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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 But there are out of date wikis that say to do this and some distros might turn this option on by default.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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