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[alsa-devel] ASoC: about the array to cache registers
by Joonyoung Shim 24 Jul '09

24 Jul '09

Hi, Many ASoC codec drivers have an array to cache registers. The array is not used any longer after memcpy to codec->reg_cache. I think that memcpy is unnecessary, and it a waste of the memory. How about continue using the array instead of memcpy? For example, i modified such following patch in wm8731.c diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 1560028..43f8de3 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -36,7 +36,6 @@ struct snd_soc_codec_device soc_codec_dev_wm8731; /* codec private data */ struct wm8731_priv { struct snd_soc_codec codec; - u16 reg_cache[WM8731_CACHEREGNUM]; unsigned int sysclk; }; @@ -50,7 +49,7 @@ static int wm8731_spi_write(struct spi_device *spi, const char *data, int len); * using 2 wire for device control, so we cache them instead. * There is no point in caching the reset register */ -static const u16 wm8731_reg[WM8731_CACHEREGNUM] = { +static u16 wm8731_reg[WM8731_CACHEREGNUM] = { 0x0097, 0x0097, 0x0079, 0x0079, 0x000a, 0x0008, 0x009f, 0x000a, 0x0000, 0x0000 @@ -586,9 +585,7 @@ static int wm8731_register(struct wm8731_priv *wm8731) codec->dai = &wm8731_dai; codec->num_dai = 1; codec->reg_cache_size = WM8731_CACHEREGNUM; - codec->reg_cache = &wm8731->reg_cache; - - memcpy(codec->reg_cache, wm8731_reg, sizeof(wm8731_reg)); + codec->reg_cache = &wm8731_reg; ret = wm8731_reset(codec); if (ret < 0) {

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[alsa-devel] get two mono channel from stereo line
by arif setiawan 24 Jul '09

24 Jul '09

Hi, We have a beagleboard with one audio line-in and one audio line-out. From the documentation both are stereo line. Now we want to use four mono audio line for our system. two input from radio and microphone, two output to radio and speaker. We use ALSA and we would like to know how we can achieve this either by plugin or programming alsa. Thank you for your help Arif

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[alsa-devel] Is snd_pcm_hw_params_set_period_time_near the same as sampling period?
by Guilherme Longo 24 Jul '09

24 Jul '09

Hi mates. I am trying to understand a piece of a code I am studing . What does exactly the functions snd_pcm_hw_params_set_period_time_near sets? In my understand period time should be the time between 2 consecutive samples, but the following code does not appears to be calculating that. buffer_time = 500000 // What I believe that means 0.5s as US is the unit used. if (period_time == 0 && period_frames == 0) { if (buffer_time > 0) period_time = buffer_time / 4; else period_frames = buffer_frames / 4; } Is this period time the same as period describled in here: http://www.fon.hum.uva.nl/praat/manual/sampling_period.html Thanks!

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[alsa-devel] get two mono channel from stereo line
by arif setiawan 24 Jul '09

24 Jul '09

Hi, We have a beagleboard with one audio line-in and one audio line-out. From the documentation both are stereo line. Now we want to use four mono audio line for our system. two input from radio and microphone, two output to radio and speaker. We use ALSA and we would like to know how we can achieve this either by plugin or programming alsa. Thank you for your help Arif

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[alsa-devel] multi-channel playback regression
by Wu Fengguang 24 Jul '09

24 Jul '09

Hi Takashi, When doing multi-channel playback tests on IbexPeak, I found that the following patch makes the playback enter an infinite loop, repeatedly playing a range of ~0.5s audio content. (Seems that some buffer pointer can never advance.) Thanks, Fengguang --- 79452f0a28aa5a40522c487b42a5fc423647ad98 Author: Takashi Iwai <tiwai(a)suse.de> Date: Wed Jul 22 12:51:51 2009 +0200 ALSA: pcm - Fix regressions with VMware VMware tends to report PCM positions and period updates at utterly wrong timing. This screws up the recent PCM core code that tries to correct the position based on the irq timing. Now, when a backward irq position is detected, skip the update instead of rebasing. (This is almost the old behavior before 2.6.30.) Signed-off-by: Takashi Iwai <tiwai(a)suse.de> diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 333e4dd..3b673e2 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -244,18 +244,27 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) delta = new_hw_ptr - hw_ptr_interrupt; } if (delta < 0) { - delta += runtime->buffer_size; + if (runtime->periods == 1) + delta += runtime->buffer_size; if (delta < 0) { hw_ptr_error(substream, "Unexpected hw_pointer value " "(stream=%i, pos=%ld, intr_ptr=%ld)\n", substream->stream, (long)pos, (long)hw_ptr_interrupt); +#if 1 + /* simply skipping the hwptr update seems more + * robust in some cases, e.g. on VMware with + * inaccurate timer source + */ + return 0; /* skip this update */ +#else /* rebase to interrupt position */ hw_base = new_hw_ptr = hw_ptr_interrupt; /* align hw_base to buffer_size */ hw_base -= hw_base % runtime->buffer_size;

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[alsa-devel] snd_pcm_hw_params _set_period_time_near
by Guilherme Longo 24 Jul '09

24 Jul '09

Mates, I can´t find the real meaning of this function. At first I thought that it was used to set the time between 2 consecutive samples, but it is not. Does anyone could gimme some help with that? Tks

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[alsa-devel] SoC codec read callback definition
by Lopez Cruz, Misael 23 Jul '09

23 Jul '09

read callback (of snd_soc_codec struct) is defined as a function that returns unsigned int: struct snd_soc_codec { ... unsigned int (*read)(struct snd_soc_codec *, unsigned int); int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); ... } However, actual implementation of that callback in many CODEC drivers may return a negative error value (i.e. when a register is out of cache reg size). Should the callback definition be changed to return _int_ instead of _unsigned int_? Some drivers (like wm8900, wm9081) handle invalid registers with BUG_ON, in that case it's not necessary to return the negative error code. What should be the best way to handle this? Or is it ok as currently implemented? Well, it was just a thought. -Misa

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[alsa-devel] [PATCH] ALSA: snd_usb_caiaq: add support for Audio2DJ
by Daniel Mack 23 Jul '09

23 Jul '09

This adds support for Native Instrument's freshly announced Audio2DJ sound device hardware. Version number bumped to 1.3.19. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/Kconfig | 1 + sound/usb/caiaq/audio.c | 1 + sound/usb/caiaq/device.c | 8 +++++++- sound/usb/caiaq/device.h | 1 + 4 files changed, 10 insertions(+), 1 deletions(-) diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index 523aec1..73525c0 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -48,6 +48,7 @@ config SND_USB_CAIAQ * Native Instruments Kore Controller * Native Instruments Kore Controller 2 * Native Instruments Audio Kontrol 1 + * Native Instruments Audio 2 DJ * Native Instruments Audio 4 DJ * Native Instruments Audio 8 DJ * Native Instruments Guitar Rig Session I/O diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index 8f9b60c..121af06 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -646,6 +646,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_GUITARRIGMOBILE): dev->samplerates |= SNDRV_PCM_RATE_192000; /* fall thru */ + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO2DJ): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ): dev->samplerates |= SNDRV_PCM_RATE_88200; diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index de38108..83e6c13 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -35,13 +35,14 @@ #include "input.h" MODULE_AUTHOR("Daniel Mack <daniel(a)caiaq.de>"); -MODULE_DESCRIPTION("caiaq USB audio, version 1.3.18"); +MODULE_DESCRIPTION("caiaq USB audio, version 1.3.19"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, RigKontrol3}," "{Native Instruments, Kore Controller}," "{Native Instruments, Kore Controller 2}," "{Native Instruments, Audio Kontrol 1}," + "{Native Instruments, Audio 2 DJ}," "{Native Instruments, Audio 4 DJ}," "{Native Instruments, Audio 8 DJ}," "{Native Instruments, Session I/O}," @@ -121,6 +122,11 @@ static struct usb_device_id snd_usb_id_table[] = { .idVendor = USB_VID_NATIVEINSTRUMENTS, .idProduct = USB_PID_AUDIO4DJ }, + { + .match_flags = USB_DEVICE_ID_MATCH_DEVICE, + .idVendor = USB_VID_NATIVEINSTRUMENTS, + .idProduct = USB_PID_AUDIO2DJ + }, { /* terminator */ } }; diff --git a/sound/usb/caiaq/device.h b/sound/usb/caiaq/device.h index ece7351..44e3edf 100644 --- a/sound/usb/caiaq/device.h +++ b/sound/usb/caiaq/device.h @@ -10,6 +10,7 @@ #define USB_PID_KORECONTROLLER 0x4711 #define USB_PID_KORECONTROLLER2 0x4712 #define USB_PID_AK1 0x0815 +#define USB_PID_AUDIO2DJ 0x041c #define USB_PID_AUDIO4DJ 0x0839 #define USB_PID_AUDIO8DJ 0x1978 #define USB_PID_SESSIONIO 0x1915 -- 1.6.3.1

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[alsa-devel] [PATCH 4/4] Let wm97xx-battery use irq to detect charger state
by Marek Vasut 23 Jul '09

23 Jul '09

Hi, I split the "[PATCH 3/4] Convert WM97xx driver to use platform_data" patch into three smaller as Mark requested. These three will be: [PATCH 2/4] Make wm97xx-core pass platform_data to battery, ts [PATCH 3/4] Convert WM97xx battery to use platform_data [PATCH 4/4] Let wm97xx-battery use irq to detect charger state

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[alsa-devel] [PATCH 3/4] Convert WM97xx battery to use platform_data
by Marek Vasut 23 Jul '09

23 Jul '09

Hi, I split the "[PATCH 3/4] Convert WM97xx driver to use platform_data" patch into three smaller as Mark requested. These three will be: [PATCH 2/4] Make wm97xx-core pass platform_data to battery, ts [PATCH 3/4] Convert WM97xx battery to use platform_data [PATCH 4/4] Let wm97xx-battery use irq to detect charger state

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