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Re: [alsa-devel] Nvidia hdmi problems
by Takashi Iwai 21 Oct '08

21 Oct '08

At Tue, 21 Oct 2008 09:52:46 +0200, Nir Tzachar wrote: > > On Tue, Oct 21, 2008 at 9:50 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Tue, 21 Oct 2008 09:39:49 +0200, > > Nir Tzachar wrote: > >> > >> On Tue, Oct 21, 2008 at 9:34 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > >> > At Tue, 21 Oct 2008 09:30:21 +0200, > >> > Nir Tzachar wrote: > >> >> > >> >> Hello. > >> >> > >> >> On Tue, Oct 21, 2008 at 9:18 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > >> >> > At Mon, 20 Oct 2008 16:35:01 +0200, > >> >> > Nir Tzachar wrote: > >> >> >> > >> >> >> Hello, > >> >> >> > >> >> >> I have been trying to get audio through hdmi to work for some time > >> >> >> now. I have cloned the sound-2.6 repository to get the latest drivers, > >> >> >> which should support hdmi, AFAIK. > >> >> > > >> >> > Try more later version, from my sound git tree, or even Linus tree. > >> >> > git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git > >> >> > >> >> I tried yesterday's version. Apart from not having hdmi output, it had > >> >> problems with volume control. > >> > > >> > Please elaborate? > >> > >> I performed a git-clone on the repository. The head was > >> b47d892509f2d908974781443d29b11dcd29d809. > > > > Better to pull onto the latest Linus tree. > > That is, clone linux-2.6.git, then pull from sound-2.6.git. > > ok, will do this. > > >> When I played a movie in mplayer and touch the volume controls, > >> mplayer got stuck. I can perform other tests. I'll also try to update > >> the tree. > > > > Which device are you using? HDMI or other? > > Also, please check whether this happens with Linus tree and 2.6.27.x, > > too. > > Using the analog device, and it works under 2.6.27.1 flawlessly. OK, then it must be a regression. If it works with Linus tree, the problem is the post 2.6.28 stuff. If it doesn't work with Linus tree, too, it's likely changes to patch_realtek.c... thanks, Takashi

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[alsa-devel] [PATCH] pci: use pci_ioremap_bar() in sound/
by Arjan van de Ven 21 Oct '08

21 Oct '08

From 6666b6edd4a0c496932ef1640b742cc72f9c57d8 Mon Sep 17 00:00:00 2001 From: Arjan van de Ven <arjan(a)linux.intel.com> Date: Sun, 28 Sep 2008 16:20:09 -0700 Subject: [PATCH] pci: use pci_ioremap_bar() in sound/ Use the newly introduced pci_ioremap_bar() function in sound/. pci_ioremap_bar() just takes a pci device and a bar number, with the goal of making it really hard to get wrong, while also having a central place to stick sanity checks. CC: tiwai(a)suse.de Signed-off-by: Arjan van de Ven <arjan(a)linux.intel.com> --- sound/pci/ad1889.c | 2 +- sound/pci/atiixp.c | 2 +- sound/pci/atiixp_modem.c | 2 +- sound/pci/au88x0/au88x0.c | 3 +-- sound/pci/bt87x.c | 3 +-- sound/pci/cs4281.c | 4 ++-- sound/pci/cs5530.c | 2 +- sound/pci/hda/hda_intel.c | 2 +- sound/pci/mixart/mixart.c | 3 +-- 9 files changed, 10 insertions(+), 13 deletions(-) diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c index 92f3a97..a7f38e6 100644 --- a/sound/pci/ad1889.c +++ b/sound/pci/ad1889.c @@ -932,7 +932,7 @@ snd_ad1889_create(struct snd_card *card, goto free_and_ret; chip->bar = pci_resource_start(pci, 0); - chip->iobase = ioremap_nocache(chip->bar, pci_resource_len(pci, 0)); + chip->iobase = pci_ioremap_bar(pci, 0); if (chip->iobase == NULL) { printk(KERN_ERR PFX "unable to reserve region.\n"); err = -EBUSY; diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index 085a52b..226fe82 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -1609,7 +1609,7 @@ static int __devinit snd_atiixp_create(struct snd_card *card, return err; } chip->addr = pci_resource_start(pci, 0); - chip->remap_addr = ioremap_nocache(chip->addr, pci_resource_len(pci, 0)); + chip->remap_addr = pci_ioremap_bar(pci, 0); if (chip->remap_addr == NULL) { snd_printk(KERN_ERR "AC'97 space ioremap problem\n"); snd_atiixp_free(chip); diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c index 2f10630..0e6e5cc 100644 --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -1252,7 +1252,7 @@ static int __devinit snd_atiixp_create(struct snd_card *card, return err; } chip->addr = pci_resource_start(pci, 0); - chip->remap_addr = ioremap_nocache(chip->addr, pci_resource_len(pci, 0)); + chip->remap_addr = pci_ioremap_bar(pci, 0); if (chip->remap_addr == NULL) { snd_printk(KERN_ERR "AC'97 space ioremap problem\n"); snd_atiixp_free(chip); diff --git a/sound/pci/au88x0/au88x0.c b/sound/pci/au88x0/au88x0.c index 68368e4..a36d4d1 100644 --- a/sound/pci/au88x0/au88x0.c +++ b/sound/pci/au88x0/au88x0.c @@ -180,8 +180,7 @@ snd_vortex_create(struct snd_card *card, struct pci_dev *pci, vortex_t ** rchip) if ((err = pci_request_regions(pci, CARD_NAME_SHORT)) != 0) goto regions_out; - chip->mmio = ioremap_nocache(pci_resource_start(pci, 0), - pci_resource_len(pci, 0)); + chip->mmio = pci_ioremap_bar(pci, 0); if (!chip->mmio) { printk(KERN_ERR "MMIO area remap failed.\n"); err = -ENOMEM; diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c index 3aa8d97..1aa1c04 100644 --- a/sound/pci/bt87x.c +++ b/sound/pci/bt87x.c @@ -749,8 +749,7 @@ static int __devinit snd_bt87x_create(struct snd_card *card, pci_disable_device(pci); return err; } - chip->mmio = ioremap_nocache(pci_resource_start(pci, 0), - pci_resource_len(pci, 0)); + chip->mmio = pci_ioremap_bar(pci, 0); if (!chip->mmio) { snd_printk(KERN_ERR "cannot remap io memory\n"); err = -ENOMEM; diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c index ef9308f..192e784 100644 --- a/sound/pci/cs4281.c +++ b/sound/pci/cs4281.c @@ -1382,8 +1382,8 @@ static int __devinit snd_cs4281_create(struct snd_card *card, chip->ba0_addr = pci_resource_start(pci, 0); chip->ba1_addr = pci_resource_start(pci, 1); - chip->ba0 = ioremap_nocache(chip->ba0_addr, pci_resource_len(pci, 0)); - chip->ba1 = ioremap_nocache(chip->ba1_addr, pci_resource_len(pci, 1)); + chip->ba0 = pci_ioremap_bar(pci, 0); + chip->ba1 = pci_ioremap_bar(pci, 1); if (!chip->ba0 || !chip->ba1) { snd_cs4281_free(chip); return -ENOMEM; diff --git a/sound/pci/cs5530.c b/sound/pci/cs5530.c index 7ff8b68..4d9378d 100644 --- a/sound/pci/cs5530.c +++ b/sound/pci/cs5530.c @@ -132,7 +132,7 @@ static int __devinit snd_cs5530_create(struct snd_card *card, } chip->pci_base = pci_resource_start(pci, 0); - mem = ioremap_nocache(chip->pci_base, pci_resource_len(pci, 0)); + mem = pci_ioremap_bar(pci, 0); if (mem == NULL) { kfree(chip); pci_disable_device(pci); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 9f316c1..f080f8c 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2158,7 +2158,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, } chip->addr = pci_resource_start(pci, 0); - chip->remap_addr = ioremap_nocache(chip->addr, pci_resource_len(pci,0)); + chip->remap_addr = pci_ioremap_bar(pci, 0); if (chip->remap_addr == NULL) { snd_printk(KERN_ERR SFX "ioremap error\n"); err = -ENXIO; diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index 2d0dce6..ae7601f 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -1314,8 +1314,7 @@ static int __devinit snd_mixart_probe(struct pci_dev *pci, } for (i = 0; i < 2; i++) { mgr->mem[i].phys = pci_resource_start(pci, i); - mgr->mem[i].virt = ioremap_nocache(mgr->mem[i].phys, - pci_resource_len(pci, i)); + mgr->mem[i].virt = pci_ioremap_bar(pci, i); if (!mgr->mem[i].virt) { printk(KERN_ERR "unable to remap resource 0x%lx\n", mgr->mem[i].phys); -- 1.5.5.1 -- Arjan van de Ven Intel Open Source Technology Centre For development, discussion and tips for power savings, visit http://www.lesswatts.org

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Re: [alsa-devel] Nvidia hdmi problems
by Takashi Iwai 21 Oct '08

21 Oct '08

At Tue, 21 Oct 2008 09:39:49 +0200, Nir Tzachar wrote: > > On Tue, Oct 21, 2008 at 9:34 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Tue, 21 Oct 2008 09:30:21 +0200, > > Nir Tzachar wrote: > >> > >> Hello. > >> > >> On Tue, Oct 21, 2008 at 9:18 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > >> > At Mon, 20 Oct 2008 16:35:01 +0200, > >> > Nir Tzachar wrote: > >> >> > >> >> Hello, > >> >> > >> >> I have been trying to get audio through hdmi to work for some time > >> >> now. I have cloned the sound-2.6 repository to get the latest drivers, > >> >> which should support hdmi, AFAIK. > >> > > >> > Try more later version, from my sound git tree, or even Linus tree. > >> > git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git > >> > >> I tried yesterday's version. Apart from not having hdmi output, it had > >> problems with volume control. > > > > Please elaborate? > > I performed a git-clone on the repository. The head was > b47d892509f2d908974781443d29b11dcd29d809. Better to pull onto the latest Linus tree. That is, clone linux-2.6.git, then pull from sound-2.6.git. > When I played a movie in mplayer and touch the volume controls, > mplayer got stuck. I can perform other tests. I'll also try to update > the tree. Which device are you using? HDMI or other? Also, please check whether this happens with Linus tree and 2.6.27.x, too. thanks, Takashi

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Re: [alsa-devel] Nvidia hdmi problems
by Takashi Iwai 21 Oct '08

21 Oct '08

At Tue, 21 Oct 2008 09:30:21 +0200, Nir Tzachar wrote: > > Hello. > > On Tue, Oct 21, 2008 at 9:18 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Mon, 20 Oct 2008 16:35:01 +0200, > > Nir Tzachar wrote: > >> > >> Hello, > >> > >> I have been trying to get audio through hdmi to work for some time > >> now. I have cloned the sound-2.6 repository to get the latest drivers, > >> which should support hdmi, AFAIK. > > > > Try more later version, from my sound git tree, or even Linus tree. > > git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git > > I tried yesterday's version. Apart from not having hdmi output, it had > problems with volume control. Please elaborate? Takashi

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[alsa-devel] Nvidia hdmi problems
by Nir Tzachar 21 Oct '08

21 Oct '08

Hello, I have been trying to get audio through hdmi to work for some time now. I have cloned the sound-2.6 repository to get the latest drivers, which should support hdmi, AFAIK. Motherboard is Gigabyte GA-73PVM-S2H, with nforce 630i and realtek alc889a. aplay lists the hdmi device: > aplay -L default:CARD=NVidia HDA NVidia, ALC885 Analog Default Audio Device front:CARD=NVidia,DEV=0 HDA NVidia, ALC885 Analog Front speakers surround40:CARD=NVidia,DEV=0 HDA NVidia, ALC885 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=NVidia,DEV=0 HDA NVidia, ALC885 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=NVidia,DEV=0 HDA NVidia, ALC885 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=NVidia,DEV=0 HDA NVidia, ALC885 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=NVidia,DEV=0 HDA NVidia, ALC885 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=NVidia,DEV=0 HDA NVidia, ALC885 Digital IEC958 (S/PDIF) Digital Audio Output hdmi:CARD=NVidia,DEV=0 HDA NVidia HDMI Audio Output null Discard all samples (playback) or generate zero samples (capture) but, > aplay -D hdmi test.wav aplay: main:583: audio open error: No such file or directory fails when trying to open /dev/snd/pcmC0D3p only the following devices exists: /dev/snd/seq /dev/snd/controlC0 /dev/snd/pcmC0D0c /dev/snd/pcmC0D0p /dev/snd/pcmC0D1c /dev/snd/pcmC0D1p /dev/snd/pcmC0D2c /dev/snd/timer I am using alsa-lib and alsa-utils 1.0.17 . Does anyone have an idea? Thanks.

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[alsa-devel] snd_pcsp locking mess
by Stas Sergeev 21 Oct '08

21 Oct '08

bugme-daemon(a)bugzilla.kernel.org wrote: > ------- Comment #17 from tiwai(a)suse.de 2008-05-18 10:22 ------- > Anyway, what I meant is that the part touching I/O ports could be a faster path > than softirq. And the hrtimer code can use a bit simpler locks. The hrtimer > callback needs just one thing about DMA buffer - the buffer is allocated and > the address doesn't change during the callback. So, we'd need the lock for the > hr callback and for the PCM prepare, hwparams and free callbacks, i.e. to > assure that the hrtimer callback is finished when these PCM callbacks may > change the DMA buffer. Thus this lock doesn't have to be PCM substream lock. I wonder if it would be better to introduce such a lock in an alsa core rather than in a driver. The same way as snd_pcm_stream_lock(). > The current lock mess may come from the PCM trigger (STOP) called from > snd_pcm_period_elapsed(). So, putting snd_pcm_period_elapsed() out of hrtimer > lock is anyway necessary. A tasklet for snd_pcm_period_elapsed() is needed for > this reason, too. Wouldn't it be better to just make snd_pcm_period_elapsed() to not take the stream lock? There is a requirement right now to drop the stream lock before calling this. I still beleive this is the root of all the headache. If you need to drop it, then you either accept the race or need a second lock, and the troubles ensue.

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[alsa-devel] [PATCH1/2] hda: add access to jack functions
by Matthew Ranostay 21 Oct '08

21 Oct '08

Removed 'static' declaration from get_jack_location() to be accessed via the codec patches. Also added jack location and type reporting functions for use in the jack abstraction layer implementations. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 4a08c31..10f26e0 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -383,6 +383,9 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, /* * get widget capabilities */ +const char *get_jack_connectivity(u32 cfg); +const char *get_jack_type(u32 cfg); +const char *get_jack_location(u32 cfg); static inline u32 get_wcaps(struct hda_codec *codec, hda_nid_t nid) { if (nid < codec->start_nid || diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 64ab19f..b77c1b1 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -145,7 +145,7 @@ static void print_pcm_caps(struct snd_info_buffer *buffer, print_pcm_formats(buffer, stream); } -static const char *get_jack_location(u32 cfg) +const char *get_jack_location(u32 cfg) { static char *bases[7] = { "N/A", "Rear", "Front", "Left", "Right", "Top", "Bottom", @@ -171,6 +171,26 @@ static const char *get_jack_location(u32 cfg) return "UNKNOWN"; } +const char *get_jack_connectivity(u32 cfg) +{ + static char *jack_locations[4] = { "Ext", "Int", "Sep", "Oth" }; + + return jack_locations[(cfg >> (AC_DEFCFG_LOCATION_SHIFT + 4)) & 3]; +} + +const char *get_jack_type(u32 cfg) +{ + static char *jack_types[16] = { + "Line Out", "Speaker", "HP Out", "CD", + "SPDIF Out", "Digital Out", "Modem Line", "Modem Hand", + "Line In", "Aux", "Mic", "Telephony", + "SPDIF In", "Digitial In", "Reserved", "Other" + }; + + return jack_types[(cfg & AC_DEFCFG_DEVICE) + >> AC_DEFCFG_DEVICE_SHIFT]; +} + static const char *get_jack_connection(u32 cfg) { static char *names[16] = { @@ -206,13 +226,6 @@ static void print_pin_caps(struct snd_info_buffer *buffer, int *supports_vref) { static char *jack_conns[4] = { "Jack", "N/A", "Fixed", "Both" }; - static char *jack_types[16] = { - "Line Out", "Speaker", "HP Out", "CD", - "SPDIF Out", "Digital Out", "Modem Line", "Modem Hand", - "Line In", "Aux", "Mic", "Telephony", - "SPDIF In", "Digitial In", "Reserved", "Other" - }; - static char *jack_locations[4] = { "Ext", "Int", "Sep", "Oth" }; unsigned int caps, val; caps = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); @@ -274,8 +287,8 @@ static void print_pin_caps(struct snd_info_buffer *buffer, caps = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); snd_iprintf(buffer, " Pin Default 0x%08x: [%s] %s at %s %s\n", caps, jack_conns[(caps & AC_DEFCFG_PORT_CONN) >> AC_DEFCFG_PORT_CONN_SHIFT], - jack_types[(caps & AC_DEFCFG_DEVICE) >> AC_DEFCFG_DEVICE_SHIFT], - jack_locations[(caps >> (AC_DEFCFG_LOCATION_SHIFT + 4)) & 3], + get_jack_type(caps), + get_jack_connectivity(caps), get_jack_location(caps)); snd_iprintf(buffer, " Conn = %s, Color = %s\n", get_jack_connection(caps),

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Re: [alsa-devel] Audio on Linux is a mess
by Marek Vasut 20 Oct '08

20 Oct '08

You got what you paid for. Maybe if you change your attitude and supply some information, someone might consider looking in your issue, othervise it would be a problem since I forgot my crystal ball back at home. On Monday 20 of October 2008 16:41:45 Fred . wrote: > Yeah, fix the Linux audio so that if an application behaves wrong, or > does something stupid, then after I close the application, the system > should be silent. > It shouldn't continue make sound even after I quit the application. > And PulseAudio shouldn't crash because one application did something > even if it was stupid. > > On Mon, Oct 20, 2008 at 8:35 AM, Marek Vasut <marek.vasut(a)gmail.com> wrote: > > On Friday 17 of October 2008 15:24:20 Fred . wrote: > >> ALSA lib pulse.c:272:(pulse_connect) PulseAudio: Unable to connect: > >> Timeout ALSA lib pulse.c:272:(pulse_connect) PulseAudio: Unable to > >> connect: Timeout AL lib: alsa.c:344: Could not open playback device > >> 'default': Connection refused AL lib: oss.c:179: Could not open > >> /dev/dsp: Device or resource busy File > >> "/var/lib/python-support/python2.5/pyglet/media/drivers/openal/__init__. > >>py" , line 340, in driver_init_device = alc.alcOpenDevice(device_name) > >> > >> Audio on Linux is a mess. > >> I launch an Python script that uses audio then it breaks, and when the > >> Python script is closed, I still hear the audio skipping and repeating > >> like, > >> "kakakakakakaka" in infinite loop. > >> And PulseAudio seems to be frozen. > >> Then I kill PulseAudio and finally the audio noise stops. > >> > >> Oh, and I heard OpenAL downsamples stereo to mono on Linux due to a bug. > >> > >> Whole audio on Linux is just a big mess. > > > > Please behave yourself. > > Is there any way we can help you (which I doubt, but ...)? > > > >> _______________________________________________ > >> 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] Audio on Linux is a mess
by Fred . 20 Oct '08

20 Oct '08

ALSA lib pulse.c:272:(pulse_connect) PulseAudio: Unable to connect: Timeout ALSA lib pulse.c:272:(pulse_connect) PulseAudio: Unable to connect: Timeout AL lib: alsa.c:344: Could not open playback device 'default': Connection refused AL lib: oss.c:179: Could not open /dev/dsp: Device or resource busy File "/var/lib/python-support/python2.5/pyglet/media/drivers/openal/__init__.py", line 340, in driver_init_device = alc.alcOpenDevice(device_name) Audio on Linux is a mess. I launch an Python script that uses audio then it breaks, and when the Python script is closed, I still hear the audio skipping and repeating like, "kakakakakakaka" in infinite loop. And PulseAudio seems to be frozen. Then I kill PulseAudio and finally the audio noise stops. Oh, and I heard OpenAL downsamples stereo to mono on Linux due to a bug. Whole audio on Linux is just a big mess.

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[alsa-devel] No pcm device?
by John Kacur 20 Oct '08

20 Oct '08

I'm not sure if the hardware I have makes this problem simple for me, or difficult, because it seems to differ from other ASoC code. Basically, I have a i2s as transport, connected to max5556 driving the output. (and a few other chips related to amplification and muting) There is however no sound card, thus no PCM device. Now the question is how to proceed? In other words what kind of minimal PCM layer do I need if any? Thanks

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