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[alsa-devel] [PATCH] ALSA: hda: Use LPIB for another mainboard
by Daniel Chen 31 May '10

31 May '10

Hi, Attached is a patch against my latest LPIB patches that makes the original reporter's playback and capture audible by default. Please apply. This patch is applicable to all 2.6.3x stable series. Best, -Dan

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[alsa-devel] [PATCH] ALSA: hda: Use mbp31 quirk for an iMac model
by Daniel Chen 31 May '10

31 May '10

Hi, Attached is a patch against sound-2.6 master enabling audible playback and capture for the original reporter's iMac. Please apply. This patch is applicable to all 2.6.3x stable series. Best, -Dan

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[alsa-devel] [PATCH] ALSA: hda: Use LPIB for an ASUS device
by Daniel Chen 31 May '10

31 May '10

Hi, Attached is a patch against sound-2.6 master that enables the original reporter to adjust capture volume through PulseAudio without the daemon crashing. This patch is applicable to all 2.6.3x stable series. Best, -Dan

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[alsa-devel] [PATCH] pcm: Define G723 3-bit and 5-bit formats
by Ben Collins 31 May '10

31 May '10

pcm: Define G723 3-bit and 5-bit formats This defines the 24bps and 40bps (8khz sample rate) G.723 codec formats. They are going to be used once I submit the driver for an mpeg4/g723 compression card. I've updated the signed value to -1 as per Takashi's comments since these are non-linear formats. Signed-off-by: Ben Collins <bcollins(a)bluecherry.net> Acked-by: Takashi Iwai <tiwai(a)suse.de> diff --git a/include/sound/asound.h b/include/sound/asound.h index 0985955..62e0857 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -212,7 +212,11 @@ typedef int __bitwise snd_pcm_format_t; #define SNDRV_PCM_FORMAT_S18_3BE ((__force snd_pcm_format_t) 41) /* in three bytes */ #define SNDRV_PCM_FORMAT_U18_3LE ((__force snd_pcm_format_t) 42) /* in three bytes */ #define SNDRV_PCM_FORMAT_U18_3BE ((__force snd_pcm_format_t) 43) /* in three bytes */ -#define SNDRV_PCM_FORMAT_LAST SNDRV_PCM_FORMAT_U18_3BE +#define SNDRV_PCM_FORMAT_G723_24 ((__force snd_pcm_format_t) 44) /* 8 samples in 3 bytes */ +#define SNDRV_PCM_FORMAT_G723_24_1B ((__force snd_pcm_format_t) 45) /* 1 sample in 1 byte */ +#define SNDRV_PCM_FORMAT_G723_40 ((__force snd_pcm_format_t) 46) /* 8 Samples in 5 bytes */ +#define SNDRV_PCM_FORMAT_G723_40_1B ((__force snd_pcm_format_t) 47) /* 1 sample in 1 byte */ +#define SNDRV_PCM_FORMAT_LAST SNDRV_PCM_FORMAT_G723_40_1B #ifdef SNDRV_LITTLE_ENDIAN #define SNDRV_PCM_FORMAT_S16 SNDRV_PCM_FORMAT_S16_LE diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 8b611a5..37b5c3c 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -173,6 +173,10 @@ struct snd_pcm_ops { #define SNDRV_PCM_FMTBIT_U18_3LE (1ULL << SNDRV_PCM_FORMAT_U18_3LE) #define SNDRV_PCM_FMTBIT_S18_3BE (1ULL << SNDRV_PCM_FORMAT_S18_3BE) #define SNDRV_PCM_FMTBIT_U18_3BE (1ULL << SNDRV_PCM_FORMAT_U18_3BE) +#define SNDRV_PCM_FMTBIT_G723_24 (1ULL << SNDRV_PCM_FORMAT_G723_24) +#define SNDRV_PCM_FMTBIT_G723_24_1B (1ULL << SNDRV_PCM_FORMAT_G723_24_1B) +#define SNDRV_PCM_FMTBIT_G723_40 (1ULL << SNDRV_PCM_FORMAT_G723_40) +#define SNDRV_PCM_FMTBIT_G723_40_1B (1ULL << SNDRV_PCM_FORMAT_G723_40_1B) #ifdef SNDRV_LITTLE_ENDIAN #define SNDRV_PCM_FMTBIT_S16 SNDRV_PCM_FMTBIT_S16_LE diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c index ea2bf82..434af3c 100644 --- a/sound/core/pcm_misc.c +++ b/sound/core/pcm_misc.c @@ -128,6 +128,14 @@ static struct pcm_format_data pcm_formats[SNDRV_PCM_FORMAT_LAST+1] = { .width = 4, .phys = 4, .le = -1, .signd = -1, .silence = {}, }, + [SNDRV_PCM_FORMAT_G723_24] = { + .width = 3, .phys = 3, .le = -1, .signd = -1, + .silence = {}, + }, + [SNDRV_PCM_FORMAT_G723_40] = { + .width = 5, .phys = 5, .le = -1, .signd = -1, + .silence = {}, + }, /* FIXME: the following three formats are not defined properly yet */ [SNDRV_PCM_FORMAT_MPEG] = { .le = -1, .signd = -1, @@ -186,6 +194,14 @@ static struct pcm_format_data pcm_formats[SNDRV_PCM_FORMAT_LAST+1] = { .width = 18, .phys = 24, .le = 0, .signd = 0, .silence = { 0x02, 0x00, 0x00 }, }, + [SNDRV_PCM_FORMAT_G723_24_1B] = { + .width = 3, .phys = 8, .le = -1, .signd = -1, + .silence = {}, + }, + [SNDRV_PCM_FORMAT_G723_40_1B] = { + .width = 5, .phys = 8, .le = -1, .signd = -1, + .silence = {}, + }, }; -- Bluecherry: http://www.bluecherrydvr.com/ SwissDisk : http://www.swissdisk.com/ Ubuntu : http://www.ubuntu.com/ My Blog : http://ben-collins.blogspot.com/

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[alsa-devel] [PATCH 1/4] snd-usb-caiaq: Restore 'Control vinyl' input mode on A4DJ
by Mark Hills 31 May '10

31 May '10

This feature was undocumented on early A4DJ units. It is indicated by lighting both the 'line' and 'phono' lamps at the same time. Newer units document this and the newer Windows drivers enable this for all units, so restore the functionality. This patch simplifies the code and changes the mode mapping to match the A8DJ, favouring simpler code and consistency over keeping the existing mapping. Both 'Control vinyl' and 'Phono' input modes enable the hardware preamp. The difference is the input impedance. This reverts commit 9a9527e. Acked-by: Daniel Mack <daniel(a)caiaq.de> Signed-off-by: Mark Hills <mark(a)pogo.org.uk> --- sound/usb/caiaq/control.c | 30 ++---------------------------- 1 files changed, 2 insertions(+), 28 deletions(-) diff --git a/sound/usb/caiaq/control.c b/sound/usb/caiaq/control.c index 36ed703..70c3866 100644 --- a/sound/usb/caiaq/control.c +++ b/sound/usb/caiaq/control.c @@ -42,21 +42,12 @@ static int control_info(struct snd_kcontrol *kcontrol, switch (dev->chip.usb_id) { case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ): - if (pos == 0) { - /* current input mode of A8DJ */ - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 2; - return 0; - } - break; - case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ): if (pos == 0) { - /* current input mode of A4DJ */ + /* current input mode of A8DJ and A4DJ */ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; + uinfo->value.integer.max = 2; return 0; } break; @@ -86,14 +77,6 @@ static int control_get(struct snd_kcontrol *kcontrol, struct snd_usb_caiaqdev *dev = caiaqdev(chip->card); int pos = kcontrol->private_value; - if (dev->chip.usb_id == - USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ)) { - /* A4DJ has only one control */ - /* do not expose hardware input mode 0 */ - ucontrol->value.integer.value[0] = dev->control_state[0] - 1; - return 0; - } - if (pos & CNT_INTVAL) ucontrol->value.integer.value[0] = dev->control_state[pos & ~CNT_INTVAL]; @@ -113,15 +96,6 @@ static int control_put(struct snd_kcontrol *kcontrol, unsigned char cmd = EP1_CMD_WRITE_IO; switch (dev->chip.usb_id) { - case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ): { - /* A4DJ has only one control */ - /* do not expose hardware input mode 0 */ - dev->control_state[0] = ucontrol->value.integer.value[0] + 1; - snd_usb_caiaq_send_command(dev, EP1_CMD_WRITE_IO, - dev->control_state, sizeof(dev->control_state)); - return 1; - } - case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1): cmd = EP1_CMD_DIMM_LEDS; break; -- 1.7.1

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[alsa-devel] [PATCH] ALSA: hda: Use LPIB for a Shuttle device
by Daniel Chen 31 May '10

31 May '10

Hi, Attached please find a patch against current sound-2.6 master that fixes audio anomalies and PulseAudio crashing by specifying LPIB method explicitly. Please apply. This patch is applicable to all 2.6.3x stable series. Best, -Dan

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[alsa-devel] [PATCH] ALSA: usb/mixer.c: remove unused variable
by Daniel Mack 31 May '10

31 May '10

This was a development left-over. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Clemens Ladisch <clemens(a)ladisch.de> --- sound/usb/mixer.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index fd3e08d..a35dd74 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1970,8 +1970,6 @@ static void snd_usb_mixer_proc_read(struct snd_info_entry *entry, static void snd_usb_mixer_interrupt_v2(struct usb_mixer_interface *mixer, int attribute, int value, int index) { - struct usb_mixer_elem_info *info; - switch (attribute) { case UAC2_CS_CUR: snd_usb_mixer_notify_id(mixer, index); -- 1.7.1

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Re: [alsa-devel] Problem with snd_hda_intel from 05-27-2010 snapshot
by Jaroslav Kysela 30 May '10

30 May '10

On Thu, 27 May 2010, VDR User wrote: > On Thu, May 27, 2010 at 12:37 AM, Jaroslav Kysela <perex(a)perex.cz> wrote: >> Check any suspicious lines in /var/log/messages and follow proc xrun_debug >> settings noted in bug above. I need at least 20 first lines after the >> playback is started. > > By this do you mean compile the drivers with pcm-xrun-debug > --with-debug=verbose --enable-verbose-procfs options? Also, is there > a way to disable all the debug logging in maybe > /etc/modprobe.d/alsa-base.conf? My system is installed on a CF and I > can't have alsa hammering it with writes is they're not needed but the > only way I know to turn the debug logging off is to not compile it in > the first place. I asked for one run, so you can recompile ALSA driver without debugging support after test. I think that the problem might be that the WALLCLK register gives wrong values for your hardware (but the Intel specification says that implementation of this register is MANDATORY). Here is a patch which might show the bug: diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 170610e..a1e7c97 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1891,6 +1891,7 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev) int stream; wallclk = azx_readl(chip, WALLCLK) - azx_dev->start_wallclk; + printk("wallclk: %u, %u\n", wallclk, azx_readl(chip, WALLCLK)); if (wallclk < (azx_dev->period_wallclk * 2) / 3) return -1; /* bogus (too early) interrupt */ >> Also, use 'aplay -Dplughw:0' from the alsa-utils package to trigger the >> playback. > > The hdmi device I'm using is 1,3: > > test:~$ aplay -l > **** List of PLAYBACK Hardware Devices **** > card 0: NVidia [HDA NVidia], device 0: ALC883 Analog [ALC883 Analog] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: NVidia [HDA NVidia], device 1: ALC883 Digital [ALC883 Digital] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 1: NVidia_1 [HDA NVidia], device 3: NVIDIA HDMI [NVIDIA HDMI] > Subdevices: 0/1 > Subdevice #0: subdevice #0 > > So what exactly do I need to do then with aplay? Do I need some file > to play? Sorry, I've never used Linux as a desktop so I'm only > familiar with running VDR from command line. If you use HDMI device, use 'aplay plughw:1,0 <some_wav_file>'. Some_wav_file means a .wav type audio file. Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

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[alsa-devel] How to support software mixing on QUALCOMM 72XX without mmap?
by Berg.Tian 29 May '10

29 May '10

Hello,guys. I'm porting ASLA1.0.23 to qualcomm 72xx base on ARM11. The first instance,"aplay test.wav", works now, but the second instance should fail with "device busy" error message. I have tried plug dmix and dshare "aply -D plug:mycard test.wav", but met a fail again for the soc chip doesn't support mmap. Of course,pcm_mycard is defined in /etc/asound.conf. In my pc ,both above cmds work well . Though,the document shows that: NOTE: For ALSA 1.0.9rc2 and higher you don't need to setup dmix for analogue output. Dmix is enabled by default for soundcards which don't support hardware mixing. You still need to set it up for digital outputs. My questions are: 1)Is software mixing implemented in ALSA Driver,or in ALSA LIB? 2)What could I do to make the software mixing work? Is there any special configure which need to adjust? And when testing on pc whth HD audio, I noticed that A)when aplay instances runs WITHOUT dmix, the result of"cat /etc/one_of_my_aplay_pid/maps" should like: ... b331d000-b731e000 rw-s 00000000 00:15 862346 /dev/shm/pulse-shm-2882420837 ... B)when aplay instances runs WITH dmix,the result is: ... b7511000-b7519000 rw-s 00000000 00:0f 4560 /dev/snd/pcmC0D0p b7519000-b751a000 rw-s 81000000 00:0f 4560 /dev/snd/pcmC0D0p b751a000-b751b000 r--s 80000000 00:0f 4560 /dev/snd/pcmC0D0p ... So,the third question: 3) From A) and B),can I make a result that the mmap feacture is not necessary for the buildin software mixing ? Expect your help,please berg

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Re: [alsa-devel] Mixer volume/decibel mapping
by Clemens Ladisch 29 May '10

29 May '10

Raymond Yau wrote: > 2010/5/21 Clemens Ladisch <clemens(a)ladisch.de> > > DB_LINEAR means that the mixer control value is a multiplier for the > > PCM sample value, as James wrote. A DB_LINEAR mapping has > > 0 == -infinity dB, full scale == 0 dB, and the mapping between mixer > > control values and corresponding dB values it _not_ linear. > > pci/oxygen/oxygen.c:static const DECLARE_TLV_DB_LINEAR(ak4396_db_scale, TLV_DB_GAIN_MUTE, 0); > pci/oxygen/hifier.c:static const DECLARE_TLV_DB_LINEAR(ak4396_db_scale, TLV_DB_GAIN_MUTE, 0); > > Refer to AK4396 datasheet > > Output Volume > > The AK4396 includes channel independent digital output volumes (ATT) with > 256 levels at linear step including MUTE. > These volumes are in front of the DAC and can attenuate the input data from > 0dB to –48dB and mute. > > dB range is -48dB to 0dB only > > ATT7-0: Attenuation Level > ATT = 20 log10 (ATT_DATA / 255) [dB] > FFH : 0dB (Default) > 00H : Mute This is the section of the datasheet on which the above declarations are based. Now if you have some remark or question, you should remember to include it in your mail. Regards, Clemens

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