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

2 3

[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)

1 0

[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?

4 4

[alsa-devel] dont know if bug in /tmp/217usbkeyboardbug.log
by linuxcbon 11 Oct '07

11 Oct '07

hi, I have no idea if this is a bug : tv card is hauppauge wintv model 44346, tuner model is Philips FI1216MF MK2 I found in /tmp/217usbkeyboardbug.log Last 20 lines of dmesg: tveeprom 2-0050: has radio bttv0: Hauppauge eeprom indicates model#44346 bttv0: using tuner=3 bttv0: i2c: checking for MSP34xx @ 0x80... found msp3400 2-0040: MSP3415G-B8 found @ 0x80 (bt878 #0 [sw]) msp3400 2-0040: MSP3415G-B8 supports nicam and radio, mode is autodetect and autoselect bttv0: i2c: checking for TDA9875 @ 0xb0... not found bttv0: i2c: checking for TDA7432 @ 0x8a... not found bttv0: i2c: checking for TDA9887 @ 0x86... not found tuner 2-0061: chip found @ 0xc2 (bt878 #0 [sw]) tuner 2-0061: type set to 3 (Philips (SECAM+PAL_BG) (FI1216MF, FM1216MF, FR1216MF)) tuner 2-0061: type set to 3 (Philips (SECAM+PAL_BG) (FI1216MF, FM1216MF, FR1216MF)) bttv0: registered device video0 bttv0: registered device vbi0 bttv0: registered device radio0 bttv0: PLL: 28636363 => 35468950 .. ok ALSA /mnt/hda3/sources/sources116/alsa-driver-1.0.15rc2/pci/bt87x.c:872: unknown card 0x878-0x70:0x13eb ALSA /mnt/hda3/sources/sources116/alsa-driver-1.0.15rc2/pci/bt87x.c:875: please mail id, board name, and, if it works, the correct digital_rate option to <alsa-devel(a)alsa-project.org> ACPI: PCI Interrupt 0000:01:04.1[A] -> Link [LNK1] -> GSI 5 (level, low) -> IRQ 5 ALSA /mnt/hda3/sources/sources116/alsa-driver-1.0.15rc2/pci/bt87x.c:943: bt87x0: Using board 0, analog, digital (rate 32000 Hz) I hope this can help. regards _____________________________________________________________________________ Ne gardez plus qu'une seule adresse mail ! Copiez vos mails vers Yahoo! Mail

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[alsa-devel] vaio_mux's num_items is not equal to the actual items
by zhejiang 11 Oct '07

11 Oct '07

In patch_sigmatel.c, static struct hda_input_mux vaio_mux = { .num_items = 2, .items = { /* { "HP", 0x0 }, */ { "Mic Jack", 0x1 }, { "Internal Mic", 0x2 }, { "PCM", 0x3 }, } }; Why is the num_items not 3 ? The "PCM" won't be selected with num_times = 2.

2 1

[alsa-devel] Creative Labs XFi drivers.
by Tobin Davis 11 Oct '07

11 Oct '07

Creative Labs has released a proprietary driver base for their XFi cards. I downloaded it because it claims to have compatibility with alsa. In fact, it requires alsa include files to be able to compile. I haven't looked at the source yet (this information is all in the Readme.txt in the main download) and probably won't (especially since I don't own a Creative Labs XFi card). What I don't understand, is are they allowed to use our code to build their drivers? They use alsa core modules to interoperate (from what I can tell in the Readme.txt). Also, this line in the License.txt is especially troubling: > In the event that the Software includes Linux code or drivers, then, unless expressly stated to the contrary, such Linux > code and drivers are proprietary to Creative and all rights therein are reserved. I'm deleting all but the big tarball I downloaded. I'll delete it later. -- Tobin Davis <tdavis(a)dsl-only.net>

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[alsa-devel] Alsa 1.0.15rc2 and oops with Dell M1210
by zhejiang 10 Oct '07

10 Oct '07

Seems that the spec->input_mux contains no item at all. So system maybe crash in the snd_hda_input_mux_info() when try to copy the items[].label,which is null pointer. May you replace the function stac92xx_mux_enum_info() in alsa-driver-1.0.15rc2/alsa-kernel/pci/hda/patch_sigmatel.c ? static int stac92xx_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct sigmatel_spec *spec = codec->spec; printk("spec->input_mux->num_items is %d \n",spec->input_mux->num_items); if(!spec->input_mux->num_items) return 0; return snd_hda_input_mux_info(spec->input_mux, uinfo); } Does it still crash? What's the number of num_items? The pin_config for M1210 didn't include any input pin. I wonder why.

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Re: [alsa-devel] chmod with USB sound device?
by william estrada 10 Oct '07

10 Oct '07

Harald, I revisited /etc/security/console.perms.d/50-default.perms and got it to work. For the list, I changed: In /etc/security/console.perms.d/50-default.perms > # permission definitions . > <console> 0666 <sound> 0666 root And In /etc/udev/rules.d/50-udev.rules > KERNEL=="snd/*", MODE="0666", OWNER="wre00:root" Then I did the udevtrigger. Then i did the udevtrigger, plugged in my USB sound device and got this: > [root@laptop wre00]# ls -lrt /dev/snd > total 0 > crw-rw-rw- 1 wre00 root 116, 2 Oct 10 09:45 timer > crw-rw-rw- 1 wre00 root 116, 3 Oct 10 09:45 seq > crw-rw-rw- 1 wre00 root 116, 4 Oct 10 09:45 pcmC0D0p > crw-rw-rw- 1 wre00 root 116, 5 Oct 10 09:45 pcmC0D0c > crw-rw-rw- 1 wre00 root 116, 6 Oct 10 09:45 controlC0 > crw-rw-rw- 1 wre00 root 116, 7 Oct 10 10:28 pcmC1D0p > crw-rw-rw- 1 wre00 root 116, 8 Oct 10 10:28 pcmC1D0c > crw-rw-rw- 1 wre00 root 116, 9 Oct 10 10:28 controlC1 I guess both changes are required but it now works. Thanks for your time. William Estrada MrUmunhum(a)popdial.com Mt-Umunhum-Wireless.net ( http://Mt-Umunhum-Wireless.net ) Ymessenger: MrUmunhum Harald Hoyer wrote: > william estrada wrote: >> Hi guys, >> >> I'm trying to run 'chmod' on a USB sound card using 'Hot Plug' udev >> support. >> From reading, I think I need to update my >> '/etc/udev/rules.d/50-udev.rules'? >> I have made these changes: >>> # audio devices . >>> KERNEL=="snd/*", MODE="0666" > > You may have a look in pam_console... > /etc/security/console.perms > /etc/security/console.perms.d/* >

1 0

[alsa-devel] snd-hdsp & digiface
by Fernando Lopez-Lezcano 10 Oct '07

10 Oct '07

Hi... We are using both digiface and multiface interfaces here at ccrma and we have a (small) problem of consistency in the channel numbering because of the way they are connected. Both cards are connected digitally to digital mixers, but in the case of the multiface the digital channels (which would be the ones users use) are 9-16 while on the digiface the first eight are 1-8[*]. Is there a way to change the ordering in the multiface so that the digital interface is presented as ports 1-8 instead of 9-16? (even a hack to the driver would be welcome) - and preferably one that does not need the default interface to be changed from hw:0 in jack... -- Fernando [*] accounts are shared so any configuration you do on one machine for 1-8 will be wrong for the other where the real outputs are 9-16.

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Re: [alsa-devel] different user causes 'no such device'
by william estrada 10 Oct '07

10 Oct '07

Thanks, I did a 'chmod -R 666 /dev/snd/' and it worked. Now how do I make the hot plugged USB sound device use these setting?? Thanks for your time. William Estrada MrUmunhum(a)popdial.com Mt-Umunhum-Wireless.net ( http://Mt-Umunhum-Wireless.net ) Ymessenger: MrUmunhum Takashi Iwai wrote: > At Wed, 03 Oct 2007 12:55:27 -0700, > william estrada wrote: >> Hi guys, >> >> I have what seams like a interesting problem. I have written an ALSA app >> to record wav files and it runs while I am logged in an terminal as the >> logged in Gnome user or root. But when I su to another user, it can't >> find the default sound card. >> >> I am running FC6. I tried changing own of /dev/dsp to 'nobody', that did >> not work. >> >> Anyone know of this problem and a fix for it. >> >> Thanks for your time. >> >> -- >> William Estrada >> MrUmunhum(a)popdial.com >> Mt-Umunhum-Wireless.net ( http://Mt-Umunhum-Wireless.net ) >> Ymessenger: MrUmunhum >> >>> MSGrec - Version 1.1, Mt-Umunhum-Wireless.net >>> WavMSG: Paused: enter 'q' to stop, <ENTER> to start. >>> ALSA lib confmisc.c:848:(snd_func_card_driver) cannot find card '0' >>> ALSA lib conf.c:3500:(_snd_config_evaluate) function snd_func_card_driver returned error: No such device >>> ALSA lib confmisc.c:397:(snd_func_concat) error evaluating strings >>> ALSA lib conf.c:3500:(_snd_config_evaluate) function snd_func_concat returned error: No such device >>> ALSA lib confmisc.c:1248:(snd_func_refer) error evaluating name >>> ALSA lib conf.c:3500:(_snd_config_evaluate) function snd_func_refer returned error: No such device >>> ALSA lib conf.c:3972:(snd_config_expand) Evaluate error: No such device >>> ALSA lib pcm.c:2145:(snd_pcm_open_noupdate) Unknown PCM default > > ALSA-native apps access only ALSA device files, usually /dev/snd/*, > not /dev/dsp*, those for OSS. In this case, it should be > /dev/snd/controlC* files. Check the premission of these files. > > > Takashi >

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[alsa-devel] Add quirk support for Dell XPS1330 and Inspiron 1420.
by Tim Gardner 10 Oct '07

10 Oct '07

-- Tim Gardner tim.gardner(a)ubuntu.com

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[alsa-devel] [PATCH] stac9200 eapd fixes
by Brian Hinz 10 Oct '07

10 Oct '07

Hi, The following patch addresses alsa bug #0002948 and it's (many) duplicates. Please consider it for inclusion in the next release. Thanks, -brian --- alsa-kernel/pci/hda/hda_codec.c 2007-09-24 21:46:11.000000000 -0400 +++ alsa-kernel/pci/hda/hda_codec.c 2007-09-24 22:00:15.000000000 -0400 @@ -1629,11 +1629,30 @@ power_state); nid = codec->start_nid; + unsigned int pincap; for (i = 0; i < codec->num_nodes; i++, nid++) { - if (get_wcaps(codec, nid) & AC_WCAP_POWER) - snd_hda_codec_write(codec, nid, 0, + if (get_wcaps(codec, nid) & AC_WCAP_POWER) { + pincap = snd_hda_param_read(codec, nid, + AC_PAR_PIN_CAP); + /* + * don't power down the widget if it controls eapd + * and EAPD_BTLENABLE is set. + */ + if (pincap & AC_PINCAP_EAPD) { + int eapd = snd_hda_codec_read(codec, nid, + 0, AC_VERB_GET_EAPD_BTLENABLE, 0); + eapd &= 0x02; + if (power_state != AC_PWRST_D3 || !eapd) { + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_POWER_STATE, + power_state); + } + } else { + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, power_state); + } + } } if (power_state == AC_PWRST_D0) { --- alsa-kernel/pci/hda/patch_sigmatel.c 2007-09-24 21:46:12.000000000 -0400 +++ alsa-kernel/pci/hda/patch_sigmatel.c 2007-09-26 20:57:55.000000000 -0400 @@ -378,6 +378,13 @@ {} }; +static struct hda_verb stac9200_eapd_init[] = { + /* set dac0mux for dac converter */ + { 0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, + { 0x08, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, + {} +}; + static struct hda_verb stac925x_core_init[] = { /* set dac0mux for dac converter */ { 0x06, AC_VERB_SET_CONNECT_SEL, 0x00}, @@ -2492,7 +2499,23 @@ spec->num_dmics = 0; spec->num_adcs = 1; - spec->init = stac9200_core_init; + /* + * EAPD needs to be set on resume + */ + switch (codec->subsystem_id) { + case 0x107b0205: /* Gateway S-7110M */ + spec->init = stac9200_eapd_init; + break; + case 0x107b0317: /* Gateway MT3423, MX3412, MX3410, MX3414 */ + spec->init = stac9200_eapd_init; + break; + case 0x107b0318: /* Gateway ML3109, MT3707 */ + spec->init = stac9200_eapd_init; + break; + default: + spec->init = stac9200_core_init; + break; + } spec->mixer = stac9200_mixer; err = stac9200_parse_auto_config(codec);

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[PATCH] UBUNTU: Re-add quirk support for Dell XPS 1330 and Inspiron 1420.
by Tim Gardner 09 Oct '07

09 Oct '07

Signed-off-by: Tim Gardner <tim.gardner(a)canonical.com> --- .../alsa-kernel/pci/hda/patch_sigmatel.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/updates/alsa/alsa-driver/alsa-kernel/pci/hda/patch_sigmatel.c b/updates/alsa/alsa-driver/alsa-kernel/pci/hda/patch_sigmatel.c index 27360d2..f714de1 100644 --- a/updates/alsa/alsa-driver/alsa-kernel/pci/hda/patch_sigmatel.c +++ b/updates/alsa/alsa-driver/alsa-kernel/pci/hda/patch_sigmatel.c @@ -1097,11 +1097,13 @@ static struct snd_pci_quirk stac927x_cfg_tbl[] = { SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2003, "Intel D965", STAC_D965_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2002, "Intel D965", STAC_D965_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2001, "Intel D965", STAC_D965_3ST), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f3, "Dell Inspiron 1420", STAC_D965_3ST), /* Dell 3 stack systems */ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01dd, "Dell Dimension E520", STAC_DELL_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01ed, "Dell ", STAC_DELL_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f4, "Dell ", STAC_DELL_3ST), /* 965 based 5 stack systems */ + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0209, "Dell XPS 1330", STAC_D965_5ST), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2301, "Intel D965", STAC_D965_5ST), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2302, "Intel D965", STAC_D965_5ST), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2303, "Intel D965", STAC_D965_5ST), -- 1.5.2.5 --------------080009030006050203010104--

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[alsa-devel] ALSA Support for Maya44 PCI?
by Piotr Makowski 09 Oct '07

09 Oct '07

There is already a programmer working on the driver, but thanks for your interest! You can expect the driver for ESI MAYA44 PCI card soon. There will be also an official announcement on alsa-devel when the programmer will start his work. Regards, Piotr Makowski 2007/10/9, Nicolas Rodriguez <nicolas.rod(a)gmail.com>: > I bought a Maya44 PCI 4-channel soundcard and I'll like to use it in Linux. > I couldn't get it to work with Ubuntu (I don't remember the ALSA version, > sorry). Do you support this card? If you don't, I can create the driver for > it. I've read somewhere that ESI-PRO can give the card specifications to > developers interested in build drivers for their cards. I've sent them an > email asking for the card specifications for I've got no response. What do > you recommend? > > Thanks in advance, > Nico. > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

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[alsa-devel] [PATCH] [RFC] sis7019: Initial support for the SiS 7019 Audio Accelerator
by Dave Dillow 09 Oct '07

09 Oct '07

Basic audio support for the SiS 7019 Audio Accelerator as found in the SiS 55x SoC. There is currently no synth or power management support at the moment, but the audio playback has been tested with several channel and rate combinations, as well as various period and buffer configurations. Signed-off-by: David Dillow <dave(a)thedillows.org> --- include/sound/sis7019.h | 342 ++++++++++++++ sound/pci/Kconfig | 10 + sound/pci/Makefile | 2 + sound/pci/sis7019.c | 1197 +++++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 1551 insertions(+), 0 deletions(-) diff --git a/include/sound/sis7019.h b/include/sound/sis7019.h new file mode 100644 index 0000000..d395abb --- /dev/null +++ b/include/sound/sis7019.h @@ -0,0 +1,342 @@ +#ifndef __sis7019_h__ +#define __sis7019_h__ + +/* + * Definitions for SiS7019 Audio Accelerator + * + * Copyright (C) 2004-2007, David Dillow + * Written by David Dillow <dave(a)thedillows.org> + * Inspired by the Trident 4D-WaveDX/NX driver. + * + * All rights reserved. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation, version 2. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + + +/* General Control Register */ +#define SIS_GCR 0x00 +#define SIS_GCR_MACRO_POWER_DOWN 0x80000000 +#define SIS_GCR_MODEM_ENABLE 0x00010000 +#define SIS_GCR_SOFTWARE_RESET 0x00000001 + +/* General Interrupt Enable Register */ +#define SIS_GIER 0x04 +#define SIS_GIER_MODEM_TIMER_IRQ_ENABLE 0x00100000 +#define SIS_GIER_MODEM_RX_DMA_IRQ_ENABLE 0x00080000 +#define SIS_GIER_MODEM_TX_DMA_IRQ_ENABLE 0x00040000 +#define SIS_GIER_AC97_GPIO1_IRQ_ENABLE 0x00020000 +#define SIS_GIER_AC97_GPIO0_IRQ_ENABLE 0x00010000 +#define SIS_GIER_AC97_SAMPLE_TIMER_IRQ_ENABLE 0x00000010 +#define SIS_GIER_AUDIO_GLOBAL_TIMER_IRQ_ENABLE 0x00000008 +#define SIS_GIER_AUDIO_RECORD_DMA_IRQ_ENABLE 0x00000004 +#define SIS_GIER_AUDIO_PLAY_DMA_IRQ_ENABLE 0x00000002 +#define SIS_GIER_AUDIO_WAVE_ENGINE_IRQ_ENABLE 0x00000001 + +/* General Interrupt Status Register */ +#define SIS_GISR 0x08 +#define SIS_GISR_MODEM_TIMER_IRQ_STATUS 0x00100000 +#define SIS_GISR_MODEM_RX_DMA_IRQ_STATUS 0x00080000 +#define SIS_GISR_MODEM_TX_DMA_IRQ_STATUS 0x00040000 +#define SIS_GISR_AC97_GPIO1_IRQ_STATUS 0x00020000 +#define SIS_GISR_AC97_GPIO0_IRQ_STATUS 0x00010000 +#define SIS_GISR_AC97_SAMPLE_TIMER_IRQ_STATUS 0x00000010 +#define SIS_GISR_AUDIO_GLOBAL_TIMER_IRQ_STATUS 0x00000008 +#define SIS_GISR_AUDIO_RECORD_DMA_IRQ_STATUS 0x00000004 +#define SIS_GISR_AUDIO_PLAY_DMA_IRQ_STATUS 0x00000002 +#define SIS_GISR_AUDIO_WAVE_ENGINE_IRQ_STATUS 0x00000001 + +/* DMA Control Register */ +#define SIS_DMA_CSR 0x10 +#define SIS_DMA_CSR_PCI_SETTINGS 0x0000001d +#define SIS_DMA_CSR_CONCURRENT_ENABLE 0x00000200 +#define SIS_DMA_CSR_PIPELINE_ENABLE 0x00000100 +#define SIS_DMA_CSR_RX_DRAIN_ENABLE 0x00000010 +#define SIS_DMA_CSR_RX_FILL_ENABLE 0x00000008 +#define SIS_DMA_CSR_TX_DRAIN_ENABLE 0x00000004 +#define SIS_DMA_CSR_TX_LOWPRI_FILL_ENABLE 0x00000002 +#define SIS_DMA_CSR_TX_HIPRI_FILL_ENABLE 0x00000001 + +/* Playback Channel Start Registers */ +#define SIS_PLAY_START_A_REG 0x14 +#define SIS_PLAY_START_B_REG 0x18 + +/* Playback Channel Stop Registers */ +#define SIS_PLAY_STOP_A_REG 0x1c +#define SIS_PLAY_STOP_B_REG 0x20 + +/* Recording Channel Start Register */ +#define SIS_RECORD_START_REG 0x24 + +/* Recording Channel Stop Register */ +#define SIS_RECORD_STOP_REG 0x28 + +/* Playback Interrupt Status Registers */ +#define SIS_PISR_A 0x2c +#define SIS_PISR_B 0x30 + +/* Recording Interrupt Status Register */ +#define SIS_RISR 0x34 + +/* AC97 AC-link Playback Source Register */ +#define SIS_AC97_PSR 0x40 +#define SIS_AC97_PSR_MODEM_HEADSET_SRC_MIXER 0x0f000000 +#define SIS_AC97_PSR_MODEM_LINE2_SRC_MIXER 0x00f00000 +#define SIS_AC97_PSR_MODEM_LINE1_SRC_MIXER 0x000f0000 +#define SIS_AC97_PSR_PCM_LFR_SRC_MIXER 0x0000f000 +#define SIS_AC97_PSR_PCM_SURROUND_SRC_MIXER 0x00000f00 +#define SIS_AC97_PSR_PCM_CENTER_SRC_MIXER 0x000000f0 +#define SIS_AC97_PSR_PCM_LR_SRC_MIXER 0x0000000f + +/* AC97 AC-link Command Register */ +#define SIS_AC97_CMD 0x50 +#define SIS_AC97_CMD_DATA_MASK 0xffff0000 +#define SIS_AC97_CMD_REG_MASK 0x0000ff00 +#define SIS_AC97_CMD_CODEC3_READ 0x0000000d +#define SIS_AC97_CMD_CODEC3_WRITE 0x0000000c +#define SIS_AC97_CMD_CODEC2_READ 0x0000000b +#define SIS_AC97_CMD_CODEC2_WRITE 0x0000000a +#define SIS_AC97_CMD_CODEC_READ 0x00000009 +#define SIS_AC97_CMD_CODEC_WRITE 0x00000008 +#define SIS_AC97_CMD_CODEC_WARM_RESET 0x00000005 +#define SIS_AC97_CMD_CODEC_COLD_RESET 0x00000004 +#define SIS_AC97_CMD_DONE 0x00000000 + +/* AC97 AC-link Semaphore Register */ +#define SIS_AC97_SEMA 0x54 +#define SIS_AC97_SEMA_BUSY 0x00000001 +#define SIS_AC97_SEMA_RELEASE 0x00000000 + +/* AC97 AC-link Status Register */ +#define SIS_AC97_STATUS 0x58 +#define SIS_AC97_STATUS_AUDIO_D2_INACT_SECS 0x03f00000 +#define SIS_AC97_STATUS_MODEM_ALIVE 0x00002000 +#define SIS_AC97_STATUS_AUDIO_ALIVE 0x00001000 +#define SIS_AC97_STATUS_CODEC3_READY 0x00000400 +#define SIS_AC97_STATUS_CODEC2_READY 0x00000200 +#define SIS_AC97_STATUS_CODEC_READY 0x00000100 +#define SIS_AC97_STATUS_WARM_RESET 0x00000080 +#define SIS_AC97_STATUS_COLD_RESET 0x00000040 +#define SIS_AC97_STATUS_POWERED_DOWN 0x00000020 +#define SIS_AC97_STATUS_NORMAL 0x00000010 +#define SIS_AC97_STATUS_READ_EXPIRED 0x00000004 +#define SIS_AC97_STATUS_SEMAPHORE 0x00000002 +#define SIS_AC97_STATUS_BUSY 0x00000001 + +/* AC97 AC-link Audio Configuration Register */ +#define SIS_AC97_CONF 0x5c +#define SIS_AC97_CONF_AUDIO_ALIVE 0x80000000 +#define SIS_AC97_CONF_WARM_RESET_ENABLE 0x40000000 +#define SIS_AC97_CONF_PR6_ENABLE 0x20000000 +#define SIS_AC97_CONF_PR5_ENABLE 0x10000000 +#define SIS_AC97_CONF_PR4_ENABLE 0x08000000 +#define SIS_AC97_CONF_PR3_ENABLE 0x04000000 +#define SIS_AC97_CONF_PR2_PR7_ENABLE 0x02000000 +#define SIS_AC97_CONF_PR0_PR1_ENABLE 0x01000000 +#define SIS_AC97_CONF_AUTO_PM_ENABLE 0x00800000 +#define SIS_AC97_CONF_PCM_LFE_ENABLE 0x00080000 +#define SIS_AC97_CONF_PCM_SURROUND_ENABLE 0x00040000 +#define SIS_AC97_CONF_PCM_CENTER_ENABLE 0x00020000 +#define SIS_AC97_CONF_PCM_LR_ENABLE 0x00010000 +#define SIS_AC97_CONF_PCM_CAP_MIC_ENABLE 0x00002000 +#define SIS_AC97_CONF_PCM_CAP_LR_ENABLE 0x00001000 +#define SIS_AC97_CONF_PCM_CAP_MIC_FROM_CODEC3 0x00000200 +#define SIS_AC97_CONF_PCM_CAP_LR_FROM_CODEC3 0x00000100 +#define SIS_AC97_CONF_CODEC3_PM_VRM 0x00000080 +#define SIS_AC97_CONF_CODEC_PM_VRM 0x00000040 +#define SIS_AC97_CONF_CODEC3_VRA_ENABLE 0x00000020 +#define SIS_AC97_CONF_CODEC_VRA_ENABLE 0x00000010 +#define SIS_AC97_CONF_CODEC3_PM_EAC 0x00000008 +#define SIS_AC97_CONF_CODEC_PM_EAC 0x00000004 +#define SIS_AC97_CONF_CODEC3_EXISTS 0x00000002 +#define SIS_AC97_CONF_CODEC_EXISTS 0x00000001 + +/* Playback Channel Sync Group registers */ +#define SIS_PLAY_SYNC_GROUP_A 0x80 +#define SIS_PLAY_SYNC_GROUP_B 0x84 +#define SIS_PLAY_SYNC_GROUP_C 0x88 +#define SIS_PLAY_SYNC_GROUP_D 0x8c +#define SIS_MIXER_SYNC_GROUP 0x90 + +/* Wave Engine Config and Control Register */ +#define SIS_WECCR 0xa0 +#define SIS_WECCR_TESTMODE_MASK 0x00300000 +#define SIS_WECCR_TESTMODE_NORMAL 0x00000000 +#define SIS_WECCR_TESTMODE_BYPASS_NSO_ALPHA 0x00100000 +#define SIS_WECCR_TESTMODE_BYPASS_FC 0x00200000 +#define SIS_WECCR_TESTMODE_BYPASS_WOL 0x00300000 +#define SIS_WECCR_RESONANCE_DELAY_MASK 0x00060000 +#define SIS_WECCR_RESONANCE_DELAY_NONE 0x00000000 +#define SIS_WECCR_RESONANCE_DELAY_FC_1F00 0x00020000 +#define SIS_WECCR_RESONANCE_DELAY_FC_1E00 0x00040000 +#define SIS_WECCR_RESONANCE_DELAY_FC_1C00 0x00060000 +#define SIS_WECCR_IGNORE_CHANNEL_PARMS 0x00010000 +#define SIS_WECCR_COMMAND_CHANNEL_ID_MASK 0x0003ff00 +#define SIS_WECCR_COMMAND_MASK 0x00000007 +#define SIS_WECCR_COMMAND_NONE 0x00000000 +#define SIS_WECCR_COMMAND_DONE 0x00000000 +#define SIS_WECCR_COMMAND_PAUSE 0x00000001 +#define SIS_WECCR_COMMAND_TOGGLE_VEG 0x00000002 +#define SIS_WECCR_COMMAND_TOGGLE_MEG 0x00000003 +#define SIS_WECCR_COMMAND_TOGGLE_VEG_MEG 0x00000004 + +/* Wave Engine Volume Control Register */ +#define SIS_WEVCR 0xa4 +#define SIS_WEVCR_LEFT_MUSIC_ATTENUATION_MASK 0xff000000 +#define SIS_WEVCR_RIGHT_MUSIC_ATTENUATION_MASK 0x00ff0000 +#define SIS_WEVCR_LEFT_WAVE_ATTENUATION_MASK 0x0000ff00 +#define SIS_WEVCR_RIGHT_WAVE_ATTENUATION_MASK 0x000000ff + +/* Wave Engine Interrupt Status Registers */ +#define SIS_WEISR_A 0xa8 +#define SIS_WEISR_B 0xac + + +/* Playback DMA parameters (paramter RAM) */ +#define SIS_PLAY_DMA_OFFSET 0x0000 +#define SIS_PLAY_DMA_SIZE 0x10 +#define SIS_PLAY_DMA_ADDR(addr, num) \ + ((num * SIS_PLAY_DMA_SIZE) + (addr) + SIS_PLAY_DMA_OFFSET) + +#define SIS_PLAY_DMA_FORMAT_CSO 0x00 +#define SIS_PLAY_DMA_FORMAT_UNSIGNED 0x00080000 +#define SIS_PLAY_DMA_FORMAT_8BIT 0x00040000 +#define SIS_PLAY_DMA_FORMAT_MONO 0x00020000 +#define SIS_PLAY_DMA_CSO_MASK 0x0000ffff +#define SIS_PLAY_DMA_BASE 0x04 +#define SIS_PLAY_DMA_CONTROL 0x08 +#define SIS_PLAY_DMA_STOP_AT_SSO 0x04000000 +#define SIS_PLAY_DMA_RELEASE 0x02000000 +#define SIS_PLAY_DMA_LOOP 0x01000000 +#define SIS_PLAY_DMA_INTR_AT_SSO 0x00080000 +#define SIS_PLAY_DMA_INTR_AT_ESO 0x00040000 +#define SIS_PLAY_DMA_INTR_AT_LEO 0x00020000 +#define SIS_PLAY_DMA_INTR_AT_MLP 0x00010000 +#define SIS_PLAY_DMA_LEO_MASK 0x0000ffff +#define SIS_PLAY_DMA_SSO_ESO 0x0c +#define SIS_PLAY_DMA_SSO_MASK 0xffff0000 +#define SIS_PLAY_DMA_ESO_MASK 0x0000ffff + +/* Capture DMA parameters (paramter RAM) */ +#define SIS_CAPTURE_DMA_OFFSET 0x0800 +#define SIS_CAPTURE_DMA_SIZE 0x10 +#define SIS_CAPTURE_DMA_ADDR(addr, num) \ + ((num * SIS_CAPTURE_DMA_SIZE) + (addr) + SIS_CAPTURE_DMA_OFFSET) + +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_0 0 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_1 1 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_2 2 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_3 3 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_4 4 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_5 5 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_6 6 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_7 7 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_8 8 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_9 9 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_10 10 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_11 11 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_12 12 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_13 13 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_14 14 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_15 15 +#define SIS_CAPTURE_CHAN_AC97_PCM_IN 16 +#define SIS_CAPTURE_CHAN_AC97_MIC_IN 17 +#define SIS_CAPTURE_CHAN_AC97_LINE1_IN 18 +#define SIS_CAPTURE_CHAN_AC97_LINE2_IN 19 +#define SIS_CAPTURE_CHAN_AC97_HANDSE_IN 20 + +#define SIS_CAPTURE_DMA_FORMAT_CSO 0x00 +#define SIS_CAPTURE_DMA_MONO_MODE_MASK 0xc0000000 +#define SIS_CAPTURE_DMA_MONO_MODE_AVG 0x00000000 +#define SIS_CAPTURE_DMA_MONO_MODE_LEFT 0x40000000 +#define SIS_CAPTURE_DMA_MONO_MODE_RIGHT 0x80000000 +#define SIS_CAPTURE_DMA_FORMAT_UNSIGNED 0x00080000 +#define SIS_CAPTURE_DMA_FORMAT_8BIT 0x00040000 +#define SIS_CAPTURE_DMA_FORMAT_MONO 0x00020000 +#define SIS_CAPTURE_DMA_CSO_MASK 0x0000ffff +#define SIS_CAPTURE_DMA_BASE 0x04 +#define SIS_CAPTURE_DMA_CONTROL 0x08 +#define SIS_CAPTURE_DMA_STOP_AT_SSO 0x04000000 +#define SIS_CAPTURE_DMA_RELEASE 0x02000000 +#define SIS_CAPTURE_DMA_LOOP 0x01000000 +#define SIS_CAPTURE_DMA_INTR_AT_LEO 0x00020000 +#define SIS_CAPTURE_DMA_INTR_AT_MLP 0x00010000 +#define SIS_CAPTURE_DMA_LEO_MASK 0x0000ffff +#define SIS_CAPTURE_DMA_RESERVED 0x0c + + +/* Mixer routing list start pointer (parameter RAM) */ +#define SIS_MIXER_START_OFFSET 0x1000 +#define SIS_MIXER_START_SIZE 0x04 +#define SIS_MIXER_START_ADDR(addr, num) \ + ((num * SIS_MIXER_START_SIZE) + (addr) + SIS_MIXER_START_OFFSET) + +#define SIS_MIXER_START_MASK 0x0000007f + +/* Mixer routing table (parameter RAM) */ +#define SIS_MIXER_OFFSET 0x1400 +#define SIS_MIXER_SIZE 0x04 +#define SIS_MIXER_ADDR(addr, num) \ + ((num * SIS_MIXER_SIZE) + (addr) + SIS_MIXER_OFFSET) + +#define SIS_MIXER_RIGHT_ATTENUTATION_MASK 0xff000000 +#define SIS_MIXER_RIGHT_NO_ATTEN 0xff000000 +#define SIS_MIXER_LEFT_ATTENUTATION_MASK 0xff000000 +#define SIS_MIXER_LEFT_NO_ATTEN 0xff000000 +#define SIS_MIXER_NEXT_ENTRY_MASK 0x00007f00 +#define SIS_MIXER_NEXT_ENTRY_NONE 0x00000000 +#define SIS_MIXER_DEST_MASK 0x0000007f +#define SIS_MIXER_DEST_0 0x00000020 +#define SIS_MIXER_DEST_1 0x00000021 +#define SIS_MIXER_DEST_2 0x00000022 +#define SIS_MIXER_DEST_3 0x00000023 +#define SIS_MIXER_DEST_4 0x00000024 +#define SIS_MIXER_DEST_5 0x00000025 +#define SIS_MIXER_DEST_6 0x00000026 +#define SIS_MIXER_DEST_7 0x00000027 +#define SIS_MIXER_DEST_8 0x00000028 +#define SIS_MIXER_DEST_9 0x00000029 +#define SIS_MIXER_DEST_10 0x0000002a +#define SIS_MIXER_DEST_11 0x0000002b +#define SIS_MIXER_DEST_12 0x0000002c +#define SIS_MIXER_DEST_13 0x0000002d +#define SIS_MIXER_DEST_14 0x0000002e +#define SIS_MIXER_DEST_15 0x0000002f + +/* Wave Engine Control Parameters (parameter RAM) */ +#define SIS_WAVE_OFFSET 0x2000 +#define SIS_WAVE_SIZE 0x40 +#define SIS_WAVE_ADDR(addr, num) \ + ((num * SIS_WAVE_SIZE) + (addr) + SIS_WAVE_OFFSET) + +#define SIS_WAVE_GENERAL 0x00 +#define SIS_WAVE_GENERAL_WAVE_VOLUME 0x80000000 +#define SIS_WAVE_GENERAL_MUSIC_VOLUME 0x00000000 +#define SIS_WAVE_GENERAL_VOLUME_MASK 0x7f000000 +#define SIS_WAVE_GENERAL_ARTICULATION 0x04 +#define SIS_WAVE_GENERAL_ARTICULATION_DELTA_MASK 0x3fff0000 +#define SIS_WAVE_ARTICULATION 0x08 +#define SIS_WAVE_TIMER 0x0c +#define SIS_WAVE_GENERATOR 0x10 +#define SIS_WAVE_CHANNEL_CONTROL 0x14 +#define SIS_WAVE_CHANNEL_CONTROL_FIRST_SAMPLE 0x80000000 +#define SIS_WAVE_CHANNEL_CONTROL_AMP_ENABLE 0x40000000 +#define SIS_WAVE_CHANNEL_CONTROL_FILTER_ENABLE 0x20000000 +#define SIS_WAVE_CHANNEL_CONTROL_INTERPOLATE_ENABLE 0x10000000 +#define SIS_WAVE_LFO_EG_CONTROL 0x18 +#define SIS_WAVE_LFO_EG_CONTROL_2 0x1c +#define SIS_WAVE_LFO_EG_CONTROL_3 0x20 +#define SIS_WAVE_LFO_EG_CONTROL_4 0x24 + +#endif /* __sis7019_h__ */ diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index c6b4410..801bb48 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -705,6 +705,16 @@ config SND_RME9652 To compile this driver as a module, choose M here: the module will be called snd-rme9652. +config SND_SIS7019 + tristate "SiS 7019 Audio Accelerator" + depends on SND + select SND_AC97_CODEC + help + Say Y here to include support for the SiS 7019 Audio Accelerator. + + To compile this driver as a module, choose M here: the module + will be called snd-sis7019. + config SND_SONICVIBES tristate "S3 SonicVibes" depends on SND diff --git a/sound/pci/Makefile b/sound/pci/Makefile index cd76e02..1904a11 100644 --- a/sound/pci/Makefile +++ b/sound/pci/Makefile @@ -23,6 +23,7 @@ snd-intel8x0m-objs := intel8x0m.o snd-maestro3-objs := maestro3.o snd-rme32-objs := rme32.o snd-rme96-objs := rme96.o +snd-sis7019-objs := sis7019.o snd-sonicvibes-objs := sonicvibes.o snd-via82xx-objs := via82xx.o snd-via82xx-modem-objs := via82xx_modem.o @@ -48,6 +49,7 @@ obj-$(CONFIG_SND_INTEL8X0M) += snd-intel8x0m.o obj-$(CONFIG_SND_MAESTRO3) += snd-maestro3.o obj-$(CONFIG_SND_RME32) += snd-rme32.o obj-$(CONFIG_SND_RME96) += snd-rme96.o +obj-$(CONFIG_SND_SIS7019) += snd-sis7019.o obj-$(CONFIG_SND_SONICVIBES) += snd-sonicvibes.o obj-$(CONFIG_SND_VIA82XX) += snd-via82xx.o obj-$(CONFIG_SND_VIA82XX_MODEM) += snd-via82xx-modem.o diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c new file mode 100644 index 0000000..3a74f64 --- /dev/null +++ b/sound/pci/sis7019.c @@ -0,0 +1,1197 @@ +/* + * Driver for SiS7019 Audio Accelerator + * + * Copyright (C) 2004-2007, David Dillow + * Written by David Dillow <dave(a)thedillows.org> + * Inspired by the Trident 4D-WaveDX/NX driver. + * + * All rights reserved. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation, version 2. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <sound/driver.h> +#include <linux/init.h> +#include <linux/pci.h> +#include <linux/time.h> +#include <linux/moduleparam.h> +#include <linux/interrupt.h> +#include <linux/delay.h> +#include <sound/core.h> +#include <sound/ac97_codec.h> +#include <sound/initval.h> +#include <sound/sis7019.h> + +MODULE_AUTHOR("David Dillow <dave(a)thedillows.org>"); +MODULE_DESCRIPTION("SiS 7019"); +MODULE_LICENSE("GPL"); +MODULE_SUPPORTED_DEVICE("{{SiS,SiS7019 Audio Accelerator}}"); + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for SiS7019 Audio Accelerator."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for SiS7019 Audio Accelerator."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable SiS7019 Audio Accelerator."); + +static struct pci_device_id snd_sis7019_ids[] = { + { PCI_DEVICE(PCI_VENDOR_ID_SI, 0x7019) }, + { 0, } +}; + +MODULE_DEVICE_TABLE(pci, snd_sis7019_ids); + +/* There are three timing modes for the voices. + * + * For both playback and capture, when the buffer is two periods long, + * we use the hardware's built-in Mid-Loop Interrupt and End-Loop Interrupt + * to let us know when the periods have ended. + * + * When performing playback with more than two periods per buffer, we set + * the "Stop Sample Offset" and tell the hardware to interrupt us when we + * reach it. We then update the offset and continue on until we are + * interrupted for the next period. + * + * Capture channels do not have a SSO, so we allocate a playback channel to + * use as a timer for the capture periods. We use the End-Loop Interrupt + * to clock out the periods, and adjust the Loop End Offset of the timing + * channel to maintain synchronization. This algorithm came from the + * Trident driver. + * + * FIXME: The timing channel is configured to give quite a bit of + * attenuation (~96dB), but it may be possible to gain a bit more, or + * even better, play from a page of silence. + * + * FIXME: It'd be nice to make use of some of the synth features in the + * hardware, but a woeful lack of documentation is a significant roadblock. + */ +struct voice { + u16 flags; +#define VOICE_IN_USE 1 +#define VOICE_CAPTURE 2 +#define VOICE_SSO_TIMING 4 +#define VOICE_SYNC_TIMING 8 + u16 sync_cso; + u16 period_size; + u16 buffer_size; + u32 sso; + struct snd_pcm_substream *substream; + struct voice *timing; + void __iomem *ctrl_base; + void __iomem *wave_base; + void __iomem *sync_base; + int num; +}; + +struct sis7019 { + long unsigned ioport; + void __iomem *ioaddr; + int irq; + int codecs_present; + + struct pci_dev *pci; + struct snd_pcm *pcm; + struct snd_card *card; + struct snd_ac97 *ac97[3]; + + /* hw_lock protects access to the registers and wave engine parameters + * voice_lock protects allocation/freeing of the voice descriptions + */ + spinlock_t hw_lock; + spinlock_t voice_lock; + + struct voice voices[64]; + struct voice capture_voice; +}; + +#define SIS_PRIMARY_CODEC_PRESENT 0x0001 +#define SIS_SECONDARY_CODEC_PRESENT 0x0002 +#define SIS_TERTIARY_CODEC_PRESENT 0x0004 + +static struct snd_pcm_hardware sis_playback_hw_info = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_SYNC_START), + .formats = (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE), + .rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 4000, + .rate_max = 48000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = 0x3ffdc, + .period_bytes_min = 8, + .period_bytes_max = 0x3ffdc, + .periods_min = 1, + .periods_max = 0x1ffe, +}; + +static struct snd_pcm_hardware sis_capture_hw_info = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_SYNC_START), + .formats = (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE), + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 4000, + .rate_max = 48000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = 0x3ffdc, + .period_bytes_min = 8, + .period_bytes_max = 0x3ffdc, + .periods_min = 1, + .periods_max = 0x1ffe, +}; + +static void sis_update_end_offsets(struct voice *voice) +{ + void __iomem *base = voice->ctrl_base; + + if (voice->flags & VOICE_SSO_TIMING) { + /* Calculate the new SSO */ + voice->sso += voice->period_size; + if (voice->sso >= voice->buffer_size) + voice->sso -= voice->buffer_size; + + /* The SSO is in the upper 16 bits of the register. */ + writew(voice->sso & 0xffff, base + SIS_PLAY_DMA_SSO_ESO + 2); + } else if (voice->flags & VOICE_SYNC_TIMING) { + u16 cso = readw(voice->sync_base + SIS_CAPTURE_DMA_FORMAT_CSO); + int target = voice->sync_cso; + int sync; + u16 leo; + + /* Calculate our relative offset between the target and + * the actual CSO value. Since we're operating in a loop, + * if the value is more than half way around, we can + * consider ourselves wrapped. + */ + sync = cso; + sync -= target; + if (sync > (voice->buffer_size / 2)) + sync -= voice->buffer_size; + + /* If sync is negative, then we interrupted too early, and + * we'll need to come back in a few samples and try again. + * There's a minimum wait, as it takes some time for the DMA + * engine to startup, etc... + * + * FIXME: the first interrupt seems to come when we're at + * offset 0, so we have a huge early entry. Not a huge issue, + * but it would be nice if it didn't happen. + */ + if (sync < 0) { + leo = -sync; + if (leo < 16) + leo = 16; + writew(leo, base + SIS_PLAY_DMA_CONTROL); + return; + } + + /* Ok, we interrupted right on time, or (hopefully) just + * a bit late. We'll adjst our next waiting period based + * on how close we got. + * + * We need to stay just behind the actual channel to ensure + * it really is past a period when we get our interrupt -- + * otherwise we'll fall into the early code above and have + * minimum wait time, which could cause issues. + * + * But we don't want to be too late, as we don't want to + * cut into the user's time to refresh the buffers too + * much, especially if we're using small periods. + */ + if (sync < 12) + leo = voice->period_size + 2; + else + leo = voice->period_size - 7; + + sync = voice->sync_cso + voice->period_size; + if (sync >= voice->buffer_size) + sync -= voice->buffer_size; + + voice->sync_cso = sync; + writew(leo, base + SIS_PLAY_DMA_CONTROL); + } +} + +static irqreturn_t sis_interrupt(int irq, void *dev) +{ + struct sis7019 *sis = dev; + unsigned long io = sis->ioport; + struct voice *voice; + u32 intr, status, orig; + int bit; + + /* We only use the DMA interrupts, and we don't enable any other + * source of interrupts. But, it is possible to see an interupt + * status that didn't actually interrupt us, so eliminate anything + * we're not expecting to avoid falsely claiming an IRQ, and an + * ensuing endless loop. + */ + intr = inl(io + SIS_GISR); + intr &= SIS_GISR_AUDIO_PLAY_DMA_IRQ_STATUS | + SIS_GISR_AUDIO_RECORD_DMA_IRQ_STATUS; + if (!intr) + return IRQ_NONE; + + spin_lock(&sis->hw_lock); + do { + status = inl(io + SIS_PISR_A); + if (status) { + orig = status; + voice = sis->voices; + while (status) { + bit = __ffs(status); + status >>= bit + 1; + voice += bit; + snd_pcm_period_elapsed(voice->substream); + sis_update_end_offsets(voice); + voice++; + } + outl(orig, io + SIS_PISR_A); + } + + status = inl(io + SIS_PISR_B); + if (status) { + orig = status; + voice = &sis->voices[32]; + while (status) { + bit = __ffs(status); + status >>= bit + 1; + voice += bit; + snd_pcm_period_elapsed(voice->substream); + sis_update_end_offsets(voice); + voice++; + } + outl(orig, io + SIS_PISR_B); + } + + status = inl(io + SIS_RISR); + if (status) { + voice = &sis->capture_voice; + if (!voice->timing) + snd_pcm_period_elapsed(voice->substream); + + outl(status, io + SIS_RISR); + } + + outl(intr, io + SIS_GISR); + intr = inl(io + SIS_GISR); + intr &= SIS_GISR_AUDIO_PLAY_DMA_IRQ_STATUS | + SIS_GISR_AUDIO_RECORD_DMA_IRQ_STATUS; + } while (intr); + spin_unlock(&sis->hw_lock); + + return IRQ_HANDLED; +} + +static u32 sis_rate_to_delta(unsigned int rate) +{ + u32 delta; + + /* This was copied from the trident driver, but it seems its gotten + * around a bit... nevertheless, it works well. + * + * We special case 44100 and 8000 since rounding with the equation + * does not give us an accurate enough value. For 11025 and 22050 + * the equation gives us the best answer. All other frequencies will + * also use the equation. JDW + */ + if (rate == 44100) + delta = 0xeb3; + else if (rate == 8000) + delta = 0x2ab; + else if (rate == 48000) + delta = 0x1000; + else + delta = (((rate << 12) + 24000) / 48000) & 0x0000ffff; + return delta; +} + +static void sis_free_voice(struct sis7019 *sis, struct voice *voice) +{ + unsigned long flags; + + spin_lock_irqsave(&sis->voice_lock, flags); + if (voice->timing) { + voice->timing->flags = ~(VOICE_IN_USE | VOICE_SSO_TIMING | + VOICE_SYNC_TIMING); + voice->timing = NULL; + } + voice->flags &= ~(VOICE_IN_USE | VOICE_SSO_TIMING | VOICE_SYNC_TIMING); + spin_unlock_irqrestore(&sis->voice_lock, flags); +} + +static struct voice *sis_alloc_playback_voice(struct sis7019 *sis) +{ + struct voice *voice; + unsigned long flags; + int i; + + spin_lock_irqsave(&sis->voice_lock, flags); + + for (i = 0; i < 64; i++) { + voice = &sis->voices[i]; + if (voice->flags & VOICE_IN_USE) + continue; + voice->flags |= VOICE_IN_USE; + goto found_one; + } + voice = NULL; + +found_one: + spin_unlock_irqrestore(&sis->voice_lock, flags); + + return voice; +} + +static int sis_alloc_timing_voice(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + int needed; + + /* If the period size is exactly one half of the buffer, we don't + * need a timing voice, as we can use the middle loop interrupt + * to clock out the periods. + */ + needed = (params_period_size(hw_params) != + (params_buffer_size(hw_params) / 2)); + + if (needed && !voice->timing) { + voice->timing = sis_alloc_playback_voice(sis); + if (!voice->timing) + return -ENOMEM; + voice->timing->substream = substream; + } else if (!needed && voice->timing) { + sis_free_voice(sis, voice); + voice->timing = NULL; + } + + return 0; +} + +static int sis_playback_open(struct snd_pcm_substream *substream) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice; + + voice = sis_alloc_playback_voice(sis); + if (!voice) + return -EAGAIN; + + voice->substream = substream; + runtime->private_data = voice; + runtime->hw = sis_playback_hw_info; + snd_pcm_set_sync(substream); + return 0; +} + +static int sis_substream_close(struct snd_pcm_substream *substream) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + + sis_free_voice(sis, voice); + return 0; +} + +static int sis_playback_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); +} + +static int sis_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static int sis_pcm_playback_prepare(struct snd_pcm_substream *substream) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + void __iomem *ctrl_base = voice->ctrl_base; + void __iomem *wave_base = voice->wave_base; + u32 format, dma_addr, control, sso_eso, delta, reg; + u16 leo; + + /* Prepare the parameters for this channel before we take the lock. + */ + format = 0; + if (snd_pcm_format_width(runtime->format) == 8) + format |= SIS_PLAY_DMA_FORMAT_8BIT; + if (!snd_pcm_format_signed(runtime->format)) + format |= SIS_PLAY_DMA_FORMAT_UNSIGNED; + if (runtime->channels == 1) + format |= SIS_PLAY_DMA_FORMAT_MONO; + + dma_addr = runtime->dma_addr; + leo = runtime->buffer_size - 1; + if (runtime->period_size == (runtime->buffer_size / 2)) { + control = leo | SIS_PLAY_DMA_LOOP; + control |= SIS_PLAY_DMA_INTR_AT_LEO | SIS_PLAY_DMA_INTR_AT_MLP; + sso_eso = leo; + } else { + voice->flags |= VOICE_SSO_TIMING; + voice->sso = runtime->period_size; + voice->period_size = runtime->period_size; + voice->buffer_size = runtime->buffer_size; + + control = leo | SIS_PLAY_DMA_LOOP; + control |= SIS_PLAY_DMA_INTR_AT_SSO | SIS_PLAY_DMA_INTR_AT_LEO; + sso_eso = leo | (runtime->period_size - 1) << 16; + } + + delta = sis_rate_to_delta(runtime->rate); + + + /* Ok, we're ready to go, so take the lock and set up the channel. + */ + spin_lock_irq(&sis->hw_lock); + + writel(format, ctrl_base + SIS_PLAY_DMA_FORMAT_CSO); + writel(dma_addr, ctrl_base + SIS_PLAY_DMA_BASE); + writel(control, ctrl_base + SIS_PLAY_DMA_CONTROL); + writel(sso_eso, ctrl_base + SIS_PLAY_DMA_SSO_ESO); + + for (reg = 0; reg < SIS_WAVE_SIZE; reg += 4) + writel(0, wave_base + reg); + + writel(SIS_WAVE_GENERAL_WAVE_VOLUME, wave_base + SIS_WAVE_GENERAL); + writel(delta << 16, wave_base + SIS_WAVE_GENERAL_ARTICULATION); + writel(SIS_WAVE_CHANNEL_CONTROL_FIRST_SAMPLE | + SIS_WAVE_CHANNEL_CONTROL_AMP_ENABLE | + SIS_WAVE_CHANNEL_CONTROL_INTERPOLATE_ENABLE, + wave_base + SIS_WAVE_CHANNEL_CONTROL); + + /* Force PCI writes to post. */ + readl(ctrl_base); + spin_unlock_irq(&sis->hw_lock); + + return 0; +} + +static int sis_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + unsigned long io = sis->ioport; + struct snd_pcm_substream *s; + struct voice *voice; + void *chip; + int starting; + u32 record = 0; + u32 play[2] = { 0, 0 }; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: + starting = 1; + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: + starting = 0; + break; + default: + return -EINVAL; + } + + snd_pcm_group_for_each_entry(s, substream) { + /* Make sure it is for us... */ + chip = snd_pcm_substream_chip(s); + if (chip != sis) + continue; + + voice = s->runtime->private_data; + if (voice->flags & VOICE_CAPTURE) { + record |= 1 << voice->num; + voice = voice->timing; + } + + /* voice could be NULL if this a recording stream, and it + * doesn't have an external timing channel. + */ + if (voice) + play[voice->num / 32] |= 1 << (voice->num & 0x1f); + + snd_pcm_trigger_done(s, substream); + } + + spin_lock(&sis->hw_lock); + if (starting) { + if (record) + outl(record, io + SIS_RECORD_START_REG); + if (play[0]) + outl(play[0], io + SIS_PLAY_START_A_REG); + if (play[1]) + outl(play[1], io + SIS_PLAY_START_B_REG); + } else { + if (record) + outl(record, io + SIS_RECORD_STOP_REG); + if (play[0]) + outl(play[0], io + SIS_PLAY_STOP_A_REG); + if (play[1]) + outl(play[1], io + SIS_PLAY_STOP_B_REG); + } + spin_unlock(&sis->hw_lock); + return 0; +} + +static snd_pcm_uframes_t sis_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + u32 cso; + + cso = readl(voice->ctrl_base + SIS_PLAY_DMA_FORMAT_CSO); + cso &= 0xffff; + return cso; +} + +static int sis_capture_open(struct snd_pcm_substream *substream) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = &sis->capture_voice; + unsigned long flags; + + /* FIXME: The driver only supports recording from one channel + * at the moment, but it could support more. + */ + spin_lock_irqsave(&sis->voice_lock, flags); + if (voice->flags & VOICE_IN_USE) + voice = NULL; + else + voice->flags |= VOICE_IN_USE; + spin_unlock_irqrestore(&sis->voice_lock, flags); + + if (!voice) + return -EAGAIN; + + voice->substream = substream; + runtime->private_data = voice; + runtime->hw = sis_capture_hw_info; + runtime->hw.rates = sis->ac97[0]->rates[AC97_RATES_ADC]; + snd_pcm_limit_hw_rates(runtime); + snd_pcm_set_sync(substream); + return 0; +} + +static int sis_capture_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + int rc; + + rc = snd_ac97_set_rate(sis->ac97[0], AC97_PCM_LR_ADC_RATE, + params_rate(hw_params)); + if (rc) + goto out; + + rc = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + if (rc < 0) + goto out; + + rc = sis_alloc_timing_voice(substream, hw_params); + +out: + return rc; +} + +static int sis_pcm_capture_prepare(struct snd_pcm_substream *substream) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + struct voice *timing = voice->timing; + void __iomem *rec_base = voice->ctrl_base; + void __iomem *play_base = NULL; + void __iomem *wave_base = NULL; + u32 format, dma_addr, rec_ctrl; + u16 leo; + + /* The following variables are only used if there is a timing channel. + */ + u32 uninitialized_var(timing_ctrl); + u32 uninitialized_var(sso_eso); + u32 uninitialized_var(delta); + + format = 0; + if (snd_pcm_format_width(runtime->format) == 8) + format = SIS_CAPTURE_DMA_FORMAT_8BIT; + if (!snd_pcm_format_signed(runtime->format)) + format |= SIS_CAPTURE_DMA_FORMAT_UNSIGNED; + if (runtime->channels == 1) + format |= SIS_CAPTURE_DMA_FORMAT_MONO; + + dma_addr = runtime->dma_addr; + leo = runtime->buffer_size - 1; + rec_ctrl = leo | SIS_CAPTURE_DMA_LOOP; + if (timing) { + /* We're using a timing voice, so the capture voice will + * not need to generate interrupts. + */ + timing->flags |= VOICE_SYNC_TIMING; + timing->sync_cso = runtime->period_size; + timing->period_size = runtime->period_size; + timing->buffer_size = runtime->buffer_size; + timing->sync_base = voice->ctrl_base; + + timing_ctrl = leo + 16; + timing_ctrl |= SIS_PLAY_DMA_LOOP | SIS_PLAY_DMA_INTR_AT_LEO; + sso_eso = leo; + + delta = sis_rate_to_delta(runtime->rate); + + play_base = timing->ctrl_base; + wave_base = timing->wave_base; + } else { + /* We've just got two periods, so we can use the capture + * voice's interrupt generation. + */ + rec_ctrl |= SIS_CAPTURE_DMA_INTR_AT_MLP; + rec_ctrl |= SIS_CAPTURE_DMA_INTR_AT_LEO; + } + + /* We're ready to program the channel(s), so take the lock and go. + */ + spin_lock_irq(&sis->hw_lock); + + writel(format, rec_base + SIS_CAPTURE_DMA_FORMAT_CSO); + writel(dma_addr, rec_base + SIS_CAPTURE_DMA_BASE); + writel(rec_ctrl, rec_base + SIS_CAPTURE_DMA_CONTROL); + + if (play_base) { + u32 reg; + + writel(format, play_base + SIS_PLAY_DMA_FORMAT_CSO); + writel(dma_addr, play_base + SIS_PLAY_DMA_BASE); + writel(timing_ctrl, play_base + SIS_PLAY_DMA_CONTROL); + writel(sso_eso, play_base + SIS_PLAY_DMA_SSO_ESO); + + for (reg = 0; reg < SIS_WAVE_SIZE; reg += 4) + writel(0, wave_base + reg); + + /* We don't want the timing channel to generate any sound, + * so set it to maximum attenuation (31.75 dB), and run it + * through the music amplifier, which has been set to add + * an additional 64 dB of attenuation. That will be 95.76 dB + * total. + * + * FIXME: While this works well in practice, purists may + * wish to investigate using the mixer lists to further + * attenuate the timing channel -- or set aside a page + * of zeros for silence. This may also free up the music + * amplifier for other purposes. + */ + writel(SIS_WAVE_GENERAL_MUSIC_VOLUME | (0x7f << 24), + wave_base + SIS_WAVE_GENERAL); + writel(delta << 16, wave_base + SIS_WAVE_GENERAL_ARTICULATION); + writel(SIS_WAVE_CHANNEL_CONTROL_FIRST_SAMPLE | + SIS_WAVE_CHANNEL_CONTROL_AMP_ENABLE | + SIS_WAVE_CHANNEL_CONTROL_INTERPOLATE_ENABLE, + wave_base + SIS_WAVE_CHANNEL_CONTROL); + } + + readl(rec_base + 0); + + spin_unlock_irq(&sis->hw_lock); + return 0; +} + +static struct snd_pcm_ops sis_playback_ops = { + .open = sis_playback_open, + .close = sis_substream_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = sis_playback_hw_params, + .hw_free = sis_hw_free, + .prepare = sis_pcm_playback_prepare, + .trigger = sis_pcm_trigger, + .pointer = sis_pcm_pointer, +}; + +static struct snd_pcm_ops sis_capture_ops = { + .open = sis_capture_open, + .close = sis_substream_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = sis_capture_hw_params, + .hw_free = sis_hw_free, + .prepare = sis_pcm_capture_prepare, + .trigger = sis_pcm_trigger, + .pointer = sis_pcm_pointer, +}; + +static int __devinit sis_pcm_create(struct sis7019 *sis) +{ + struct snd_pcm *pcm; + int rc; + + rc = snd_pcm_new(sis->card, "SiS 7019", 0, 40, 1, &pcm); + if (rc) + return rc; + + pcm->private_data = sis; + strcpy(pcm->name, "SiS 7019"); + sis->pcm = pcm; + + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &sis_playback_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &sis_capture_ops); + + /* Try to preallocate some memory, but it's not the end of the + * world if this fails. + */ + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(sis->pci), 64*1024, 128*1024); + + return 0; +} + +static unsigned short sis_ac97_rw(struct sis7019 *sis, int codec, u32 cmd) +{ + unsigned long io = sis->ioport; + unsigned long flags; + unsigned short val = 0xffff; + u16 status; + u16 rdy; + int count; + const static u16 codec_ready[3] = { + SIS_AC97_STATUS_CODEC_READY, + SIS_AC97_STATUS_CODEC2_READY, + SIS_AC97_STATUS_CODEC3_READY, + }; + + rdy = codec_ready[codec]; + + spin_lock_irqsave(&sis->hw_lock, flags); + + /* Get the AC97 semaphore ... + */ + count = 0xffff; + while ((inw(io + SIS_AC97_SEMA) & SIS_AC97_SEMA_BUSY) && --count) + udelay(1); + + if (!count) + goto timeout; + + /* ... and wait for any outstanding commands to complete ... + */ + count = 0xffff; + do { + status = inw(io + SIS_AC97_STATUS); + if ((status & rdy) && !(status & SIS_AC97_STATUS_BUSY)) + break; + + udelay(1); + } while (--count); + + if (!count) + goto timeout_sema; + + /* ... before sending our command and waiting for it to finish ... + */ + outl(cmd, io + SIS_AC97_CMD); + udelay(10); + + count = 0xffff; + while ((inw(io + SIS_AC97_STATUS) & SIS_AC97_STATUS_BUSY) && --count) + udelay(1); + + /* ... and reading the results (if any). + */ + val = inl(io + SIS_AC97_CMD) >> 16; + +timeout_sema: + outl(SIS_AC97_SEMA_RELEASE, io + SIS_AC97_SEMA); +timeout: + spin_unlock_irqrestore(&sis->hw_lock, flags); + + if (!count) { + printk(KERN_ERR "sis7019: ac97 codec %d timeout cmd 0x%08x\n", + codec, cmd); + } + + return val; +} + +static void sis_ac97_write(struct snd_ac97 *ac97, unsigned short reg, + unsigned short val) +{ + const static u32 cmd[3] = { + SIS_AC97_CMD_CODEC_WRITE, + SIS_AC97_CMD_CODEC2_WRITE, + SIS_AC97_CMD_CODEC3_WRITE, + }; + sis_ac97_rw(ac97->private_data, ac97->num, + (val << 16) | (reg << 8) | cmd[ac97->num]); +} + +static unsigned short sis_ac97_read(struct snd_ac97 *ac97, unsigned short reg) +{ + const static u32 cmd[3] = { + SIS_AC97_CMD_CODEC_READ, + SIS_AC97_CMD_CODEC2_READ, + SIS_AC97_CMD_CODEC3_READ, + }; + return sis_ac97_rw(ac97->private_data, ac97->num, + (reg << 8) | cmd[ac97->num]); +} + +static int __devinit sis_mixer_create(struct sis7019 *sis) +{ + struct snd_ac97_bus *bus; + struct snd_ac97_template ac97; + static struct snd_ac97_bus_ops ops = { + .write = sis_ac97_write, + .read = sis_ac97_read, + }; + int rc; + + memset(&ac97, 0, sizeof(ac97)); + ac97.private_data = sis; + + rc = snd_ac97_bus(sis->card, 0, &ops, NULL, &bus); + if (!rc && sis->codecs_present & SIS_PRIMARY_CODEC_PRESENT) + rc = snd_ac97_mixer(bus, &ac97, &sis->ac97[0]); + ac97.num = 1; + if (!rc && (sis->codecs_present & SIS_SECONDARY_CODEC_PRESENT)) + rc = snd_ac97_mixer(bus, &ac97, &sis->ac97[1]); + ac97.num = 2; + if (!rc && (sis->codecs_present & SIS_TERTIARY_CODEC_PRESENT)) + rc = snd_ac97_mixer(bus, &ac97, &sis->ac97[2]); + + /* If we return an error here, then snd_card_free() should + * free up any ac97 codecs that got created, as well as the bus. + */ + return rc; +} + +static int sis_chip_free(struct sis7019 *sis) +{ + /* Reset the chip, and disable all interrputs. + */ + outl(SIS_GCR_SOFTWARE_RESET, sis->ioport + SIS_GCR); + udelay(10); + outl(0, sis->ioport + SIS_GCR); + outl(0, sis->ioport + SIS_GIER); + + /* Now, free everything we allocated. + */ + if (sis->irq >= 0) + free_irq(sis->irq, sis); + + if (sis->ioaddr) + iounmap(sis->ioaddr); + + pci_release_regions(sis->pci); + pci_disable_device(sis->pci); + + return 0; +} + +static int sis_dev_free(struct snd_device *dev) +{ + struct sis7019 *sis = dev->device_data; + return sis_chip_free(sis); +} + +static int sis_chip_init(struct sis7019 *sis) +{ + unsigned long io = sis->ioport; + void __iomem *ioaddr = sis->ioaddr; + u16 status; + int count; + int i; + + /* Reset the audio controller + */ + outl(SIS_GCR_SOFTWARE_RESET, io + SIS_GCR); + udelay(10); + outl(0, io + SIS_GCR); + + /* Get the AC-link semaphore, and reset the codecs + */ + count = 0xffff; + while ((inw(io + SIS_AC97_SEMA) & SIS_AC97_SEMA_BUSY) && --count) + udelay(1); + + if (!count) + return -EIO; + + outl(SIS_AC97_CMD_CODEC_COLD_RESET, io + SIS_AC97_CMD); + udelay(10); + + count = 0xffff; + while ((inw(io + SIS_AC97_STATUS) & SIS_AC97_STATUS_BUSY) && --count) + udelay(1); + + /* Now that we've finished the reset, find out what's attached. + */ + status = inl(io + SIS_AC97_STATUS); + if (status & SIS_AC97_STATUS_CODEC_READY) + sis->codecs_present |= SIS_PRIMARY_CODEC_PRESENT; + if (status & SIS_AC97_STATUS_CODEC2_READY) + sis->codecs_present |= SIS_SECONDARY_CODEC_PRESENT; + if (status & SIS_AC97_STATUS_CODEC3_READY) + sis->codecs_present |= SIS_TERTIARY_CODEC_PRESENT; + + /* All done, let go of the semaphore, and check for errors + */ + outl(SIS_AC97_SEMA_RELEASE, io + SIS_AC97_SEMA); + if (!sis->codecs_present || !count) + return -EIO; + + /* Let the hardware know that the audio driver is alive, + * and enable PCM slots on the AC-link for L/R playback (3 & 4) and + * record channels. We're going to want to use Variable Rate Audio + * for recording, to avoid needlessly resampling from 48kHZ. + */ + outl(SIS_AC97_CONF_AUDIO_ALIVE, io + SIS_AC97_CONF); + outl(SIS_AC97_CONF_AUDIO_ALIVE | SIS_AC97_CONF_PCM_LR_ENABLE | + SIS_AC97_CONF_PCM_CAP_MIC_ENABLE | + SIS_AC97_CONF_PCM_CAP_LR_ENABLE | + SIS_AC97_CONF_CODEC_VRA_ENABLE, io + SIS_AC97_CONF); + + /* All AC97 PCM slots should be sourced from sub-mixer 0. + */ + outl(0, io + SIS_AC97_PSR); + + /* There is only one valid DMA setup for a PCI environment. + */ + outl(SIS_DMA_CSR_PCI_SETTINGS, io + SIS_DMA_CSR); + + /* Reset the syncronization groups for all of the channels + * to be asyncronous. If we start doing SPDIF or 5.1 sound, etc. + * we'll need to change how we handle these. Until then, we just + * assign sub-mixer 0 to all playback channels, and avoid any + * attenuation on the audio. + */ + outl(0, io + SIS_PLAY_SYNC_GROUP_A); + outl(0, io + SIS_PLAY_SYNC_GROUP_B); + outl(0, io + SIS_PLAY_SYNC_GROUP_C); + outl(0, io + SIS_PLAY_SYNC_GROUP_D); + outl(0, io + SIS_MIXER_SYNC_GROUP); + + for (i = 0; i < 64; i++) { + writel(i, SIS_MIXER_START_ADDR(ioaddr, i)); + writel(SIS_MIXER_RIGHT_NO_ATTEN | SIS_MIXER_LEFT_NO_ATTEN | + SIS_MIXER_DEST_0, SIS_MIXER_ADDR(ioaddr, i)); + } + + /* Don't attenuate any audio set for the wave amplifier. + * Do maximum attenuation on audio set to use the music amplifier, + * as we'll set any timing channels to use that. + */ + outl(0xffff0000, SIS_WEVCR); + + /* Go ahead and enable the DMA interrupts. They won't go live + * until we start a channel. + */ + outl(SIS_GIER_AUDIO_PLAY_DMA_IRQ_ENABLE | + SIS_GIER_AUDIO_RECORD_DMA_IRQ_ENABLE, io + SIS_GIER); + + return 0; +} + +static int __devinit sis_chip_create(struct snd_card *card, + struct pci_dev *pci) +{ + struct sis7019 *sis = card->private_data; + struct voice *voice; + static struct snd_device_ops ops = { + .dev_free = sis_dev_free, + }; + int rc; + int i; + + rc = pci_enable_device(pci); + if (rc) + goto error_out; + + /* FIXME: Not sure of our DMA limit, use 32 bit mask for now. + */ + if (pci_set_dma_mask(pci, DMA_32BIT_MASK) < 0) { + printk(KERN_ERR "sis7019: architecture does not support " + "32-bit PCI busmaster DMA"); + goto error_out_enabled; + } + + memset(sis, 0, sizeof(*sis)); + spin_lock_init(&sis->hw_lock); + spin_lock_init(&sis->voice_lock); + sis->card = card; + sis->pci = pci; + sis->irq = -1; + sis->ioport = pci_resource_start(pci, 0); + + rc = pci_request_regions(pci, "SiS 7019"); + if (rc) { + printk(KERN_ERR "sis7019: unable request regions\n"); + goto error_out_enabled; + } + + rc = -EIO; + sis->ioaddr = ioremap(pci_resource_start(pci, 1), 0x4000); + if (!sis->ioaddr) { + printk(KERN_ERR "sis7019: unable to remap MMIO, aborting\n"); + goto error_out_cleanup; + } + + rc = sis_chip_init(sis); + if (rc) + goto error_out_cleanup; + + if (request_irq(pci->irq, sis_interrupt, IRQF_DISABLED|IRQF_SHARED, + "SiS 7019", sis)) { + printk(KERN_ERR "unable to allocate irq %d\n", sis->irq); + goto error_out_cleanup; + } + + sis->irq = pci->irq; + pci_set_master(pci); + + for (i = 0; i < 64; i++) { + voice = &sis->voices[i]; + voice->num = i; + voice->ctrl_base = SIS_PLAY_DMA_ADDR(sis->ioaddr, i); + voice->wave_base = SIS_WAVE_ADDR(sis->ioaddr, i); + } + + voice = &sis->capture_voice; + voice->flags = VOICE_CAPTURE; + voice->num = SIS_CAPTURE_CHAN_AC97_PCM_IN; + voice->ctrl_base = SIS_CAPTURE_DMA_ADDR(sis->ioaddr, voice->num); + + rc = snd_device_new(card, SNDRV_DEV_LOWLEVEL, sis, &ops); + if (rc) + goto error_out_cleanup; + + snd_card_set_dev(card, &pci->dev); + + return 0; + +error_out_cleanup: + sis_chip_free(sis); + +error_out_enabled: + pci_disable_device(pci); + +error_out: + return rc; +} + +static int __devinit snd_sis7019_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) +{ + struct snd_card *card; + struct sis7019 *sis; + static int dev; + int rc; + + rc = -ENODEV; + if (dev >= SNDRV_CARDS) + goto error_out; + + rc = -ENOENT; + if (!enable[dev]) { + dev++; + goto error_out; + } + + rc = -ENOMEM; + card = snd_card_new(index[dev], id[dev], THIS_MODULE, sizeof(*sis)); + if (!card) + goto error_out; + + rc = sis_chip_create(card, pci); + if (rc) + goto card_error_out; + + sis = card->private_data; + strcpy(card->driver, "SiS 7019"); + strcpy(card->shortname, "SiS 7019"); + + rc = sis_mixer_create(sis); + if (rc) + goto card_error_out; + + rc = sis_pcm_create(sis); + if (rc) + goto card_error_out; + + snprintf(card->longname, sizeof(card->longname), + "%s Audio Accelerator with %s at 0x%lx, irq %d", + card->shortname, snd_ac97_get_short_name(sis->ac97[0]), + sis->ioport, sis->irq); + + rc = snd_card_register(card); + if (rc) + goto card_error_out; + + pci_set_drvdata(pci, card); + dev++; + return 0; + +card_error_out: + snd_card_free(card); + +error_out: + return rc; +} + +static void __devexit snd_sis7019_remove(struct pci_dev *pci) +{ + snd_card_free(pci_get_drvdata(pci)); + pci_set_drvdata(pci, NULL); +} + +static struct pci_driver sis7019_driver = { + .name = "SiS7019", + .id_table = snd_sis7019_ids, + .probe = snd_sis7019_probe, + .remove = __devexit_p(snd_sis7019_remove), + + /* FIXME: need to support power management */ +}; + +static int __init sis7019_init(void) +{ + return pci_register_driver(&sis7019_driver); +} + +static void __exit sis7019_exit(void) +{ + pci_unregister_driver(&sis7019_driver); +} + +module_init(sis7019_init); +module_exit(sis7019_exit); -- 1.5.0.6

3 5

Re: [alsa-devel] different user causes 'no such device'
by william estrada 09 Oct '07

09 Oct '07

Thanks, I will try this. William Estrada MrUmunhum(a)popdial.com Mt-Umunhum-Wireless.net ( http://Mt-Umunhum-Wireless.net ) Ymessenger: MrUmunhum Takashi Iwai wrote: > At Wed, 03 Oct 2007 12:55:27 -0700, > william estrada wrote: >> Hi guys, >> >> I have what seams like a interesting problem. I have written an ALSA app >> to record wav files and it runs while I am logged in an terminal as the >> logged in Gnome user or root. But when I su to another user, it can't >> find the default sound card. >> >> I am running FC6. I tried changing own of /dev/dsp to 'nobody', that did >> not work. >> >> Anyone know of this problem and a fix for it. >> >> Thanks for your time. >> >> -- >> William Estrada >> MrUmunhum(a)popdial.com >> Mt-Umunhum-Wireless.net ( http://Mt-Umunhum-Wireless.net ) >> Ymessenger: MrUmunhum >> >>> MSGrec - Version 1.1, Mt-Umunhum-Wireless.net >>> WavMSG: Paused: enter 'q' to stop, <ENTER> to start. >>> ALSA lib confmisc.c:848:(snd_func_card_driver) cannot find card '0' >>> ALSA lib conf.c:3500:(_snd_config_evaluate) function snd_func_card_driver returned error: No such device >>> ALSA lib confmisc.c:397:(snd_func_concat) error evaluating strings >>> ALSA lib conf.c:3500:(_snd_config_evaluate) function snd_func_concat returned error: No such device >>> ALSA lib confmisc.c:1248:(snd_func_refer) error evaluating name >>> ALSA lib conf.c:3500:(_snd_config_evaluate) function snd_func_refer returned error: No such device >>> ALSA lib conf.c:3972:(snd_config_expand) Evaluate error: No such device >>> ALSA lib pcm.c:2145:(snd_pcm_open_noupdate) Unknown PCM default > > ALSA-native apps access only ALSA device files, usually /dev/snd/*, > not /dev/dsp*, those for OSS. In this case, it should be > /dev/snd/controlC* files. Check the premission of these files. > > > Takashi >

1 0

[alsa-devel] ALSA Support for Maya44 PCI?
by Nicolas Rodriguez 09 Oct '07

09 Oct '07

I bought a Maya44 PCI 4-channel soundcard and I'll like to use it in Linux. I couldn't get it to work with Ubuntu (I don't remember the ALSA version, sorry). Do you support this card? If you don't, I can create the driver for it. I've read somewhere that ESI-PRO can give the card specifications to developers interested in build drivers for their cards. I've sent them an email asking for the card specifications for I've got no response. What do you recommend? Thanks in advance, Nico.

1 0

[alsa-devel] Envy24control s/pdif input bugfix
by Stephan Zaubzer 09 Oct '07

09 Oct '07

Hi! This patch against the stable version of envy24control fixes a bug where the S/Pdif input switch doesn't work on Terratec DMX 6fire cards. This bug first appeared in version 1.0.11rc3. diff -ubr alsa-tools-1.0.14patch/envy24control/hardware.c alsa-tools-1.0.14/envy24control/hardware.c --- alsa-tools-1.0.14patch/envy24control/hardware.c 2007-09-23 15:58:53.000000000 +0200 +++ alsa-tools-1.0.14/envy24control/hardware.c 2007-06-04 13:27:25.000000000 +0200 @@ -775,7 +775,7 @@ snd_ctl_elem_value_set_interface(volume_rate, SND_CTL_ELEM_IFACE_MIXER); snd_ctl_elem_value_set_name(volume_rate, "Multi Track Volume Rate"); - if (card_is_dmx6fire) { + if (! card_is_dmx6fire) { snd_ctl_elem_value_set_interface(spdif_input, SND_CTL_ELEM_IFACE_MIXER); snd_ctl_elem_value_set_name(spdif_input, "Optical Digital Input Switch"); } else { Regards Stephan

3 4

Re: [alsa-devel] [Alsa-user] Asus M2NPV-VM snd_hda_intel problem - also with Alsa 1.0.15rc3
by Ma Begaj 09 Oct '07

09 Oct '07

> Firstly, I don't have the same board. Similar, but different. > > I have an Asus A8N-VM CSM board, but it has the same AD1986A chip on it. > I didn't have the same issue with sound muting, but I did have issues > with surround (it just didn't work). I don't remember the details of how > > > --- patch_analog.c.backup 2007-04-14 11:01:17.209146249 +0100 > +++ patch_analog.c 2007-04-14 13:37:47.005206476 +0100 > @@ -895,14 +895,14 @@ > case AD1986A_3STACK: > spec->num_mixers = 2; > spec->mixers[1] = ad1986a_3st_mixers; > - spec->num_init_verbs = 3; > +/* spec->num_init_verbs = 3; > spec->init_verbs[1] = ad1986a_3st_init_verbs; > - spec->init_verbs[2] = ad1986a_ch2_init; > + spec->init_verbs[2] = ad1986a_ch2_init; */ > spec->channel_mode = ad1986a_modes; > spec->num_channel_mode = ARRAY_SIZE(ad1986a_modes); > spec->need_dac_fix = 1; > - spec->multiout.max_channels = 2; > - spec->multiout.num_dacs = 1; > + spec->multiout.max_channels = 6; > + spec->multiout.num_dacs = 3; > break; > case AD1986A_LAPTOP: > spec->mixers[0] = ad1986a_laptop_mixers; > thanks for your try. it helped partly ... I tried 3 patches (incl. yours) and only yours helped partly at least. I patched Alsa 1.0.15rc3. Sound mutes as soon as I change volume, and I can only rmmod/modprobe snd-hda-intel to get it back. But Surround is working with your patch as I can see: "speaker-test -Dplug:surround51 -c6 -t waw" recognizes correctly all speakears. "speaker-test -c6 -t waw" does not recognize the speakers correctly, but it probably should not, or ?! MythTv is playing stereo files only on the right speakers. I do not now is this a MythTv or Alsa problem. probably MythTv and I should create a .asoundrc which should correct this somehow. I did not try hg, because I thought that Alsa 1.0.15rc3 is very new. Thanks. M.

1 1

[alsa-devel] Capturing of Samsung Q1 Ultra (ALC262) does not work
by Pengcheng Zou 09 Oct '07

09 Oct '07

Hi, capture function of the Samsung Q1 Ultra (using ALC262 codec) does not work. I tried the most recent ALSA build (alsa-driver-1.0.15rc3), the playback works well, however the recording does not work. Samsung Q1 (using AD1986A codec) has been supported already, wonder if there is any patches for Q1 Ultra? please note Samsung released 2 types of UPMC, Q1 and Q1 Ultra. the Q1 ultra is the one I have: http://computers.pricegrabber.com/tablet-pcs/m/37616779/details/ TIA, -- Pengcheng

3 2

[alsa-devel] snd_pcm_drain and lock-ups
by Mike Gorse 08 Oct '07

08 Oct '07

Hi all, I've followed Arve Knudsen over here. I am getting a lock-up when calling snd_pcm_drain(). This happens when using espeak, which is using portaudio. Portaudio calls snd_pcm_drain while trying to recover from a XRUN. snd_pcm_drain is calling snd_pcm_rate_drain in my case. snd_pcm_rate_drain then calls snd_pcm_wait. snd_pcm_wait first checks if snd_pcm_mmap_avail(pcm) >= pcm->avail_min. snd_pcm_map_avail is returning a number less than 300, often between 135 and 140 but not always, where pcm->avail_min == 1024. So it does not check for the XRUN state since it is conditional on snd_pcm_mmap_avail returning >= 1024. This is on my laptop, which is running Linux 2.6.22 and alsa-lib 1.0.15-rc1 and uses the snd-intel8x0 driver. Does this give people enough information to tell what might be going on? I have a small test program that causes the lock-up on my laptop where the rate plugin gets used (it sometimes locks up immediately, sometimes after a few seconds), but it needs espeak and portaudio to be installed. Thanks, -- Mike Gorse / AIM:linvortex / http://mgorse.freeshell.org --

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Re: [alsa-devel] [Alsa-user] How to get set/fixed sample rate of ALSA device?
by Rene Herman 08 Oct '07

08 Oct '07

On 10/08/2007 12:41 AM, Mark Constable wrote: > Any chance that your program could take an arg to investigate > other cards (or perhaps plugins) and also return an errno so > it could be used in scripts ? Only if you promise to never drop CCs again (although I don't know about plugins). > # gcc -W -Wall -o snd_rate snd_rate.c -l asound > snd_rate.c: In function 'main': > snd_rate.c:14: warning: the address of ‘params’ will always evaluate as 'true' That's the ALSA snd_pcm_hw_params_alloca(). No such warning here, but I've made it a regular malloc. > # snd_rate > snd_pcm_open: Device or resource busy > > In fact, why isn't there a single simple "alsainfo" type > program that tells a user some (or a lot) of info about > their hardware and alsa-lib install ? I guess because noone wrote one :-) It seems by the way that dir != 0 _can_ be returned for hw devices, and this just ignores it. Just some rounding issue I guess. Rene. /* gcc -W -Wall -o alsa-rate alsa-rate.c -lasound */ #include <stdio.h> #include <alsa/asoundlib.h> int main(int argc, char *argv[]) { char *device; snd_pcm_t *handle; snd_pcm_hw_params_t *params; unsigned int min; unsigned int max; int dir; int err; device = argc > 1 ? argv[1] : "hw:0,0"; err = snd_pcm_open(&handle, device, SND_PCM_STREAM_PLAYBACK, 0); if (err < 0) { fprintf(stderr, "snd_pcm_open: %s\n", snd_strerror(err)); return err; } err = snd_pcm_hw_params_malloc(&params); if (err < 0) { fprintf(stderr, "snd_pcm_hw_params_alloc: %s\n", snd_strerror(err)); return err; } err = snd_pcm_hw_params_any(handle, params); if (err < 0) { fprintf(stderr, "snd_pcm_hw_params_any: %s\n", snd_strerror(err)); return err; } err = snd_pcm_close(handle); if (err < 0) { fprintf(stderr, "snd_pcm_close: %s\n", snd_strerror(err)); return err; } err = snd_pcm_hw_params_get_rate_min(params, &min, &dir); if (err < 0) { fprintf(stderr, "snd_pcm_hw_params_get_rate_min: %s\n", snd_strerror(err)); return err; } err = snd_pcm_hw_params_get_rate_max(params, &max, &dir); if (err < 0) { fprintf(stderr, "snd_pcm_hw_params_get_rate_max: %s\n", snd_strerror(err)); return err; } snd_pcm_hw_params_free(params); printf("%u-%u\n", min, max); return 0; }

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[alsa-devel] Use-before-initialization in alsa-oss
by Alexander E. Patrakov 06 Oct '07

06 Oct '07

dsp->oss_format was used before it was set correctly spotted by running "aoss flite_time 10:00" Signed-Off-By: Alexander E. Patrakov --- alsa-oss-1.0.14.orig/alsa/pcm.c 2007-09-29 16:12:44.000000000 +0600 +++ alsa-oss-1.0.14/alsa/pcm.c 2007-09-29 16:13:06.000000000 +0600 @@ -215,10 +215,10 @@ unsigned int rate, periods_min; if (!pcm) continue; + dsp->format = oss_format_to_alsa(dsp->oss_format); str->frame_bytes = snd_pcm_format_physical_width(dsp->format) * dsp->channels / 8; snd_pcm_hw_params_alloca(&hw); snd_pcm_hw_params_any(pcm, hw); - dsp->format = oss_format_to_alsa(dsp->oss_format); err = snd_pcm_hw_params_set_format(pcm, hw, dsp->format); if (err < 0) -- Alexander E. Patrakov

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[alsa-devel] ASoC v2 on PowerPC
by Jon Smirl 05 Oct '07

05 Oct '07

I'm looking into write a ASoC v2 driver for the Efika. Efika uses a Freescale MPC5200B and a STAC9766. This is a practice driver before writing one for our target hardware (which doesn't work yet, same CPU but different codec). I've brought the changes from wolfson/asoc-v2-dev into my powerpc tree and got a couple of config errors with ARCH=powerpc. sound/soc/pxa/Kconfig:23:warning: 'select' used by config symbol 'SND_PXA2XX_SOC_SSP' refers to undefined symbol 'PXA_SSP' sound/soc/au1x/Kconfig:35:warning: 'select' used by config symbol 'SND_SOC_EXM1200_SSI' refers to undefined symbol 'SND_SOC_CS4251X' Suggestions on the best way to start? Target hardware also has a WM8804 on it. -- Jon Smirl jonsmirl(a)gmail.com

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[alsa-devel] Why was the built-in software volume control removed?
by kyle cronan 04 Oct '07

04 Oct '07

I think I may have posted this to the wrong list. There wasn't any response yet on alsa-user, so I'm reposting here. Summary: I'm seeing a lot of glitches ever since the built-in software volume was removed. What was the reason for this change? Softvol doesn't work nearly as well for me. ________________ Hi everyone, I recently updated my media center which is running Fedora devel. It pulled down a new kernel with updated ALSA drivers for my HDA audio chip (Intel 82801H + ALC883). I use IEC958 output to a pair of studio monitors (good sound with minimal fuss). The first thing I noticed about the update was that my software volume control was no longer saving state. It was resetting to something around 80% every time an app opened the audio device, a level which, in my setup, is quite loud! Then I saw that in alsamixer I had no master volume control, not even a PCM control! So I started experimented with my settings, and I found that after I selected "PCM" under "Default Mixer Tracks" in gnome-sound-properties I had a PCM volume control again (still no master), and apps using alsa output would once again remember the system volume. Don't know what that gnome setting actually did, but I've not been able to get it back to the original state, even when I ctrl-click to deselect "PCM" in the list. However, apps that use legacy OSS output still blast me with nearly full volume every time. I've also noticed another glitch: using the volume control applet to adjust my PCM volume, the audio sometimes skips. If I quickly drag the slider back and forth, it always does. I'm pretty sure I would have noticed this if it occurred with my previous kernel (haven't had a chance to test that yet). Anyway, looking into this problem, I found the following note in the changes document at http://www.alsa-project.org/main/index.php/Changes_v1.0.15rc2_v1.0.15rc3 Sound Core ... - Remove the built-in software volume control in favour of softvol. So, my (main) question is: is there a good reason that the built-in software volume was removed? I didn't find anything about it in the archives for this list or -devel. I'm really not in favor of this change, because It used to work flawlessly for me, whereas now I'm running into all these strange problems with my IEC958 output. Thanks, Kyle Cronan <kyle(a)pbx.org>

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[alsa-devel] Tascam US122 on Arm
by ST 04 Oct '07

04 Oct '07

Hi I am a little irritated that there has been no replies to my questions to this list (http://mailman.alsa-project.org/pipermail/alsa-devel/2007-September/003387.…, http://mailman.alsa-project.org/pipermail/alsa-devel/2007-September/003083.…). Am i on the wrong list? Please tell me if so. Probably its also that i didn't make clear in my last post that the tascam driver works on arm with my US122 sound card, for ca. 51 seconds. After that i get the following error: Sequence Error!(hcd_frame=6 ep=8in;wait=65538,frame=2). Most propably some urb of usb-frame 65538 is still missing. Cause could be too long delays in usb-hcd interrupt handling. I also tried insmod ./2.6.18/kernel/sound/usb/usx2y/snd-usb-usx2y.ko nrpacks=X with X=[1,2,4,8]. I thought that if it is really an CPU performance problem than probably a bigger buffer would eliminate the problem. However the error persisted. Even commenting out /* #define USX2Y_NRPACKS_VARIABLE y */ and setting #define USX2Y_NRPACKS 1 didn't help. So my question is now, is the 250Mhz ARM really to slow or is there any chance that there is an error in either the usb stack or the driver exposed only on the ARM architecture? Things speaking for the latter are: *Error allways in the 65536-65540 value. *Error happens most of the times after about 50secs. Since i do not know much about USB any hint is apreciated. ST

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[alsa-devel] about sound/core/pcm_lib.c:1997: capture read error (DMA or IRQ trouble?)
by Akio 04 Oct '07

04 Oct '07

Dear ALL: after long time amr-wb codec loopback test. (above 90 hours) I got the following kernel message. it comes from snd_pcm_lib_read1() function. ALSA sound/core/pcm_lib.c:1997: capture read error (DMA or IRQ trouble?) ALSA sound/core/pcm_lib.c:1997: capture read error (DMA or IRQ trouble?) ALSA sound/core/pcm_lib.c:1997: capture read error (DMA or IRQ trouble?) ALSA sound/core/pcm_lib.c:1997: capture read error (DMA or IRQ trouble?) ALSA sound/core/pcm_lib.c:1997: capture read error (DMA or IRQ trouble?) ALSA sound/core/pcm_lib.c:1997: capture read error (DMA or IRQ trouble?) What should I do? My platform is ARM926, with Kernel 2.6.19.2 ALSA version is 1.0.13. Best Regards, Akio

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[alsa-devel] different user causes 'no such device'
by william estrada 03 Oct '07

03 Oct '07

Hi guys, I have what seams like a interesting problem. I have written an ALSA app to record wav files and it runs while I am logged in an terminal as the logged in Gnome user or root. But when I su to another user, it can't find the default sound card. I am running FC6. I tried changing own of /dev/dsp to 'nobody', that did not work. Anyone know of this problem and a fix for it. Thanks for your time. -- William Estrada MrUmunhum(a)popdial.com Mt-Umunhum-Wireless.net ( http://Mt-Umunhum-Wireless.net ) Ymessenger: MrUmunhum > MSGrec - Version 1.1, Mt-Umunhum-Wireless.net > WavMSG: Paused: enter 'q' to stop, <ENTER> to start. > ALSA lib confmisc.c:848:(snd_func_card_driver) cannot find card '0' > ALSA lib conf.c:3500:(_snd_config_evaluate) function snd_func_card_driver returned error: No such device > ALSA lib confmisc.c:397:(snd_func_concat) error evaluating strings > ALSA lib conf.c:3500:(_snd_config_evaluate) function snd_func_concat returned error: No such device > ALSA lib confmisc.c:1248:(snd_func_refer) error evaluating name > ALSA lib conf.c:3500:(_snd_config_evaluate) function snd_func_refer returned error: No such device > ALSA lib conf.c:3972:(snd_config_expand) Evaluate error: No such device > ALSA lib pcm.c:2145:(snd_pcm_open_noupdate) Unknown PCM default

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