November 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: bells: Up clock rate to 48kHz and 512fs
by Dimitris Papastamos 02 Nov '12

02 Nov '12

The current WM0010 firmware images expect a 48kHz * 512fs clkin. Change-Id: I46f3f030175ebcbe9be75d01482379021510f459 Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/bells.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/bells.c b/sound/soc/samsung/bells.c index 3d7ace7..c8ac240 100644 --- a/sound/soc/samsung/bells.c +++ b/sound/soc/samsung/bells.c @@ -27,8 +27,8 @@ */ #define MCLK_RATE 24576000 -#define SYS_AUDIO_RATE 44100 -#define SYS_MCLK_RATE (SYS_AUDIO_RATE * 256) +#define SYS_AUDIO_RATE 48000 +#define SYS_MCLK_RATE (SYS_AUDIO_RATE * 512) #define DAI_AP_DSP 0 #define DAI_DSP_CODEC 1 -- 1.8.0

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[alsa-devel] hda: Cirrus: fixes for CS4207 coef setup
by Alexander Stein 02 Nov '12

02 Nov '12

Hi, I'm noticing some problems using the CS4207 codec with out custom board. Output works fine so far, but I noticed that input on ADC1 (blue) is just mono and input on ADC2 (pink) is reversed stereo. Both tests were done by playing a wav on an output and directly feeding the input. After some investigations, I found out, that coef in the patch_cirrus driver don't match the CS4207 datasheet (CS4207-F4). As the driver also mentions CS4206 I'm aware that my pathces might not be suitable in general, but I couldn't find any datasheet for CS4206 so I can't compare the coef settings. Summarized here: * The beep bits were written to the wrong index (See CS4207-F4 6.15.6.5 BeepConfiguration(CIR=0004h)) * The DMIC enable signals were written to the wrong index (Same chapter as above) As the drivers sets 2 DMIC bits each, while CS4207 has only 1 for each, the coef interface might be different on both codec. Maybe someone with more detailed knowledge about the CS4206 can put his comments. Best regards, Alexander

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[alsa-devel] limitations on number of alsa controls for a card
by Vinod Koul 02 Nov '12

02 Nov '12

Hi, Is there any limitation on maximum number of controls we can have. I had some platform side controls to represent DSP and adding them made some of codec controls to vanish. And looks like if I register these platform controls not all codec controls are visible in userpsace. Is there nay max limit I am hitting? -- ~Vinod

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[alsa-devel] Transmitting ac3 content via spdif
by Fabio Estevam 02 Nov '12

02 Nov '12

Hi, I am trying to transmit ac3 content via spdif on a mx6 (on a yet non-mainlined driver). Tried to force non-audio mode via iecset: # iecset audio 0 control "IEC958 Playback Default" (index -1) not found root@imx6qsabrelite:~# aplay -l **** List of PLAYBACK Hardware Devices **** card 0: sgtl5000audio [sgtl5000-audio], device 0: HiFi sgtl5000-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: imxhdmisoc [imx-hdmi-soc], device 0: IMX HDMI TX mxc-hdmi-soc-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 Transmitting a wav file works fine though. Any suggestions as to how provide the "IEC958 Playback Default" control? Thanks, Fabio Estevam

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[alsa-devel] [PATCH RFT] ASoC: si476x: Add missing break for SNDRV_PCM_FORMAT_S8 switch case
by Axel Lin 01 Nov '12

01 Nov '12

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/si476x.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c index 38145ba..dadf04d 100644 --- a/sound/soc/codecs/si476x.c +++ b/sound/soc/codecs/si476x.c @@ -159,6 +159,7 @@ static int si476x_codec_hw_params(struct snd_pcm_substream *substream, switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: width = SI476X_PCM_FORMAT_S8; + break; case SNDRV_PCM_FORMAT_S16_LE: width = SI476X_PCM_FORMAT_S16_LE; break; -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: mxs-saif: Fix channel swap for 24-bit format
by Fabio Estevam 01 Nov '12

01 Nov '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> Playing 24-bit format file leads to channel swap on mx28 and the reason is that the current driver performs one write/read to/from the SAIF_DATA register to trigger the transfer. This approach works fine for S16_LE case because SAIF_DATA is a 32-bit register and thus is capable of storing the 16-bit left and right channels, but for the S24_LE case it can only store one channel, so in order to not lose the FIFO sync an extra read/write is needed. Reported-by: Dan Winner <DWinner(a)tc-helicon.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> Tested-by: Dan Winner <DWinner(a)tc-helicon.com> --- sound/soc/mxs/mxs-saif.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index aa037b2..05a875c 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -523,16 +523,22 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd, if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { /* - * write a data to saif data register to trigger - * the transfer + * write data to saif data register to trigger + * the transfer. + * For 24-bit format the 32-bit FIFO register stores + * only one channel, so we need to write twice. */ __raw_writel(0, saif->base + SAIF_DATA); + __raw_writel(0, saif->base + SAIF_DATA); } else { /* - * read a data from saif data register to trigger - * the receive + * read data from saif data register to trigger + * the receive. + * For 24-bit format the 32-bit FIFO register stores + * only one channel, so we need to read twice. */ __raw_readl(saif->base + SAIF_DATA); + __raw_readl(saif->base + SAIF_DATA); } master_saif->ongoing = 1; -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: atmel_ssc_dai: make module reloadable
by Alexandre Belloni 01 Nov '12

01 Nov '12

As the platform_devices were never unregistered, it was not possible to unload/reload the module. Signed-off-by: Alexandre Belloni <alexandre.belloni(a)piout.net> --- sound/soc/atmel/atmel_ssc_dai.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 354341e..07ee57b 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -155,6 +155,8 @@ static struct atmel_ssc_info ssc_info[NUM_SSC_DEVICES] = { #endif }; +static struct platform_device *dma_pdev; +static struct platform_device *ssc_pdevs[NUM_SSC_DEVICES]; /* * SSC interrupt handler. Passes PDC interrupts to the DMA @@ -796,6 +798,12 @@ static __devinit int asoc_ssc_probe(struct platform_device *pdev) static int __devexit asoc_ssc_remove(struct platform_device *pdev) { + int i; + platform_device_unregister(dma_pdev); + + for(i = 0; i < NUM_SSC_DEVICES; i++) + platform_device_unregister(ssc_pdevs[i]); + snd_soc_unregister_dai(&pdev->dev); return 0; } @@ -816,7 +824,6 @@ static struct platform_driver asoc_ssc_driver = { int atmel_ssc_set_audio(int ssc_id) { struct ssc_device *ssc; - static struct platform_device *dma_pdev; struct platform_device *ssc_pdev; int ret; @@ -855,6 +862,8 @@ int atmel_ssc_set_audio(int ssc_id) if (ret < 0) platform_device_put(ssc_pdev); + ssc_pdevs[ssc_id] = ssc_pdev; + return ret; } EXPORT_SYMBOL_GPL(atmel_ssc_set_audio); -- 1.7.9.5

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[alsa-devel] Issue when upgrading from 1.0.22 to 1.0.25
by Nick Harezga 01 Nov '12

01 Nov '12

I am attempting to upgrade from Ubuntu version 10.04 to version 12.04. Version 10.04 has ALSA version 1.0.22 and version 12.04 has ALSA 1.0.25. I have a simple test application that captures microphone input and then plays the audio through the speaker. When run with ALSA 1.0.22 the program works as expected. When I run the program with ALSA 1.0.25, I get a bad pipe error indicating buffer underrun when calling writes. Both systems have the same hardware (Dell Latitude E6400 laptop). I tried looking through the change logs to determine if there were any obvious changes that stood out to me, but couldn't find anything. I have attached my code. Any suggestions? Program is compiled using "g++ audio_test.cpp -o audio_test -lasound" Audio sample rate: 8000 Buffer size: 1800 Format: Signed 16bit LE, interleaved

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[alsa-devel] [PATCH v3 1/2] ASoC: mxs-saif: Remove platform data
by Fabio Estevam 01 Nov '12

01 Nov '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> All MXS users have been converted to device tree and the board files have been removed. No need to keep platform data in the driver. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v2: - Also remove 'if (np)' as dt is the only probing mechanism. Changes since v1: - Also remove mxs_saif_platform_data and sound/saif.h, which was used only to include mxs_saif_platform_data definition. include/sound/saif.h | 16 ---------------- sound/soc/mxs/mxs-saif.c | 42 +++++++++++++++--------------------------- 2 files changed, 15 insertions(+), 43 deletions(-) delete mode 100644 include/sound/saif.h diff --git a/include/sound/saif.h b/include/sound/saif.h deleted file mode 100644 index f22f3e1..0000000 diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index aa037b2..ab5bdec 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -32,7 +32,6 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> -#include <sound/saif.h> #include <asm/mach-types.h> #include <mach/hardware.h> #include <mach/mxs.h> @@ -652,10 +651,9 @@ static int __devinit mxs_saif_probe(struct platform_device *pdev) struct device_node *np = pdev->dev.of_node; struct resource *iores, *dmares; struct mxs_saif *saif; - struct mxs_saif_platform_data *pdata; struct pinctrl *pinctrl; int ret = 0; - + struct device_node *master; if (!np && pdev->id >= ARRAY_SIZE(mxs_saif)) return -EINVAL; @@ -664,31 +662,21 @@ static int __devinit mxs_saif_probe(struct platform_device *pdev) if (!saif) return -ENOMEM; - if (np) { - struct device_node *master; - saif->id = of_alias_get_id(np, "saif"); - if (saif->id < 0) - return saif->id; - /* - * If there is no "fsl,saif-master" phandle, it's a saif - * master. Otherwise, it's a slave and its phandle points - * to the master. - */ - master = of_parse_phandle(np, "fsl,saif-master", 0); - if (!master) { - saif->master_id = saif->id; - } else { - saif->master_id = of_alias_get_id(master, "saif"); - if (saif->master_id < 0) - return saif->master_id; - } + saif->id = of_alias_get_id(np, "saif"); + if (saif->id < 0) + return saif->id; + /* + * If there is no "fsl,saif-master" phandle, it's a saif + * master. Otherwise, it's a slave and its phandle points + * to the master. + */ + master = of_parse_phandle(np, "fsl,saif-master", 0); + if (!master) { + saif->master_id = saif->id; } else { - saif->id = pdev->id; - pdata = pdev->dev.platform_data; - if (pdata && !pdata->master_mode) - saif->master_id = pdata->master_id; - else - saif->master_id = saif->id; + saif->master_id = of_alias_get_id(master, "saif"); + if (saif->master_id < 0) + return saif->master_id; } if (saif->master_id < 0 || saif->master_id >= ARRAY_SIZE(mxs_saif)) { -- 1.7.9.5

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[alsa-devel] macmini5,3 and imac27 ssid conflict
by Raymond Yau 01 Nov '12

01 Nov '12

>>> http://www.alsa-project.org/db/?f=5e7f45089ce82985c003fa2608bed82726ebbdf6 > > how many audio jacks do your macmini have ? > > > > seem only line in support jack detect, internal mic support balanced but > > ext mic does not support jack detect > > There are two audio jacks: > > 1. An analogue headset/external speakers jack. Note that in order for the external speakers to work, I need to use HDA Analyser to change the settings otherwise things remain stuck on the internal speaker. I did not try an analogue headset. https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/892697 > > 2. An analogue line-in jack. I tried this with a powered condenser microphone - I can hear a recording, but the level is too low for practical use. > > I'm not sure there is an internal microphone connected - shouting at the mac mini box does not produce an audible recording :P http://git.kernel.org/?p=linux/kernel/git/tiwai/sound.git;a=commitdiff;h=6d… Advanced information - PCI Vendor/Device/Subsystem ID's !!------------------------------------------------------- 00:1b.0 0403: 8086:1c20 (rev 05) Subsystem: 8086:7270 your macmini pci subsystem is also 8086:7270 which is same as imac27 the pin default of your macmini (init_pin_config) is replaced by the pin config of model imac27 (driver_pin_config) Codec: Cirrus Logic CS4206 Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x10134206 Subsystem Id: 0x106b2100 is the subsystem id in codec unique ? Sysfs Files !!----------- /sys/class/sound/hwC0D0/init_pin_configs: 0x09 0x400000f0 0x0a 0x002b4040 0x0b 0x90100130 0x0c 0x018b3020 0x0d 0x90a00110 0x0e 0x400000f0 0x0f 0x01cbe030 0x10 0x014be060 0x12 0x01ab9070 0x15 0x400000f0 /sys/class/sound/hwC0D0/driver_pin_configs: 0x09 0x012b4050 0x0a 0x90100140 0x0b 0x90100142 0x0c 0x018b3020 0x0d 0x90a00110 0x0e 0x400000f0 0x0f 0x01cbe030 0x10 0x014be060 0x12 0x01ab9070 0x15 0x400000f0 /sys/class/sound/hwC0D0/user_pin_configs: /sys/class/sound/hwC0D0/init_verbs:

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