1.7.1

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[alsa-devel] [PATCH] ALSA: hda - Fix ALC660-VD/ALC861-VD capture/playback mixers
by Herton Ronaldo Krzesinski 25 Nov '10

25 Nov '10

The mixer nids passed to alc_auto_create_input_ctls are wrong: 0x15 is a pin, and 0x09 is the ADC on both ALC660-VD/ALC861-VD. Thus with current code, input playback volume/switches and input source mixer controls are not created, and recording doesn't work. Select correct mixers, 0x0b (input playback mixer) and 0x22 (capture source mixer). Reference: https://qa.mandriva.com/show_bug.cgi?id=61159 Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com.br> --- sound/pci/hda/patch_realtek.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b4b3afe..5aafae9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -16952,7 +16952,7 @@ static struct alc_config_preset alc861vd_presets[] = { static int alc861vd_auto_create_input_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { - return alc_auto_create_input_ctls(codec, cfg, 0x15, 0x09, 0); + return alc_auto_create_input_ctls(codec, cfg, 0x0b, 0x22, 0); } -- 1.7.3.2

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Re: [alsa-devel] [PATCH 5/8] ASoC: smdk_wm9713: fix resource leak in smdk_init error path
by JASWINDERSINGH BRAR 25 Nov '10

25 Nov '10

------- Original Message ------- Sender : Axel Lin<axel.lin(a)gmail.com> Date : Nov 25, 2010 16:11 (GMT+09:00) Title : [PATCH 5/8] ASoC: smdk_wm9713: fix resource leak in smdk_init error path > Fix the error path to properly free allocated resources. Acked-by: Jassi Brar <jassi.brar(a)samsung.com>

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[alsa-devel] asoc register read/write calls
by Koul, Vinod 25 Nov '10

25 Nov '10

Mark, In the asoc framework, do the codec register read-write calls need to be atomic? Will it work in non-atomic context? ~Vinod

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[alsa-devel] [PATCH] ASoC: Add WM8737 ALC support
by Mark Brown 24 Nov '10

24 Nov '10

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8737.c | 42 ++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 42 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c index a325fe6..2301338 100644 --- a/sound/soc/codecs/wm8737.c +++ b/sound/soc/codecs/wm8737.c @@ -78,6 +78,8 @@ static const unsigned int micboost_tlv[] = { static const DECLARE_TLV_DB_SCALE(pga_tlv, -9750, 50, 1); static const DECLARE_TLV_DB_SCALE(adc_tlv, -600, 600, 0); static const DECLARE_TLV_DB_SCALE(ng_tlv, -7800, 600, 0); +static const DECLARE_TLV_DB_SCALE(alc_max_tlv, -1200, 600, 0); +static const DECLARE_TLV_DB_SCALE(alc_target_tlv, -1800, 100, 0); static const char *micbias_enum_text[] = { "25%", @@ -103,6 +105,38 @@ static const char *high_cutoff_text[] = { static const struct soc_enum high_3d = SOC_ENUM_SINGLE(WM8737_3D_ENHANCE, 5, 2, high_cutoff_text); +static const char *alc_fn_text[] = { + "Disabled", "Right", "Left", "Stereo" +}; + +static const struct soc_enum alc_fn = + SOC_ENUM_SINGLE(WM8737_ALC1, 7, 4, alc_fn_text); + +static const char *alc_hold_text[] = { + "0", "2.67ms", "5.33ms", "10.66ms", "21.32ms", "42.64ms", "85.28ms", + "170.56ms", "341.12ms", "682.24ms", "1.364s", "2.728s", "5.458s", + "10.916s", "21.832s", "43.691s" +}; + +static const struct soc_enum alc_hold = + SOC_ENUM_SINGLE(WM8737_ALC2, 0, 16, alc_hold_text); + +static const char *alc_atk_text[] = { + "8.4ms", "16.8ms", "33.6ms", "67.2ms", "134.4ms", "268.8ms", "537.6ms", + "1.075s", "2.15s", "4.3s", "8.6s" +}; + +static const struct soc_enum alc_atk = + SOC_ENUM_SINGLE(WM8737_ALC3, 0, 11, alc_atk_text); + +static const char *alc_dcy_text[] = { + "33.6ms", "67.2ms", "134.4ms", "268.8ms", "537.6ms", "1.075s", "2.15s", + "4.3s", "8.6s", "17.2s", "34.41s" +}; + +static const struct soc_enum alc_dcy = + SOC_ENUM_SINGLE(WM8737_ALC3, 4, 11, alc_dcy_text); + static const struct snd_kcontrol_new wm8737_snd_controls[] = { SOC_DOUBLE_R_TLV("Mic Boost Volume", WM8737_AUDIO_PATH_L, WM8737_AUDIO_PATH_R, 6, 3, 0, micboost_tlv), @@ -132,6 +166,14 @@ SOC_SINGLE_TLV("3D ADC Volume", WM8737_3D_ENHANCE, 7, 1, 1, adc_tlv), SOC_SINGLE("Noise Gate Switch", WM8737_NOISE_GATE, 0, 1, 0), SOC_SINGLE_TLV("Noise Gate Threshold Volume", WM8737_NOISE_GATE, 2, 7, 0, ng_tlv), + +SOC_ENUM("ALC", alc_fn), +SOC_SINGLE_TLV("ALC Max Gain Volume", WM8737_ALC1, 4, 7, 0, alc_max_tlv), +SOC_SINGLE_TLV("ALC Target Volume", WM8737_ALC1, 0, 15, 0, alc_target_tlv), +SOC_ENUM("ALC Hold Time", alc_hold), +SOC_SINGLE("ALC ZC Switch", WM8737_ALC2, 4, 1, 0), +SOC_ENUM("ALC Attack Time", alc_atk), +SOC_ENUM("ALC Decay Time", alc_dcy), }; static const char *linsel_text[] = { -- 1.7.1

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