October 2007 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] hda: STAC9228 DMIC Support
by Matthew Ranostay 18 Oct '07

18 Oct '07

Added DMIC support for STAC9228 cards, as well as adding terminators to other hda_nid DMIC arrays, and added a new Vendor ID tag. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r e91f57f8127a pci/hda/hda_codec.c --- a/pci/hda/hda_codec.c Tue Oct 09 11:58:41 2007 +0200 +++ b/pci/hda/hda_codec.c Wed Oct 17 11:59:47 2007 -0400 @@ -55,6 +55,7 @@ static struct hda_vendor_id hda_vendor_i { 0x10ec, "Realtek" }, { 0x1057, "Motorola" }, { 0x1106, "VIA" }, + { 0x111d, "IDT" }, { 0x11d4, "Analog Devices" }, { 0x13f6, "C-Media" }, { 0x14f1, "Conexant" }, diff -r e91f57f8127a pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Tue Oct 09 11:58:41 2007 +0200 +++ b/pci/hda/patch_sigmatel.c Wed Oct 17 13:06:58 2007 -0400 @@ -182,8 +182,9 @@ static hda_nid_t stac925x_dac_nids[1] = 0x02, }; -static hda_nid_t stac925x_dmic_nids[1] = { - 0x15, +static hda_nid_t stac925x_dmic_nids[2] = { + 0x15, + 0x00, /* terminator */ }; static hda_nid_t stac922x_adc_nids[2] = { @@ -202,6 +203,11 @@ static hda_nid_t stac927x_mux_nids[3] = 0x15, 0x16, 0x17 }; +static hda_nid_t stac927x_dmic_nids[3] = { + 0x13, 0x14, + 0x00, /* terminator */ +}; + static hda_nid_t stac9205_adc_nids[2] = { 0x12, 0x13 }; @@ -210,8 +216,9 @@ static hda_nid_t stac9205_mux_nids[2] = 0x19, 0x1a }; -static hda_nid_t stac9205_dmic_nids[2] = { +static hda_nid_t stac9205_dmic_nids[3] = { 0x17, 0x18, + 0x00, /* terminator */ }; static hda_nid_t stac9200_pin_nids[8] = { @@ -2716,7 +2723,6 @@ static int patch_stac927x(struct hda_cod spec->mux_nids = stac927x_mux_nids; spec->num_muxes = ARRAY_SIZE(stac927x_mux_nids); spec->num_adcs = ARRAY_SIZE(stac927x_adc_nids); - spec->num_dmics = 0; spec->init = d965_core_init; spec->mixer = stac927x_mixer; break; @@ -2725,7 +2731,6 @@ static int patch_stac927x(struct hda_cod spec->mux_nids = stac927x_mux_nids; spec->num_muxes = ARRAY_SIZE(stac927x_mux_nids); spec->num_adcs = ARRAY_SIZE(stac927x_adc_nids); - spec->num_dmics = 0; spec->init = d965_core_init; spec->mixer = stac927x_mixer; break; @@ -2734,9 +2739,18 @@ static int patch_stac927x(struct hda_cod spec->mux_nids = stac927x_mux_nids; spec->num_muxes = ARRAY_SIZE(stac927x_mux_nids); spec->num_adcs = ARRAY_SIZE(stac927x_adc_nids); - spec->num_dmics = 0; spec->init = stac927x_core_init; spec->mixer = stac927x_mixer; + } + + switch (codec->subsystem_id) { + case 0x1028020A: /* STAC 9228 */ + case 0x10280209: /* STAC 9228 */ + spec->dmic_nids = stac927x_dmic_nids; + spec->num_dmics = 2; + break; + default: + spec->num_dmics = 0; } spec->multiout.dac_nids = spec->dac_nids; @@ -2798,7 +2812,7 @@ static int patch_stac9205(struct hda_cod spec->mux_nids = stac9205_mux_nids; spec->num_muxes = ARRAY_SIZE(stac9205_mux_nids); spec->dmic_nids = stac9205_dmic_nids; - spec->num_dmics = ARRAY_SIZE(stac9205_dmic_nids); + spec->num_dmics = 2; spec->dmux_nid = 0x1d; spec->init = stac9205_core_init;

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[alsa-devel] alsa-driver-1.0.15rc2 build issues on RHEL4/CentOS4
by James Pearson 18 Oct '07

18 Oct '07

alsa-driver-1.0.15rc2 won't build cleanly on RHEL4/CentOS4 (2.6.9 based kernel). I've included the relevant errors below after running: configure --with-redhat make -i The problem with misc/ac97_bus.c, I've already reported at: <http://mailman.alsa-project.org/pipermail/alsa-devel/2007-September/003119.…> James Pearson Build errors: CC [M] /tmp/alsa-driver-1.0.15rc2/misc/ac97_bus.o In file included from /tmp/alsa-driver-1.0.15rc2/misc/ac97_bus.c:3: /tmp/alsa-driver-1.0.15rc2/misc/../alsa-kernel/ac97_bus.c: In function `ac97_bus_suspend': /tmp/alsa-driver-1.0.15rc2/misc/../alsa-kernel/ac97_bus.c:35: error: too few arguments to function /tmp/alsa-driver-1.0.15rc2/misc/../alsa-kernel/ac97_bus.c: In function `ac97_bus_resume': /tmp/alsa-driver-1.0.15rc2/misc/../alsa-kernel/ac97_bus.c:45: error: too few arguments to function make[4]: [/tmp/alsa-driver-1.0.15rc2/misc/ac97_bus.o] Error 1 (ignored) CC [M] /tmp/alsa-driver-1.0.15rc2/pci/cs5535audio/cs5535audio_pm.o In file included from /tmp/alsa-driver-1.0.15rc2/pci/cs5535audio/cs5535audio_pm.c:2: /tmp/alsa-driver-1.0.15rc2/pci/cs5535audio/../../alsa-kernel/pci/cs5535audio/cs5535audio_pm.c: In function `snd_cs5535audio_suspend': /tmp/alsa-driver-1.0.15rc2/pci/cs5535audio/../../alsa-kernel/pci/cs5535audio/cs5535audio_pm.c:77: error: void value not ignored as it ought to be /tmp/alsa-driver-1.0.15rc2/pci/cs5535audio/../../alsa-kernel/pci/cs5535audio/cs5535audio_pm.c: In function `snd_cs5535audio_resume': /tmp/alsa-driver-1.0.15rc2/pci/cs5535audio/../../alsa-kernel/pci/cs5535audio/cs5535audio_pm.c:95: error: void value not ignored as it ought to be make[5]: [/tmp/alsa-driver-1.0.15rc2/pci/cs5535audio/cs5535audio_pm.o] Error 1 (ignored) LD [M] /tmp/alsa-driver-1.0.15rc2/pci/cs5535audio/snd-cs5535audio.o ld: /tmp/alsa-driver-1.0.15rc2/pci/cs5535audio/cs5535audio_pm.o: No such file: No such file or directory make[5]: [/tmp/alsa-driver-1.0.15rc2/pci/cs5535audio/snd-cs5535audio.o] Error 1 (ignored) Building modules, stage 2. MODPOST *** Warning: "class_device_destroy" [/tmp/alsa-driver-1.0.15rc2/pci/asihpi/snd-asihpi.ko] undefined! *** Warning: "ac97_bus_type" [/tmp/alsa-driver-1.0.15rc2/pci/ac97/snd-ac97-codec.ko] undefined! *** Warning: "platform_device_alloc" [/tmp/alsa-driver-1.0.15rc2/drivers/snd-portman2x4.ko] undefined! *** Warning: "platform_device_add" [/tmp/alsa-driver-1.0.15rc2/drivers/snd-portman2x4.ko] undefined! *** Warning: "platform_device_put" [/tmp/alsa-driver-1.0.15rc2/drivers/snd-portman2x4.ko] undefined! *** Warning: "platform_device_alloc" [/tmp/alsa-driver-1.0.15rc2/drivers/snd-mts64.ko] undefined! *** Warning: "platform_device_add" [/tmp/alsa-driver-1.0.15rc2/drivers/snd-mts64.ko] undefined! *** Warning: "platform_device_put" [/tmp/alsa-driver-1.0.15rc2/drivers/snd-mts64.ko] undefined!

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[alsa-devel] [patch] snd_pcm_dmix_close: raise semaphore if unable to discard
by Mike Gorse 18 Oct '07

18 Oct '07

This patch causes snd_pcm_dmix_close() to up a semaphore after downing it if it is unable to discard it. It prevents some deadlock that I am getting when a couple of applications interact and one of them closes the device and later re-opens it. -- --- src/pcm/pcm_dmix.c.orig 2007-10-17 20:19:47.000000000 -0400 +++ src/pcm/pcm_dmix.c 2007-10-17 20:50:46.000000000 -0400 @@ -652,8 +652,11 @@ if (dmix->client) snd_pcm_direct_client_discard(dmix); shm_sum_discard(dmix); - if (snd_pcm_direct_shm_discard(dmix)) - snd_pcm_direct_semaphore_discard(dmix); + if (snd_pcm_direct_shm_discard(dmix)) { + if (snd_pcm_direct_semaphore_discard(dmix)) { + snd_pcm_direct_semaphore_up(dmix, DIRECT_IPC_SEM_CLIENT); + } + } else snd_pcm_direct_semaphore_up(dmix, DIRECT_IPC_SEM_CLIENT); free(dmix->bindings);

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[alsa-devel] [patch] Fix bug with bt87x driver's load_all option
by Trent Piepho 18 Oct '07

18 Oct '07

snd-bt87x: Make the load_all option work correctly If the load_all option was turned on all cards would be treated as unknown, even those which are in the database. Of course, if the card is in the database there is no reason to use the load_all option. It's there to force loading when the card isn't in the database. But there are out of date wikis that say to do this and some distros might turn this option on by default. So, we keep the load_all option from turning known cards into unknown cards. Signed-off-by: Trent Piepho <xyzzy(a)speakeasy.org> diff --git a/pci/bt87x.c b/pci/bt87x.c --- a/pci/bt87x.c +++ b/pci/bt87x.c @@ -1021,7 +1021,7 @@ static int __devinit snd_bt87x_detect_ca int i; const struct pci_device_id *supported; - supported = pci_match_device(&driver, pci); + supported = pci_match_id(snd_bt87x_ids, pci); if (supported && supported->driver_data > 0) return supported->driver_data;

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[alsa-devel] [PATCH] ALSA - protect Dreamcast PCM driver (AICA) from G2 bus effects
by Adrian McMenamin 18 Oct '07

18 Oct '07

The G2 bus on the SEGA Dreamcast connects both the maple peripheral bus and the AICA sound memory. DMA requests on one can cause the other to timeout on memory operations. This patch prevents maple interrupts from causing hiccoughs in the AICA sound (maple bus code will land in 2.6.24). There are other cleanups for this (AICA) code - but this is in effect a regression fix rather than a cleanup. Signed-off-by: Adrian McMenamin <adrian(a)mcmen.demon.co.uk> diff --git a/sound/sh/aica.c b/sound/sh/aica.c index 7397865..407981d 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -106,11 +106,14 @@ static void spu_write_wait(void) static void spu_memset(u32 toi, u32 what, int length) { int i; + unsigned long flags; snd_assert(length % 4 == 0, return); for (i = 0; i < length; i++) { if (!(i % 8)) spu_write_wait(); + local_irq_save(flags); writel(what, toi + SPU_MEMORY_BASE); + local_irq_restore(flags); toi++; } } @@ -118,6 +121,7 @@ static void spu_memset(u32 toi, u32 what, int length) /* spu_memload - write to SPU address space */ static void spu_memload(u32 toi, void *from, int length) { + unsigned long flags; u32 *froml = from; u32 __iomem *to = (u32 __iomem *) (SPU_MEMORY_BASE + toi); int i; @@ -128,7 +132,9 @@ static void spu_memload(u32 toi, void *from, int length) if (!(i % 8)) spu_write_wait(); val = *froml; + local_irq_save(flags); writel(val, to); + local_irq_restore(flags); froml++; to++; } @@ -138,28 +144,36 @@ static void spu_memload(u32 toi, void *from, int length) static void spu_disable(void) { int i; + unsigned long flags; u32 regval; spu_write_wait(); regval = readl(ARM_RESET_REGISTER); regval |= 1; spu_write_wait(); + local_irq_save(flags); writel(regval, ARM_RESET_REGISTER); + local_irq_restore(flags); for (i = 0; i < 64; i++) { spu_write_wait(); regval = readl(SPU_REGISTER_BASE + (i * 0x80)); regval = (regval & ~0x4000) | 0x8000; spu_write_wait(); + local_irq_save(flags); writel(regval, SPU_REGISTER_BASE + (i * 0x80)); + local_irq_restore(flags); } } /* spu_enable - set spu registers to enable sound output */ static void spu_enable(void) { + unsigned long flags; u32 regval = readl(ARM_RESET_REGISTER); regval &= ~1; spu_write_wait(); + local_irq_save(flags); writel(regval, ARM_RESET_REGISTER); + local_irq_restore(flags); } /* @@ -168,25 +182,34 @@ static void spu_enable(void) */ static void spu_reset(void) { + unsigned long flags; spu_disable(); spu_memset(0, 0, 0x200000 / 4); /* Put ARM7 in endless loop */ + local_irq_save(flags); ctrl_outl(0xea000002, SPU_MEMORY_BASE); + local_irq_restore(flags); spu_enable(); } /* aica_chn_start - write to spu to start playback */ static void aica_chn_start(void) { + unsigned long flags; spu_write_wait(); + local_irq_save(flags); writel(AICA_CMD_KICK | AICA_CMD_START, (u32 *) AICA_CONTROL_POINT); + local_irq_restore(flags); } /* aica_chn_halt - write to spu to halt playback */ static void aica_chn_halt(void) { + unsigned long flags; spu_write_wait(); + local_irq_save(flags); writel(AICA_CMD_KICK | AICA_CMD_STOP, (u32 *) AICA_CONTROL_POINT); + local_irq_restore(flags); } /* ALSA code below */ @@ -213,12 +236,13 @@ static int aica_dma_transfer(int channels, int buffer_size, int q, err, period_offset; struct snd_card_aica *dreamcastcard; struct snd_pcm_runtime *runtime; - err = 0; + unsigned long flags; dreamcastcard = substream->pcm->private_data; period_offset = dreamcastcard->clicks; period_offset %= (AICA_PERIOD_NUMBER / channels); runtime = substream->runtime; for (q = 0; q < channels; q++) { + local_irq_save(flags); err = dma_xfer(AICA_DMA_CHANNEL, (unsigned long) (runtime->dma_area + (AICA_BUFFER_SIZE * q) / @@ -228,9 +252,12 @@ static int aica_dma_transfer(int channels, int buffer_size, AICA_CHANNEL0_OFFSET + q * CHANNEL_OFFSET + AICA_PERIOD_SIZE * period_offset, buffer_size / channels, AICA_DMA_MODE); - if (unlikely(err < 0)) + if (unlikely(err < 0)) { + local_irq_restore(flags); break; + } dma_wait_for_completion(AICA_DMA_CHANNEL); + local_irq_restore(flags); } return err; }

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[alsa-devel] RME HDSP driver enhancements for HDSPe cards
by Ptitjes 18 Oct '07

18 Oct '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, First, sorry for the cross-post, but as i did not know where to start, i wanted to ask to both lists. I bought a RME HDSPe PCI card. I should receive it in the following days. My plan is to: 1) Determine whether the current driver works as is with the HDSPe cards and make my report to both lists. 2) Enhance the current driver in order to take profit of the new feature (32bit float format) However, i am new to Alsa, Jack and HDSP driver internal code structure. So i wanted to collect informations and hints as where to start from and be the most efficient in starting the work as i can. Could you please give me links as to: - - General Alsa driver structure and data flow - - Work done by Alsa HDSP driver and work done by Jack (I ever thought most of the work is done by Jack as my current HDSP PCCard do not work with Alsa only) - - Parts of the Alsa HDSP driver that would a priori need changes. - - Parts of Jack that would a priori need changes. Thank you very much for your help. I hope i did not bother everyone with my newbie questions. Best regards, Didier. -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHFdRLVE1mALQL3JsRAvp1AKCV7L7DdX1doenajmXFX8vnYkbbqQCgkbuS GZgsAKdtGyiwfP4vNYYdZb4= =vvgU -----END PGP SIGNATURE-----

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[alsa-devel] 1.0.15rc3 patch_analog.s bug + fix
by Peter Skensved 17 Oct '07

17 Oct '07

Hi, I've come across a problem with the 1.0.15rc3 alsa driver on my Lenovo 3000 N100 laptop using the auto mute feature - it appears to work backwards ! Plugging in the headphones unmutes the external speakers and unplugging the heaphones mutes them ... Changing the if statement in line ~600 in patch_analog.c in ad1986a_update_hp to if (!spec->jack_preset) makes it work the intended way. ( spec->jack_present changes state _before_ the call to ad1986a_hp_automute ) peter ---- Peter Skensved Email : peter(a)SNO.Phy.QueensU.CA Dept. of Physics, Queen's University, Kingston, Ontario, Canada

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[alsa-devel] Recent ALSA version needs special configuration for Terratec Phase 26 USB
by Janne Himanka 17 Oct '07

17 Oct '07

Hello, I run Debian unstable on my Athlon XP1800 machine, and updating libasound2 package from version 1.0.13-2 to 1.0.14 (and later) made my Terratec Phase 26 USB soundcard useless. Whenever I tried to play something, the program would crash with a segmentation fault. When I downgraded to 1.0.13, everything would work again. It took a lot of head scratching and experimentation to find out that adding cards.USB-Audio.pcm.use_dmix."USB2448" no to my .asoundrc (it is otherwise empty) is needed for it to work. After adding that, it seems to work as it did before. The string in hyphens varies according to the mode one selects from the Terratec front panel. There are four modes with different word length and bitrates. The Phase 26 is adaptive; it doesn't need converting for a specific format. Bitrates from 32 KHz to 96 KHz and word lengths from 16 to 32 bits just work. I'm writing you this because a comment in USB-Audio.conf says: # If your device requires such a definition to work correctly, please report it # to <alsa-devel(a)alsa-project.org>. If you need any additional info, I'll be glad to help. Regards, Janne Himanka

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[alsa-devel] Asus Eeepc missing in alsa models list?
by Herton Ronaldo Krzesinski 17 Oct '07

17 Oct '07

Looking at the change 291702f017efdfe556cb87b8530eb7d1ff08cbae, may be the addition of the Asus Eeepc model was missed? The following patch fixes it if it was not intended originally: diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 4b48c2e..e2a1985 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -843,6 +843,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. 3stack-6ch-dig 3-stack (6-channel) with SPDIF 6stack-dig 6-stack with SPDIF lenovo-101e Lenovo laptop + eeepc-p701 ASUS Eeepc auto auto-config reading BIOS (default) ALC882/885 diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 53b0428..2ceaf4c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12058,6 +12058,7 @@ static const char *alc662_models[ALC662_MODEL_LAST] = { [ALC662_3ST_6ch] = "3stack-6ch", [ALC662_5ST_DIG] = "6stack-dig", [ALC662_LENOVO_101E] = "lenovo-101e", + [ALC662_ASUS_EEEPC_P701] = "eeepc-p701", [ALC662_AUTO] = "auto", }; -- []'s Herton

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Re: [alsa-devel] hda-intel: no soundcard with current linus' git tree
by Maxim Levitsky 17 Oct '07

17 Oct '07

On Wednesday 17 October 2007 00:18:42 Thomas Meyer wrote: > $ dmesg > > [schnipp] > > ACPI: PCI Interrupt 0000:00:1b.0[A] -> GSI 22 (level, low) -> IRQ 21 > PCI: Enabling bus mastering for device 0000:00:1b.0 > PCI: Setting latency timer of device 0000:00:1b.0 to 64 > hda_codec: STAC922x, Apple subsys_id=106b0200 > ACPI: PCI interrupt for device 0000:00:1b.0 disabled > HDA Intel: probe of 0000:00:1b.0 failed with error -16 > > [schnipp] > > > - > To unsubscribe from this list: send the line "unsubscribe linux-kernel" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html > Please read the FAQ at http://www.tux.org/lkml/ > Please bisect this (the alsa merge happened just short time ago) Which kernel did work? Best regards, Maxim Levitsky

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[alsa-devel] About the buffer_time and period_time
by Fengwei Yin 17 Oct '07

17 Oct '07

Hi all, I met a problem about period_time/buffer_time on PXA with ASOC enabled. The kernel version is 2.6.21. The Audio stream is 32k which is not supported by PXA ASOC driver. Please refer following RATES definition for PXA2xx. #define PXA2XX_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000) So I set plughw as default ALSA device enable auto samplerate conversion. The code to set the period_time/buffer_time is as following: err = snd_pcm_hw_params_set_rate_near(hPCM, params, &rate, 0); if (err < 0) { error("Rate not available: %d\n", rate); return err; } err = snd_pcm_hw_params_get_buffer_time_max(params, &buffer_time, 0); if (buffer_time > 500000) buffer_time = 500000; buffer_time = 400000; period_time = buffer_time/4; err = snd_pcm_hw_params_set_buffer_time_near(hPCM, params, &buffer_time, 0); if (err < 0) { error("Buffer time not available: %d, error number: %d\n", buffer_time, err); return err; } err = snd_pcm_hw_params_set_period_time_near(hPCM, params, &period_time, 0); if (err < 0) { error("Period time not available: %d\n", period_time); return err; } If I set buffer_time to 400000, I will get: " Buffer time not available: 400000, error number: -22 Cannot init audio driver. Floating point exception " If I set buffer time to 500000, buffer_time can be set correctly. But audio playback failed. If I set period_time before setting buffer_time, Everything is OK. Any thought about this problem. Thanks. Regards Yin, Fengwei

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[alsa-devel] Intel HDA digital out distortion
by pr0t0cod7 16 Oct '07

16 Oct '07

Hi, On Monday 23 July 2007 23:09:24 Ingmar Stieger wrote: > a couple of weeks ago, I bought an ASUS mainboard with integrated > Intel > HDA and an AD1988B codec. Unfortunately, the digital / SPDIF output > gets distorted with stock 1.0.14 drivers. It does not matter what the > volume levels of the various mixers are set to. I have the same issues. Any sound is distorted through spdif output (seems more pronounced with low frequencies.) Sound is not distorted: - using passthough (playing AC3) - when volume is set to the very minimal (no perceptible distortion) - with analog output My motherboard: ASUS P5B deluxe with integrated soundchip Kernel: 2.6.22 Alsa: Intel HD audio kernel built module asound.conf: http://pastebin.com/f2c54b47b > I think I might have found the source for this problem. Maybe it's > only a workaround, but it solved the issue and digital out now works > fine for me, so I thought I'd share. The patch worked for me, no more distortion :) Thanks for sharing, Alex.

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[alsa-devel] fsc esprimo v5505 (snd_intel_hda) - output in /var/log/messages
by Jan-Simon Möller 16 Oct '07

16 Oct '07

Hi! What do these messages mean ? Oct 15 09:26:22 legolas kernel: ALSA sound/pci/hda/hda_intel.c:1185: azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 Oct 15 09:26:22 legolas kernel: ALSA sound/pci/hda/hda_codec.c:623: hda_codec_setup_stream: NID=0x13, stream=0x5, channel=0, format=0x11 Oct 15 09:26:22 legolas kernel: ALSA sound/pci/hda/hda_codec.c:623: hda_codec_setup_stream: NID=0x19, stream=0x5, channel=0, format=0x11 Oct 15 09:26:22 legolas kernel: ALSA sound/pci/hda/hda_codec.c:623: hda_codec_setup_stream: NID=0x19, stream=0x0, channel=0, format=0x0 Oct 15 09:26:22 legolas kernel: ALSA sound/pci/hda/hda_codec.c:623: hda_codec_setup_stream: NID=0x13, stream=0x0, channel=0, format=0x0 Oct 15 09:26:22 legolas kernel: ALSA sound/pci/hda/hda_intel.c:1185: azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 Oct 15 09:26:22 legolas kernel: ALSA sound/pci/hda/hda_codec.c:623: hda_codec_setup_stream: NID=0x13, stream=0x5, channel=0, format=0x11 Oct 15 09:26:22 legolas kernel: ALSA sound/pci/hda/hda_codec.c:623: hda_codec_setup_stream: NID=0x19, stream=0x5, channel=0, format=0x11 Oct 15 09:26:22 legolas kernel: ALSA sound/pci/hda/hda_codec.c:623: hda_codec_setup_stream: NID=0x19, stream=0x0, channel=0, format=0x0 Oct 15 09:26:22 legolas kernel: ALSA sound/pci/hda/hda_codec.c:623: hda_codec_setup_stream: NID=0x13, stream=0x0, channel=0, format=0x0 Thanks ! Jan

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Re: [alsa-devel] [Alsa-user] How to get set/fixed sample rate of ALSA device?
by Rene Herman 16 Oct '07

16 Oct '07

On 10/07/2007 07:04 PM, Peteris Krisjanis wrote: >> "Current sample rate" only has meaning with the device open , and if I >> understand you right, a "cat >> /proc/asound/card0/pcm0{p,c}/sub0/hw_params will then get you what you >> want as to the "get" part (substitute for 0 in the above as required). > > I actually meant what kind of sample rate is set in sound card. For > example, my onboard sound card has fixed Hz. I would like to know such > information _before_ I touch sound card :) Okay, you want the information that the driver author supplied in the struct snd_pcm_hardware for the stream. I've in fact also wanted that information on a number of occasions and just looked at the driver source then. > So you suggest to open device via any means and then read that file? I > did this and it works. > I guess I can even try to implement it in app and check for sample > rate everytime before I play. > > But is this only solution? It's rather hack, and I would like to have > more reliable method. You might consider this less of of a hack -- it uses the ALSA API to query the information from the PCM handle directly. Please see the library docoumentation for other information you can extract from a hw_params struct. (I don't suppose you can ever get anything but dir == 0 for "hw" devices). Introducing a /proc/asound/card0/pcm0{p,c}/hw_limits or similar might not be a bad idea though? Rene /* gcc -W -Wall -o snd_rate snd_rate.c -l asound */ #include <stdio.h> #include <alsa/asoundlib.h> int main(void) { snd_pcm_t *handle; snd_pcm_hw_params_t *params; unsigned int val; int dir; int err; snd_pcm_hw_params_alloca(&params); err = snd_pcm_open(&handle, "hw:0,0", SND_PCM_STREAM_PLAYBACK, 0); if (err < 0) { fprintf(stderr, "snd_pcm_open: %s\n", snd_strerror(err)); return -1; } err = snd_pcm_hw_params_any(handle, params); if (err < 0) { fprintf(stderr, "snd_pcm_hw_params_any: %s\n", snd_strerror(err)); return -1; } err = snd_pcm_hw_params_get_rate_min(params, &val, &dir); if (err < 0) { fprintf(stderr, "snd_pcm_hw_params_get_rate_min: %s\n", snd_strerror(err)); return -1; } printf("min: (%c) %u\n", "<=>"[dir + 1], val); err = snd_pcm_hw_params_get_rate_max(params, &val, &dir); if (err < 0) { fprintf(stderr, "snd_pcm_hw_params_get_rate_max: %s\n", snd_strerror(err)); return -1; } printf("max: (%c) %u\n", "<=>"[dir + 1], val); snd_pcm_close(handle); return 0; }

4 7

[alsa-devel] intel-hda patch for Dell Latitude D531
by glogow＠fbihome.de 16 Oct '07

16 Oct '07

Hi I have attached three files to this mail: 1. patch_sigmatel.c.diff I have a Latitude D531 which has a ATI SB600 with a STAC 9205. The chip is detected, but no config was read with the latest drivers. The problems are the dmic_nids arrays, which are ignored for auto config lookup. This calles is_in_nid_list, which expects a 0 terminated list, but these arrays are not. Additionally I updated the dell_9205_m43_pin_configs with values from the latest win drivers. 2. create_dell_tables.pl When looking into the ini files and comparing the nids and config values of the various tables, I got really confused, so I wrote a litte script to extract the values and build the table. This is the extended version, after I got my sound working. It extracts all pin infos from all ini files, generates the apropriate tables and prepends the Version.txt from the driver, resultiung in a kind of C header. Maybe it's usefull for you. 3. patch_sigmatel_dell.h Output from running create_dell_tables.pl on the latest drivers for my D531 from Dell. HTH Jan-Marek Glogowski ---------------------------------------------------------------- This message was sent using IMP, the Internet Messaging Program.

3 3

[alsa-devel] ALSA 1.0.15 final
by Jaroslav Kysela 16 Oct '07

16 Oct '07

Hello all, delayed but ... all ALSA packages 1.0.15 final are available for download on http://www.alsa-project.org . The changelog is available on web for 1.0.14 -> 1.0.15. I will make also available 1.0.15rc3 -> 1.0.15 changelog in next few hours. Have fun, Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project

1 0

[alsa-devel] How to detect 'dead air' while recording?
by william estrada 15 Oct '07

15 Oct '07

Hi group, I want to filter out 'dead air' while recording. What is the best way to do this? I don't want to use JACK, just ALSA. Any URLs? -- William Estrada MrUmunhum(a)popdial.com Mt-Umunhum-Wireless.net ( http://Mt-Umunhum-Wireless.net ) Ymessenger: MrUmunhum

1 0

[alsa-devel] LGPL Licensing
by WOODEN John 15 Oct '07

15 Oct '07

I wish to use either the ALSA or SALSA API within my application. I also intend to commercially distribute my application and sell it. What affect does ALSA's and/or SALSA's Lesser GPL license have upon applications that utilize their API's? Do I have to make the source code of my application freely available if I choose to utilize the ALSA or SALSA API? Thanks, John. IMPORTANT NOTICE: This e-mail and any attachment to it are intended only to be read or used by the named addressee. It is confidential and may contain legally privileged information. No confidentiality or privilege is waived or lost by any mistaken transmission to you. The RTA is not responsible for any unauthorised alterations to this e-mail or attachment to it. Views expressed in this message are those of the individual sender, and are not necessarily the views of the RTA. If you receive this e-mail in error, please immediately delete it from your system and notify the sender. You must not disclose, copy or use any part of this e-mail if you are not the intended recipient.

2 1

Re: [alsa-devel] [Pkg-alsa-devel] Bug#429326: Bug#437827: libasound2: Sound creaks badly when playing DVDs or OGG files
by Elimar Riesebieter 15 Oct '07

15 Oct '07

forwarded 429326 alsa-devel(a)alsa-project.org thanks On Sun, 14 Oct 2007 the mental interface of Loïc Minier told: > On Sun, Oct 14, 2007, Elimar Riesebieter wrote: > > From my point of view it is not reasonable to patch Debian packages, > > which of course will break 32bit only sound cards. So we've to find > > a way to force a solution directly on alsa-devel. Who will forward > > the _Debian_bug to alsa-devel(a)alsa-project.org ? > > I don't understand your remark about patching the Debian package? Why > would reverting the bit shift macros break anything? > > The attached patch for alsa-lib should be enough to fix this, upstream > should revert the broken change in the bit shifts. Yeah, we can provide it. But shouldn't it be a solution for all ALSA users? This is Debian #429326. As well reported as https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3360. We have to look for a general solution as the distributors don't need to patch sources. > Alsa #3360; GNOME #436192; LP #116990; Mandriva #33908; Revert HG r2264 > (changeset 23c4c0f5de40) as the bit shifts overflow when the volume is high > for 32 bits to 16 bits resampling > > diff -r 23c4c0f5de40 -r 883ae69471d2 src/pcm/plugin_ops.h > --- a/src/pcm/plugin_ops.h Fri Mar 09 17:06:23 2007 +0100 > +++ b/src/pcm/plugin_ops.h Fri Mar 09 15:30:38 2007 +0100 > @@ -85,8 +85,6 @@ static inline u_int32_t sx24s(u_int32_t > #define _put_triple(ptr,val) _put_triple_be(ptr,val) > #define _put_triple_s(ptr,val) _put_triple_le(ptr,val) > #endif > - > -#define shift_down(val, bits) (((val) + (1 << ((bits) - 1))) >> (bits)) > > #ifdef COPY_LABELS > static void *copy_labels[5] = { > @@ -258,14 +256,14 @@ conv_xxx1_009x: as_u32(dst) = sx24s((u_i > conv_xxx1_009x: as_u32(dst) = sx24s((u_int32_t)(as_u8c(src) ^ 0x80) << 8); goto CONV_END; > conv_xxx1_9000: as_u32(dst) = (u_int32_t)(as_u8c(src) ^ 0x80) << 24; goto CONV_END; > conv_xxx1_0009: as_u32(dst) = (u_int32_t)(as_u8c(src) ^ 0x80); goto CONV_END; > -conv_xx12_xxx1: as_u8(dst) = shift_down(as_u16c(src), 8); goto CONV_END; > +conv_xx12_xxx1: as_u8(dst) = as_u16c(src) >> 8; goto CONV_END; > conv_xx12_xx12: as_u16(dst) = as_u16c(src); goto CONV_END; > conv_xx12_xx21: as_u16(dst) = bswap_16(as_u16c(src)); goto CONV_END; > conv_xx12_x120: as_u32(dst) = sx24((u_int32_t)as_u16c(src) << 8); goto CONV_END; > conv_xx12_021x: as_u32(dst) = sx24s((u_int32_t)bswap_16(as_u16c(src)) << 8); goto CONV_END; > conv_xx12_1200: as_u32(dst) = (u_int32_t)as_u16c(src) << 16; goto CONV_END; > conv_xx12_0021: as_u32(dst) = (u_int32_t)bswap_16(as_u16c(src)); goto CONV_END; > -conv_xx12_xxx9: as_u8(dst) = shift_down(as_u16c(src), 8) ^ 0x80; goto CONV_END; > +conv_xx12_xxx9: as_u8(dst) = (as_u16c(src) >> 8) ^ 0x80; goto CONV_END; > conv_xx12_xx92: as_u16(dst) = as_u16c(src) ^ 0x8000; goto CONV_END; > conv_xx12_xx29: as_u16(dst) = bswap_16(as_u16c(src)) ^ 0x80; goto CONV_END; > conv_xx12_x920: as_u32(dst) = sx24((u_int32_t)(as_u16c(src) ^ 0x8000) << 8); goto CONV_END; > @@ -284,57 +282,57 @@ conv_xx12_01Ax: as_u32(dst) = sx24s((u_i > conv_xx12_01Ax: as_u32(dst) = sx24s((u_int32_t)(as_u16c(src) ^ 0x80) << 8); goto CONV_END; > conv_xx12_A100: as_u32(dst) = (u_int32_t)bswap_16(as_u16c(src) ^ 0x80) << 16; goto CONV_END; > conv_xx12_001A: as_u32(dst) = (u_int32_t)(as_u16c(src) ^ 0x80); goto CONV_END; > -conv_x123_xxx1: as_u8(dst) = shift_down(as_u32c(src), 16); goto CONV_END; > -conv_x123_xx12: as_u16(dst) = shift_down(as_u32c(src), 8); goto CONV_END; > -conv_x123_xx21: as_u16(dst) = bswap_16(shift_down(as_u32c(src), 8)); goto CONV_END; > +conv_x123_xxx1: as_u8(dst) = as_u32c(src) >> 16; goto CONV_END; > +conv_x123_xx12: as_u16(dst) = as_u32c(src) >> 8; goto CONV_END; > +conv_x123_xx21: as_u16(dst) = bswap_16(as_u32c(src) >> 8); goto CONV_END; > conv_x123_x123: as_u32(dst) = sx24(as_u32c(src)); goto CONV_END; > conv_x123_321x: as_u32(dst) = sx24s(bswap_32(as_u32c(src))); goto CONV_END; > conv_x123_1230: as_u32(dst) = as_u32c(src) << 8; goto CONV_END; > -conv_x123_0321: as_u32(dst) = shift_down(bswap_32(as_u32c(src)), 8); goto CONV_END; > - conv_x123_xxx9: as_u8(dst) = shift_down(as_u32c(src), 16) ^ 0x80; goto CONV_END; > -conv_x123_xx92: as_u16(dst) = shift_down(as_u32c(src), 8) ^ 0x8000; goto CONV_END; > -conv_x123_xx29: as_u16(dst) = bswap_16(shift_down(as_u32c(src), 8)) ^ 0x80; goto CONV_END; > +conv_x123_0321: as_u32(dst) = bswap_32(as_u32c(src)) >> 8; goto CONV_END; > +conv_x123_xxx9: as_u8(dst) = (as_u32c(src) >> 16) ^ 0x80; goto CONV_END; > +conv_x123_xx92: as_u16(dst) = (as_u32c(src) >> 8) ^ 0x8000; goto CONV_END; > +conv_x123_xx29: as_u16(dst) = bswap_16(as_u32c(src) >> 8) ^ 0x80; goto CONV_END; > conv_x123_x923: as_u32(dst) = sx24(as_u32c(src) ^ 0x800000); goto CONV_END; > conv_x123_329x: as_u32(dst) = sx24s(bswap_32(as_u32c(src)) ^ 0x8000); goto CONV_END; > conv_x123_9230: as_u32(dst) = (as_u32c(src) ^ 0x800000) << 8; goto CONV_END; > -conv_x123_0329: as_u32(dst) = shift_down(bswap_32(as_u32c(src)), 8) ^ 0x80; goto CONV_END; > -conv_123x_xxx3: as_u8(dst) = shift_down(as_u32c(src), 8) & 0xff; goto CONV_END; > -conv_123x_xx32: as_u16(dst) = bswap_16(shift_down(as_u32c(src), 8)); goto CONV_END; > -conv_123x_xx23: as_u16(dst) = shift_down(as_u32c(src), 8) & 0xffff; goto CONV_END; > +conv_x123_0329: as_u32(dst) = (bswap_32(as_u32c(src)) >> 8) ^ 0x80; goto CONV_END; > +conv_123x_xxx3: as_u8(dst) = (as_u32c(src) >> 8) & 0xff; goto CONV_END; > +conv_123x_xx32: as_u16(dst) = bswap_16(as_u32c(src) >> 8); goto CONV_END; > +conv_123x_xx23: as_u16(dst) = (as_u32c(src) >> 8) & 0xffff; goto CONV_END; > conv_123x_x321: as_u32(dst) = sx24(bswap_32(as_u32c(src))); goto CONV_END; > conv_123x_123x: as_u32(dst) = sx24s(as_u32c(src)); goto CONV_END; > conv_123x_3210: as_u32(dst) = bswap_32(as_u32c(src)) << 8; goto CONV_END; > -conv_123x_0123: as_u32(dst) = shift_down(as_u32c(src), 8); goto CONV_END; > -conv_123x_xxxB: as_u8(dst) = (shift_down(as_u32c(src), 8) & 0xff) ^ 0x80; goto CONV_END; > -conv_123x_xxB2: as_u16(dst) = bswap_16(shift_down(as_u32c(src), 8) ^ 0x80); goto CONV_END; > -conv_123x_xx2B: as_u16(dst) = (shift_down(as_u32c(src), 8) & 0xffff) ^ 0x80; goto CONV_END; > +conv_123x_0123: as_u32(dst) = as_u32c(src) >> 8; goto CONV_END; > +conv_123x_xxxB: as_u8(dst) = ((as_u32c(src) >> 8) & 0xff) ^ 0x80; goto CONV_END; > +conv_123x_xxB2: as_u16(dst) = bswap_16((as_u32c(src) >> 8) ^ 0x80); goto CONV_END; > +conv_123x_xx2B: as_u16(dst) = ((as_u32c(src) >> 8) & 0xffff) ^ 0x80; goto CONV_END; > conv_123x_xB21: as_u32(dst) = sx24(bswap_32(as_u32c(src)) ^ 0x800000); goto CONV_END; > conv_123x_12Bx: as_u32(dst) = sx24s(as_u32c(src) ^ 0x8000); goto CONV_END; > conv_123x_B210: as_u32(dst) = bswap_32(as_u32c(src) ^ 0x8000) << 8; goto CONV_END; > -conv_123x_012B: as_u32(dst) = shift_down(as_u32c(src), 8) ^ 0x80; goto CONV_END; > -conv_1234_xxx1: as_u8(dst) = shift_down(as_u32c(src), 24); goto CONV_END; > -conv_1234_xx12: as_u16(dst) = shift_down(as_u32c(src), 16); goto CONV_END; > -conv_1234_xx21: as_u16(dst) = bswap_16(shift_down(as_u32c(src), 16)); goto CONV_END; > -conv_1234_x123: as_u32(dst) = sx24(shift_down(as_u32c(src), 8)); goto CONV_END; > +conv_123x_012B: as_u32(dst) = (as_u32c(src) >> 8) ^ 0x80; goto CONV_END; > +conv_1234_xxx1: as_u8(dst) = as_u32c(src) >> 24; goto CONV_END; > +conv_1234_xx12: as_u16(dst) = as_u32c(src) >> 16; goto CONV_END; > +conv_1234_xx21: as_u16(dst) = bswap_16(as_u32c(src) >> 16); goto CONV_END; > +conv_1234_x123: as_u32(dst) = sx24(as_u32c(src) >> 8); goto CONV_END; > conv_1234_321x: as_u32(dst) = sx24s(bswap_32(as_u32c(src)) << 8); goto CONV_END; > conv_1234_1234: as_u32(dst) = as_u32c(src); goto CONV_END; > conv_1234_4321: as_u32(dst) = bswap_32(as_u32c(src)); goto CONV_END; > -conv_1234_xxx9: as_u8(dst) = shift_down(as_u32c(src), 24) ^ 0x80; goto CONV_END; > -conv_1234_xx92: as_u16(dst) = shift_down(as_u32c(src), 16) ^ 0x8000; goto CONV_END; > -conv_1234_xx29: as_u16(dst) = bswap_16(shift_down(as_u32c(src), 16)) ^ 0x80; goto CONV_END; > -conv_1234_x923: as_u32(dst) = sx24(shift_down(as_u32c(src), 8) ^ 0x800000); goto CONV_END; > +conv_1234_xxx9: as_u8(dst) = (as_u32c(src) >> 24) ^ 0x80; goto CONV_END; > +conv_1234_xx92: as_u16(dst) = (as_u32c(src) >> 16) ^ 0x8000; goto CONV_END; > +conv_1234_xx29: as_u16(dst) = bswap_16(as_u32c(src) >> 16) ^ 0x80; goto CONV_END; > +conv_1234_x923: as_u32(dst) = sx24((as_u32c(src) >> 8) ^ 0x800000); goto CONV_END; > conv_1234_329x: as_u32(dst) = sx24s((bswap_32(as_u32c(src)) ^ 0x80) << 8); goto CONV_END; > conv_1234_9234: as_u32(dst) = as_u32c(src) ^ 0x80000000; goto CONV_END; > conv_1234_4329: as_u32(dst) = bswap_32(as_u32c(src)) ^ 0x80; goto CONV_END; > conv_1234_xxx4: as_u8(dst) = as_u32c(src) & 0xff; goto CONV_END; > conv_1234_xx43: as_u16(dst) = bswap_16(as_u32c(src)); goto CONV_END; > conv_1234_xx34: as_u16(dst) = as_u32c(src) & 0xffff; goto CONV_END; > -conv_1234_x432: as_u32(dst) = sx24(shift_down(bswap_32(as_u32c(src)), 8)); goto CONV_END; > +conv_1234_x432: as_u32(dst) = sx24(bswap_32(as_u32c(src)) >> 8); goto CONV_END; > conv_1234_234x: as_u32(dst) = sx24s(as_u32c(src) << 8); goto CONV_END; > conv_1234_xxxC: as_u8(dst) = (as_u32c(src) & 0xff) ^ 0x80; goto CONV_END; > conv_1234_xxC3: as_u16(dst) = bswap_16(as_u32c(src) ^ 0x80); goto CONV_END; > conv_1234_xx3C: as_u16(dst) = (as_u32c(src) & 0xffff) ^ 0x80; goto CONV_END; > -conv_1234_xC32: as_u32(dst) = sx24(shift_down(bswap_32(as_u32c(src)), 8) ^ 0x800000); goto CONV_END; > +conv_1234_xC32: as_u32(dst) = sx24((bswap_32(as_u32c(src)) >> 8) ^ 0x800000); goto CONV_END; > conv_1234_23Cx: as_u32(dst) = sx24s((as_u32c(src) ^ 0x80) << 8); goto CONV_END; > conv_1234_C321: as_u32(dst) = bswap_32(as_u32c(src) ^ 0x80); goto CONV_END; > conv_1234_123C: as_u32(dst) = as_u32c(src) ^ 0x80; goto CONV_END; > @@ -384,26 +382,26 @@ get16_12_92: sample = as_u16c(src) ^ 0x8 > get16_12_92: sample = as_u16c(src) ^ 0x8000; goto GET16_END; > get16_12_21: sample = bswap_16(as_u16c(src)); goto GET16_END; > get16_12_A1: sample = bswap_16(as_u16c(src) ^ 0x80); goto GET16_END; > -get16_0123_12: sample = shift_down(as_u32c(src), 8); goto GET16_END; > -get16_0123_92: sample = shift_down(as_u32c(src), 8) ^ 0x8000; goto GET16_END; > -get16_1230_32: sample = bswap_16(shift_down(as_u32c(src), 8)); goto GET16_END; > -get16_1230_B2: sample = bswap_16(shift_down(as_u32c(src), 8) ^ 0x80); goto GET16_END; > -get16_1234_12: sample = shift_down(as_u32c(src), 16); goto GET16_END; > -get16_1234_92: sample = shift_down(as_u32c(src), 16) ^ 0x8000; goto GET16_END; > +get16_0123_12: sample = as_u32c(src) >> 8; goto GET16_END; > +get16_0123_92: sample = (as_u32c(src) >> 8) ^ 0x8000; goto GET16_END; > +get16_1230_32: sample = bswap_16(as_u32c(src) >> 8); goto GET16_END; > +get16_1230_B2: sample = bswap_16((as_u32c(src) >> 8) ^ 0x80); goto GET16_END; > +get16_1234_12: sample = as_u32c(src) >> 16; goto GET16_END; > +get16_1234_92: sample = (as_u32c(src) >> 16) ^ 0x8000; goto GET16_END; > get16_1234_43: sample = bswap_16(as_u32c(src)); goto GET16_END; > get16_1234_C3: sample = bswap_16(as_u32c(src) ^ 0x80); goto GET16_END; > -get16_123_12: sample = shift_down(_get_triple(src), 8); goto GET16_END; > -get16_123_92: sample = shift_down(_get_triple(src), 8) ^ 0x8000; goto GET16_END; > -get16_123_32: sample = shift_down(_get_triple_s(src), 8); goto GET16_END; > -get16_123_B2: sample = shift_down(_get_triple_s(src), 8) ^ 0x8000; goto GET16_END; > -get16_123_12_20: sample = shift_down(_get_triple(src), 4); goto GET16_END; > -get16_123_92_20: sample = shift_down(_get_triple(src), 4) ^ 0x8000; goto GET16_END; > -get16_123_32_20: sample = shift_down(_get_triple_s(src), 4); goto GET16_END; > -get16_123_B2_20: sample = shift_down(_get_triple_s(src), 4) ^ 0x8000; goto GET16_END; > -get16_123_12_18: sample = shift_down(_get_triple(src), 2); goto GET16_END; > -get16_123_92_18: sample = shift_down(_get_triple(src), 2) ^ 0x8000; goto GET16_END; > -get16_123_32_18: sample = shift_down(_get_triple_s(src), 2); goto GET16_END; > -get16_123_B2_18: sample = shift_down(_get_triple_s(src), 2) ^ 0x8000; goto GET16_END; > +get16_123_12: sample = _get_triple(src) >> 8; goto GET16_END; > +get16_123_92: sample = (_get_triple(src) >> 8) ^ 0x8000; goto GET16_END; > +get16_123_32: sample = _get_triple_s(src) >> 8; goto GET16_END; > +get16_123_B2: sample = (_get_triple_s(src) >> 8) ^ 0x8000; goto GET16_END; > +get16_123_12_20: sample = _get_triple(src) >> 4; goto GET16_END; > +get16_123_92_20: sample = (_get_triple(src) >> 4) ^ 0x8000; goto GET16_END; > +get16_123_32_20: sample = _get_triple_s(src) >> 4; goto GET16_END; > +get16_123_B2_20: sample = (_get_triple_s(src) >> 4) ^ 0x8000; goto GET16_END; > +get16_123_12_18: sample = _get_triple(src) >> 2; goto GET16_END; > +get16_123_92_18: sample = (_get_triple(src) >> 2) ^ 0x8000; goto GET16_END; > +get16_123_32_18: sample = _get_triple_s(src) >> 2; goto GET16_END; > +get16_123_B2_18: sample = (_get_triple_s(src) >> 2) ^ 0x8000; goto GET16_END; > } > #endif > > @@ -431,8 +429,8 @@ static void *put16_labels[4 * 2 * 2] = { > > #ifdef PUT16_END > while (0) { > -put16_12_1: as_u8(dst) = shift_down(sample, 8); goto PUT16_END; > -put16_12_9: as_u8(dst) = shift_down(sample, 8) ^ 0x80; goto PUT16_END; > +put16_12_1: as_u8(dst) = sample >> 8; goto PUT16_END; > +put16_12_9: as_u8(dst) = (sample >> 8) ^ 0x80; goto PUT16_END; > put16_12_12: as_u16(dst) = sample; goto PUT16_END; > put16_12_92: as_u16(dst) = sample ^ 0x8000; goto PUT16_END; > put16_12_21: as_u16(dst) = bswap_16(sample); goto PUT16_END; > @@ -502,8 +500,8 @@ get32_12_A100: sample = (u_int32_t)bswap > get32_12_A100: sample = (u_int32_t)bswap_16(as_u16c(src) ^ 0x80) << 16; goto GET32_END; > get32_0123_1230: sample = as_u32c(src) << 8; goto GET32_END; > get32_0123_9230: sample = (as_u32c(src) << 8) ^ 0x80000000; goto GET32_END; > -get32_1230_3210: sample = bswap_32(shift_down(as_u32c(src), 8)); goto GET32_END; > -get32_1230_B210: sample = bswap_32(shift_down(as_u32c(src), 8) ^ 0x80); goto GET32_END; > +get32_1230_3210: sample = bswap_32(as_u32c(src) >> 8); goto GET32_END; > +get32_1230_B210: sample = bswap_32((as_u32c(src) >> 8) ^ 0x80); goto GET32_END; > get32_1234_1234: sample = as_u32c(src); goto GET32_END; > get32_1234_9234: sample = as_u32c(src) ^ 0x80000000; goto GET32_END; > get32_1234_4321: sample = bswap_32(as_u32c(src)); goto GET32_END; > @@ -570,32 +568,32 @@ static void *put32_labels[4 * 2 * 2 + 4 > > #ifdef PUT32_END > while (0) { > -put32_1234_1: as_u8(dst) = shift_down(sample, 24); goto PUT32_END; > -put32_1234_9: as_u8(dst) = shift_down(sample, 24) ^ 0x80; goto PUT32_END; > -put32_1234_12: as_u16(dst) = shift_down(sample, 16); goto PUT32_END; > -put32_1234_92: as_u16(dst) = shift_down(sample, 16) ^ 0x8000; goto PUT32_END; > -put32_1234_21: as_u16(dst) = bswap_16(shift_down(sample, 16)); goto PUT32_END; > -put32_1234_29: as_u16(dst) = bswap_16(shift_down(sample, 16)) ^ 0x80; goto PUT32_END; > -put32_1234_0123: as_u32(dst) = sx24(shift_down(sample, 8)); goto PUT32_END; > -put32_1234_0923: as_u32(dst) = sx24(shift_down(sample, 8) ^ 0x800000); goto PUT32_END; > +put32_1234_1: as_u8(dst) = sample >> 24; goto PUT32_END; > +put32_1234_9: as_u8(dst) = (sample >> 24) ^ 0x80; goto PUT32_END; > +put32_1234_12: as_u16(dst) = sample >> 16; goto PUT32_END; > +put32_1234_92: as_u16(dst) = (sample >> 16) ^ 0x8000; goto PUT32_END; > +put32_1234_21: as_u16(dst) = bswap_16(sample >> 16); goto PUT32_END; > +put32_1234_29: as_u16(dst) = bswap_16(sample >> 16) ^ 0x80; goto PUT32_END; > +put32_1234_0123: as_u32(dst) = sx24(sample >> 8); goto PUT32_END; > +put32_1234_0923: as_u32(dst) = sx24((sample >> 8) ^ 0x800000); goto PUT32_END; > put32_1234_3210: as_u32(dst) = sx24s(bswap_32(sample) << 8); goto PUT32_END; > put32_1234_3290: as_u32(dst) = sx24s((bswap_32(sample) ^ 0x80) << 8); goto PUT32_END; > put32_1234_1234: as_u32(dst) = sample; goto PUT32_END; > put32_1234_9234: as_u32(dst) = sample ^ 0x80000000; goto PUT32_END; > put32_1234_4321: as_u32(dst) = bswap_32(sample); goto PUT32_END; > put32_1234_4329: as_u32(dst) = bswap_32(sample) ^ 0x80; goto PUT32_END; > -put32_1234_123: _put_triple(dst, shift_down(sample, 8)); goto PUT32_END; > -put32_1234_923: _put_triple(dst, shift_down(sample ^ 0x80000000, 8)); goto PUT32_END; > -put32_1234_321: _put_triple_s(dst, shift_down(sample, 8)); goto PUT32_END; > -put32_1234_329: _put_triple_s(dst, shift_down(sample ^ 0x80000000, 8)); goto PUT32_END; > -put32_1234_123_20: _put_triple(dst, shift_down(sample, 12)); goto PUT32_END; > -put32_1234_923_20: _put_triple(dst, shift_down(sample ^ 0x80000000, 12)); goto PUT32_END; > -put32_1234_321_20: _put_triple_s(dst, shift_down(sample, 12)); goto PUT32_END; > -put32_1234_329_20: _put_triple_s(dst, shift_down(sample ^ 0x80000000, 12)); goto PUT32_END; > -put32_1234_123_18: _put_triple(dst, shift_down(sample, 14)); goto PUT32_END; > -put32_1234_923_18: _put_triple(dst, shift_down(sample ^ 0x80000000, 14)); goto PUT32_END; > -put32_1234_321_18: _put_triple_s(dst, shift_down(sample, 14)); goto PUT32_END; > -put32_1234_329_18: _put_triple_s(dst, shift_down(sample ^ 0x80000000, 14)); goto PUT32_END; > +put32_1234_123: _put_triple(dst, sample >> 8); goto PUT32_END; > +put32_1234_923: _put_triple(dst, (sample ^ 0x80000000) >> 8); goto PUT32_END; > +put32_1234_321: _put_triple_s(dst, sample >> 8); goto PUT32_END; > +put32_1234_329: _put_triple_s(dst, (sample ^ 0x80000000) >> 8); goto PUT32_END; > +put32_1234_123_20: _put_triple(dst, sample >> 12); goto PUT32_END; > +put32_1234_923_20: _put_triple(dst, (sample ^ 0x80000000) >> 12); goto PUT32_END; > +put32_1234_321_20: _put_triple_s(dst, sample >> 12); goto PUT32_END; > +put32_1234_329_20: _put_triple_s(dst, (sample ^ 0x80000000) >> 12); goto PUT32_END; > +put32_1234_123_18: _put_triple(dst, sample >> 14); goto PUT32_END; > +put32_1234_923_18: _put_triple(dst, (sample ^ 0x80000000) >> 14); goto PUT32_END; > +put32_1234_321_18: _put_triple_s(dst, sample >> 14); goto PUT32_END; > +put32_1234_329_18: _put_triple_s(dst, (sample ^ 0x80000000) >> 14); goto PUT32_END; > } > #endif > > @@ -1081,4 +1079,3 @@ norms_32_s32s: _norms(src, dst, 32, 1, > #undef _put_triple_le > #undef _put_triple_be > > -#undef shift_down -- Excellent day for drinking heavily. Spike the office water cooler;-)

2 1

[alsa-devel] intel-hda patch for sigmatel codec volume knob
by zhejiang 15 Oct '07

15 Oct '07

Hi, I found that STAC_VOLKNOB hardwired the KNOB nid to 0x24. It is okay for stac9205 and stac927x. But the VolumeKnob nid of stac9220-9221 is 0x16. diff -Nur a/alsa-driver-1.0.15rc3/alsa-kernel/pci/hda/patch_sigmatel.c b/alsa-driver-1.0.15rc3/alsa-kernel/pci/hda/patch_sigmatel.c --- a/alsa-driver-1.0.15rc3/alsa-kernel/pci/hda/patch_sigmatel.c 2007-09-21 16:41:28.000000000 +0800 +++ b/alsa-driver-1.0.15rc3/alsa-kernel/pci/hda/patch_sigmatel.c 2007-10-15 15:13:14.000000000 +0800 @@ -351,7 +351,7 @@ static int stac92xx_volknob_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - ucontrol->value.integer.value[0] = kcontrol->private_value; + ucontrol->value.integer.value[0] = kcontrol->private_value & 0xFFFF; return 0; } @@ -360,14 +360,15 @@ { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - if (kcontrol->private_value == ucontrol->value.integer.value[0]) + if ((kcontrol->private_value & 0xFFFF) == ucontrol->value.integer.value[0]) return 0; - kcontrol->private_value = ucontrol->value.integer.value[0]; + kcontrol->private_value &= 0xFFFF0000; + kcontrol->private_value += ucontrol->value.integer.value[0]; - snd_hda_codec_write_cache(codec, 0x24, 0, + snd_hda_codec_write_cache(codec, kcontrol->private_value >> 16, 0, AC_VERB_SET_VOLUME_KNOB_CONTROL, - kcontrol->private_value | 0x80); + (kcontrol->private_value & 0xFFFF) | 0x80); return 1; } @@ -433,7 +434,7 @@ .private_value = verb_read | (verb_write << 16), \ } -#define STAC_VOLKNOB \ +#define STAC_VOLKNOB(knob_nid) \ { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = "Master Playback Volume", \ @@ -441,7 +442,7 @@ .info = stac92xx_volknob_info, \ .get = stac92xx_volknob_get, \ .put = stac92xx_volknob_put, \ - .private_value = 127, \ + .private_value = 127 | (knob_nid << 16), \ } @@ -474,7 +475,7 @@ }, STAC_INPUT_SOURCE(2), STAC_ANALOG_LOOPBACK(0xFE0, 0x7E0), - STAC_VOLKNOB, + STAC_VOLKNOB(0x24), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1b, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1d, 0x0, HDA_OUTPUT), @@ -490,7 +491,7 @@ /* This needs to be generated dynamically based on sequence */ static struct snd_kcontrol_new stac922x_mixer[] = { STAC_INPUT_SOURCE(2), - STAC_VOLKNOB, + STAC_VOLKNOB(0x16), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x17, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x17, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x0, 0x12, 0x0, HDA_OUTPUT), @@ -504,7 +505,7 @@ static struct snd_kcontrol_new stac927x_mixer[] = { STAC_INPUT_SOURCE(3), - STAC_VOLKNOB, + STAC_VOLKNOB(0x24), STAC_ANALOG_LOOPBACK(0xFEB, 0x7EB), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x18, 0x0, HDA_INPUT),

3 2

[alsa-devel] Fwd: FSC Esprimo Mobile V5505 - info, partly working
by Jan-Simon Möller 13 Oct '07

13 Oct '07

Hi ! I got a new FSC Esprimo Mobile V5505 (full system info attached). When i use "options snd-hda-intel model=fujitsu-c1410 enable=1 index=0" sound works, but it sounds not good and smooth. Chip is a conexant afaik. Attachements: * Full lspci -vv -nn Whats needed to improve sound and make the mixer available ? If any other data is needed -> please ping ! Sincerely yours Jan-Simon Möller dl9pf(a)gmx.de

2 3

[alsa-devel] Patch for Intel-HDA ALC262 on FSC Lifebook C1410
by Veli-Matti Lintu 12 Oct '07

12 Oct '07

Hello, After error and trial I got sound working on Fujitsu-Siemens Computers Lifebook C1410. Attached is a patch that works for me, but probably needs some cleaning. The laptop uses an ALC262 chip that is also used on other FSC models, but the needed settings seem to be somewhat different from the existing Fujitsu-settings in patch_realtek.c. If I understand correctly, the chips use same subsystem ids, so the patch does not automatically detect the correct chipset. Maybe someone more knowledgeable can help with this? The patch names the internal mic as Int Mic and external mic that can be plugged in as Ext Mic. There's some more background information in Ubuntu's Launchpad: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/125790 The patch is tested with kernel 2.6.22.9. I am not really familiar with alsa development, so I cannot say which version of the alsa drivers that kernel version uses. The environment where the testing was done is running Ubuntu 7.04 with alsa version 1.0.13. Only updated component is the kernel. The patch needs model=fujitsu-c1410 parameter to be passed to the module: /etc/modprobe.d/alsa-base: options snd-hda-intel model=fujitsu-c1410 The subsystem 10cf:1397 is mapped in patch_realtek.c alc262_cfg_tbl to ALC262_FUJITSU and ALC262_FUJITSU_C1410 would need to be mapped to same ids: ------------------------------------------------------------------- sound/pci/hda/patch_realtek.c: SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU), ------------------------------------------------------------------- lspci -nv (on C1410) 00:1b.0 0403: 8086:27d8 (rev 02) Subsystem: 10cf:1397 Flags: bus master, fast devsel, latency 0, IRQ 22 Memory at f0640000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Capabilities: [70] Express Unknown type IRQ 0 Capabilities: [100] Virtual Channel Capabilities: [130] Unknown (5) ------------------------------------------------------------------- I hope someone with more knowledge about alsa drivers could have a look at this. The patch has been in internal use for a few weeks and people seem to be happy with it, so I hope also others can get their sound working. The patch and hw information follow: --------------------------------------------------------------------------------------------- --- linux-2.6.22.9/sound/pci/hda/patch_realtek.c.orig 2007-09-26 21:03:01.000000000 +0300 +++ linux-2.6.22.9/sound/pci/hda/patch_realtek.c 2007-10-07 13:28:12.000000000 +0300 @@ -89,6 +89,7 @@ ALC262_HIPPO, ALC262_HIPPO_1, ALC262_FUJITSU, + ALC262_FUJITSU_C1410, ALC262_HP_BPC, ALC262_HP_BPC_D7000_WL, ALC262_HP_BPC_D7000_WF, @@ -7068,6 +7069,15 @@ }, }; +static struct hda_input_mux alc262_fujitsu_c1410_capture_source = { + .num_items = 3, + .items = { + { "Ext Mic", 0x0 }, + { "Int Mic", 0x1 }, + { "CD", 0x4 }, + }, +}; + static struct hda_input_mux alc262_HP_capture_source = { .num_items = 5, .items = { @@ -7182,6 +7192,43 @@ { } /* end */ }; +static struct snd_kcontrol_new alc262_fujitsu_c1410_mixer[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Volume", + .info = snd_hda_mixer_amp_volume_info, + .get = snd_hda_mixer_amp_volume_get, + .put = alc262_fujitsu_master_vol_put, + .tlv = { .c = snd_hda_mixer_amp_tlv }, + .private_value = HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, HDA_OUTPUT), + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = snd_hda_mixer_amp_switch_info, + .get = snd_hda_mixer_amp_switch_get, + .put = alc262_fujitsu_master_sw_put, + .private_value = HDA_COMPOSE_AMP_VAL(0x0d, 3, 0, HDA_OUTPUT), + }, + + HDA_CODEC_VOLUME("Headphones Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphones Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x15, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME("Ext Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT), + + HDA_CODEC_VOLUME("Ext Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Ext Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + + HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), + { } /* end */ +}; + /* additional init verbs for Benq laptops */ static struct hda_verb alc262_EAPD_verbs[] = { {0x20, AC_VERB_SET_COEF_INDEX, 0x07}, @@ -7581,6 +7628,7 @@ [ALC262_HIPPO] = "hippo", [ALC262_HIPPO_1] = "hippo_1", [ALC262_FUJITSU] = "fujitsu", + [ALC262_FUJITSU_C1410] = "fujitsu-c1410", [ALC262_HP_BPC] = "hp-bpc", [ALC262_HP_BPC_D7000_WL]= "hp-bpc-d7000", [ALC262_BENQ_ED8] = "benq", @@ -7659,6 +7707,18 @@ .input_mux = &alc262_fujitsu_capture_source, .unsol_event = alc262_fujitsu_unsol_event, }, + [ALC262_FUJITSU_C1410] = { + .mixers = { alc262_fujitsu_c1410_mixer }, + .init_verbs = { alc262_init_verbs, alc262_EAPD_verbs, alc262_fujitsu_unsol_verbs }, + .num_dacs = ARRAY_SIZE(alc262_dac_nids), + .dac_nids = alc262_dac_nids, + .hp_nid = 0x03, + .dig_out_nid = ALC262_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc262_modes), + .channel_mode = alc262_modes, + .input_mux = &alc262_fujitsu_c1410_capture_source, + .unsol_event = alc262_fujitsu_unsol_event, + }, [ALC262_HP_BPC] = { .mixers = { alc262_HP_BPC_mixer }, .init_verbs = { alc262_HP_BPC_init_verbs }, --------------------------------------------------------------------------------------------- ALSA Audio Debug v0.1.0 - su 7.10.2007 15:13:09 +0300 http://alsa.opensrc.org/aadebug http://www.gnu.org/licenses/gpl.txt Kernel ---------------------------------------------------- Linux oppilas 2.6.22.9-c1410 #2 SMP Sun Oct 7 11:57:15 EEST 2007 i686 GNU/Linux Loaded Modules -------------------------------------------- snd_hda_intel 257560 1 snd_pcm_oss 44928 0 snd_mixer_oss 17664 1 snd_pcm_oss snd_pcm 80772 2 snd_hda_intel,snd_pcm_oss snd_seq_dummy 4740 0 snd_seq_oss 33408 0 snd_seq_midi 9600 0 snd_rawmidi 25856 1 snd_seq_midi snd_seq_midi_event 8448 2 snd_seq_oss,snd_seq_midi snd_seq 53360 6 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_seq_midi_event snd_timer 24196 2 snd_pcm,snd_seq snd_seq_device 9356 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_rawmidi,snd_seq snd 54916 11 snd_hda_intel,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_seq_oss,snd_rawmidi,snd_seq,snd_timer,snd_seq_device snd_page_alloc 11400 2 snd_hda_intel,snd_pcm Modprobe Conf --------------------------------------------- Warning: module config file does not exist This means any kernel modules will not be auto loaded See your linux distro docs on how to create this file Proc Asound ----------------------------------------------- Advanced Linux Sound Architecture Driver Version 1.0.14 (Thu May 31 09:03:25 2007 UTC). 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xf0640000 irq 22 2: : timer 3: : sequencer 4: [ 0- 6]: digital audio playback 5: [ 0- 6]: digital audio capture 6: [ 0- 2]: digital audio capture 7: [ 0- 1]: digital audio playback 8: [ 0- 0]: digital audio playback 9: [ 0- 0]: digital audio capture 10: [ 0] : control 00-06: Si3054 Modem : Si3054 Modem : playback 1 : capture 1 00-02: ALC262 Analog : ALC262 Analog : capture 2 00-01: ALC262 Digital : ALC262 Digital : playback 1 00-00: ALC262 Analog : ALC262 Analog : playback 1 : capture 2 Client info cur clients : 3 peak clients : 3 max clients : 192 Client 0 : "System" [Kernel] Port 0 : "Timer" (Rwe-) Port 1 : "Announce" (R-e-) Connecting To: 15:0 Client 14 : "Midi Through" [Kernel] Port 0 : "Midi Through Port-0" (RWe-) Client 15 : "OSS sequencer" [Kernel] Port 0 : "Receiver" (-we-) Connected From: 0:1 Dev Snd --------------------------------------------------- controlC0 pcmC0D0p pcmC0D2c pcmC0D6p timer pcmC0D0c pcmC0D1p pcmC0D6c seq CPU ------------------------------------------------------- model name : Intel(R) Core(TM)2 CPU T5600 @ 1.83GHz cpu MHz : 1000.000 model name : Intel(R) Core(TM)2 CPU T5600 @ 1.83GHz cpu MHz : 1000.000 RAM ------------------------------------------------------- MemTotal: 1026252 kB SwapTotal: 2441872 kB Hardware -------------------------------------------------- 00:00.0 Host bridge: Intel Corporation Mobile 945GM/PM/GMS/940GML and 945GT Express Memory Controller Hub (rev 03) 00:02.0 VGA compatible controller: Intel Corporation Mobile 945GM/GMS/940GML Express Integrated Graphics Controller (rev 03) 00:02.1 Display controller: Intel Corporation Mobile 945GM/GMS/940GML Express Integrated Graphics Controller (rev 03) 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 02) 00:1c.0 PCI bridge: Intel Corporation 82801G (ICH7 Family) PCI Express Port 1 (rev 02) 00:1c.1 PCI bridge: Intel Corporation 82801G (ICH7 Family) PCI Express Port 2 (rev 02) 00:1c.2 PCI bridge: Intel Corporation 82801G (ICH7 Family) PCI Express Port 3 (rev 02) 00:1d.0 USB Controller: Intel Corporation 82801G (ICH7 Family) USB UHCI #1 (rev 02) 00:1d.1 USB Controller: Intel Corporation 82801G (ICH7 Family) USB UHCI #2 (rev 02) 00:1d.2 USB Controller: Intel Corporation 82801G (ICH7 Family) USB UHCI #3 (rev 02) 00:1d.3 USB Controller: Intel Corporation 82801G (ICH7 Family) USB UHCI #4 (rev 02) 00:1d.7 USB Controller: Intel Corporation 82801G (ICH7 Family) USB2 EHCI Controller (rev 02) 00:1e.0 PCI bridge: Intel Corporation 82801 Mobile PCI Bridge (rev e2) 00:1f.0 ISA bridge: Intel Corporation 82801GBM (ICH7-M) LPC Interface Bridge (rev 02) 00:1f.1 IDE interface: Intel Corporation 82801G (ICH7 Family) IDE Controller (rev 02) 00:1f.2 SATA controller: Intel Corporation 82801GBM/GHM (ICH7 Family) Serial ATA Storage Controller AHCI (rev 02) 00:1f.3 SMBus: Intel Corporation 82801G (ICH7 Family) SMBus Controller (rev 02) 02:00.0 Ethernet controller: Marvell Technology Group Ltd. 88E8055 PCI-E Gigabit Ethernet Controller (rev 12) 05:00.0 Network controller: Intel Corporation PRO/Wireless 3945ABG Network Connection (rev 02) 08:03.0 CardBus bridge: Ricoh Co Ltd RL5c476 II (rev b3) 08:03.1 FireWire (IEEE 1394): Ricoh Co Ltd R5C552 IEEE 1394 Controller (rev 08) 08:03.2 Generic system peripheral [0805]: Ricoh Co Ltd R5C822 SD/SDIO/MMC/MS/MSPro Host Adapter (rev 17) 00:00.0 0600: 8086:27a0 (rev 03) Subsystem: 10cf:1378 Flags: bus master, fast devsel, latency 0 Capabilities: [e0] Vendor Specific Information 00:02.0 0300: 8086:27a2 (rev 03) (prog-if 00 [VGA]) Subsystem: 10cf:1381 Flags: bus master, fast devsel, latency 0, IRQ 19 Memory at f0300000 (32-bit, non-prefetchable) [size=512K] I/O ports at 1800 [size=8] Memory at e0000000 (32-bit, prefetchable) [size=256M] Memory at f0400000 (32-bit, non-prefetchable) [size=256K] Capabilities: [90] Message Signalled Interrupts: Mask- 64bit- Queue=0/0 Enable- Capabilities: [d0] Power Management version 2 00:02.1 0380: 8086:27a6 (rev 03) Subsystem: 10cf:1381 Flags: bus master, fast devsel, latency 0 Memory at f0380000 (32-bit, non-prefetchable) [size=512K] Capabilities: [d0] Power Management version 2 00:1b.0 0403: 8086:27d8 (rev 02) Subsystem: 10cf:1397 Flags: bus master, fast devsel, latency 0, IRQ 22 Memory at f0640000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Capabilities: [70] Express Unknown type IRQ 0 Capabilities: [100] Virtual Channel Capabilities: [130] Unknown (5) 00:1c.0 0604: 8086:27d0 (rev 02) (prog-if 00 [Normal decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=02, subordinate=02, sec-latency=0 I/O behind bridge: 00002000-00002fff Memory behind bridge: f0000000-f00fffff Prefetchable memory behind bridge: 0000000054000000-00000000540fffff Capabilities: [40] Express Root Port (Slot+) IRQ 0 Capabilities: [80] Message Signalled Interrupts: Mask- 64bit- Queue=0/0 Enable+ Capabilities: [90] Subsystem: 10cf:138b Capabilities: [a0] Power Management version 2 Capabilities: [100] Virtual Channel Capabilities: [180] Unknown (5) 00:1c.1 0604: 8086:27d2 (rev 02) (prog-if 00 [Normal decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=03, subordinate=04, sec-latency=0 Capabilities: [40] Express Root Port (Slot+) IRQ 0 Capabilities: [80] Message Signalled Interrupts: Mask- 64bit- Queue=0/0 Enable+ Capabilities: [90] Subsystem: 10cf:138b Capabilities: [a0] Power Management version 2 Capabilities: [100] Virtual Channel Capabilities: [180] Unknown (5) 00:1c.2 0604: 8086:27d4 (rev 02) (prog-if 00 [Normal decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=05, subordinate=07, sec-latency=0 Memory behind bridge: 54100000-541fffff Capabilities: [40] Express Root Port (Slot+) IRQ 0 Capabilities: [80] Message Signalled Interrupts: Mask- 64bit- Queue=0/0 Enable+ Capabilities: [90] Subsystem: 10cf:138b Capabilities: [a0] Power Management version 2 Capabilities: [100] Virtual Channel Capabilities: [180] Unknown (5) 00:1d.0 0c03: 8086:27c8 (rev 02) (prog-if 00 [UHCI]) Subsystem: 10cf:1389 Flags: bus master, medium devsel, latency 0, IRQ 20 I/O ports at 1820 [size=32] 00:1d.1 0c03: 8086:27c9 (rev 02) (prog-if 00 [UHCI]) Subsystem: 10cf:1389 Flags: bus master, medium devsel, latency 0, IRQ 18 I/O ports at 1840 [size=32] 00:1d.2 0c03: 8086:27ca (rev 02) (prog-if 00 [UHCI]) Subsystem: 10cf:1389 Flags: bus master, medium devsel, latency 0, IRQ 21 I/O ports at 1860 [size=32] 00:1d.3 0c03: 8086:27cb (rev 02) (prog-if 00 [UHCI]) Subsystem: 10cf:1389 Flags: bus master, medium devsel, latency 0, IRQ 19 I/O ports at 1880 [size=32] 00:1d.7 0c03: 8086:27cc (rev 02) (prog-if 20 [EHCI]) Subsystem: 10cf:138a Flags: bus master, medium devsel, latency 0, IRQ 20 Memory at f0644000 (32-bit, non-prefetchable) [size=1K] Capabilities: [50] Power Management version 2 Capabilities: [58] Debug port 00:1e.0 0604: 8086:2448 (rev e2) (prog-if 01 [Subtractive decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=08, subordinate=0c, sec-latency=32 I/O behind bridge: 00003000-00003fff Memory behind bridge: f0200000-f02fffff Prefetchable memory behind bridge: 0000000050000000-0000000053ffffff Capabilities: [50] Subsystem: 10cf:1383 00:1f.0 0601: 8086:27b9 (rev 02) Subsystem: 10cf:1384 Flags: bus master, medium devsel, latency 0 Capabilities: [e0] Vendor Specific Information 00:1f.1 0101: 8086:27df (rev 02) (prog-if 8a [Master SecP PriP]) Subsystem: 10cf:1385 Flags: bus master, medium devsel, latency 0, IRQ 21 I/O ports at 01f0 [size=8] I/O ports at 03f4 [size=1] I/O ports at 0170 [size=8] I/O ports at 0374 [size=1] I/O ports at 1810 [size=16] 00:1f.2 0106: 8086:27c5 (rev 02) (prog-if 01 [AHCI 1.0]) Subsystem: 10cf:1387 Flags: bus master, 66MHz, medium devsel, latency 0, IRQ 220 I/O ports at 18d0 [size=8] I/O ports at 18c4 [size=4] I/O ports at 18c8 [size=8] I/O ports at 18c0 [size=4] I/O ports at 18b0 [size=16] Memory at f0644400 (32-bit, non-prefetchable) [size=1K] Capabilities: [80] Message Signalled Interrupts: Mask- 64bit- Queue=0/0 Enable+ Capabilities: [70] Power Management version 2 00:1f.3 0c05: 8086:27da (rev 02) Subsystem: 10cf:1388 Flags: medium devsel, IRQ 11 I/O ports at 18e0 [size=32] 02:00.0 0200: 11ab:4363 (rev 12) Subsystem: 10cf:139a Flags: bus master, fast devsel, latency 0, IRQ 219 Memory at f0000000 (64-bit, non-prefetchable) [size=16K] I/O ports at 2000 [size=256] [virtual] Expansion ROM at 54000000 [disabled] [size=128K] Capabilities: [48] Power Management version 3 Capabilities: [50] Vital Product Data Capabilities: [5c] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable+ Capabilities: [e0] Express Legacy Endpoint IRQ 0 Capabilities: [100] Advanced Error Reporting 05:00.0 0280: 8086:4222 (rev 02) Subsystem: 8086:1001 Flags: bus master, fast devsel, latency 0, IRQ 21 Memory at 54100000 (32-bit, non-prefetchable) [size=4K] Capabilities: [c8] Power Management version 2 Capabilities: [d0] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Capabilities: [e0] Express Legacy Endpoint IRQ 0 Capabilities: [100] Advanced Error Reporting Capabilities: [140] Device Serial Number b5-ec-71-ff-ff-77-1b-00 08:03.0 0607: 1180:0476 (rev b3) Subsystem: 10cf:1254 Flags: bus master, medium devsel, latency 168, IRQ 19 Memory at f0201000 (32-bit, non-prefetchable) [size=4K] Bus: primary=08, secondary=09, subordinate=0c, sec-latency=176 Memory window 0: 50000000-53fff000 (prefetchable) Memory window 1: 58000000-5bfff000 I/O window 0: 00003000-000030ff I/O window 1: 00003400-000034ff 16-bit legacy interface ports at 0001 08:03.1 0c00: 1180:0552 (rev 08) (prog-if 10 [OHCI]) Subsystem: 10cf:1255 Flags: bus master, medium devsel, latency 32, IRQ 22 Memory at f0200000 (32-bit, non-prefetchable) [size=2K] Capabilities: [dc] Power Management version 2 08:03.2 0805: 1180:0822 (rev 17) Subsystem: 10cf:1323 Flags: bus master, medium devsel, latency 32, IRQ 19 Memory at f0200800 (32-bit, non-prefetchable) [size=256] Capabilities: [80] Power Management version 2 --------------------------------------------------------------------------------------------- Veli-Matti Lintu

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[alsa-devel] snd-hda-intel: no sound on ASUS F3Sa
by Luca Tettamanti 11 Oct '07

11 Oct '07

Hi, I can't enable sound on a (new) ASUS F3Sa laptop with this sound device: 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03) Subsystem: ASUSTeK Computer Inc. Unknown device 1339 Control: I/O- Mem+ BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- Interrupt: pin A routed to IRQ 22 Region 0: Memory at febf8000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Address: 0000000000000000 Data: 0000 Capabilities: [70] Express Unknown type IRQ 0 Device: Supported: MaxPayload 128 bytes, PhantFunc 0, ExtTag- Device: Latency L0s <64ns, L1 <1us Device: Errors: Correctable- Non-Fatal- Fatal- Unsupported- Device: RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ Device: MaxPayload 128 bytes, MaxReadReq 128 bytes Link: Supported Speed unknown, Width x0, ASPM unknown, Port 0 Link: Latency L0s <64ns, L1 <1us Link: ASPM Disabled CommClk- ExtSynch- Link: Speed unknown, Width x0 Codec: kronos@nb:~$ cat /proc/asound/card0/codec#0 | head -n1 Codec: Realtek ALC660-VD (full output attached) without arguments I get: ACPI: PCI Interrupt 0000:00:1b.0[A] -> GSI 22 (level, low) -> IRQ 22 PCI: Setting latency timer of device 0000:00:1b.0 to 64 codec_mask = 0x3 hda_codec: model '3stack-660' is selected for config 1043:1339 (Asus G1) while with model=auto: ACPI: PCI Interrupt 0000:00:1b.0[A] -> GSI 22 (level, low) -> IRQ 22 PCI: Setting latency timer of device 0000:00:1b.0 to 64 codec_mask = 0x3 hda_codec: model 'auto' is selected autoconfig: line_outs=1 (0x14/0x0/0x0/0x0/0x0) speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) hp_outs=1 (0x1b/0x0/0x0/0x0/0x0) inputs: mic=0x18, fmic=0x19, line=0x0, fline=0x0, cd=0x0, aux=0x0 I also tried all other possible models without success; I can't get any output on the built-in speakers nor on the headphones. I tried both 2.6.23-git (from a couple of days ago) and ALSA 1.0.15rc3. Luca (CC me, I'm not subscribed) -- "Sei l'unica donna della mia vita". (Adamo)

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[alsa-devel] How to enforce 32bit access?
by Joachim Foerster 11 Oct '07

11 Oct '07

Hi ALSA devs, (based on a Xilinx ML403, PPC405, OPB bus) I have a custom sound controller, which allows 32bit write accesses to its IO memory, only. The controller supports S16_LE, S16_BE, U16_LE, U16_BE with min. and max. 2 channels (stereo), only! => So a frame has to be 32bits wide and one frame has to be written at one point of time (= one 32bit bus access). Now, playing a 16kHz, S16_LE, Mono .wav-file with aplay invokes the conversion framework of the alsa-lib. The following line of code (of pcm_plugin.h) is executed: conv_xx12_xx12: as_u16(dst) = as_u16c(src); This is a 16bit access and thus fails with a Bus error/Kernel Oops. Therefore my question is: In my driver for the controller, how do I tell ALSA, that it is only allowed to write in 32bit quantities? Is this possible at all? Or do I have to use a special .asoundrc file? Thanks, Joachim PS: 16bit, signed/unsigned, LE/BE, stereo (!) .wav files can be played without any problem.

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[alsa-devel] alsa 1.0.15
by paul blakeley 11 Oct '07

11 Oct '07

Can someone please tell me when version 1.0.15 will be released and then become the current stable package?

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