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[alsa-devel] ASoC updates for 2.6.35
by Mark Brown 13 Jul '10

13 Jul '10

The following changes since commit 1c5474a65bf15a4cb162dfff86d6d0b5a08a740c: Linux 2.6.35-rc5 (2010-07-12 14:55:33 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.35 Kuninori Morimoto (2): ASoC: fsi: fixup clock inversion operation ASoC: fsi: fixup wrong value setting order of TDM Small, targetted fixes for a specific platform. sound/soc/sh/fsi.c | 27 +++++++++++---------------- 1 files changed, 11 insertions(+), 16 deletions(-)

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[alsa-devel] FIxed alsa-tools' envy24control missing peak level meters and "Reset Peaks"
by Niels Mayer 13 Jul '10

13 Jul '10

I've been long annoyed by alsa-tools' envy24control (*) lack of peak-level indication in it's metering, and the non-implementation of its "Reset Peaks" button: https://bugzilla.redhat.com/show_bug.cgi?id=602903 I've fixed that annoyance. Now, a small white line appears above the highest level on each meter, and can be reset via "Reset Peaks" button. Original source: http://git.alsa-project.org/?p=alsa-tools.git;a=blob_plain;f=envy24control/… Patch: http://nielsmayer.com/npm/602903.patch To run this version of envy24control grab the most recent stable release ( ftp://ftp.alsa-project.org/pub/tools/alsa-tools-1.0.23.tar.bz2 ) or git pull from trunk of the "alsa-tools" project ( http://alsa-project.org/main/index.php/GIT_Server ). After unpacking and assuming you've got the patch in ~/602903.patch do: >> cd alsa-tools-1.0.23 >> cat ~/602903.patch | patch -p1 it should give message "patching file envy24control/levelmeters.c" ... follow the directions to compile alsa-tools.... Can whoever's in charge of these things commit this change "upstream" and get it to propagate to the distros? Alternately, give me commit access and I'll do it myself, if people agree this patch is worth having. Next steps: (1) Think about building-in meter ballistics from Fons Adriaensen's jkmeter ( http://www.kokkinizita.net/linuxaudio/downloads/jkmeter-0.4.0.tar.bz2 ). Which helps implent Bob Katz' "K System" ( http://www.digido.com/level-practices-part-2-includes-the-k-system.html ). Although it probably most makes sense to implement such metering on the main "digital mixer" meter of envy24control, if it's not too inefficient, I'd want it available on all meters, despite the warning from Fons ( http://old.nabble.com/First-release-of-jkmeter-td18798950.html ). > This is the type of meter you want for live recording, > mixing and mastering. It probably makes no sense to > use it on all tracks of a DAW, where keeping digital > level within limits is the main purpose of metering. (2) Fix https://bugzilla.redhat.com/show_bug.cgi?id=602900 which causes 'envy24control' to crash when given "-D" argument. and behave incorrectly when given a correct non-numeric name for an audio device: e.g. >> envy24control --card M66 >> invalid card type (driver is USB-Audio) >> envy24control --card M66 --device M66 >> Segmentation fault (core dumped) (3) Apply more reasonable defaults and automatically size to correct dimensions. I use, for example: >> envy24control --card 2 --outputs 4 --input 4 --pcm_output 8 --view_spdif_playback --midichannel 1 --midienhanced --window_width 1275 --tall_eq_mixer_heights 1 & (4) Add midi output, in addition to midi input, which would monitor the hardware peak metering from the ice1712 ("amixer -c M66 cget iface=PCM,name='Multi Track Peak',numid=45") and output a MIDI note-on on the corresponding MIDI channel whenever a specified threshold value was exceeded. This could be used to implement "automatic gain control" in association with the midi-controllable ADC gain provided by envy24control. Niels http://nielsmayer.com PS: (*) http://alsa.opensrc.org/index.php/Envy24control Cards supported by this fix: * M Audio Delta 1010 * M Audio Delta 1010LT * M Audio Delta DiO 2496 * M Audio Delta 66 * M Audio Delta 44 * M Audio Delta 410 * M Audio Audiophile 2496 * TerraTec EWS 88MT * TerraTec EWS 88D * TerraTec EWX 24/96 * TerraTec DMX 6Fire * TerraTec Phase 88 * Hoontech SoundTrack DSP 24 * Hoontech SoundTrack DSP 24 Value * Hoontech SoundTrack DSP 24 Media 7.1 * Event Electronics, EZ8 * Digigram VX442 * Lionstracs, Mediastaton * Terrasoniq TS 88

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[alsa-devel] [PATCH] ASoC: au1x: fix section mismatch in psc-i2s.c
by Manuel Lauss 13 Jul '10

13 Jul '10

Annotate platform probe callback with __devinit instead of plain __init. Signed-off-by: Manuel Lauss <manuel.lauss(a)googlemail.com> --- sound/soc/au1x/psc-i2s.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index 495be6e..24454c9 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -300,7 +300,7 @@ struct snd_soc_dai au1xpsc_i2s_dai = { }; EXPORT_SYMBOL(au1xpsc_i2s_dai); -static int __init au1xpsc_i2s_drvprobe(struct platform_device *pdev) +static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev) { struct resource *r; unsigned long sel; -- 1.7.1.1 -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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[alsa-devel] [PATCH 34/36] sound/soc/blackfin: Remove unnecessary casts of private_data
by Joe Perches 13 Jul '10

13 Jul '10

Signed-off-by: Joe Perches <joe(a)perches.com> --- sound/soc/blackfin/bf5xx-ac97.c | 6 ++---- sound/soc/blackfin/bf5xx-tdm.c | 6 ++---- 2 files changed, 4 insertions(+), 8 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index 523b7fc..c0eba51 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -255,8 +255,7 @@ EXPORT_SYMBOL_GPL(soc_ac97_ops); #ifdef CONFIG_PM static int bf5xx_ac97_suspend(struct snd_soc_dai *dai) { - struct sport_device *sport = - (struct sport_device *)dai->private_data; + struct sport_device *sport = dai->private_data; pr_debug("%s : sport %d\n", __func__, dai->id); if (!dai->active) @@ -271,8 +270,7 @@ static int bf5xx_ac97_suspend(struct snd_soc_dai *dai) static int bf5xx_ac97_resume(struct snd_soc_dai *dai) { int ret; - struct sport_device *sport = - (struct sport_device *)dai->private_data; + struct sport_device *sport = dai->private_data; pr_debug("%s : sport %d\n", __func__, dai->id); if (!dai->active) diff --git a/sound/soc/blackfin/bf5xx-tdm.c b/sound/soc/blackfin/bf5xx-tdm.c index 4b36012..24c1426 100644 --- a/sound/soc/blackfin/bf5xx-tdm.c +++ b/sound/soc/blackfin/bf5xx-tdm.c @@ -210,8 +210,7 @@ static int bf5xx_tdm_set_channel_map(struct snd_soc_dai *dai, #ifdef CONFIG_PM static int bf5xx_tdm_suspend(struct snd_soc_dai *dai) { - struct sport_device *sport = - (struct sport_device *)dai->private_data; + struct sport_device *sport = dai->private_data; if (!dai->active) return 0; @@ -225,8 +224,7 @@ static int bf5xx_tdm_suspend(struct snd_soc_dai *dai) static int bf5xx_tdm_resume(struct snd_soc_dai *dai) { int ret; - struct sport_device *sport = - (struct sport_device *)dai->private_data; + struct sport_device *sport = dai->private_data; if (!dai->active) return 0; -- 1.7.1.337.g6068.dirty

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[alsa-devel] [patch 0/2] Kirkwood audio fixes
by arnaud.patard＠rtp-net.org 13 Jul '10

13 Jul '10

Hi, While working on the t5325 audio support, I found two small buglets in the kirkwood ASoC support. The first one is aiming at fixing mono audio playback and should fix mono playback for all platforms. The second is adding support for muting/unmuting kirkwood audio. It's mainly usefull for the t5325 which has a bootloader playing with this (hardware default is not muted). Arnaud

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[alsa-devel] [PATCH 1/6 v3] ASoC: fsi: modify format area definition on flags
by Kuninori Morimoto 13 Jul '10

13 Jul '10

There is no necessity that each bit in this area has the meaning. This patch modify it to sequence number Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v2 -> v3 o don't use << 0 shift 2/6 - 6/6 doesn't depend on 1/6 patch include/sound/sh_fsi.h | 12 ++++++------ 1 files changed, 6 insertions(+), 6 deletions(-) diff --git a/include/sound/sh_fsi.h b/include/sound/sh_fsi.h index c022736..3ecdb96 100644 --- a/include/sound/sh_fsi.h +++ b/include/sound/sh_fsi.h @@ -55,12 +55,12 @@ #define SH_FSI_GET_IFMT(x) ((x >> 8) & SH_FSI_FMT_MASK) #define SH_FSI_GET_OFMT(x) ((x >> 0) & SH_FSI_FMT_MASK) -#define SH_FSI_FMT_MONO (1 << 0) -#define SH_FSI_FMT_MONO_DELAY (1 << 1) -#define SH_FSI_FMT_PCM (1 << 2) -#define SH_FSI_FMT_I2S (1 << 3) -#define SH_FSI_FMT_TDM (1 << 4) -#define SH_FSI_FMT_TDM_DELAY (1 << 5) +#define SH_FSI_FMT_MONO 0 +#define SH_FSI_FMT_MONO_DELAY 1 +#define SH_FSI_FMT_PCM 2 +#define SH_FSI_FMT_I2S 3 +#define SH_FSI_FMT_TDM 4 +#define SH_FSI_FMT_TDM_DELAY 5 #define SH_FSI_IFMT_TDM_CH(x) \ (SH_FSI_IFMT(TDM) | SH_FSI_SET_CH_I(x)) -- 1.7.0.4

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[alsa-devel] [PATCH 0/2] ASoC: fsi: fixup patches
by Kuninori Morimoto 13 Jul '10

13 Jul '10

Dear Mark These are v2 bug fix patches for FSI driver. Kuninori Morimoto (2): ASoC: fsi: fixup clock inversion operation ASoC: fsi: fixup wrong value setting order of TDM v1 -> v2 It is bug fix only Best regards -- Kuninori Morimoto

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[alsa-devel] [PATCH 1/4] ASoC: DaVinci: remove requirement that dma_params is 1st in structure
by Troy Kisky 13 Jul '10

13 Jul '10

Remove requirement that dma_params is 1st in the structures davinci_audio_dev and davinci_mcbsp_dev. Signed-off-by: Troy Kisky <troy.kisky(a)boundarydevices.com> --- sound/soc/davinci/davinci-i2s.c | 6 +----- sound/soc/davinci/davinci-mcasp.c | 1 + sound/soc/davinci/davinci-mcasp.h | 5 ----- sound/soc/davinci/davinci-pcm.c | 7 ++++--- 4 files changed, 6 insertions(+), 13 deletions(-) diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 4ae7070..9e69a4e 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -98,11 +98,6 @@ enum { }; struct davinci_mcbsp_dev { - /* - * dma_params must be first because rtd->dai->cpu_dai->private_data - * is cast to a pointer of an array of struct davinci_pcm_dma_params in - * davinci_pcm_open. - */ struct davinci_pcm_dma_params dma_params[2]; void __iomem *base; #define MOD_DSP_A 0 @@ -547,6 +542,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = res->start; davinci_i2s_dai.private_data = dev; + davinci_i2s_dai.dma_data = dev->dma_params; ret = snd_soc_register_dai(&davinci_i2s_dai); if (ret != 0) goto err_free_mem; diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 5d1f98a..5023915 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -907,6 +907,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dma_data->channel = res->start; davinci_mcasp_dai[pdata->op_mode].private_data = dev; + davinci_mcasp_dai[pdata->op_mode].dma_data = dev->dma_params; davinci_mcasp_dai[pdata->op_mode].dev = &pdev->dev; ret = snd_soc_register_dai(&davinci_mcasp_dai[pdata->op_mode]); diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 9d179cc..582c924 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -39,11 +39,6 @@ enum { }; struct davinci_audio_dev { - /* - * dma_params must be first because rtd->dai->cpu_dai->private_data - * is cast to a pointer of an array of struct davinci_pcm_dma_params in - * davinci_pcm_open. - */ struct davinci_pcm_dma_params dma_params[2]; void __iomem *base; int sample_rate; diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index c73a915..c2c2e85 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -238,10 +238,11 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream) struct davinci_runtime_data *prtd; int ret = 0; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct davinci_pcm_dma_params *pa = rtd->dai->cpu_dai->private_data; - struct davinci_pcm_dma_params *params = &pa[substream->stream]; - if (!params) + struct davinci_pcm_dma_params *pa = rtd->dai->cpu_dai->dma_data; + struct davinci_pcm_dma_params *params; + if (!pa) return -ENODEV; + params = &pa[substream->stream]; snd_soc_set_runtime_hwparams(substream, &davinci_pcm_hardware); /* ensure that buffer size is a multiple of period size */ -- 1.5.6.3

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[alsa-devel] [PATCH] ASoC: TWL4030: Fix for digital loopback gain range
by Peter Ujfalusi 13 Jul '10

13 Jul '10

When the gain is configured using dB value it was not possible to use -24dB since the loopback got muted instead of -24dB. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/twl4030.c | 7 ++++--- 1 files changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 8d36bfa..d7f0048 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -540,10 +540,11 @@ static const struct snd_kcontrol_new twl4030_dapm_abypassl2_control = static const struct snd_kcontrol_new twl4030_dapm_abypassv_control = SOC_DAPM_SINGLE("Switch", TWL4030_REG_VDL_APGA_CTL, 2, 1, 0); -/* Digital bypass gain, 0 mutes the bypass */ +/* Digital bypass gain, mute instead of -30dB */ static const unsigned int twl4030_dapm_dbypass_tlv[] = { - TLV_DB_RANGE_HEAD(2), - 0, 3, TLV_DB_SCALE_ITEM(-2400, 0, 1), + TLV_DB_RANGE_HEAD(3), + 0, 1, TLV_DB_SCALE_ITEM(-3000, 600, 1), + 2, 3, TLV_DB_SCALE_ITEM(-2400, 0, 0), 4, 7, TLV_DB_SCALE_ITEM(-1800, 600, 0), }; -- 1.7.1

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[alsa-devel] [PATCH] ASoC: TWL4030: DAC power optimization
by Peter Ujfalusi 13 Jul '10

13 Jul '10

Restructure the DAPM connections in order to enable only the needed DAC (out of four in twl4030 series). I need to keep the 'AIF Enable' supply connected to the L2/R2 digital path, since the digital loopback needs AIF and APLL running. If no valid route available, than none of the DAC will be powered, but the AIF and APLL is going to be enabled. Furthermore, if only one audio path have valid route, than only the corresponding DAC will be powered. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/twl4030.c | 15 +++++++++------ 1 files changed, 9 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index d7f0048..6fd6d0b 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1460,8 +1460,11 @@ static const struct snd_soc_dapm_route intercon[] = { /* Supply for the digital part (APLL) */ {"Digital Voice Playback Mixer", NULL, "APLL Enable"}, - {"Digital R1 Playback Mixer", NULL, "AIF Enable"}, - {"Digital L1 Playback Mixer", NULL, "AIF Enable"}, + {"DAC Left1", NULL, "AIF Enable"}, + {"DAC Right1", NULL, "AIF Enable"}, + {"DAC Left2", NULL, "AIF Enable"}, + {"DAC Right1", NULL, "AIF Enable"}, + {"Digital R2 Playback Mixer", NULL, "AIF Enable"}, {"Digital L2 Playback Mixer", NULL, "AIF Enable"}, @@ -1532,10 +1535,10 @@ static const struct snd_soc_dapm_route intercon[] = { /* outputs */ /* Must be always connected (for AIF and APLL) */ - {"Virtual HiFi OUT", NULL, "Digital L1 Playback Mixer"}, - {"Virtual HiFi OUT", NULL, "Digital R1 Playback Mixer"}, - {"Virtual HiFi OUT", NULL, "Digital L2 Playback Mixer"}, - {"Virtual HiFi OUT", NULL, "Digital R2 Playback Mixer"}, + {"Virtual HiFi OUT", NULL, "DAC Left1"}, + {"Virtual HiFi OUT", NULL, "DAC Right1"}, + {"Virtual HiFi OUT", NULL, "DAC Left2"}, + {"Virtual HiFi OUT", NULL, "DAC Right2"}, /* Must be always connected (for APLL) */ {"Virtual Voice OUT", NULL, "Digital Voice Playback Mixer"}, /* Physical outputs */ -- 1.7.1.1

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