June 2009 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] M-Audio Transit: alsamixer shows 100% and -0.5db at the same time !?
by Klaus Schulz 08 Jun '09

08 Jun '09

Hi folks. My M-Audio Transit shows at 100% and 0.5db attentuation in alsamixer at the same time. See below printout. What could be the explanation for this behaviour? THX \Klaus amixer -Dhw:0 Simple mixer control 'PCM',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 254 Mono: Front Left: Playback 254 [100%] [0.50dB] [on] Front Right: Playback 254 [100%] [0.50dB] [on]

3 2

[alsa-devel] [PATCH v4 2/4] ARM: DaVinci: ASoC: Introduce platform driver model for dm644x, dm355
by Chaithrika U S 08 Jun '09

08 Jun '09

Introduce the platform driver model to get platform data for dm355 and dm644x. Register platform driver and acquire the resources in the probe function Since the platform specific code had been moved from machine driver to dm<soc>.c Signed-off-by: Naresh Medisetty <naresh(a)ti.com> Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- This patch applies against ALSA mainline - that's the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git sound/soc/davinci/davinci-i2s.c | 117 ++++++++++++++++++++++----------------- 1 files changed, 67 insertions(+), 50 deletions(-) diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index b1ea52f..500d2f5 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -436,16 +436,40 @@ static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } -static int davinci_i2s_probe(struct platform_device *pdev, - struct snd_soc_dai *dai) +#define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 + +static struct snd_soc_dai_ops davinci_i2s_dai_ops = { + .startup = davinci_i2s_startup, + .trigger = davinci_i2s_trigger, + .hw_params = davinci_i2s_hw_params, + .set_fmt = davinci_i2s_set_dai_fmt, + +}; + +struct snd_soc_dai davinci_i2s_dai = { + .name = "davinci-i2s", + .id = 0, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = &davinci_i2s_dai_ops, + +}; +EXPORT_SYMBOL_GPL(davinci_i2s_dai); + +static int davinci_i2s_probe(struct platform_device *pdev) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_card *card = socdev->card; - struct snd_soc_dai *cpu_dai = card->dai_link->cpu_dai; + struct snd_platform_data *pdata = pdev->dev.platform_data; struct davinci_mcbsp_dev *dev; - struct resource *mem, *ioarea; - struct evm_snd_platform_data *pdata; - int ret; + struct resource *mem, *ioarea, *res; + int ret = 0; mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!mem) { @@ -466,9 +490,7 @@ static int davinci_i2s_probe(struct platform_device *pdev, goto err_release_region; } - cpu_dai->private_data = dev; - - dev->clk = clk_get(&pdev->dev, NULL); + dev->clk = clk_get(&pdev->dev, pdata->clk_name); if (IS_ERR(dev->clk)) { ret = -ENODEV; goto err_free_mem; @@ -476,18 +498,35 @@ static int davinci_i2s_probe(struct platform_device *pdev, clk_enable(dev->clk); dev->base = (void __iomem *)IO_ADDRESS(mem->start); - pdata = pdev->dev.platform_data; dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK] = &davinci_i2s_pcm_out; - dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->channel = pdata->tx_dma_ch; dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->dma_addr = (dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DXR_REG); dev->dma_params[SNDRV_PCM_STREAM_CAPTURE] = &davinci_i2s_pcm_in; - dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->channel = pdata->rx_dma_ch; dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->dma_addr = (dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DRR_REG); + /* first TX, then RX */ + res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!res) { + dev_err(&pdev->dev, "no DMA resource\n"); + goto err_free_mem; + } + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->channel = res->start; + + res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!res) { + dev_err(&pdev->dev, "no DMA resource\n"); + goto err_free_mem; + } + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->channel = res->start; + + davinci_i2s_dai.private_data = dev; + ret = snd_soc_register_dai(&davinci_i2s_dai); + if (ret != 0) + goto err_free_mem; + return 0; err_free_mem: @@ -498,62 +537,40 @@ err_release_region: return ret; } -static void davinci_i2s_remove(struct platform_device *pdev, - struct snd_soc_dai *dai) +static int davinci_i2s_remove(struct platform_device *pdev) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_card *card = socdev->card; - struct snd_soc_dai *cpu_dai = card->dai_link->cpu_dai; - struct davinci_mcbsp_dev *dev = cpu_dai->private_data; + struct davinci_mcbsp_dev *dev = davinci_i2s_dai.private_data; struct resource *mem; + snd_soc_unregister_dai(&davinci_i2s_dai); clk_disable(dev->clk); clk_put(dev->clk); dev->clk = NULL; - kfree(dev); - mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); release_mem_region(mem->start, (mem->end - mem->start) + 1); -} -#define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 - -static struct snd_soc_dai_ops davinci_i2s_dai_ops = { - .startup = davinci_i2s_startup, - .trigger = davinci_i2s_trigger, - .hw_params = davinci_i2s_hw_params, - .set_fmt = davinci_i2s_set_dai_fmt, -}; + return 0; +} -struct snd_soc_dai davinci_i2s_dai = { - .name = "davinci-i2s", - .id = 0, - .probe = davinci_i2s_probe, - .remove = davinci_i2s_remove, - .playback = { - .channels_min = 2, - .channels_max = 2, - .rates = DAVINCI_I2S_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .capture = { - .channels_min = 2, - .channels_max = 2, - .rates = DAVINCI_I2S_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = &davinci_i2s_dai_ops, +static struct platform_driver davinci_mcbsp_driver = { + .probe = davinci_i2s_probe, + .remove = davinci_i2s_remove, + .driver = { + .name = "davinci-asp", + .owner = THIS_MODULE, + }, }; -EXPORT_SYMBOL_GPL(davinci_i2s_dai); static int __init davinci_i2s_init(void) { - return snd_soc_register_dai(&davinci_i2s_dai); + return platform_driver_register(&davinci_mcbsp_driver); } module_init(davinci_i2s_init); static void __exit davinci_i2s_exit(void) { - snd_soc_unregister_dai(&davinci_i2s_dai); + platform_driver_unregister(&davinci_mcbsp_driver); } module_exit(davinci_i2s_exit); -- 1.5.6

3 2

[alsa-devel] [PATCH] ASoC: Allow 32bits registers
by Daniel Ribeiro 08 Jun '09

08 Jun '09

Replace the remaining unsigned shorts with unsigned ints. Tested with pcap2 codec (25 bits registers). Signed-off-by: Daniel Ribeiro <drwyrm(a)gmail.com> --- include/sound/soc.h | 4 ++-- sound/soc/soc-core.c | 28 ++++++++++++++-------------- sound/soc/soc-dapm.c | 16 ++++++++-------- 3 files changed, 24 insertions(+), 24 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index a40bc6f..33b7de1 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -212,9 +212,9 @@ void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count, /* codec register bit access */ int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg, - unsigned short mask, unsigned short value); + unsigned int mask, unsigned int value); int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg, - unsigned short mask, unsigned short value); + unsigned int mask, unsigned int value); int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, struct snd_ac97_bus_ops *ops, int num); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1cd149b..c9f19d0 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1262,10 +1262,10 @@ EXPORT_SYMBOL_GPL(snd_soc_free_ac97_codec); * Returns 1 for change else 0. */ int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg, - unsigned short mask, unsigned short value) + unsigned int mask, unsigned int value) { int change; - unsigned short old, new; + unsigned int old, new; mutex_lock(&io_mutex); old = snd_soc_read(codec, reg); @@ -1292,10 +1292,10 @@ EXPORT_SYMBOL_GPL(snd_soc_update_bits); * Returns 1 for change else 0. */ int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg, - unsigned short mask, unsigned short value) + unsigned int mask, unsigned int value) { int change; - unsigned short old, new; + unsigned int old, new; mutex_lock(&io_mutex); old = snd_soc_read(codec, reg); @@ -1580,7 +1580,7 @@ int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned short val, bitmask; + unsigned int val, bitmask; for (bitmask = 1; bitmask < e->max; bitmask <<= 1) ; @@ -1609,8 +1609,8 @@ int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned short val; - unsigned short mask, bitmask; + unsigned int val; + unsigned int mask, bitmask; for (bitmask = 1; bitmask < e->max; bitmask <<= 1) ; @@ -1646,7 +1646,7 @@ int snd_soc_get_value_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned short reg_val, val, mux; + unsigned int reg_val, val, mux; reg_val = snd_soc_read(codec, e->reg); val = (reg_val >> e->shift_l) & e->mask; @@ -1685,8 +1685,8 @@ int snd_soc_put_value_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned short val; - unsigned short mask; + unsigned int val; + unsigned int mask; if (ucontrol->value.enumerated.item[0] > e->max - 1) return -EINVAL; @@ -1846,7 +1846,7 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; - unsigned short val, val2, val_mask; + unsigned int val, val2, val_mask; val = (ucontrol->value.integer.value[0] & mask); if (invert) @@ -1912,7 +1912,7 @@ int snd_soc_get_volsw_2r(struct snd_kcontrol *kcontrol, unsigned int reg2 = mc->rreg; unsigned int shift = mc->shift; int max = mc->max; - unsigned int mask = (1<<fls(max))-1; + unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; ucontrol->value.integer.value[0] = @@ -1952,7 +1952,7 @@ int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol, unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; int err; - unsigned short val, val2, val_mask; + unsigned int val, val2, val_mask; val_mask = mask << shift; val = (ucontrol->value.integer.value[0] & mask); @@ -2044,7 +2044,7 @@ int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); unsigned int reg = mc->reg; int min = mc->min; - unsigned short val; + unsigned int val; val = (ucontrol->value.integer.value[0]+min) & 0xff; val |= ((ucontrol->value.integer.value[1]+min) & 0xff) << 8; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 735903a..75c372f 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -227,7 +227,7 @@ static int dapm_connect_mixer(struct snd_soc_codec *codec, static int dapm_update_bits(struct snd_soc_dapm_widget *widget) { int change, power; - unsigned short old, new; + unsigned int old, new; struct snd_soc_codec *codec = widget->codec; /* check for valid widgets */ @@ -1205,7 +1205,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; - unsigned short val, val2, val_mask; + unsigned int val, val2, val_mask; int ret; val = (ucontrol->value.integer.value[0] & mask); @@ -1269,7 +1269,7 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned short val, bitmask; + unsigned int val, bitmask; for (bitmask = 1; bitmask < e->max; bitmask <<= 1) ; @@ -1297,8 +1297,8 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned short val, mux; - unsigned short mask, bitmask; + unsigned int val, mux; + unsigned int mask, bitmask; int ret = 0; for (bitmask = 1; bitmask < e->max; bitmask <<= 1) @@ -1356,7 +1356,7 @@ int snd_soc_dapm_get_value_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned short reg_val, val, mux; + unsigned int reg_val, val, mux; reg_val = snd_soc_read(widget->codec, e->reg); val = (reg_val >> e->shift_l) & e->mask; @@ -1396,8 +1396,8 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned short val, mux; - unsigned short mask; + unsigned int val, mux; + unsigned int mask; int ret = 0; if (ucontrol->value.enumerated.item[0] > e->max - 1) -- tg: (81ee1ba..) asoc/allow-32bits-registers (depends on: master) -- Daniel Ribeiro

2 2

[alsa-devel] ASoC update for 2.6.31
by Mark Brown 08 Jun '09

08 Jun '09

The following changes since commit e3509ff0fb9df53e45cd68488e3b463a80455db7: Daniel Mack (1): ASoC: fix NULL pointer dereference in soc_suspend() are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31 Mark Brown (1): ASoC: Apostrophe patrol Troy Kisky (1): ASoC: codec tlv320aic23 fix bogus divide by 0 message sound/soc/codecs/tlv320aic23.c | 10 +++++----- sound/soc/soc-dapm.c | 10 +++++----- 2 files changed, 10 insertions(+), 10 deletions(-)

2 1

[alsa-devel] Audio driver for dbAu1250 Development board
by Manjunath A M 08 Jun '09

08 Jun '09

Hi I am trying to make audio up in dbAu1250 Development board, I downloaded linux-2.6.29.4 kernel from the site "http://www.linux-mips.org/pub/linux/mips/kernel/v2.6/ " and applied the patch "asoc-v2.6.29-rc8-2009-03-17.patch"downloaded from "http://opensource.wolfsonmicro.com/~git/asoc/" but while compiling i am getting the bellow errors ********Linux compilation errors************* sound/soc/au1x/exmmb-ac97.c:24:38: error: asm/mach-exm1200/exm1200.h: No such file or directory sound/soc/au1x/exmmb-ac97.c:26:25: error: linux/exm32.h: No such file or directory sound/soc/au1x/exmmb-ac97.c:58: error: subscripted value is neither array nor pointer sound/soc/au1x/exmmb-ac97.c:65: error: variable âsnd_soc_machine_exm97â has initializer but incomplete type sound/soc/au1x/exmmb-ac97.c:66: error: unknown field ânameâ specified in initializer sound/soc/au1x/exmmb-ac97.c:66: warning: excess elements in struct initializer sound/soc/au1x/exmmb-ac97.c:66: warning: (near initialization for âsnd_soc_machine_exm97â) sound/soc/au1x/exmmb-ac97.c:67: error: unknown field âdai_linkâ specified in initializer sound/soc/au1x/exmmb-ac97.c:67: warning: excess elements in struct initializer sound/soc/au1x/exmmb-ac97.c:67: warning: (near initialization for âsnd_soc_machine_exm97â) sound/soc/au1x/exmmb-ac97.c:68: error: unknown field ânum_linksâ specified in initializer sound/soc/au1x/exmmb-ac97.c:68: warning: excess elements in struct initializer sound/soc/au1x/exmmb-ac97.c:68: warning: (near initialization for âsnd_soc_machine_exm97â) sound/soc/au1x/exmmb-ac97.c:73: error: unknown field âmachineâ specified in initializer sound/soc/au1x/exmmb-ac97.c:73: warning: initialization from incompatible pointer type sound/soc/au1x/exmmb-ac97.c:74: error: unknown field âplatformâ specified in initializer sound/soc/au1x/exmmb-ac97.c:74: warning: initialization from incompatible pointer type sound/soc/au1x/exmmb-ac97.c:80: error: âPSC1_BASEâ undeclared here (not in a function) sound/soc/au1x/exmmb-ac97.c: In function âpsc_to_ac97â: sound/soc/au1x/exmmb-ac97.c:107: error: implicit declaration of function âau_readlâ sound/soc/au1x/exmmb-ac97.c:110: error: implicit declaration of function âau_writelâ sound/soc/au1x/exmmb-ac97.c:111: error: implicit declaration of function âau_syncâ sound/soc/au1x/exmmb-ac97.c:115: error: âPSC_CTLâ undeclared (first use in this function) sound/soc/au1x/exmmb-ac97.c:115: error: (Each undeclared identifier is reported only once sound/soc/au1x/exmmb-ac97.c:115: error: for each function it appears in.) sound/soc/au1x/exmmb-ac97.c:117: error: âPSC_SELâ undeclared (first use in this function) sound/soc/au1x/exmmb-ac97.c:122: error: âPSC_AC97RSTâ undeclared (first use in this function) sound/soc/au1x/exmmb-ac97.c:124: error: implicit declaration of function âmsleepâ sound/soc/au1x/exmmb-ac97.c:133: error: âPSC_AC97STATâ undeclared (first use in this function) sound/soc/au1x/exmmb-ac97.c:140: error: âPSC_AC97CFGâ undeclared (first use in this function) sound/soc/au1x/exmmb-ac97.c: In function âexm1200_mobo_ac97_initâ: sound/soc/au1x/exmmb-ac97.c:160: error: implicit declaration of function âau_readwâ sound/soc/au1x/exmmb-ac97.c:160: error: âEXM1200_FPGA_BRDCTRLâ undeclared (first use in this function) sound/soc/au1x/exmmb-ac97.c:162: error: implicit declaration of function âau_writewâ make[3]: *** [sound/soc/au1x/exmmb-ac97.o] Error 1 make[2]: *** [sound/soc/au1x] Error 2 **************************************************** before applying the patch the board was working but the audio was not working.. can you suggest me which kernel source you used and how you got the audio working for your board. can you send me the sample kernel image "vmlinux.srec" image for testing.. Thanks Manjunath.AM The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it.

3 6

Re: [alsa-devel] ALSA patch Nvidia 8 channel hdmi audio
by Takashi Iwai 08 Jun '09

08 Jun '09

At Thu, 4 Jun 2009 08:18:23 +0200, Alan Alan wrote: > > 2009/6/4 Takashi Iwai <tiwai(a)suse.de> > > At Thu, 4 Jun 2009 01:28:24 +0200, > Alan Alan wrote: > > > > Hello Takashi ! > > > > I tried to apply the patch that removes restrictions to 16bits 48khz > format. > > Well now we have the following: > > 16bit 48khz works > > 32bit 48khz works > > 32bit 44khz, 96khz, 192khz don't work > > 24bit 44khz, 48khz, 96khz, 192khz don't work > > 16bit 44khz, 96khz, 192khz don't work > > > > I only get > > Sample format not available for playback: > > Setting of hwparams failed: > > As you can find in alsa-info output, it seems that the codec supports > only 48kHz and 88.2kHz, strangely. I don't know whether it's a > (hardware) bug or not. > > The codec surely supports up to 192khz as it works on Windows. > > I think this line is completely wrong as here it says: It is not "wrong". These are values that the codec reports, so it's more or less correct as the h/w information. > rates [0xc0]: 48000 88200 Wei, is it right? Or is anything missing here? > bits [0xf]: 8 16 20 24 > > 8, 20, 24bits are supported, but they all give an error message. 24bit here means 24bit in 4 bytes, thus it's essentially 32bit format. 24bit in 3 bytes format isn't supported on HD-audio. > 32 bits are not supposed to be supported but it works with 48 and 88.2khz. > > How can we change the software side to accept 96 or 192khz spample formats and > 24 bits ? As long as the codec reports so, it's no "software" problem. > I think it is not hardware limited and we just need to change the allowed > formats to be played. If it's just a hardware problem that reports wrong values, you can override the information, simply replacing the bits my patch disabled to the right values. Takashi > > > > > I enclosed to pastebin my alsainfo script output. > > > > http://pastebin.com/m77dcfb3f > > > > Thanks for the help ! > > > > 2009/6/3 Takashi Iwai <tiwai(a)suse.de> > > > > At Wed, 3 Jun 2009 11:34:00 +0200, > > Alan Alan wrote: > > > > > > Hello Takashi ! > > > > > > What should this patch do ? > > > > The patch removes the restriction patch_nvhdmi.c sets. Instead, > > the driver tries to detect the supported format, rate, etc from the > > codec widget information. > > > > > Is it a cure for the 96, 192khz or 24bit sample playback problem ? > > > > Possibly. > > > > > If so, will it resample or natievly pass the stream to the > amplifier ? > > > > It just enables the native support, possibly. > > > > And, don't forget to give back the alsa-info.sh output (run with > > --no-upload option). > > > > Takashi > > > > > > > > Thank you > > > > > > Alan > > > > > > 2009/6/3 Takashi Iwai <tiwai(a)suse.de> > > > > > > At Wed, 3 Jun 2009 10:00:24 +0200, > > > Alan Alan wrote: > > > > > > > > Hello Wei ! > > > > > > > > Thanks for replying. > > > > I tested it with speaker-test and xbmc babylon. Both had > problems > > with > > > > channel mapping. > > > > I used this command: > > > > > > > > speaker-test -Dhdmi -c6 > > > > or > > > > speaker-test -Dhdmi -c8 > > > > > > > > alternatively you could try > > > > > > > > speaker-test -Dhw:0,3 -c6 > > > > > > > > > > > > I had to rund Xorg first and have the command from an xterm, > > becasue > > > from a > > > > normal terminal it would not work. > > > > I think that is normal as a limitation of hdmi. > > > > > > > > What about the support for 24bits 192khz sound ? > > > > > > > > speaker-test -Dhdmi -c6 -r192000 does not work. > > > > > > How about the patch below? > > > > > > Also, could you give alsa-info.sh output? > > > > > > Takashi > > > > > > --- > > > diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/ > > patch_nvhdmi.c > > > index f5792e2..48443de 100644 > > > --- a/sound/pci/hda/patch_nvhdmi.c > > > +++ b/sound/pci/hda/patch_nvhdmi.c > > > @@ -258,9 +258,11 @@ static struct hda_pcm_stream > > > nvhdmi_pcm_digital_playback_8ch = { > > > .channels_min = 2, > > > .channels_max = 8, > > > .nid = Nv_Master_Convert_nid, > > > +#if 0 > > > .rates = SNDRV_PCM_RATE_48000, > > > .maxbps = 16, > > > .formats = SNDRV_PCM_FMTBIT_S16_LE, > > > +#endif > > > .ops = { > > > .open = nvhdmi_dig_playback_pcm_open, > > > .close = nvhdmi_dig_playback_pcm_close_8ch, > > > @@ -273,9 +275,11 @@ static struct hda_pcm_stream > > > nvhdmi_pcm_digital_playback_2ch = { > > > .channels_min = 2, > > > .channels_max = 2, > > > .nid = Nv_Master_Convert_nid, > > > +#if 0 > > > .rates = SNDRV_PCM_RATE_48000, > > > .maxbps = 16, > > > .formats = SNDRV_PCM_FMTBIT_S16_LE, > > > +#endif > > > .ops = { > > > .open = nvhdmi_dig_playback_pcm_open, > > > .close = nvhdmi_dig_playback_pcm_close_2ch, > > > > > > > > > > > >

3 7

[alsa-devel] [PATCH v4 0/4] ARM: DaVinci: ASoC: Audio support for DM646x
by Chaithrika U S 07 Jun '09

07 Jun '09

From: Chaithrika U S <chaithrika(a)ti.com> Add audio support for DM646x Signed-off-by: Naresh Medisetty <naresh(a)ti.com> Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- This patch set adds audio support for DM646x This updated version has included most of the review comments received for the previous version of the patch set. Patch 1: Registers the platform devices for ASP on dm355, dm644x and dm646x This patch applies to the DaVinci git tree available at http://git.kernel.org/?p=linux/kernel/git/khilman/linux-davinci.git Patch 2: Introduces platform driver model for dm644x and dm355 Applies against ALSA mainline - to the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git Patch 3: Adds driver support for the two instances of McASP on TI's DM646x Applies against ALSA mainline - to the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git Patch 4: Add machine driver support for DM646x Applies against ALSA mainline - to the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git These patches are based on the work done by Steve Chen <schen(a)mvista.com> and Pavel Kiryukhin <pkiryukhin(a)ru.mvista.com> on Monstavista 2.6.18 kernel.

2 1

[alsa-devel] Testers wanted: New SB X-Fi driver
by Gernot Kohlhaas 07 Jun '09

07 Jun '09

Hi! I found a little strange thing: Recording from microphone is only possible, when you unmute the "Line-In" playback-device (and set microphone as capture device, of course). Oh, and by the way: Is there any possibility to apply a low pass filter to the subwoofer-channel without using ladspa (xmms and audacious won't play songs with my low pass filter. Mplayer, xine and aplay are working fine). Thanks again for this driver :) -- http://www.gnupp.de/verschluesselung/warum.html Key ID: 0x4C0B2250 Key: http://home.arcor.de/mr_maniac/ANDERES/public.gpg

1 0

[alsa-devel] [RFC/PATCH] asoc: pxa-ssp.c -> enchance the I2S enchancement :)
by Daniel Ribeiro 06 Jun '09

06 Jun '09

Hi, this patch is what i need to use Paul's pxa-ssp.c I2S enchancements on my board. Basically, it doesn't enforce the "fake" network mode, as i dont need it. And i need _real_ network mode, so i prefer to set SSCR0_MOD only when really using network mode. And it uses PXA_SSP_FRM_??FS to actually set SSCR0_DataSize() too. This allows me to use 32bit frames for 16 bit stereo audio. (No, I don't want to fake network mode for this, and SSCR0_MOD doesn't work as expected when PXA is slave). This patch is incremental to the recent patch from Paul Shen. Signed-off-by: Daniel Ribeiro <drwyrm(a)gmail.com> diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index ed0906e..762cb93 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -579,7 +579,7 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: case SND_SOC_DAIFMT_LEFT_J: - sscr0 |= SSCR0_PSP | SSCR0_MOD; + sscr0 |= SSCR0_PSP; sscr1 |= SSCR1_RWOT | SSCR1_TRAIL; /* See hw_params() */ @@ -655,7 +655,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, int dma = 0, chn = params_channels(params); u32 sscr0; u32 sspsp; - int frame_width; + int frame_width = 0; int width = snd_pcm_format_physical_width(params_format(params)); int ttsa = ssp_read_reg(ssp, SSTSA) & 0xf; @@ -698,7 +698,6 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(16)); break; } - ssp_write_reg(ssp, SSCR0, sscr0); switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: @@ -742,6 +741,15 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, break; } + if (frame_width) { + sscr0 &= ~(SSCR0_EDSS | SSCR0_DataSize(16)); + if (frame_width % 48 == 0) + sscr0 |= SSCR0_EDSS | SSCR0_DataSize(8); + else + sscr0 |= SSCR0_EDSS | SSCR0_DataSize(16); + } + ssp_write_reg(ssp, SSCR0, sscr0); + /* When we use a network mode, we always require TDM slots * - complain loudly and fail if they've not been set up yet. */ -- Daniel Ribeiro

1 0

[alsa-devel] Testers wanted: New SB X-Fi driver
by MonsieurWeller 06 Jun '09

06 Jun '09

Hello. Works fine, but the external rack of the "x-fi fatality" didn't seems to work. I plug my headphone on it and i have no sound. Thanks.

2 1