October 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] sound: soc: fsl: Fix memory leak in imx-audmux.c
by Felipe Pena 13 Oct '13

13 Oct '13

When audmux_clk is used and clk_prepare_enable function succeed, the memory alloc'd to buf variable is leaked Signed-off-by: Felipe Pena <felipensp(a)gmail.com> --- sound/soc/fsl/imx-audmux.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index d3bf71a..977759a 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -66,19 +66,20 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf, size_t count, loff_t *ppos) { ssize_t ret; - char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL); + char *buf; int port = (int)file->private_data; u32 pdcr, ptcr; - if (!buf) - return -ENOMEM; - if (audmux_clk) { ret = clk_prepare_enable(audmux_clk); if (ret) return ret; } + buf = kmalloc(PAGE_SIZE, GFP_KERNEL); + if (!buf) + return -ENOMEM; + ptcr = readl(audmux_base + IMX_AUDMUX_V2_PTCR(port)); pdcr = readl(audmux_base + IMX_AUDMUX_V2_PDCR(port)); -- 1.7.10.4

1 0

[alsa-devel] Recording devices opened mono still sending stereo causing corrupt audio
by Sean M. Pappalardo - D.J. Pegasus 12 Oct '13

12 Oct '13

(Re-sending including alsa-devel and more information.) Hello there. I'm a developer for Mixxx, the cross-platform open-source DJ software which uses PortAudio and we've been trying for a long time to track down the cause of "horrible mic sound" as documented in our bug: https://bugs.launchpad.net/mixxx/+bug/900364 The resulting recorded audio sounds about 50-60% lower pitch with crackles and it only happens with the ALSA API on Linux and only with certain (mostly Intel-based) audio devices. I have just discovered by testing in Audacity (which also uses PA) that the problem occurs when PA (and therefore ALSA) is asked to open just one channel of a sound input device yet the device (or ALSA) seems to stream two anyway. I have been able to reproduce the problem in Audacity (which also uses PA) when selecting the ALSA API and mono recording, but it works correctly when I change it to stereo recording. I'm suspecting there's a miscommunication between PA and ALSA or ALSA and the sound hardware where either the lower level is saying its input(s) can only be opened in stereo and the higher level (or the application above it) ignores that, or the lower level says it's fine opening mono but then misbehaves. How can I find out which, whether it's ALSA, PA, or the sound hardware at fault? Sincerely, Sean M. Pappalardo "D.J. Pegasus" Mixxx Developer - Controller Specialist

5 6

[alsa-devel] [PATCH] ASoC: rcar: add ID check on rsnd_dai_get()
by Kuninori Morimoto 11 Oct '13

11 Oct '13

checking id in rsnd_dai_get() is good idea Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 28c24fc..b234ed6 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -318,6 +318,9 @@ int rsnd_dai_id(struct rsnd_priv *priv, struct rsnd_dai *rdai) struct rsnd_dai *rsnd_dai_get(struct rsnd_priv *priv, int id) { + if ((id < 0) || (id >= rsnd_dai_nr(priv))) + return NULL; + return priv->rdai + id; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: rcar: fixup generation checker
by Kuninori Morimoto 11 Oct '13

11 Oct '13

Current rcar is using rsnd_is_gen1/gen2() to checking its IP generation, but it needs data mask. This patch fixes it up. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/rcar_snd.h | 1 + sound/soc/sh/rcar/rsnd.h | 4 ++-- 2 files changed, 3 insertions(+), 2 deletions(-) diff --git a/include/sound/rcar_snd.h b/include/sound/rcar_snd.h index fe66533..fb0a312 100644 --- a/include/sound/rcar_snd.h +++ b/include/sound/rcar_snd.h @@ -68,6 +68,7 @@ struct rsnd_scu_platform_info { * * A : generation */ +#define RSND_GEN_MASK (0xF << 0) #define RSND_GEN1 (1 << 0) /* fixme */ #define RSND_GEN2 (2 << 0) /* fixme */ diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 3868aaf..3b71c3f 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -216,8 +216,8 @@ int rsnd_gen_path_exit(struct rsnd_priv *priv, void __iomem *rsnd_gen_reg_get(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg); -#define rsnd_is_gen1(s) ((s)->info->flags & RSND_GEN1) -#define rsnd_is_gen2(s) ((s)->info->flags & RSND_GEN2) +#define rsnd_is_gen1(s) (((s)->info->flags & RSND_GEN_MASK) == RSND_GEN1) +#define rsnd_is_gen2(s) (((s)->info->flags & RSND_GEN_MASK) == RSND_GEN2) /* * R-Car ADG -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: rcar: fixup rsnd_platform_call() return value
by Kuninori Morimoto 11 Oct '13

11 Oct '13

Un-implemented platform callback is not error. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index fc83f0f..28c24fc 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -103,7 +103,7 @@ * rsnd_platform functions */ #define rsnd_platform_call(priv, dai, func, param...) \ - (!(priv->info->func) ? -ENODEV : \ + (!(priv->info->func) ? 0 : \ priv->info->func(param)) /* -- 1.7.9.5

2 1

Re: [alsa-devel] [PATCH v4 0/5] dma: mxs-dma bugfixes and cleanup
by Fabio Estevam 11 Oct '13

11 Oct '13

Hi Markus, [Also including alsa-devel and some other folks involved with mxs audio] On Fri, Oct 11, 2013 at 9:12 AM, Markus Pargmann <mpa(a)pengutronix.de> wrote: > Hi, > > v4 adds another patch to increase sound stability. Patch 3 ("dma: mxs-dma: Fix > channel reset hardware bug") decreases the number of channel stalls, but after > hours of repeated playback the bug still appears. The new patch ("dma: mxs: Use > semaphores for cyclic DMA") replaces reset channel usage with hardware > semaphore counter mechanism. How is the bug you are trying to solve manifest and how to reproduce it? I have been investigating an issue about sound stability on mx28 as well. The usecase I am doing is: while true do arecord -D hw:0,1 -d 10 -f S16_LE -r 44100 -c 2 | aplay sleep 2 done (Record 10 seconds, sleep and keep in this loop forever). The problem I see is that randomly the captured audio becomes extremely noisy and loud (throughout the whole 10 seconds) and then in the subsequent loop it becomes normal. This is very random and it happens at a 1 or 2% failure rate. I tested your v3, but it did not help to fix this particular issue. I also tried experimenting with the SAIF IOMUX settings: index cda19c8..463fa92 100644 --- a/arch/arm/boot/dts/imx28.dtsi +++ b/arch/arm/boot/dts/imx28.dtsi @@ -545,9 +545,9 @@ MX28_PAD_SAIF0_BITCLK__SAIF0_BITCLK MX28_PAD_SAIF0_SDATA0__SAIF0_SDATA0 >; - fsl,drive-strength = <MXS_DRIVE_12mA>; + fsl,drive-strength = <MXS_DRIVE_4mA>; fsl,voltage = <MXS_VOLTAGE_HIGH>; - fsl,pull-up = <MXS_PULL_ENABLE>; + fsl,pull-up = <MXS_PULL_DISABLE>; }; saif0_pins_b: saif0@1 { @@ -557,9 +557,9 @@ MX28_PAD_SAIF0_BITCLK__SAIF0_BITCLK MX28_PAD_SAIF0_SDATA0__SAIF0_SDATA0 >; - fsl,drive-strength = <MXS_DRIVE_12mA>; + fsl,drive-strength = <MXS_DRIVE_4mA>; fsl,voltage = <MXS_VOLTAGE_HIGH>; - fsl,pull-up = <MXS_PULL_ENABLE>; + fsl,pull-up = <MXS_PULL_DISABLE>; }; saif1_pins_a: saif1@0 { @@ -567,9 +567,9 @@ fsl,pinmux-ids = < MX28_PAD_SAIF1_SDATA0__SAIF1_SDATA0 >; - fsl,drive-strength = <MXS_DRIVE_12mA>; + fsl,drive-strength = <MXS_DRIVE_4mA>; fsl,voltage = <MXS_VOLTAGE_HIGH>; - fsl,pull-up = <MXS_PULL_ENABLE>; + fsl,pull-up = <MXS_PULL_DISABLE>; }; and my tests are showing a much better stability. Not able to get this problem after several hours. Regards, Fabio Estevam

3 2

[alsa-devel] [PATCH 1/5] ARM: Kconfig: Move if ARCH_S3C64XX statement to mach-s3c64xx/Kconfig
by Tomasz Figa 11 Oct '13

11 Oct '13

All other platforms have this condition checked inside their own Kconfig files, so for consistency this patch makes it this way for mach-s3c64xx as well. Signed-off-by: Tomasz Figa <tomasz.figa(a)gmail.com> --- arch/arm/Kconfig | 2 -- arch/arm/mach-s3c64xx/Kconfig | 4 ++++ 2 files changed, 4 insertions(+), 2 deletions(-) diff --git a/arch/arm/Kconfig b/arch/arm/Kconfig index b766dad..dc51f8a 100644 --- a/arch/arm/Kconfig +++ b/arch/arm/Kconfig @@ -995,9 +995,7 @@ source "arch/arm/mach-sti/Kconfig" source "arch/arm/mach-s3c24xx/Kconfig" -if ARCH_S3C64XX source "arch/arm/mach-s3c64xx/Kconfig" -endif source "arch/arm/mach-s5p64x0/Kconfig" diff --git a/arch/arm/mach-s3c64xx/Kconfig b/arch/arm/mach-s3c64xx/Kconfig index bd14e3a..0e23910 100644 --- a/arch/arm/mach-s3c64xx/Kconfig +++ b/arch/arm/mach-s3c64xx/Kconfig @@ -3,6 +3,8 @@ # # Licensed under GPLv2 +if ARCH_S3C64XX + # temporary until we can eliminate all drivers using it. config PLAT_S3C64XX bool @@ -322,3 +324,5 @@ config MACH_S3C64XX_DT board. Note: This is under development and not all peripherals can be supported with this machine file. + +endif -- 1.8.3.2

5 9

[alsa-devel] [PATCH] pcm: use the same set of uint_XX types everywhere
by John Spencer 11 Oct '13

11 Oct '13

previously, pcm.h prototyped the functions snd_pcm_format_silence* with u_int_XX types (legacy BSD types from sys/types.h), while the implementation in pcm_misc.c used the C99 stdint.h types uint_XX. this caused compilation errors on systems where the typedef for u_int_XX and uint_XX is not exactly the same. we fix it by using the C99 stdint.h types everywhere. Signed-off-by: John Spencer <maillist-alsa(a)barfooze.de> --- include/pcm.h | 9 +++++---- src/pcm/pcm_misc.c | 42 +++++++++++++++++++++--------------------- 2 files changed, 26 insertions(+), 25 deletions(-) diff --git a/include/pcm.h b/include/pcm.h index 95b8aed..cd847e4 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -33,6 +33,7 @@ extern "C" { #endif +#include <stdint.h> /** * \defgroup PCM PCM Interface * See the \ref pcm page for more details. @@ -1048,10 +1049,10 @@ int snd_pcm_format_width(snd_pcm_format_t format); /* in bits */ int snd_pcm_format_physical_width(snd_pcm_format_t format); /* in bits */ snd_pcm_format_t snd_pcm_build_linear_format(int width, int pwidth, int unsignd, int big_endian); ssize_t snd_pcm_format_size(snd_pcm_format_t format, size_t samples); -u_int8_t snd_pcm_format_silence(snd_pcm_format_t format); -u_int16_t snd_pcm_format_silence_16(snd_pcm_format_t format); -u_int32_t snd_pcm_format_silence_32(snd_pcm_format_t format); -u_int64_t snd_pcm_format_silence_64(snd_pcm_format_t format); +uint8_t snd_pcm_format_silence(snd_pcm_format_t format); +uint16_t snd_pcm_format_silence_16(snd_pcm_format_t format); +uint32_t snd_pcm_format_silence_32(snd_pcm_format_t format); +uint64_t snd_pcm_format_silence_64(snd_pcm_format_t format); int snd_pcm_format_set_silence(snd_pcm_format_t format, void *buf, unsigned int samples); snd_pcm_sframes_t snd_pcm_bytes_to_frames(snd_pcm_t *pcm, ssize_t bytes); diff --git a/src/pcm/pcm_misc.c b/src/pcm/pcm_misc.c index d52160c..2a8be21 100644 --- a/src/pcm/pcm_misc.c +++ b/src/pcm/pcm_misc.c @@ -363,7 +363,7 @@ ssize_t snd_pcm_format_size(snd_pcm_format_t format, size_t samples) * \param format Sample format * \return silence 64 bit word */ -u_int64_t snd_pcm_format_silence_64(snd_pcm_format_t format) +uint64_t snd_pcm_format_silence_64(snd_pcm_format_t format) { switch (format) { case SNDRV_PCM_FORMAT_S8: @@ -437,7 +437,7 @@ u_int64_t snd_pcm_format_silence_64(snd_pcm_format_t format) { union { float f[2]; - u_int64_t i; + uint64_t i; } u; u.f[0] = u.f[1] = 0.0; #ifdef SNDRV_LITTLE_ENDIAN @@ -450,7 +450,7 @@ u_int64_t snd_pcm_format_silence_64(snd_pcm_format_t format) { union { double f; - u_int64_t i; + uint64_t i; } u; u.f = 0.0; #ifdef SNDRV_LITTLE_ENDIAN @@ -463,7 +463,7 @@ u_int64_t snd_pcm_format_silence_64(snd_pcm_format_t format) { union { float f[2]; - u_int64_t i; + uint64_t i; } u; u.f[0] = u.f[1] = 0.0; #ifdef SNDRV_LITTLE_ENDIAN @@ -476,7 +476,7 @@ u_int64_t snd_pcm_format_silence_64(snd_pcm_format_t format) { union { double f; - u_int64_t i; + uint64_t i; } u; u.f = 0.0; #ifdef SNDRV_LITTLE_ENDIAN @@ -509,10 +509,10 @@ u_int64_t snd_pcm_format_silence_64(snd_pcm_format_t format) * \param format Sample format * \return silence 32 bit word */ -u_int32_t snd_pcm_format_silence_32(snd_pcm_format_t format) +uint32_t snd_pcm_format_silence_32(snd_pcm_format_t format) { assert(snd_pcm_format_physical_width(format) <= 32); - return (u_int32_t)snd_pcm_format_silence_64(format); + return (uint32_t)snd_pcm_format_silence_64(format); } /** @@ -520,10 +520,10 @@ u_int32_t snd_pcm_format_silence_32(snd_pcm_format_t format) * \param format Sample format * \return silence 16 bit word */ -u_int16_t snd_pcm_format_silence_16(snd_pcm_format_t format) +uint16_t snd_pcm_format_silence_16(snd_pcm_format_t format) { assert(snd_pcm_format_physical_width(format) <= 16); - return (u_int16_t)snd_pcm_format_silence_64(format); + return (uint16_t)snd_pcm_format_silence_64(format); } /** @@ -531,10 +531,10 @@ u_int16_t snd_pcm_format_silence_16(snd_pcm_format_t format) * \param format Sample format * \return silence 8 bit word */ -u_int8_t snd_pcm_format_silence(snd_pcm_format_t format) +uint8_t snd_pcm_format_silence(snd_pcm_format_t format) { assert(snd_pcm_format_physical_width(format) <= 8); - return (u_int8_t)snd_pcm_format_silence_64(format); + return (uint8_t)snd_pcm_format_silence_64(format); } /** @@ -550,7 +550,7 @@ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int return 0; switch (snd_pcm_format_physical_width(format)) { case 4: { - u_int8_t silence = snd_pcm_format_silence_64(format); + uint8_t silence = snd_pcm_format_silence_64(format); unsigned int samples1; if (samples % 2 != 0) return -EINVAL; @@ -559,13 +559,13 @@ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int break; } case 8: { - u_int8_t silence = snd_pcm_format_silence_64(format); + uint8_t silence = snd_pcm_format_silence_64(format); memset(data, silence, samples); break; } case 16: { - u_int16_t silence = snd_pcm_format_silence_64(format); - u_int16_t *pdata = (u_int16_t *)data; + uint16_t silence = snd_pcm_format_silence_64(format); + uint16_t *pdata = (uint16_t *)data; if (! silence) memset(data, 0, samples * 2); else { @@ -575,8 +575,8 @@ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int break; } case 24: { - u_int32_t silence = snd_pcm_format_silence_64(format); - u_int8_t *pdata = (u_int8_t *)data; + uint32_t silence = snd_pcm_format_silence_64(format); + uint8_t *pdata = (uint8_t *)data; if (! silence) memset(data, 0, samples * 3); else { @@ -595,8 +595,8 @@ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int break; } case 32: { - u_int32_t silence = snd_pcm_format_silence_64(format); - u_int32_t *pdata = (u_int32_t *)data; + uint32_t silence = snd_pcm_format_silence_64(format); + uint32_t *pdata = (uint32_t *)data; if (! silence) memset(data, 0, samples * 4); else { @@ -606,8 +606,8 @@ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int break; } case 64: { - u_int64_t silence = snd_pcm_format_silence_64(format); - u_int64_t *pdata = (u_int64_t *)data; + uint64_t silence = snd_pcm_format_silence_64(format); + uint64_t *pdata = (uint64_t *)data; if (! silence) memset(data, 0, samples * 8); else { -- 1.7.3.4

4 6

[alsa-devel] [PATCH] ALSA: hda - Sony VAIO Pro 13 (haswell) now has a working headset jack
by David Henningsson 11 Oct '13

11 Oct '13

Just got the positive confirmation from a tester: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1227093/comments/28 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 79e6fe7..bf313be 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3761,6 +3761,8 @@ static const struct hda_fixup alc269_fixups[] = { { 0x18, 0x01a1913c }, /* use as headset mic, without its own jack detect */ { } }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MIC }, [ALC269_FIXUP_ASUS_X101_FUNC] = { .type = HDA_FIXUP_FUNC, -- 1.7.9.5

2 1

[alsa-devel] [PATCH v2 1/2] ASoC: mc13783: add mixer controls
by Steffen Trumtrar 11 Oct '13

11 Oct '13

Add more controls to the alsa mixer infrastructure. Signed-off-by: Steffen Trumtrar <s.trumtrar(a)pengutronix.de> --- Changes in v2: - Fix some control names - remove controls that belong to DAPM sound/soc/codecs/mc13783.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index eedbf05..2b62737 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -541,8 +541,26 @@ static const struct soc_enum mc13783_enum_3d_mixer = static struct snd_kcontrol_new mc13783_control_list[] = { SOC_SINGLE("Loudspeaker enable", MC13783_AUDIO_RX0, 5, 1, 0), SOC_SINGLE("PCM Playback Volume", MC13783_AUDIO_RX1, 6, 15, 0), + SOC_SINGLE("PCM Playback Switch", MC13783_AUDIO_RX1, 5, 1, 0), SOC_DOUBLE("PCM Capture Volume", MC13783_AUDIO_TX, 19, 14, 31, 0), SOC_ENUM("3D Control", mc13783_enum_3d_mixer), + + SOC_SINGLE("CDCOUT Switch", MC13783_AUDIO_RX0, 18, 1, 0), + SOC_SINGLE("Earpiece Amp Switch", MC13783_AUDIO_RX0, 3, 1, 0), + SOC_DOUBLE("Headset Amp Switch", MC13783_AUDIO_RX0, 10, 9, 1, 0), + SOC_DOUBLE("Line out Amp Switch", MC13783_AUDIO_RX0, 16, 15, 1, 0), + + SOC_SINGLE("PCM Capture Mixin Switch", MC13783_AUDIO_RX0, 22, 1, 0), + SOC_SINGLE("Line in Capture Mixin Switch", MC13783_AUDIO_RX0, 23, 1, 0), + + SOC_SINGLE("CODEC Capture Volume", MC13783_AUDIO_RX1, 1, 15, 0), + SOC_SINGLE("CODEC Capture Mixin Switch", MC13783_AUDIO_RX0, 21, 1, 0), + + SOC_SINGLE("Line in Capture Volume", MC13783_AUDIO_RX1, 12, 15, 0), + SOC_SINGLE("Line in Capture Switch", MC13783_AUDIO_RX1, 10, 1, 0), + + SOC_SINGLE("MC1 Capture Bias Switch", MC13783_AUDIO_TX, 0, 1, 0), + SOC_SINGLE("MC2 Capture Bias Switch", MC13783_AUDIO_TX, 1, 1, 0), }; static int mc13783_probe(struct snd_soc_codec *codec) -- 1.8.4.rc3

2 2