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[alsa-devel] Radeon RV730 (Radeon HD 4670) not playing back
by Anders Kirchenbauer 16 Sep '09

16 Sep '09

I am unable to have sound playback through HDMI, presumably because there is no PCM codec support with the current state of my driver. An alsa-info.sh dump is below, and below that a pastebin.ca of speaker-test http://www.alsa-project.org/db/?f=2e2c6ba9f4052ef0a2defc25688067d597737af9 http://pastebin.ca/1567621 Also, changing the PCI ID to 1002aa38 in the two places in patch_atihdmi.c resulted in no sound or mixer device being available when the drivers were loaded. Went back to distro-default drivers afterwards. Only IEC958 Mute/Unmute control works, rest are permission denied. All permission denied are denied under user and root accounts. alsamixer defines card as "ATI R6xx HDMI"

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[alsa-devel] [PATCH v4 2/2] ASoC: Davinci: Add audio codec support for DM365 EVM
by miguel.aguilar＠ridgerun.com 15 Sep '09

15 Sep '09

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> This patch enables tlv320aic3101 support on DM365 EVM and it was tested on DM365 EVM rev c. Note: this patch was created based on temp/asoc branch. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- sound/soc/davinci/Kconfig | 4 ++-- sound/soc/davinci/davinci-evm.c | 7 ++++--- 2 files changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 4dfd4ad..047ee39 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -13,9 +13,9 @@ config SND_DAVINCI_SOC_MCASP tristate config SND_DAVINCI_SOC_EVM - tristate "SoC Audio support for DaVinci DM6446 or DM355 EVM" + tristate "SoC Audio support for DaVinci DM6446, DM355 or DM365 EVM" depends on SND_DAVINCI_SOC - depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM + depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM || MACH_DAVINCI_DM365_EVM select SND_DAVINCI_SOC_I2S select SND_SOC_TLV320AIC3X help diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 67414f6..7ccbe66 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -45,7 +45,8 @@ static int evm_hw_params(struct snd_pcm_substream *substream, unsigned sysclk; /* ASP1 on DM355 EVM is clocked by an external oscillator */ - if (machine_is_davinci_dm355_evm() || machine_is_davinci_dm6467_evm()) + if (machine_is_davinci_dm355_evm() || machine_is_davinci_dm6467_evm() || + machine_is_davinci_dm365_evm()) sysclk = 27000000; /* ASP0 in DM6446 EVM is clocked by U55, as configured by @@ -176,7 +177,7 @@ static struct snd_soc_dai_link da8xx_evm_dai = { .ops = &evm_ops, }; -/* davinci-evm audio machine driver */ +/* davinci dm6446, dm355 or dm365 evm audio machine driver */ static struct snd_soc_card snd_soc_card_evm = { .name = "DaVinci EVM", .platform = &davinci_soc_platform, @@ -243,7 +244,7 @@ static int __init evm_init(void) int index; int ret; - if (machine_is_davinci_evm()) { + if (machine_is_davinci_evm() || machine_is_davinci_dm365_evm()) { evm_snd_dev_data = &evm_snd_devdata; index = 0; } else if (machine_is_davinci_dm355_evm()) { -- 1.6.0.4

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[alsa-devel] [PATCH 1/6] AK4114 - fix output selector bits, fix spelling
by pavel.hofman＠ivitera.com 15 Sep '09

15 Sep '09

From: Pavel Hofman <pavel.hofman(a)ivitera.com> Signed-off-by: Pavel Hofman <pavel.hofman(a)ivitera.com> diff --git a/include/ak4114.h b/include/ak4114.h index d293d36..3ce69fd 100644 --- a/include/ak4114.h +++ b/include/ak4114.h @@ -95,13 +95,13 @@ /* AK4114_REG_IO0 */ #define AK4114_TX1E (1<<7) /* TX1 Output Enable (1 = enable) */ -#define AK4114_OPS12 (1<<2) /* Output Though Data Selector for TX1 pin */ -#define AK4114_OPS11 (1<<1) /* Output Though Data Selector for TX1 pin */ -#define AK4114_OPS10 (1<<0) /* Output Though Data Selector for TX1 pin */ +#define AK4114_OPS12 (1<<6) /* Output Data Selector for TX1 pin */ +#define AK4114_OPS11 (1<<5) /* Output Data Selector for TX1 pin */ +#define AK4114_OPS10 (1<<4) /* Output Data Selector for TX1 pin */ #define AK4114_TX0E (1<<3) /* TX0 Output Enable (1 = enable) */ -#define AK4114_OPS02 (1<<2) /* Output Though Data Selector for TX0 pin */ -#define AK4114_OPS01 (1<<1) /* Output Though Data Selector for TX0 pin */ -#define AK4114_OPS00 (1<<0) /* Output Though Data Selector for TX0 pin */ +#define AK4114_OPS02 (1<<2) /* Output Data Selector for TX0 pin */ +#define AK4114_OPS01 (1<<1) /* Output Data Selector for TX0 pin */ +#define AK4114_OPS00 (1<<0) /* Output Data Selector for TX0 pin */ /* AK4114_REG_IO1 */ #define AK4114_EFH1 (1<<7) /* Interrupt 0 pin Hold */ -- 1.5.6.3

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[alsa-devel] [PATCH v2] use set_channel_map api to reorder channels for AD1938 and AD1836
by Barry Song 15 Sep '09

15 Sep '09

From: Barry Song <barry.song(a)analog.com> Signed-off-by: Barry Song <barry.song(a)analog.com> --- sound/soc/blackfin/bf5xx-ad1836.c | 7 +++++ sound/soc/blackfin/bf5xx-ad1938.c | 9 ++++++- sound/soc/blackfin/bf5xx-tdm-pcm.c | 9 ++++-- sound/soc/blackfin/bf5xx-tdm.c | 45 +++++++++++++++++++++++++++++------ sound/soc/blackfin/bf5xx-tdm.h | 11 ++++++++ 5 files changed, 69 insertions(+), 12 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c index cd361e3..0f45a3f 100644 --- a/sound/soc/blackfin/bf5xx-ad1836.c +++ b/sound/soc/blackfin/bf5xx-ad1836.c @@ -52,6 +52,7 @@ static int bf5xx_ad1836_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + unsigned int channel_map[] = {0, 4, 1, 5, 2, 6, 3, 7}; int ret = 0; /* set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | @@ -65,6 +66,12 @@ static int bf5xx_ad1836_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; + /* set cpu DAI channel mapping */ + ret = snd_soc_dai_set_channel_map(cpu_dai, ARRAY_SIZE(channel_map), + channel_map, ARRAY_SIZE(channel_map), channel_map); + if (ret < 0) + return ret; + return 0; } diff --git a/sound/soc/blackfin/bf5xx-ad1938.c b/sound/soc/blackfin/bf5xx-ad1938.c index 08269e9..2ef1e50 100644 --- a/sound/soc/blackfin/bf5xx-ad1938.c +++ b/sound/soc/blackfin/bf5xx-ad1938.c @@ -61,6 +61,7 @@ static int bf5xx_ad1938_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + unsigned int channel_map[] = {0, 1, 2, 3, 4, 5, 6, 7}; int ret = 0; /* set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | @@ -75,7 +76,13 @@ static int bf5xx_ad1938_hw_params(struct snd_pcm_substream *substream, return ret; /* set codec DAI slots, 8 channels, all channels are enabled */ - ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xFF, 8); + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xFF, 0xFF, 8, 32); + if (ret < 0) + return ret; + + /* set cpu DAI channel mapping */ + ret = snd_soc_dai_set_channel_map(cpu_dai, ARRAY_SIZE(channel_map), + channel_map, ARRAY_SIZE(channel_map), channel_map); if (ret < 0) return ret; diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.c b/sound/soc/blackfin/bf5xx-tdm-pcm.c index ccb5e82..a8c73cb 100644 --- a/sound/soc/blackfin/bf5xx-tdm-pcm.c +++ b/sound/soc/blackfin/bf5xx-tdm-pcm.c @@ -43,7 +43,7 @@ #include "bf5xx-tdm.h" #include "bf5xx-sport.h" -#define PCM_BUFFER_MAX 0x10000 +#define PCM_BUFFER_MAX 0x8000 #define FRAGMENT_SIZE_MIN (4*1024) #define FRAGMENTS_MIN 2 #define FRAGMENTS_MAX 32 @@ -177,6 +177,9 @@ out: static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel, snd_pcm_uframes_t pos, void *buf, snd_pcm_uframes_t count) { + struct snd_pcm_runtime *runtime = substream->runtime; + struct sport_device *sport = runtime->private_data; + struct bf5xx_tdm_port *tdm_port = sport->private_data; unsigned int *src; unsigned int *dst; int i; @@ -188,7 +191,7 @@ static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel, dst += pos * 8; while (count--) { for (i = 0; i < substream->runtime->channels; i++) - *(dst + i) = *src++; + *(dst + tdm_port->tx_map[i]) = *src++; dst += 8; } } else { @@ -198,7 +201,7 @@ static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel, src += pos * 8; while (count--) { for (i = 0; i < substream->runtime->channels; i++) - *dst++ = *(src+i); + *dst++ = *(src + tdm_port->rx_map[i]); src += 8; } } diff --git a/sound/soc/blackfin/bf5xx-tdm.c b/sound/soc/blackfin/bf5xx-tdm.c index 3096bad..600987d 100644 --- a/sound/soc/blackfin/bf5xx-tdm.c +++ b/sound/soc/blackfin/bf5xx-tdm.c @@ -46,14 +46,6 @@ #include "bf5xx-sport.h" #include "bf5xx-tdm.h" -struct bf5xx_tdm_port { - u16 tcr1; - u16 rcr1; - u16 tcr2; - u16 rcr2; - int configured; -}; - static struct bf5xx_tdm_port bf5xx_tdm; static int sport_num = CONFIG_SND_BF5XX_SPORT_NUM; @@ -181,6 +173,40 @@ static void bf5xx_tdm_shutdown(struct snd_pcm_substream *substream, bf5xx_tdm.configured = 0; } +static int bf5xx_tdm_set_channel_map(struct snd_soc_dai *dai, + unsigned int tx_num, unsigned int *tx_slot, + unsigned int rx_num, unsigned int *rx_slot) +{ + int i; + unsigned int slot; + unsigned int tx_mapped = 0, rx_mapped = 0; + + if ((tx_num > BFIN_TDM_DAI_MAX_SLOTS) || + (rx_num > BFIN_TDM_DAI_MAX_SLOTS)) + return -EINVAL; + + for (i = 0; i < tx_num; i++) { + slot = tx_slot[i]; + if ((slot < BFIN_TDM_DAI_MAX_SLOTS) && + (!(tx_mapped & (1 << slot)))) { + bf5xx_tdm.tx_map[i] = slot; + tx_mapped |= 1 << slot; + } else + return -EINVAL; + } + for (i = 0; i < rx_num; i++) { + slot = rx_slot[i]; + if ((slot < BFIN_TDM_DAI_MAX_SLOTS) && + (!(rx_mapped & (1 << slot)))) { + bf5xx_tdm.rx_map[i] = slot; + rx_mapped |= 1 << slot; + } else + return -EINVAL; + } + + return 0; +} + #ifdef CONFIG_PM static int bf5xx_tdm_suspend(struct snd_soc_dai *dai) { @@ -235,6 +261,7 @@ static struct snd_soc_dai_ops bf5xx_tdm_dai_ops = { .hw_params = bf5xx_tdm_hw_params, .set_fmt = bf5xx_tdm_set_dai_fmt, .shutdown = bf5xx_tdm_shutdown, + .set_channel_map = bf5xx_tdm_set_channel_map, }; struct snd_soc_dai bf5xx_tdm_dai = { @@ -300,6 +327,8 @@ static int __devinit bfin_tdm_probe(struct platform_device *pdev) pr_err("Failed to register DAI: %d\n", ret); goto sport_config_err; } + + sport_handle->private_data = &bf5xx_tdm; return 0; sport_config_err: diff --git a/sound/soc/blackfin/bf5xx-tdm.h b/sound/soc/blackfin/bf5xx-tdm.h index 618ec3d..04189a1 100644 --- a/sound/soc/blackfin/bf5xx-tdm.h +++ b/sound/soc/blackfin/bf5xx-tdm.h @@ -9,6 +9,17 @@ #ifndef _BF5XX_TDM_H #define _BF5XX_TDM_H +#define BFIN_TDM_DAI_MAX_SLOTS 8 +struct bf5xx_tdm_port { + u16 tcr1; + u16 rcr1; + u16 tcr2; + u16 rcr2; + unsigned int tx_map[BFIN_TDM_DAI_MAX_SLOTS]; + unsigned int rx_map[BFIN_TDM_DAI_MAX_SLOTS]; + int configured; +}; + extern struct snd_soc_dai bf5xx_tdm_dai; #endif -- 1.5.6.3

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[alsa-devel] [PATCH] alsa-lib: mixer/simple_none: fix softfloat compilation
by Manuel Lauss 15 Sep '09

15 Sep '09

mixer/simple_none.c uses HAVE_SOFT_FLOAT tests to work around sqrt() uses but the definition in config.h is never picked up, leading to link errors in libasound.so when built with --with-softfloat. Signed-off-by: Manuel Lauss <manuel.lauss(a)gmail.com> --- Tested on MIPS. The pcm_ladspa plugin still calls 'sqrt()' and should be excluded from the build if --with-softfloat is specified. src/mixer/mixer_simple.h | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/src/mixer/mixer_simple.h b/src/mixer/mixer_simple.h index e88b007..bc9d20c 100644 --- a/src/mixer/mixer_simple.h +++ b/src/mixer/mixer_simple.h @@ -19,6 +19,7 @@ * */ +#include <config.h> #include "mixer_abst.h" /* make local functions really local */ -- 1.6.5.rc1

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[alsa-devel] [PATCH 1/2] ak4113 build support
by pavel.hofman＠ivitera.com 15 Sep '09

15 Sep '09

From: Pavel Hofman <pavel.hofman(a)ivitera.com> Signed-off-by: Pavel Hofman <pavel.hofman(a)ivitera.com> diff --git a/i2c/other/Makefile b/i2c/other/Makefile index 37f1603..a86990e 100644 --- a/i2c/other/Makefile +++ b/i2c/other/Makefile @@ -5,7 +5,7 @@ endif include $(SND_TOPDIR)/toplevel.config include $(SND_TOPDIR)/Makefile.conf -export-objs += ak4xxx-adda.o ak4114.o ak4117.o pt2258.o tea575x-tuner.o +export-objs += ak4xxx-adda.o ak4113.o ak4114.o ak4117.o pt2258.o tea575x-tuner.o include $(SND_TOPDIR)/alsa-kernel/i2c/other/Makefile diff --git a/i2c/other/ak4113.c b/i2c/other/ak4113.c new file mode 100644 index 0000000..41cb4ce --- /dev/null +++ b/i2c/other/ak4113.c @@ -0,0 +1,2 @@ +#include "adriver.h" +#include "../../alsa-kernel/i2c/other/ak4113.c" diff --git a/utils/insert b/utils/insert index af911d1..bfbbb83 100755 --- a/utils/insert +++ b/utils/insert @@ -395,6 +395,7 @@ case "$1" in #insert snd-i2c$SUFF #insert snd-cs8427$SUFF insert snd-ak4114$SUFF + insert snd-ak4113$SUFF insert snd-ak4xxx-adda$SUFF insert snd-ice17xx-ak4xxx$SUFF insert snd-ice1724$SUFF enable=1,1 -- 1.5.6.3

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[alsa-devel] Support for Infrasonic Quartet
by Pavel Hofman 15 Sep '09

15 Sep '09

Hi Takashi, I am sending patches for Infrasonic Quartet. The card has three external clock modes therefore I had to modify ice1724.c to support multiple spdif_master types. While working on the driver, I did some minor fixes to ak4xxx-adda.c too. For proc files I added a conversion function from int to string representing its binary format (quartet.c:get_binary). If you know of an existing utility, please let me know and I will use it instead. The binary output helps a lot with debugging. The quartet patch does not comply with checkpatch.pl in 2.6.27 since the script throws a false positive for [] in #define. Thanks a lot for comments. Best regards, Pavel.

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[alsa-devel] [PATCH] RFC: ASoC: Davinci: Fix data_type divide by zero
by Troy Kisky 15 Sep '09

15 Sep '09

This is a proposed fix for the data_type divide by 0 error. Signed-off-by: Troy Kisky <troy.kisky(a)boundarydevices.com> --- This applies on top of my last patchset to Kevin's temp/asoc branch sound/soc/davinci/davinci-i2s.c | 18 +++++++----------- sound/soc/davinci/davinci-mcasp.h | 7 ++++++- sound/soc/davinci/davinci-pcm.c | 14 ++++++-------- 3 files changed, 19 insertions(+), 20 deletions(-) diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 5f61016..252c3ea 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -123,13 +123,18 @@ static struct davinci_pcm_dma_params davinci_i2s_pcm_in = { }; struct davinci_mcbsp_dev { + /* + * dma_params must be first because rtd->dai->cpu_dai->private_data + * is cast to 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 #define MOD_DSP_B 1 int mode; u32 pcr; struct clk *clk; - struct davinci_pcm_dma_params *dma_params[2]; /* * Combining both channels into 1 element will at least double the * amount of time between servicing the dma channel, increase @@ -255,14 +260,6 @@ static void davinci_mcbsp_stop(struct davinci_mcbsp_dev *dev, int playback) toggle_clock(dev, playback); } -static int davinci_i2s_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *cpu_dai) -{ - struct davinci_mcbsp_dev *dev = cpu_dai->private_data; - cpu_dai->dma_data = dev->dma_params[substream->stream]; - return 0; -} - #define DEFAULT_BITPERSAMPLE 16 static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, @@ -393,7 +390,6 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct davinci_pcm_dma_params *dma_params = dai->dma_data; struct davinci_mcbsp_dev *dev = dai->private_data; struct snd_soc_dai *codec_dai = dev->codec_dai; struct snd_interval *i = NULL; @@ -407,6 +403,7 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, int right_first = 0; int convert_mono_stereo = 0; int playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? 1 : 0; + struct davinci_pcm_dma_params *dma_params = dev->dma_params[substream->stream]; i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS); bits_per_sample = snd_interval_value(i); @@ -529,7 +526,6 @@ static void davinci_i2s_shutdown(struct snd_pcm_substream *substream, #define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 static struct snd_soc_dai_ops davinci_i2s_dai_ops = { - .startup = davinci_i2s_startup, .shutdown = davinci_i2s_shutdown, .prepare = davinci_i2s_prepare, .trigger = davinci_i2s_trigger, diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 554354c..c6ea04d 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -39,10 +39,15 @@ enum { }; struct davinci_audio_dev { + /* + * dma_params must be first because rtd->dai->cpu_dai->private_data + * is cast to 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; struct clk *clk; - struct davinci_pcm_dma_params *dma_params[2]; unsigned int codec_fmt; /* McASP specific data */ diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 2729a80..0bd3b2e 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -265,8 +265,7 @@ static int ping_pong_dma_setup(struct snd_pcm_substream *substream) struct davinci_runtime_data *prtd = runtime->private_data; struct snd_dma_buffer *iram_dma = (struct snd_dma_buffer *)substream->dma_buffer.private_data; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct davinci_pcm_dma_params *dma_data = rtd->dai->cpu_dai->dma_data; + struct davinci_pcm_dma_params *dma_data = prtd->params; unsigned int data_type = dma_data->data_type; unsigned int acnt = dma_data->acnt; unsigned int convert_mono_stereo = dma_data->convert_mono_stereo; @@ -482,15 +481,12 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) { struct snd_dma_buffer *iram_dma; struct davinci_runtime_data *prtd = substream->runtime->private_data; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct davinci_pcm_dma_params *dma_data = rtd->dai->cpu_dai->dma_data; + struct davinci_pcm_dma_params *dma_data = prtd->params; int link; if (!dma_data) return -ENODEV; - prtd->params = dma_data; - /* Request asp master DMA channel */ link = prtd->asp_channel = edma_alloc_channel(dma_data->channel, davinci_pcm_dma_irq, substream, EVENTQ_0); @@ -660,7 +656,8 @@ davinci_pcm_pointer(struct snd_pcm_substream *substream) static int davinci_pcm_open(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct davinci_pcm_dma_params *dma_data = rtd->dai->cpu_dai->dma_data; + struct davinci_pcm_dma_params **pa = rtd->dai->cpu_dai->private_data; + struct davinci_pcm_dma_params *dma_params = pa[substream->stream]; struct snd_pcm_runtime *runtime = substream->runtime; struct davinci_runtime_data *prtd; struct snd_pcm_hardware *ppcm; @@ -668,7 +665,7 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream) ppcm = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? &pcm_hardware_playback : &pcm_hardware_capture; - allocate_sram(substream, dma_data->sram_size, ppcm); + allocate_sram(substream, dma_params->sram_size, ppcm); snd_soc_set_runtime_hwparams(substream, ppcm); /* ensure that buffer size is a multiple of period size */ ret = snd_pcm_hw_constraint_integer(runtime, @@ -681,6 +678,7 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream) return -ENOMEM; spin_lock_init(&prtd->lock); + prtd->params = dma_params; prtd->asp_channel = -1; prtd->asp_link[0] = prtd->asp_link[1] = -1; prtd->ram_channel = -1; -- 1.5.6.3

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[alsa-devel] SPDIF with X-Fi
by Daniel L. Miller 15 Sep '09

15 Sep '09

Perhaps I'm going about this the wrong way (if so, please tell me), but... My X-Fi card is now working with the new driver. I am connected to my amplifier via a fiber optic cable. Sending standard two-channel audio to the "front" alsa plug gets directed to the spdif output and sounds great. But I want to have my full surround active - particularly for movies already encoded with dts and/or ac-3. Thus far, I have not been able to get mplayer to send digital audio - I either get mplayer errors or silence. Once, and only once, I got xbmc to work with digital settings - never since. So I know it SHOULD be possible - I'm just missing something. Is anyone getting this to work? -- Daniel

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[alsa-devel] iecset with X-Fi
by Daniel L. Miller 15 Sep '09

15 Sep '09

Is the iecset utility supposed to do anything with X-Fi cards and the new driver? For me, it reports status - but seems to ignore any parameters to change the settings. -- Daniel

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