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

[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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[alsa-devel] [PATCH 0/8] ASoC: fsi: fixup patches
by Kuninori Morimoto 13 Jul '10

13 Jul '10

Dear Mark These patches are fixup for FSI driver Kuninori Morimoto (8): ASoC: fsi: modify format area definition on flags ASoC: fsi: fixup clock inversion operation ASoC: fsi: remove un-used variable on fsi_dai_startup ASoC: fsi: remove noisy CR_FMT macro ASoC: fsi: modify wrong value setting order of TDM ASoC: fsi: Change struct fsi_regs to fsi_core ASoC: fsi: Add pr_err for noticing unsupported access ASoC: fsi: Fixup for master mode Some of these are not so important for now. But it is used from "FSI - HDMI" patches which I will send in future

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[alsa-devel] issue with TWL4030
by Grazvydas Ignotas 13 Jul '10

13 Jul '10

Hi, there seems to be a problem with TWL4030 hw_param programming, which is easy to reproduce over OSS emulation with this simple program: int oss_dev = open("/dev/dsp", O_WRONLY); int stereo = 1, bits = 16, rate = 44100; ioctl(oss_dev, SNDCTL_DSP_SETFMT, &bits); ioctl(oss_dev, SNDCTL_DSP_SPEED, &rate); ioctl(oss_dev, SNDCTL_DSP_STEREO, &stereo); write(oss_dev, somebuff, sizeof(somebuff)); The above program ends up playing at 8kHz, instead of expected 44kHz. I did some call tracing and got this: [ 1092.246032] soc_pcm_hw_params, rate 8000 [ 1092.250122] twl4030_hw_params [ 1092.253234] set rate 8000 [ 1092.256103] configured 1 [ 1092.259368] soc_pcm_hw_params, rate 8000 [ 1092.263427] twl4030_hw_params [ 1092.266845] soc_pcm_hw_params, rate 44100 [ 1092.270965] twl4030_hw_params [ 1094.459106] twl4030_shutdown [ 1094.462036] configured 0 So it looks like twl4030->configured is the culprit of this, removing it solves the problem, but I guess that will break something else. Any ideas how to properly fix this? Gražvydas

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[alsa-devel] [PATCH 0/6] ASoC: fsi: modify patch
by Kuninori Morimoto 13 Jul '10

13 Jul '10

Dear Mark These are v2 patches for FSI driver almost all are same as v1 Kuninori Morimoto (6): ASoC: fsi: modify format area definition on flags ASoC: fsi: remove un-used variable on fsi_dai_startup ASoC: fsi: remove noisy CR_FMT macro ASoC: fsi: Change struct fsi_regs to fsi_core ASoC: fsi: Add pr_err for noticing unsupported access ASoC: fsi: Fixup for master mode v1 -> v2 o modiry straight start number as 0 o based on bug fix patches Best regards -- Kuninori Morimoto

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[alsa-devel] [PATCH 0/3] further BKL removal
by Arnd Bergmann 12 Jul '10

12 Jul '10

With the ioctl, block, tty, vfs and llseek series on their way into linux-next, these three patches are attacking the hardest remaining issues. If we get these ready for 2.6.36, the kernel should be almost usable with the BKL disabled. In all three cases, I'm cheating a bit, because the BKL still remains lurking in the dark corners of the three subsystems (i810/i830 for drm, OSS for sound and lockd for fs/locks.c). Arnd Cc: Christoph Hellwig <hch(a)lst.de> Cc: David Airlie <airlied(a)linux.ie> Cc: dri-devel(a)lists.freedesktop.org Cc: Frederic Weisbecker <fweisbec(a)gmail.com> Cc: Ingo Molnar <mingo(a)redhat.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: "J. Bruce Fields" <bfields(a)fieldses.org> Cc: John Kacur <jkacur(a)redhat.com> Cc: Matthew Wilcox <willy(a)linux.intel.com> Cc: Miklos Szeredi <mszeredi(a)suse.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Trond Myklebust <trond.myklebust(a)fys.uio.no> Cc: alsa-devel(a)alsa-project.org Cc: linux-fsdevel(a)vger.kernel.org Cc: linux-kernel(a)vger.kernel.org Arnd Bergmann (2): drm: kill BKL from common code sound: push BKL into open functions Matthew Wilcox (1): Remove BKL from fs/locks.c arch/um/drivers/hostaudio_kern.c | 6 ++ drivers/gpu/drm/drm_drv.c | 4 +- drivers/gpu/drm/drm_fops.c | 23 ++++---- drivers/gpu/drm/i810/i810_dma.c | 44 +++++++++----- drivers/gpu/drm/i810/i810_drv.c | 2 +- drivers/gpu/drm/i810/i810_drv.h | 1 + drivers/gpu/drm/i830/i830_dma.c | 42 +++++++++---- drivers/gpu/drm/i830/i830_drv.c | 2 +- drivers/gpu/drm/i830/i830_drv.h | 1 + fs/Kconfig | 4 + fs/locks.c | 116 ++++++++++++++++++++++-------------- include/drm/drmP.h | 2 +- sound/core/hwdep.c | 14 +++- sound/core/oss/mixer_oss.c | 19 ++++--- sound/oss/au1550_ac97.c | 26 +++++--- sound/oss/dmasound/dmasound_core.c | 28 +++++++-- sound/oss/msnd_pinnacle.c | 10 ++- sound/oss/sh_dac_audio.c | 9 ++- sound/oss/soundcard.c | 20 ++++--- sound/oss/swarm_cs4297a.c | 17 +++++- sound/oss/vwsnd.c | 8 +++ sound/sound_core.c | 6 +-- 22 files changed, 267 insertions(+), 137 deletions(-)

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[alsa-devel] [PATCH 36/36] sound/core: Remove unnecessary casts of private_data
by Joe Perches 12 Jul '10

12 Jul '10

Signed-off-by: Joe Perches <joe(a)perches.com> --- sound/core/oss/mixer_oss.c | 22 +++++++++++----------- sound/core/pcm.c | 3 +-- sound/core/timer.c | 4 ++-- 3 files changed, 14 insertions(+), 15 deletions(-) diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index f50ebf2..86afb13 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -77,7 +77,7 @@ static int snd_mixer_oss_release(struct inode *inode, struct file *file) struct snd_mixer_oss_file *fmixer; if (file->private_data) { - fmixer = (struct snd_mixer_oss_file *) file->private_data; + fmixer = file->private_data; module_put(fmixer->card->module); snd_card_file_remove(fmixer->card, file); kfree(fmixer); @@ -368,7 +368,7 @@ static int snd_mixer_oss_ioctl1(struct snd_mixer_oss_file *fmixer, unsigned int static long snd_mixer_oss_ioctl(struct file *file, unsigned int cmd, unsigned long arg) { - return snd_mixer_oss_ioctl1((struct snd_mixer_oss_file *) file->private_data, cmd, arg); + return snd_mixer_oss_ioctl1(file->private_data, cmd, arg); } int snd_mixer_oss_ioctl_card(struct snd_card *card, unsigned int cmd, unsigned long arg) @@ -582,7 +582,7 @@ static int snd_mixer_oss_get_volume1(struct snd_mixer_oss_file *fmixer, struct snd_mixer_oss_slot *pslot, int *left, int *right) { - struct slot *slot = (struct slot *)pslot->private_data; + struct slot *slot = pslot->private_data; *left = *right = 100; if (slot->present & SNDRV_MIXER_OSS_PRESENT_PVOLUME) { @@ -691,7 +691,7 @@ static int snd_mixer_oss_put_volume1(struct snd_mixer_oss_file *fmixer, struct snd_mixer_oss_slot *pslot, int left, int right) { - struct slot *slot = (struct slot *)pslot->private_data; + struct slot *slot = pslot->private_data; if (slot->present & SNDRV_MIXER_OSS_PRESENT_PVOLUME) { snd_mixer_oss_put_volume1_vol(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_PVOLUME], left, right); @@ -740,7 +740,7 @@ static int snd_mixer_oss_get_recsrc1_sw(struct snd_mixer_oss_file *fmixer, struct snd_mixer_oss_slot *pslot, int *active) { - struct slot *slot = (struct slot *)pslot->private_data; + struct slot *slot = pslot->private_data; int left, right; left = right = 1; @@ -753,7 +753,7 @@ static int snd_mixer_oss_get_recsrc1_route(struct snd_mixer_oss_file *fmixer, struct snd_mixer_oss_slot *pslot, int *active) { - struct slot *slot = (struct slot *)pslot->private_data; + struct slot *slot = pslot->private_data; int left, right; left = right = 1; @@ -766,7 +766,7 @@ static int snd_mixer_oss_put_recsrc1_sw(struct snd_mixer_oss_file *fmixer, struct snd_mixer_oss_slot *pslot, int active) { - struct slot *slot = (struct slot *)pslot->private_data; + struct slot *slot = pslot->private_data; snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_CSWITCH], active, active, 0); return 0; @@ -776,7 +776,7 @@ static int snd_mixer_oss_put_recsrc1_route(struct snd_mixer_oss_file *fmixer, struct snd_mixer_oss_slot *pslot, int active) { - struct slot *slot = (struct slot *)pslot->private_data; + struct slot *slot = pslot->private_data; snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_CROUTE], active, active, 1); return 0; @@ -813,7 +813,7 @@ static int snd_mixer_oss_get_recsrc2(struct snd_mixer_oss_file *fmixer, unsigned if (!(mixer->mask_recsrc & (1 << idx))) continue; pslot = &mixer->slots[idx]; - slot = (struct slot *)pslot->private_data; + slot = pslot->private_data; if (slot->signature != SNDRV_MIXER_OSS_SIGNATURE) continue; if (!(slot->present & SNDRV_MIXER_OSS_PRESENT_CAPTURE)) @@ -861,7 +861,7 @@ static int snd_mixer_oss_put_recsrc2(struct snd_mixer_oss_file *fmixer, unsigned if (!(mixer->mask_recsrc & (1 << idx))) continue; pslot = &mixer->slots[idx]; - slot = (struct slot *)pslot->private_data; + slot = pslot->private_data; if (slot->signature != SNDRV_MIXER_OSS_SIGNATURE) continue; if (!(slot->present & SNDRV_MIXER_OSS_PRESENT_CAPTURE)) @@ -925,7 +925,7 @@ static int snd_mixer_oss_build_test(struct snd_mixer_oss *mixer, struct slot *sl static void snd_mixer_oss_slot_free(struct snd_mixer_oss_slot *chn) { - struct slot *p = (struct slot *)chn->private_data; + struct slot *p = chn->private_data; if (p) { if (p->allocated && p->assigned) { kfree(p->assigned->name); diff --git a/sound/core/pcm.c b/sound/core/pcm.c index cbe815d..de5a0eb 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -358,8 +358,7 @@ static void snd_pcm_stream_proc_info_read(struct snd_info_entry *entry, static void snd_pcm_substream_proc_info_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { - snd_pcm_proc_info_read((struct snd_pcm_substream *)entry->private_data, - buffer); + snd_pcm_proc_info_read(entry->private_data, buffer); } static void snd_pcm_substream_proc_hw_params_read(struct snd_info_entry *entry, diff --git a/sound/core/timer.c b/sound/core/timer.c index 13afb60..5ab1014 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -962,7 +962,7 @@ static int snd_timer_s_start(struct snd_timer * timer) struct snd_timer_system_private *priv; unsigned long njiff; - priv = (struct snd_timer_system_private *) timer->private_data; + priv = timer->private_data; njiff = (priv->last_jiffies = jiffies); if (priv->correction > timer->sticks - 1) { priv->correction -= timer->sticks - 1; @@ -981,7 +981,7 @@ static int snd_timer_s_stop(struct snd_timer * timer) struct snd_timer_system_private *priv; unsigned long jiff; - priv = (struct snd_timer_system_private *) timer->private_data; + priv = timer->private_data; del_timer(&priv->tlist); jiff = jiffies; if (time_before(jiff, priv->last_expires)) -- 1.7.1.337.g6068.dirty

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[alsa-devel] [PATCH 35/36] sound/soc/davinci: Remove unnecessary casts of private_data
by Joe Perches 12 Jul '10

12 Jul '10

Signed-off-by: Joe Perches <joe(a)perches.com> --- sound/soc/davinci/davinci-pcm.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 2dc406f..def454e 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -800,7 +800,7 @@ static void davinci_pcm_free(struct snd_pcm *pcm) dma_free_writecombine(pcm->card->dev, buf->bytes, buf->area, buf->addr); buf->area = NULL; - iram_dma = (struct snd_dma_buffer *)buf->private_data; + iram_dma = buf->private_data; if (iram_dma) { sram_free(iram_dma->area, iram_dma->bytes); kfree(iram_dma); -- 1.7.1.337.g6068.dirty

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[alsa-devel] How does struct snd_pcm_hardware relate to the actual hardware
by Adam Rosenberg 12 Jul '10

12 Jul '10

I am using DMA to transfer data to SPORT1 on my Blackfin BF537. I am using a CS42448 CODEC that has 8 DACs. I am using TDM mode and have set the following: DMA Buffer Size For Each SPORT Data Line (Rx or Tx) = 131072 bytes DMA Periods = 8 static struct snd_pcm_hardware snd_pcm_hw = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_RESUME), .formats = SNDRV_PCM_FMTBIT_S32_LE, .rates = SNDRV_PCM_RATE_48000, .rate_min = 48000, .rate_max = 48000, .buffer_bytes_max = 32768, .channels_min = 2, .channels_max = 2, .period_bytes_min = 8, .period_bytes_max = 4096, .periods_min = 8, .periods_max = 8, }; Everytime I receive a DMA interrupt from the SPORT driver I call snd_pcm_period_elapsed for all of the substreams that are currently active. In the PCM Copy callback I calculate the position within the DMA buffer to write to based on the TDM slot for the channel and the position passed into the function. The audio is not being placed into the DMA buffer properly. The audio plays faster than it should and has a constant stutter. I have read the "writing an ALSA driver" guide multiple times but it does not explain the relationship between struct snd_pcm_hardware, snd_pcm_period_elapsed, and the copy callback clearly enough for me to understand what I am doing wrong. Please explain how these interact in more detail to help me fix this problem. I can also send the full source code to the CS42448 driver I am working on if more details are needed. Thank you, Adam

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