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[alsa-devel] PATCH: C-Media 6206 USB Audio
by Dan Allongo 08 Jun '09

08 Jun '09

I've been posting on the alsa-users list and I think I finally have a working patch now that I'd like to submit. The C-Media 6206 device is based off of CM106 and just needed a boot quirk added to set the registers properly for output. At least one other user can confirm that this patch works and that the SPDIF-out is also functioning now with this change. This is diff'd against today's snapshot so it should be good to go. Let me know if there's anything else that I need to do or change with this. Thanks, ---Dan --- alsa-driver/alsa-kernel/usb/usbaudio.c 2009-06-07 06:20:32.000000000 -0400 +++ alsa-driver/alsa-kernel/usb/usbaudio.c 2009-06-07 22:24:44.000000000 -0400 @@ -3291,6 +3291,21 @@ return snd_usb_cm106_write_int_reg(dev, 2, 0x8004); } +static int snd_usb_cm6206_boot_quirk(struct usb_device *dev) +{ + /* + * C-Media CM6206 is based on CM106 with two additional + * registers that are not documented in the data sheet. + * Values here are chosen based on sniffing USB traffic + * under Windows. + */ + return snd_usb_cm106_write_int_reg(dev, 0, 0x200c) + + snd_usb_cm106_write_int_reg(dev, 1, 0x3000) + + snd_usb_cm106_write_int_reg(dev, 2, 0xf800) + + snd_usb_cm106_write_int_reg(dev, 3, 0x143f) + + snd_usb_cm106_write_int_reg(dev, 4, 0x0000) + + snd_usb_cm106_write_int_reg(dev, 5, 0x3000); +} /* * Setup quirks @@ -3577,6 +3592,12 @@ goto __err_val; } + /* C-Media CM6206 / CM106-Like Sound Device */ + if (id == USB_ID(0x0d8c, 0x0102)) { + if (snd_usb_cm6206_boot_quirk(dev) < 0) + goto __err_val; + } + /* * found a config. now register to ALSA */

2 2

[PATCH - CM6206 1/1] modified: sound/usb/usbaudio.c
by Dan Allongo 08 Jun '09

08 Jun '09

Added boot quirk for C-Media CM6206 device in snd_usb_audio_probe. The function snd_usb_cm6206_boot_quirk sets up six internal 16-bit registers in order to initialize the device. Values for the registers came from sniffing USB traffic under Windows since only four of the six are documented in the datasheet for CM106 and some reserved bits were also being set. Signed-off-by: Dan Allongo <gongo2k1(a)gmail.com> diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index a8ef2cb..94e43ee 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -3291,6 +3291,24 @@ static int snd_usb_cm106_boot_quirk(struct usb_device *dev) return snd_usb_cm106_write_int_reg(dev, 2, 0x8004); } +/* + * C-Media CM6206 is based on CM106 with two additional + * registers that are not documented in the data sheet. + * Values here are chosen based on sniffing USB traffic + * under Windows. + */ +static int snd_usb_cm6206_boot_quirk(struct usb_device *dev) +{ + int err, reg; + int val[] = {0x200c,0x3000,0xf800,0x143f,0x0000,0x3000}; + + for (reg = 0; reg < 6; reg++) { + if ((err = snd_usb_cm106_write_int_reg(dev, reg, val[reg])) < 0) + return err; + } + + return err; +} /* * Setup quirks @@ -3577,6 +3595,12 @@ static void *snd_usb_audio_probe(struct usb_device *dev, goto __err_val; } + /* C-Media CM6206 / CM106-Like Sound Device */ + if (id == USB_ID(0x0d8c, 0x0102)) { + if (snd_usb_cm6206_boot_quirk(dev) < 0) + goto __err_val; + } + /* * found a config. now register to ALSA */ -- 1.5.6.5 On Mon, Jun 8, 2009 at 1:41 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > At Sun, 7 Jun 2009 22:46:51 -0400, > Dan Allongo wrote: > > > > I've been posting on the alsa-users list and I think I finally have a > > working patch now that I'd like to submit. > > The C-Media 6206 device is based off of CM106 and just needed a boot > quirk > > added to set the registers properly for output. > > At least one other user can confirm that this patch works and that the > > SPDIF-out is also functioning now with this change. > > This is diff'd against today's snapshot so it should be good to go. > > Let me know if there's anything else that I need to do or change with > this. > > Thanks for the patch! I'd like certainly to apply it, however... > > > + return snd_usb_cm106_write_int_reg(dev, 0, 0x200c) + > > + snd_usb_cm106_write_int_reg(dev, 1, 0x3000) + > > + snd_usb_cm106_write_int_reg(dev, 2, 0xf800) + > > + snd_usb_cm106_write_int_reg(dev, 3, 0x143f) + > > + snd_usb_cm106_write_int_reg(dev, 4, 0x0000) + > > + snd_usb_cm106_write_int_reg(dev, 5, 0x3000); > > Summing the return values here look weird. > You want to return an error code, right? Then use a simple if > sequence, or write a loop there. > > Also, run $LINUX/scripts/checkpatch.pl to your patch and fix errors / > warnings suggested there. Then please repost the patch with your > sign-off. > > > thanks, > > Takashi >

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[alsa-devel] [PATCH 0/7] DAPM power sequencing changes
by Mark Brown 08 Jun '09

08 Jun '09

I've implemented what I hope are some improvements to the DAPM power sequence application. Currently when it is implementing power changes DAPM will do one register write per widget. Normally this leads to multiple writes to the same register since power control tends to be clustered in a few registers, slowing things down. This can get noticable on slow or heavily contended buses, especially around resume. The overall effect of these changes is that DAPM will try to minimise the number of register writes it does when implementing power sequences. As with any substantial change in the power sequencing there is a risk that there may have been some performance regressions on some systems - my biggest concern is that some thing were being helped by the delays introduced by the repeated register writes and may require special handling to fix. My hope is that this won't happen if anything it should fix some problems by ensuring that both left and right channels of stereo paths get powered up together but without wide testing that's hard to guarantee. Any testing that you can do on your systems would be greatly appreciated. Obviously this is quite an invasive change so I don't intend to submit to ALSA until after the 2.6.31 merge window has closed. The patches can be found here for the time being: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git dapm Mark Brown (7): ASoC: Sort DAPM power sequences while building lists ASoC: Factor out DAPM sequence execution ASoC: Coalesce register writes for DAPM sequences ASoC: Sort specialised mixers and muxes together ASoC: Coalesce power updates for DAPM widgets with events ASoC: Coalesce power updates for PGAs ASoC: Make DAPM power sequence lists local variables include/sound/soc.h | 2 - sound/soc/soc-dapm.c | 302 +++++++++++++++++++++++++++++++++++++------------- 2 files changed, 227 insertions(+), 77 deletions(-)

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[alsa-devel] [PATCH] ARM: DaVinci: ASoc: Minor fixes to probe function
by chaithrika＠ti.com 08 Jun '09

08 Jun '09

From: Chaithrika U S <chaithrika(a)ti.com> Assign proper errors when platform resource claims fail. Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- Applies to branch 'davinci' of: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git sound/soc/davinci/davinci-i2s.c | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 500d2f5..2f11970 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -469,7 +469,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) struct snd_platform_data *pdata = pdev->dev.platform_data; struct davinci_mcbsp_dev *dev; struct resource *mem, *ioarea, *res; - int ret = 0; + int ret; mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!mem) { @@ -511,6 +511,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (!res) { dev_err(&pdev->dev, "no DMA resource\n"); + ret = -ENXIO; goto err_free_mem; } dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->channel = res->start; @@ -518,6 +519,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_DMA, 1); if (!res) { dev_err(&pdev->dev, "no DMA resource\n"); + ret = -ENXIO; goto err_free_mem; } dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->channel = res->start; -- 1.5.6

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[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]

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[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

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[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

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[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(-)

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[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.

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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, > > > > > > > > > > > >

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