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

[alsa-devel] [PATCH] alsa: lx6464es - support index module parameter
by Tim Blechmann 11 Jun '09

11 Jun '09

trivial patch to put the device to a specific index Signed-off-by: Tim Blechmann <tim(a)klingt.org> --- sound/pci/lx6464es/lx6464es.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c index 870bfc5..dd0f12b 100644 --- a/sound/pci/lx6464es/lx6464es.c +++ b/sound/pci/lx6464es/lx6464es.c @@ -43,6 +43,9 @@ static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for Digigram LX6464ES interface."); + static const char card_name[] = "LX6464ES"; -- 1.6.2.1

2 4

[alsa-devel] AC97 atmel
by Sedji Gaouaou 11 Jun '09

11 Jun '09

Hi, I have a patch which adds support for ATMEL AT91. I would like to know on which branch should I base my patch (linux ketchup or http://www.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git). Regards, Sedji

4 11

Re: [alsa-devel] SB X-Fi driver fixes (Re: Testers wanted:
by bill＠tellytopia.com 11 Jun '09

11 Jun '09

I am using the snd-ctxfi driver on a 64-bit version of fedora code 7. Is there anyone else working on a 64-bit system? There are some compile warnings during the compilation of the driver, but no errors. But no sound, either, so far. I have also taken the 1.0.20 library, utils and tools sources from the Alsa main downloads. In order to compile the utils I had to link the 64-bit sound libraries to /usr/lib/libasound.2.0.0, in other words: (cd /lib64; ln -s /usr/lib/libasound.2.0.0 libasound.2.0.0; ln -s /usr/lib/libasound.2.0.0 libasound.2) There is no /lib64/libasound.la Also had to install xmlto (yum install xmlto). So far I have added the snd-hda-intel driver to the blacklist but still the Soundcard Detection shows snd-hda-intel. Any ideas, anyone? Regards Bill

2 1

[alsa-devel] [PATCH v2 1/2] ASoC: change set_tdm_slot api to allow slot_size override.
by Daniel Ribeiro 11 Jun '09

11 Jun '09

Extend set_tdm_slot to allow the user to arbitrarily set the frame width and active TX/RX slots. Updates magician.c and wm9081.c for the new set_tdm_slot(). wm9081.c still doesn't handle the slot_width override, I will submit this as a separate patch after I figure how to handle the I2S case. (this series is meant for Mark's for-2.6.32 branch + "remove duplicated code" patch I sent yesterday) Signed-off-by: Daniel Ribeiro <drwyrm(a)gmail.com> --- include/sound/soc-dai.h | 5 +++-- sound/soc/codecs/wm9081.c | 5 +++-- sound/soc/pxa/magician.c | 2 +- sound/soc/pxa/pxa-ssp.c | 27 ++++++++++++++++++++------- sound/soc/soc-core.c | 9 ++++++--- 5 files changed, 33 insertions(+), 15 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 352d7ee..6692410 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -106,7 +106,7 @@ int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, - unsigned int mask, int slots); + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width); int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); @@ -140,7 +140,8 @@ struct snd_soc_dai_ops { */ int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt); int (*set_tdm_slot)(struct snd_soc_dai *dai, - unsigned int mask, int slots); + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width); int (*set_tristate)(struct snd_soc_dai *dai, int tristate); /* diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 86fc57e..c47d573 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1206,8 +1206,9 @@ static int wm9081_set_sysclk(struct snd_soc_dai *codec_dai, return 0; } +/* FIXME: Needs to handle slot_width */ static int wm9081_set_tdm_slot(struct snd_soc_dai *dai, - unsigned int mask, int slots) + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { struct snd_soc_codec *codec = dai->codec; unsigned int aif1 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_1); @@ -1219,7 +1220,7 @@ static int wm9081_set_tdm_slot(struct snd_soc_dai *dai, aif1 |= (slots - 1) << WM9081_AIFDAC_TDM_MODE_SHIFT; - switch (mask) { + switch (rx_mask) { case 1: break; case 2: diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index c89a3cd..34437d8 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -188,7 +188,7 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 1); + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 0, 1, width); if (ret < 0) return ret; diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index e22c5ce..613cccb 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -375,21 +375,34 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, * Set the active slots in TDM/Network mode */ static int pxa_ssp_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, - unsigned int mask, int slots) + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { struct ssp_priv *priv = cpu_dai->private_data; struct ssp_device *ssp = priv->dev.ssp; u32 sscr0; - sscr0 = ssp_read_reg(ssp, SSCR0) & ~SSCR0_SlotsPerFrm(7); + sscr0 = ssp_read_reg(ssp, SSCR0); + sscr0 &= ~(SSCR0_MOD | SSCR0_SlotsPerFrm(7) | SSCR0_EDSS | SSCR0_DSS); + + /* set slot width */ + if (slot_width > 16) + sscr0 |= SSCR0_EDSS | SSCR0_DataSize(slot_width - 16); + else + sscr0 |= SSCR0_DataSize(slot_width); - /* set number of active slots */ - sscr0 |= SSCR0_SlotsPerFrm(slots); + if (slots > 1) { + /* enable network mode */ + sscr0 |= SSCR0_MOD; + + /* set number of active slots */ + sscr0 |= SSCR0_SlotsPerFrm(slots); + + /* set active slot mask */ + ssp_write_reg(ssp, SSTSA, tx_mask); + ssp_write_reg(ssp, SSRSA, rx_mask); + } ssp_write_reg(ssp, SSCR0, sscr0); - /* set active slot mask */ - ssp_write_reg(ssp, SSTSA, mask); - ssp_write_reg(ssp, SSRSA, mask); return 0; } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e1a920c..c72a8e8 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2133,17 +2133,20 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt); /** * snd_soc_dai_set_tdm_slot - configure DAI TDM. * @dai: DAI - * @mask: DAI specific mask representing used slots. + * @tx_mask: bitmask representing active TX slots. + * @rx_mask: bitmask representing active RX slots. * @slots: Number of slots in use. + * @slot_width: Width in bits for each slot. * * Configures a DAI for TDM operation. Both mask and slots are codec and DAI * specific. */ int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, - unsigned int mask, int slots) + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { if (dai->ops && dai->ops->set_tdm_slot) - return dai->ops->set_tdm_slot(dai, mask, slots); + return dai->ops->set_tdm_slot(dai, tx_mask, rx_mask, + slots, slot_width); else return -EINVAL; } -- tg: (c787980..) asoc/set_tdm_slot (depends on: asoc/remove-code-duplication) -- Daniel Ribeiro

2 2

[alsa-devel] [PATCH v2 2/2] ASoC: pxa-ssp.c, Automatically set TDM when needed
by Daniel Ribeiro 11 Jun '09

11 Jun '09

* Automatically sets TDM mode for frame_width larger than 32 bits, if the user doesn't setup the TDM slots with set_tdm_slot(). * Reset SSCR0_EDSS and SSCR0_DSS on pxa_ssp_set_dai_fmt. * Makes SSCR0_MOD optional. * Clears SSCR1_RWOT case SSCR0_MOD is set. Signed-off-by: Daniel Ribeiro <drwyrm(a)gmail.com> --- sound/soc/pxa/pxa-ssp.c | 88 ++++++++++++++++++++++++++++------------------ 1 files changed, 53 insertions(+), 35 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 613cccb..7e72c41 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -452,8 +452,8 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, } /* reset port settings */ - sscr0 = ssp_read_reg(ssp, SSCR0) & - (SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); + sscr0 = ssp_read_reg(ssp, SSCR0) & (SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | + SSCR0_ACS | SSCR0_DSS | SSCR0_EDSS); sscr1 = SSCR1_RxTresh(8) | SSCR1_TxTresh(7); sspsp = 0; @@ -496,7 +496,7 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, case SND_SOC_DAIFMT_DSP_A: sspsp |= SSPSP_FSRT; case SND_SOC_DAIFMT_DSP_B: - sscr0 |= SSCR0_MOD | SSCR0_PSP; + sscr0 |= SSCR0_PSP; sscr1 |= SSCR1_TRAIL | SSCR1_RWOT; break; @@ -532,10 +532,18 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, struct ssp_priv *priv = cpu_dai->private_data; struct ssp_device *ssp = priv->dev.ssp; int chn = params_channels(params); - u32 sscr0; - u32 sspsp; + u32 sscr0, sscr1, sspsp; int width = snd_pcm_format_physical_width(params_format(params)); - int ttsa = ssp_read_reg(ssp, SSTSA) & 0xf; + int slot_width, frame_width = 0; + + /* check if the user explicitly set a slot_width */ + sscr0 = ssp_read_reg(ssp, SSCR0); + + if (sscr0 & (SSCR0_EDSS | SSCR0_DSS)) + slot_width = (sscr0 & SSCR0_DSS) + + (sscr0 & SSCR0_EDSS ? 17 : 1); + else + frame_width = slot_width = width * chn; /* generate correct DMA params */ if (cpu_dai->dma_data) @@ -545,35 +553,53 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, * to force 16-bit frame width on the wire (for S16_LE), even * with two channels. Use 16-bit DMA transfers for this case. */ - cpu_dai->dma_data = ssp_get_dma_params(ssp, - ((chn == 2) && (ttsa != 1)) || (width == 32), + cpu_dai->dma_data = ssp_get_dma_params(ssp, slot_width > 16, substream->stream == SNDRV_PCM_STREAM_PLAYBACK); /* we can only change the settings if the port is not in use */ - if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) + if (sscr0 & SSCR0_SSE) return 0; - /* clear selected SSP bits */ - sscr0 = ssp_read_reg(ssp, SSCR0) & ~(SSCR0_DSS | SSCR0_EDSS); - ssp_write_reg(ssp, SSCR0, sscr0); - - /* bit size */ - sscr0 = ssp_read_reg(ssp, SSCR0); - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: #ifdef CONFIG_PXA3xx - if (cpu_is_pxa3xx()) - sscr0 |= SSCR0_FPCKE; + if (width == 16 && cpu_is_pxa3xx()) + sscr0 |= SSCR0_FPCKE; #endif - sscr0 |= SSCR0_DataSize(16); - break; - case SNDRV_PCM_FORMAT_S24_LE: - sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(8)); - break; - case SNDRV_PCM_FORMAT_S32_LE: - sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(16)); - break; + + if (frame_width > 0) { + /* Not using network mode */ + if (frame_width > 16) + sscr0 |= SSCR0_EDSS | SSCR0_DataSize(frame_width - 16); + else + sscr0 |= SSCR0_DataSize(frame_width); + + if (frame_width > 32) { + /* + * Network mode is needed to support this frame width. + * We assume the wire is not networked and setup a + * fake network mode here. Use as many slots as needed + * each with 32 bits. + */ + int slots = frame_width / 32; + + sscr0 |= SSCR0_MOD; + sscr0 |= SSCR0_SlotsPerFrm(slots); + + /* + * Set active slots. Only set an active TX slot + * if we are going to use it. + */ + ssp_write_reg(ssp, SSRSA, slots - 1); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + ssp_write_reg(ssp, SSTSA, slots - 1); + } } + + /* If SSCR0_MOD is set we can't use SSCR1_RWOT */ + if (sscr0 & SSCR0_MOD) { + sscr1 = ssp_read_reg(ssp, SSCR1); + ssp_write_reg(ssp, SSCR1, sscr1 & ~SSCR1_RWOT); + } + ssp_write_reg(ssp, SSCR0, sscr0); switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { @@ -620,14 +646,6 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, break; } - /* When we use a network mode, we always require TDM slots - * - complain loudly and fail if they've not been set up yet. - */ - if ((sscr0 & SSCR0_MOD) && !ttsa) { - dev_err(&ssp->pdev->dev, "No TDM timeslot configured\n"); - return -EINVAL; - } - dump_registers(ssp); return 0; -- tg: (5231b18..) asoc/ssp-internals (depends on: asoc/set_tdm_slot) -- Daniel Ribeiro

1 0

[alsa-devel] [PATCH] ALSA: snd_usb_caiaq: set mixername
by Daniel Mack 11 Jun '09

11 Jun '09

alsamixer and friends want the mixername to be set. Even though the driver does not exports a real mixer device, export the name doesn't harm. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- sound/usb/caiaq/device.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index b9a2b31..2240624 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -35,7 +35,7 @@ #include "input.h" MODULE_AUTHOR("Daniel Mack <daniel(a)caiaq.de>"); -MODULE_DESCRIPTION("caiaq USB audio, version 1.3.15"); +MODULE_DESCRIPTION("caiaq USB audio, version 1.3.16"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, RigKontrol3}," @@ -426,6 +426,7 @@ static int __devinit init_card(struct snd_usb_caiaqdev *dev) strlcpy(card->driver, MODNAME, sizeof(card->driver)); strlcpy(card->shortname, dev->product_name, sizeof(card->shortname)); + strlcpy(card->mixername, dev->product_name, sizeof(card->mixername)); /* if the id was not passed as module option, fill it with a shortened * version of the product string which does not contain any -- 1.6.3.1

2 1

[alsa-devel] no sound in ubuntu 8.10 on AD1986 sound card
by sandip gangakhedkar 10 Jun '09

10 Jun '09

Hi all, I am facing a very perplexing issue with sound in Ubuntu 8.10 which I have never encountered before with linux. I do not hear any sound from my speakers. The sound card is AD 1986 (Analog Devices) and it worked wonderfully without any hiccups before on 8.10 and other Ubuntu versions. My alsa-info script is here: http://www.alsa-project.org/db/?f=7f34b4e28d97178d7f709740f91ff66ea08a9468 I suspect pulseaudio is the culprit: Output of : /var/log/syslog | grep pulseaudio is here: http://paste.ubuntu.com/192629/ Running : speaker-test -c2 -D hw:0,0 -t wav -l1 gives no sound. My motherboard is ASUS A8N-VM on a AMD 64 Athlon 3200 Desktop PC.The sound card is detected so no problems there. I have reinstalled ALSA, pulseaudio and all the related packages but to no avail. I am pretty sure this is not a problem of inadeuate support because sound was working before without any issues. Any ideas/suggestions on how to fix this problem will be greatly appreciated. I am also raising an email bug report at alsa-devel so anyone can canter around there and provide expert help :) Thanks, Sandip

3 3

[alsa-devel] [PATCH] ALSA: hda - add quirk for STAC92xx (SigmaTel STAC9205)
by Simos Xenitellis 10 Jun '09

10 Jun '09

A quirk is required for 8086:284b (rev 03) [Subsystem: 161f:2073]. The following has been tested with Alsa 1.0.20 (git master). Background details can be found at https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4561 http://forum.ubuntu-gr.org/viewtopic.php?f=38&t=5290 Tested-by: Theodora Iliopoulou <th30dr(a)gmail.com> Signed-off-by: Simos Xenitellis <simos(a)gnome.org> -- sound/pci/hda/patch_sigmatel.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 42f944b..104d739 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1590,6 +1590,8 @@ static struct snd_pci_quirk stac9200_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0xfb30, + "SigmaTel",STAC_9205_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, "DFI LanParty", STAC_REF), /* Dell laptops have BIOS problem */ @@ -5939,6 +5941,7 @@ static struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x83847662, .name = "STAC9872AK", .patch = patch_stac9872 }, { .id = 0x83847664, .name = "CXD9872AKD", .patch = patch_stac9872 }, { .id = 0x838476a0, .name = "STAC9205", .patch = patch_stac9205 }, + { .id = 0x83847698, .name = "STAC9205", .patch = patch_stac9205 }, { .id = 0x838476a1, .name = "STAC9205D", .patch = patch_stac9205 }, { .id = 0x838476a2, .name = "STAC9204", .patch = patch_stac9205 }, { .id = 0x838476a3, .name = "STAC9204D", .patch = patch_stac9205 },

2 1

Re: [alsa-devel] Testers wanted: New SB X-Fi driver
by Takashi Iwai 10 Jun '09

10 Jun '09

At Wed, 10 Jun 2009 18:21:55 +0200, MonsieurWeller wrote: > > Thanks for reply. > (Very) last little thing: i've tried to use the x-fi external rack with > my audigy2ZS (alsa driver 1.0.20 with emu10k1). And it seems to work (i > have only tried with the headphone jack). Maybe it's the same thing? Hard to tell, but it's anyway good to know. Thanks for the information! Takashi

1 0

[alsa-devel] EMU 0404USB: Odd sample rate shown during playback.
by Klaus Schulz 10 Jun '09

10 Jun '09

Hi there. The EMU 0404 is running in async USB mode as you know. When looking at the stream parameters during playback the sample rate shown is always slightly below 44100 -- something like 44098 is shown. As you write in the driver, the 0404 is returning number of samples per packet instead of the number of samples per microframes. Has this something to do with it? Question: Are we looking at a wrong output or is the samplerate really lower than 44100? THX Klaus

3 4