December 2009 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] No device for DAI omap-eac-dai error
by Belisko Marek 21 Dec '09

21 Dec '09

Hi, I'm writing new omap dai (for EAC codec base omap's). My code is based on existing omap-mcbsp.c code basically. After loadingof module I got error which appear in init function when call snd_soc_register_dai(). I browse for code but can't figure out where dai->dev should be set to don't get error message and init dai properly. Thanks for any suggestions Marek

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[alsa-devel] Installing a "different" version of Alsa
by Brian Welch 20 Dec '09

20 Dec '09

Hello, It's been a year since I posted to this list, but I'm running into a problem and can't figure out the best way to solve it, so I'm hoping someone here can help. I've got an SDK for an embedded platform that includes a specific version of Linux with ALSA built-in. On my platform, I want to use a different (newer) version of ALSA than the built-in version, so I want to build and load the binaries from the newer version and, more importantly, I want to compile my custom sound hardware driver against the newer alsa version and the SDK's Linux version. I don't really want to touch the kernel files that are in the kerenl/include/sound directory in the SDK. When I use kbuild to build my custom sound hardware driver against the SDK in the kernel, it always trys to pick up the older ALSA header files that are in the SDK, but I want to use the newer versions that are in the directory with the newer version of ALSA. What the best way to make sure I get the correct headers? Is what I'm trying "dangerous" and should I be going about this a different way? Thanks for any advice, Brian

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[alsa-devel] ASoC updates for 2.6.33
by Mark Brown 19 Dec '09

19 Dec '09

The following changes since commit 1cf86f6f9b000e98c1b7f866f99633ae67464e2f: Kuninori Morimoto (1): ASoC: ak4642: Add default return value in ak4642_modinit 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 Guennadi Liakhovetski (1): ASoC: wm8974: fix a wrong bit definition Jon Smirl (1): ASoC: Fix disable of SPDIF on STAC9766 codec sound/soc/codecs/stac9766.c | 18 +----------------- sound/soc/codecs/wm8974.c | 2 +- 2 files changed, 2 insertions(+), 18 deletions(-)

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[alsa-devel] [PATCH 0/3] McPDM driver patch series
by Candelaria Villareal, Jorge 19 Dec '09

19 Dec '09

McPDM module is the interface between the OMAP and TWL6030 audio codec. It transfers data between the two chips using a Pulse Density Modulation protocol, which allows sound to be played or recorded at speeds of up to 96000 Hz. Besides data, McPDM can transfer commands and read status to/from the codec using the frame synchronization line. 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. -- Jorge Eduardo Candelaria (2): OMAP4: IRQ: Add McPDM IRQ definition. ASoC: OMAP4: Add support for McPDM Misael Lopez Cruz (1): ASoC: OMAP4: Add McPDM platform driver arch/arm/plat-omap/include/plat/irqs.h | 2 +- sound/soc/omap/Kconfig | 3 + sound/soc/omap/Makefile | 2 + sound/soc/omap/mcpdm.c | 505 ++++++++++++++++++++++++++++++++ sound/soc/omap/mcpdm.h | 156 ++++++++++ sound/soc/omap/omap-mcpdm.c | 248 ++++++++++++++++ sound/soc/omap/omap-mcpdm.h | 29 ++ 7 files changed, 944 insertions(+), 1 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 create mode 100644 sound/soc/omap/omap-mcpdm.h

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[alsa-devel] [PATCH 7/7] sound: Use kzalloc for allocating only one thing
by Julia Lawall 19 Dec '09

19 Dec '09

From: Julia Lawall <julia(a)diku.dk> Use kzalloc rather than kcalloc(1,...) The semantic patch that makes this change is as follows: (http://coccinelle.lip6.fr/) // <smpl> @@ @@ - kcalloc(1, + kzalloc( ...) // </smpl> Signed-off-by: Julia Lawall <julia(a)diku.dk> --- sound/isa/msnd/msnd_midi.c | 2 +- sound/pci/hda/patch_realtek.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff -u -p a/sound/isa/msnd/msnd_midi.c b/sound/isa/msnd/msnd_midi.c --- a/sound/isa/msnd/msnd_midi.c 2009-03-28 23:34:55.000000000 +0100 +++ b/sound/isa/msnd/msnd_midi.c 2009-12-19 07:52:27.000000000 +0100 @@ -162,7 +162,7 @@ int snd_msndmidi_new(struct snd_card *ca err = snd_rawmidi_new(card, "MSND-MIDI", device, 1, 1, &rmidi); if (err < 0) return err; - mpu = kcalloc(1, sizeof(*mpu), GFP_KERNEL); + mpu = kzalloc(sizeof(*mpu), GFP_KERNEL); if (mpu == NULL) { snd_device_free(card, rmidi); return -ENOMEM; diff -u -p a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c --- a/sound/pci/hda/patch_realtek.c 2009-12-17 08:30:15.000000000 +0100 +++ b/sound/pci/hda/patch_realtek.c 2009-12-19 07:52:50.000000000 +0100 @@ -12857,7 +12857,7 @@ static int patch_alc268(struct hda_codec int board_config; int i, has_beep, err; - spec = kcalloc(1, sizeof(*spec), GFP_KERNEL); + spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) return -ENOMEM;

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[alsa-devel] [PATCH 0/5] AACI: cleanups
by Russell King - ARM Linux 18 Dec '09

18 Dec '09

This patch series is in preparation of adding DMA support to the AACI driver. I'd appreciate it if ALSA people could do me the curtesy of copying me on patches they dream up on ALSA drivers maintained by me, rather than running rough-shod over work that they may have queued - especially when they say "you can also do this" "yes I know, I have a patch" and then they do it anyway. If this is how they want to operate, I'll avoid working on any ALSA stuff in the future. Its just not worth me wasting my time.

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[alsa-devel] Help needed to develop quirks for USB Sound device for Roland Edirol PC-80 MIDI controller
by Alain Lauzon 18 Dec '09

18 Dec '09

Heloo I just bought a Roland Edirol PC-80 MIDI controller and found out that it was not supported on LINUX. I looked on the web and found the /usr/src/alsa/alsa-driver-xxx/usbquirks.h file to be the place to change some code to support it. I have tried many things but I don't enough info to get it right once and for all. Here is what I did : { /* PC-80 Wave */ USB_DEVICE(0x0582, 0x0092), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { /* .vendor_name = "EDIROL", */ /* .product_name = "PC-80", */ .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 0, .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = 1, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { .format = SNDRV_PCM_FORMAT_S24_3LE, .channels = 2, .iface = 0, .altsetting = 1, .altset_idx = 1, .attributes = 0, .endpoint = 0x81, .ep_attr = 0x01, .rates = SNDRV_PCM_RATE_CONTINUOUS, .rate_min = 48000, .rate_max = 48000, //(96?) } }, { .ifnum = 2, .type = QUIRK_MIDI_FIXED_ENDPOINT, .data = & (const struct snd_usb_midi_endpoint_info) { .out_cables = 0x0007, .in_cables = 0x0007 } }, { .ifnum = -1 } } } What I got working is the audio out but cannot have the audio working. Is there some resources somewhere that I could read to understand what I need to do? Thank you, Alain Lauzon -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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[alsa-devel] ASoC updates for 2.6.34
by Mark Brown 18 Dec '09

18 Dec '09

The following changes since commit bc2580061e42c323d7777029f01318f395edac0d: Julia Lawall (1): ASoC: Correct code taking the size of a pointer are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.34 Nothing terribly exciting here, almost all of it is a bunch of new drivers. Chaithrika U S (1): ASoC: DaVinci: Update suspend/resume support for McASP driver Daniel Mack (1): ASoC: Add regulator support to CS4270 codec driver Guennadi Liakhovetski (1): ASoC: wm8974: fix a wrong bit definition Ilkka Koskinen (2): ASoC: tlv320dac33: Add support for regulator framework ASoC: tpa6130a2: Add support for regulator framework Jassi Brar (2): ASoC: S3C64XX: Remove unnecessary header includes ASoC: S3C64XX: Compress and generalize the CPU driver Jon Smirl (1): ASoC: Fix disable of SPDIF on STAC9766 codec Kuninori Morimoto (4): ASoC: sh_fsi: avoid using global variable ASoC: ak4642: Add default return value in ak4642_modinit ASoC: Add DA7210 codec device support for ALSA ASoC: Add FSI-DA7210 sound support for SuperH Mark Brown (12): Merge branch 'for-2.6.33' into for-2.6.34 ASoC: Export snd_soc_update_bits_unlocked() ASoC: Display the power register in DAPM widget debugfs ASoC: Initial WM8904 CODEC driver ASoC: Push registers out of mixer power decision ASoC: Sort DAPM sequences by CODEC as well ASoC: Register the CODEC in WM8727 ASoC: Fix sorting of codecs Makefile entries ASoC: Fix build of DA7210 ASoC: Add initial WM8955 CODEC driver ASoC: Set AIF word length for WM8904 Merge branch 'for-2.6.33' into for-2.6.34 Peter Meerwald (1): ASoC: PLL computation in TLV320AIC3x SoC driver include/sound/soc.h | 3 + include/sound/wm8904.h | 57 + include/sound/wm8955.h | 26 + sound/soc/codecs/Kconfig | 12 + sound/soc/codecs/Makefile | 10 +- sound/soc/codecs/ak4642.c | 2 +- sound/soc/codecs/cs4270.c | 43 +- sound/soc/codecs/da7210.c | 589 +++++++++ sound/soc/codecs/da7210.h | 24 + sound/soc/codecs/stac9766.c | 18 +- sound/soc/codecs/tlv320aic3x.c | 75 +- sound/soc/codecs/tlv320dac33.c | 92 ++- sound/soc/codecs/tpa6130a2.c | 87 +- sound/soc/codecs/wm8727.c | 66 +- sound/soc/codecs/wm8904.c | 2555 +++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8904.h | 1681 ++++++++++++++++++++++++ sound/soc/codecs/wm8955.c | 1151 +++++++++++++++++ sound/soc/codecs/wm8955.h | 489 +++++++ sound/soc/codecs/wm8974.c | 2 +- sound/soc/davinci/davinci-mcasp.c | 18 +- sound/soc/davinci/davinci-mcasp.h | 1 + sound/soc/davinci/davinci-pcm.c | 2 +- sound/soc/s3c24xx/s3c64xx-i2s.c | 120 +-- sound/soc/sh/Kconfig | 8 + sound/soc/sh/Makefile | 2 + sound/soc/sh/fsi-da7210.c | 83 ++ sound/soc/sh/fsi.c | 115 +- sound/soc/soc-core.c | 7 +- sound/soc/soc-dapm.c | 37 +- 29 files changed, 7127 insertions(+), 248 deletions(-) create mode 100644 include/sound/wm8904.h create mode 100644 include/sound/wm8955.h create mode 100644 sound/soc/codecs/da7210.c create mode 100644 sound/soc/codecs/da7210.h create mode 100644 sound/soc/codecs/wm8904.c create mode 100644 sound/soc/codecs/wm8904.h create mode 100644 sound/soc/codecs/wm8955.c create mode 100644 sound/soc/codecs/wm8955.h create mode 100644 sound/soc/sh/fsi-da7210.c

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[alsa-devel] Play multiple sounds concurrently
by Markus Luttenberger 18 Dec '09

18 Dec '09

Hey, I want to play multiple sounds concurrently with ALSA. The audio data is mono and G.711 a-law encoded. I can successfully play a single source of audio data but the sound is distorted when I try to play two sources. Basically, this is what I'm doing: -----8<----------------- char buf1[1024]; // filled with audio data from source 1 char buf2[1024]; // filled with audio data from source 2 char wr_buf[1024]; for (int count = 0; count < 1024; count++) wr_buf[count] = (char) (0.5 * buf1[count]) + (char) (0.5 * buf2[count]); snd_pcm_writei(dev_handle, wr_buf, 1024); -----8<----------------- I think I have to use a scale value to correct the values or else clipping occures. However, this is the problem. How do I have to modify the audio bytes in order to add them together and write them into one buffer? greetings.. ..Markus

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[alsa-devel] alsa device drain power
by Michael Trimarchi 18 Dec '09

18 Dec '09

Hi all, I have a simple question is the general behavior that an alsa device start to drain power when it starts to play sound or start to recording. What happen if there is an external amplifier connect to sound device? Michael

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