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

[alsa-devel] [PATCHv2 1/6] ASoC: OMAP: data_type and sync_mode configurable in audio dma
by Candelaria Villareal, Jorge 25 Jan '10

25 Jan '10

From: Misael Lopez Cruz <x0052729(a)ti.com> Allow client drivers to set the data_type (16, 32) and the sync_mode (element, packet, etc) of the audio dma transferences. McBSP dai driver configures it for a data type of 16 bits and element sync mode. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> Signed-off-by: Jorge Eduardo Candelaria <x0107209(a)ti.com> --- sound/soc/omap/omap-mcbsp.c | 2 ++ sound/soc/omap/omap-pcm.c | 15 ++++++++------- sound/soc/omap/omap-pcm.h | 4 +++- 3 files changed, 13 insertions(+), 8 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 6bbbd2a..d297256 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -287,6 +287,8 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, omap_mcbsp_dai_dma_params[id][substream->stream].dma_req = dma; omap_mcbsp_dai_dma_params[id][substream->stream].port_addr = port; omap_mcbsp_dai_dma_params[id][substream->stream].sync_mode = sync_mode; + omap_mcbsp_dai_dma_params[id][substream->stream].data_type = + OMAP_DMA_DATA_TYPE_S16; cpu_dai->dma_data = &omap_mcbsp_dai_dma_params[id][substream->stream]; if (mcbsp_data->configured) { diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 9db2770..825db38 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -37,7 +37,8 @@ static const struct snd_pcm_hardware omap_pcm_hardware = { SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, - .formats = SNDRV_PCM_FMTBIT_S16_LE, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, .period_bytes_min = 32, .period_bytes_max = 64 * 1024, .periods_min = 2, @@ -149,6 +150,7 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) struct omap_runtime_data *prtd = runtime->private_data; struct omap_pcm_dma_data *dma_data = prtd->dma_data; struct omap_dma_channel_params dma_params; + int bytes; /* return if this is a bufferless transfer e.g. * codec <--> BT codec or GSM modem -- lg FIXME */ @@ -156,11 +158,7 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) return 0; memset(&dma_params, 0, sizeof(dma_params)); - /* - * Note: Regardless of interface data formats supported by OMAP McBSP - * or EAC blocks, internal representation is always fixed 16-bit/sample - */ - dma_params.data_type = OMAP_DMA_DATA_TYPE_S16; + dma_params.data_type = dma_data->data_type; dma_params.trigger = dma_data->dma_req; dma_params.sync_mode = dma_data->sync_mode; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { @@ -170,6 +168,7 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) dma_params.src_start = runtime->dma_addr; dma_params.dst_start = dma_data->port_addr; dma_params.dst_port = OMAP_DMA_PORT_MPUI; + dma_params.dst_fi = dma_data->packet_size; } else { dma_params.src_amode = OMAP_DMA_AMODE_CONSTANT; dma_params.dst_amode = OMAP_DMA_AMODE_POST_INC; @@ -177,6 +176,7 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) dma_params.src_start = dma_data->port_addr; dma_params.dst_start = runtime->dma_addr; dma_params.src_port = OMAP_DMA_PORT_MPUI; + dma_params.src_fi = dma_data->packet_size; } /* * Set DMA transfer frame size equal to ALSA period size and frame @@ -184,7 +184,8 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) * we can transfer the whole ALSA buffer with single DMA transfer but * still can get an interrupt at each period bounary */ - dma_params.elem_count = snd_pcm_lib_period_bytes(substream) / 2; + bytes = snd_pcm_lib_period_bytes(substream); + dma_params.elem_count = bytes >> dma_data->data_type; dma_params.frame_count = runtime->periods; omap_set_dma_params(prtd->dma_ch, &dma_params); diff --git a/sound/soc/omap/omap-pcm.h b/sound/soc/omap/omap-pcm.h index 38a821d..b19975d 100644 --- a/sound/soc/omap/omap-pcm.h +++ b/sound/soc/omap/omap-pcm.h @@ -29,8 +29,10 @@ struct omap_pcm_dma_data { char *name; /* stream identifier */ int dma_req; /* DMA request line */ unsigned long port_addr; /* transmit/receive register */ - int sync_mode; /* DMA sync mode */ void (*set_threshold)(struct snd_pcm_substream *substream); + int data_type; /* data type 8,16,32 */ + int sync_mode; /* DMA sync mode */ + int packet_size; /* packet size only in PACKET mode */ }; extern struct snd_soc_platform omap_soc_platform; -- 1.6.0.4

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[alsa-devel] ASoC updates for 2.6.33
by Mark Brown 25 Jan '10

25 Jan '10

The following changes since commit 92dcffb916d309aa01778bf8963a6932e4014d07: Linus Torvalds (1): Linux 2.6.33-rc5 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.33 The pull request includes a much bigger delta against Linus' tree, I merged up since the last batch of fixes that went into his tree didn't seem to have shown up on your fix branch for some reason. Guennadi Liakhovetski (1): ASoC: fix a memory-leak in wm8903 sound/soc/codecs/wm8903.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-)

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[alsa-devel] [PATCH v2] ad1836: reset and restore clock control mode in suspend/resume entry
by Barry Song 25 Jan '10

25 Jan '10

tests show frequent suspend/resume(frequent poweroff/on ad1836 internal components) maybe make ad1836 clock mode wrong sometimes after wakeup. This patch reset/restore ad1836 clock mode while executing PM, then ad1836 can always resume to right clock status. Signed-off-by: Barry Song <21cnbao(a)gmail.com> --- -v2: commit forgotten ad1836.h sound/soc/codecs/ad1836.c | 32 ++++++++++++++++++++++++++++++++ sound/soc/codecs/ad1836.h | 1 + 2 files changed, 33 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 2c18e3d..83add2f 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -223,6 +223,36 @@ static unsigned int ad1836_read_reg_cache(struct snd_soc_codec *codec, return reg_cache[reg]; } +#ifdef CONFIG_PM +static int ad1836_soc_suspend(struct platform_device *pdev, + pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + /* reset clock control mode */ + u16 adc_ctrl2 = codec->read(codec, AD1836_ADC_CTRL2); + adc_ctrl2 &= ~AD1836_ADC_SERFMT_MASK; + + return codec->write(codec, AD1836_ADC_CTRL2, adc_ctrl2); +} + +static int ad1836_soc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + /* restore clock control mode */ + u16 adc_ctrl2 = codec->read(codec, AD1836_ADC_CTRL2); + adc_ctrl2 |= AD1836_ADC_AUX; + + return codec->write(codec, AD1836_ADC_CTRL2, adc_ctrl2); +} +#else +#define ad1836_soc_suspend NULL +#define ad1836_soc_resume NULL +#endif + static int __devinit ad1836_spi_probe(struct spi_device *spi) { struct snd_soc_codec *codec; @@ -404,6 +434,8 @@ static int ad1836_remove(struct platform_device *pdev) struct snd_soc_codec_device soc_codec_dev_ad1836 = { .probe = ad1836_probe, .remove = ad1836_remove, + .suspend = ad1836_soc_suspend, + .resume = ad1836_soc_resume, }; EXPORT_SYMBOL_GPL(soc_codec_dev_ad1836); diff --git a/sound/soc/codecs/ad1836.h b/sound/soc/codecs/ad1836.h index 7660ee6..e9d90d3 100644 --- a/sound/soc/codecs/ad1836.h +++ b/sound/soc/codecs/ad1836.h @@ -54,6 +54,7 @@ #define AD1836_ADC_SERFMT_MASK (7 << 6) #define AD1836_ADC_SERFMT_PCK256 (0x4 << 6) #define AD1836_ADC_SERFMT_PCK128 (0x5 << 6) +#define AD1836_ADC_AUX (0x6 << 6) #define AD1836_ADC_CTRL3 14 -- 1.5.6.3

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[alsa-devel] [PATCHv2 0/6] McPDM driver patch series
by Candelaria Villareal, Jorge 25 Jan '10

25 Jan '10

McPDM module is the interface between the OMAP and TWL6030 audio codec. The interface supports the following: * Up to 5 downlink channels and 3 uplink channels * 88.2 and 96 KHz * 32 bit sample size Currently, driver is set to support up to 4 downlink channels. If 2 channels are selected, output will go to the headset, and if 4 channels are selected, output will go to the headset and speakers. Functionality to select the output at runtime will be added afterwards. Command and status mode is not supported in current patches. Changes from v1: - Device registration now takes place plat-omap/devices.c - struct resources is used to store mem and irq resources. - DMA allows packet_size to be configured. - PACKET is now used as sync_mode for McPDM -- Jorge Eduardo Candelaria (4): OMAP4: IRQ: Add McPDM IRQ definition. ARM: OMAP4: Add McPDM base address OMAP4: MCPDM: Register McPDM platform device ASoC: OMAP4: Add support for McPDM Misael Lopez Cruz (2): ASoC: OMAP: data_type and sync_mode configurable in audio dma ASoC: OMAP4: Add McPDM platform driver arch/arm/plat-omap/devices.c | 34 ++ arch/arm/plat-omap/include/plat/irqs.h | 1 + arch/arm/plat-omap/include/plat/omap44xx.h | 2 + sound/soc/omap/Kconfig | 3 + sound/soc/omap/Makefile | 2 + sound/soc/omap/mcpdm.c | 524 ++++++++++++++++++++++++++++ sound/soc/omap/mcpdm.h | 153 ++++++++ sound/soc/omap/omap-mcbsp.c | 2 + sound/soc/omap/omap-mcpdm.c | 248 +++++++++++++ sound/soc/omap/omap-mcpdm.h | 29 ++ sound/soc/omap/omap-pcm.c | 15 +- sound/soc/omap/omap-pcm.h | 4 +- 12 files changed, 1009 insertions(+), 8 deletions(-) create mode 100644 sound/soc/omap/mcpdm.c create mode 100644 sound/soc/omap/mcpdm.h create mode 100644 sound/soc/omap/omap-mcpdm.c

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[alsa-devel] snd-hda-intel w/Realtek ALC260 codec broken on FJS Celsius M440
by Martin Lucina 25 Jan '10

25 Jan '10

Hello, in the interests of helping to debug the mess that is snd-hda-intel and getting sound working properly on my machine, I'd like to report the following problem: This machine is a Fujitsu Siemens Celsius M440 desktop -- about 4 years old and running Debian lenny. The sound is onboard Intel HD Audio with a Realtek ALC260 codec. The mainboard (Fujitsu D2178) has the following jacks: Rear: (left to right) Pink - MIC - Node 0x12 in hda-analyzer Green - Speaker/Line out(?) - Node 0x0f in hda-analyzer Blue - Line in - Node 0x14 in hda-analyzer Front: (left to right) Green - Headphone - Node 0x15 in hda-analyzer Pink - MIC - Node 0x13 in hda-analyzer Internal: (at least, can confirm detail if needed) CD Internal speaker The current status with kernel 2.6.32.5 is as follows: 1) snd_hda_intel loaded with model=basic (the "slightly less broken" option) Mixer playback behaviour: Headphone -> nothing Front -> Front and Rear green jack Line -> Not tested CD -> Not tested Beep -> Beep Capture behaviour: I've only tested with a headset connected to the front jacks, with "Capture" set to 100% and "Capture Source" set to "Front Mic". In this mode I seem to get a *very weak* signal from the mic as tested by doing "arecord > ./test.wav", then "aplay ./tmp.wav". 2) snd_hda_intel loaded with model=auto (the "completely broken" option) Mixer playback behaviour: Headphone -> Internal speaker Front -> Rear green jack Beep -> Beep Capture: untested In this mode the front jacks don't seem to work at all. I'd appreciate any help on debugging this. The ideal configuration which I guess should(?) be possible is separate mixers for Front out, Rear out, Internal speaker, and working capture with both Front and Rear Mic. Also, can anyone suggest a better tool to test recording than "arecord"? Specifically something that will actually tell me if the level of the sound recorded from the Mic is "sane"? Also see the attached output from alsa-info.sh for both model=auto and model=basic. I tried a few other models but with no luck. Thanks, -mato

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[alsa-devel] [PATCH 0/5] Add Voice Codec support for DaVinci platforms
by Miguel Aguilar 25 Jan '10

25 Jan '10

Hi, The following patch series adds support for the Voice Codec in the Davinci platform and in this particular case for the DM365 EVM. The DM365 EVM has two codecs the AIC3x and the Voice Codec, however at this point is not possible to have both enabled simultaneously since the soc-core doesn't support that, so a patch for enable both simultaneously will be sent when this support is available in the soc-core. Regards, -- Miguel Angel Aguilar Ulloa Embedded Software Engineer RidgeRun Embedded Solutions miguel.aguilar(a)ridgerun.com

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[alsa-devel] [PATCH 4/5] DaVinci: DM365: Voice codec support for the DM365 SoC
by miguel.aguilar＠ridgerun.com 25 Jan '10

25 Jan '10

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> This patch adds the generic Voice Codec support for the DM365 based platforms. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- arch/arm/mach-davinci/dm365.c | 37 +++++++++++++++++++++++++++- arch/arm/mach-davinci/include/mach/dm365.h | 5 ++++ arch/arm/mach-davinci/include/mach/mux.h | 2 + 3 files changed, 43 insertions(+), 1 deletions(-) diff --git a/arch/arm/mach-davinci/dm365.c b/arch/arm/mach-davinci/dm365.c index c63afc0..b5345d5 100644 --- a/arch/arm/mach-davinci/dm365.c +++ b/arch/arm/mach-davinci/dm365.c @@ -455,7 +455,7 @@ static struct clk_lookup dm365_clks[] = { CLK(NULL, "timer3", &timer3_clk), CLK(NULL, "usb", &usb_clk), CLK("davinci_emac.1", NULL, &emac_clk), - CLK("voice_codec", NULL, &voicecodec_clk), + CLK("davinci_voicecodec", NULL, &voicecodec_clk), CLK("davinci-asp.0", NULL, &asp0_clk), CLK(NULL, "rto", &rto_clk), CLK(NULL, "mjcp", &mjcp_clk), @@ -606,6 +606,8 @@ INT_CFG(DM365, INT_NSF_DISABLE, 25, 1, 0, false) EVT_CFG(DM365, EVT2_ASP_TX, 0, 1, 0, false) EVT_CFG(DM365, EVT3_ASP_RX, 1, 1, 0, false) +EVT_CFG(DM365, EVT2_VC_TX, 0, 1, 1, false) +EVT_CFG(DM365, EVT3_VC_RX, 1, 1, 1, false) #endif }; @@ -835,6 +837,31 @@ static struct platform_device dm365_asp_device = { .resource = dm365_asp_resources, }; +static struct resource dm365_vc_resources[] = { + { + .start = DAVINCI_DM365_VC_BASE, + .end = DAVINCI_DM365_VC_BASE + SZ_1K - 1, + .flags = IORESOURCE_MEM, + }, + { + .start = DAVINCI_DMA_VC_TX, + .end = DAVINCI_DMA_VC_TX, + .flags = IORESOURCE_DMA, + }, + { + .start = DAVINCI_DMA_VC_RX, + .end = DAVINCI_DMA_VC_RX, + .flags = IORESOURCE_DMA, + }, +}; + +static struct platform_device dm365_vc_device = { + .name = "davinci_voicecodec", + .id = -1, + .num_resources = ARRAY_SIZE(dm365_vc_resources), + .resource = dm365_vc_resources, +}; + static struct resource dm365_rtc_resources[] = { { .start = DM365_RTC_BASE, @@ -991,6 +1018,14 @@ void __init dm365_init_asp(struct snd_platform_data *pdata) platform_device_register(&dm365_asp_device); } +void __init dm365_init_vc(struct snd_platform_data *pdata) +{ + davinci_cfg_reg(DM365_EVT2_VC_TX); + davinci_cfg_reg(DM365_EVT3_VC_RX); + dm365_vc_device.dev.platform_data = pdata; + platform_device_register(&dm365_vc_device); +} + void __init dm365_init_ks(struct davinci_ks_platform_data *pdata) { dm365_ks_device.dev.platform_data = pdata; diff --git a/arch/arm/mach-davinci/include/mach/dm365.h b/arch/arm/mach-davinci/include/mach/dm365.h index 3c07a88..57e51eb 100644 --- a/arch/arm/mach-davinci/include/mach/dm365.h +++ b/arch/arm/mach-davinci/include/mach/dm365.h @@ -31,8 +31,13 @@ #define DM365_RTC_BASE (0x01C69000) +#define DAVINCI_DM365_VC_BASE (0x01D0C000) +#define DAVINCI_DMA_VC_TX 2 +#define DAVINCI_DMA_VC_RX 3 + void __init dm365_init(void); void __init dm365_init_asp(struct snd_platform_data *pdata); +void __init dm365_init_vc(struct snd_platform_data *pdata); void __init dm365_init_ks(struct davinci_ks_platform_data *pdata); void __init dm365_init_rtc(void); diff --git a/arch/arm/mach-davinci/include/mach/mux.h b/arch/arm/mach-davinci/include/mach/mux.h index 137bfba..2a68c1d 100644 --- a/arch/arm/mach-davinci/include/mach/mux.h +++ b/arch/arm/mach-davinci/include/mach/mux.h @@ -327,6 +327,8 @@ enum davinci_dm365_index { /* EDMA event muxing */ DM365_EVT2_ASP_TX, DM365_EVT3_ASP_RX, + DM365_EVT2_VC_TX, + DM365_EVT3_VC_RX, DM365_EVT26_MMC0_RX, }; -- 1.6.0.4

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[alsa-devel] Underrun problem and some basic queries
by ANISH KUMAR 25 Jan '10

25 Jan '10

Dear ALSA Team, I have some queries.Please answer them.(previous post please ignore as they couldn't appear because of encoding problem) I am getting underrun problem whenever the playback is about to get finished. happening in snd_pcm_update_hw_ptr_post function: if (avail >= stop_threshold) { if (substream->runtime->status->state == SNDRV_PCM_STATE_DRAINING) snd_pcm_drain_done(substream); else xrun(substream); //here it is happening.......... return -EPIPE; } 1. As i understand if i dont call drain function this will happen.Ismy understanding correct?Because if am calling drain function then thisproblem is not happening.If the application doesn't call drain functionthen xrun is the only way to stop ALSA or does ALSA detect that thereare no more samples to play and automatically stop the play(if yes,canyou point out the source code)? 2. snd_pcm_playback_avail() returns the number of samples that stillneeds to be played?Below is the diagram which is my understanding. Ring buffer (DMA buffer created using preallocated buffer during system initalisation) ----------------------------------------------------------------- | | | | | | | ----------------------------------------------------------------- ^ ^ | | hw_ptr appl_ptr After first write call: ----------------------------------------------------------------- | | | | | | | ----------------------------------------------------------------- ^ ^ | | appl_ptr hw_ptr So my understanding is once wait_for_avail_min returns insnd_pcm_lib_write1 function.hw_ptr is increased by some amount(periodsize) but appl_ptr will increase only when wait_for_avail_minreturns.And difference of (hw_ptr+runtime->buffersize-appl_ptr) will give us unplayed samples in the case of playback? Is this whole ring buffer is filled after the snd_pcm_lib_write_transfer function is returned? 3.In some rare cases (playback case) i have seen thatwait_for_avail_min doesn't return and driver pointer function keeps ongetting called causing hw_ptr to increase.This causessnd_pcm_playback_avail() function to return a value less than avail_min(as hw_ptr is increasing and appl_ptr is stagnant as it is stuck insnd_pcm_lib_write1 function) which inturn causes wake-up fromsnd_pcm_update_hw_ptr_post() not to get called.When this type ofscenario happens?Is it a normal scenario if yes,then how will it willever come out from wait_for_avail_min? According to my understanding appl_ptr will increase only when we return from wait_for_avail_min? 4.how does alsa know that it is running short of data?As iunderstand that it does not free the memory and data is onlyoverwritten.So how does it distinguish between old and new data? 5.Burst of noise due to stale data problem reported by Jon Smirl and for which Mr Jaroslav has suggested this solution http://mailman.alsa-project.org/pipermail/alsa-devel/2009-May/017119.html.Is it implemented? Best Regards..

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[alsa-devel] Underrun problem and some basic queries
by ANISH KUMAR 25 Jan '10

25 Jan '10

Hi Takashi et al, This post of mine is just a reminder about my earlier unanswered questions which i raised about underrun. http://mailman.alsa-project.org/pipermail/alsa-devel/2010-January/024286.ht… I guess you were on holiday and i think this is a bit long list of questions which no body cared to answer.This time please answer them as this would help all the new entrants to understand ALSA. Best Regards, anish

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[alsa-devel] How much of the X-Fi (emu20k1) chip has Creative documented?
by Chris Rankin 25 Jan '10

25 Jan '10

Hi, I am thinking about upgrading my Audigy2 sound card, and was looking at the X-Fi. However, the feature matrix for X-Fi cards on the ALSA site only says "PCI". (I assume that also includes PCIe these days.) Given that there is a basic snd-ctxfi driver, it seems likely that this matrix should also include "ANALOGio", and so I was wondering if someone could tell me what the actual matrix entry for the X-Fi cards should be these days, please? Would the documentation that Creative has released allow any additional features to be implemented in the future? I am obviously hoping at least for feature parity with the Audigy2, i.e. multi-voiced hardware mixing. That "crystalizer" would seem to be one of the cards main selling points, too. However, if the driver is "feature complete" as far as Creative's documentation goes, then one of the Xonar cards would seem better value for money. Thanks for any assistance, Cheers, Chris

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