Alsa-devel December 2011

alsa-devel@alsa-project.org

88 participants
272 discussions

[alsa-devel] HDSPM - Problems with SyncIn
by Andre Schramm 19 Oct '12

19 Oct '12

Hello, I did some tests with the SyncIn-Feature of the HDSPe cards. [root@localhost ~]# lspci |grep Mult 01:00.0 Multimedia audio controller: Xilinx Corporation RME Hammerfall DSP MADI (rev d2) 02:00.0 Multimedia audio controller: Xilinx Corporation RME Hammerfall DSP MADI (rev d2) The SyncOut of card 2:00 was plugged to the SyncIn of 1:00 and 1:00 was put to AutoSync-mode automatically upon boot. Problem is that the amixer output seems to be inconsistent and differs from the proc output as well, so I'm not sure which one to trust. [root@localhost ~]# amixer -c0 contents numid=5,iface=MIXER,name='AutoSync Reference' ; type=ENUMERATED,access=r-------,values=1,items=5 ; Item #0 'Word Clock' ; Item #1 'MADI' ; Item #2 'TCO' ; Item #3 'Sync In' ; Item #4 'None' : values=3 numid=2,iface=MIXER,name='Internal Clock' ; type=ENUMERATED,access=rw------,values=1,items=9 ; Item #0 '32 kHz' ; Item #1 '44.1 kHz' ; Item #2 '48 kHz' ; Item #3 '64 kHz' ; Item #4 '88.2 kHz' ; Item #5 '96 kHz' ; Item #6 '128 kHz' ; Item #7 '176.4 kHz' ; Item #8 '192 kHz' : values=2 numid=4,iface=MIXER,name='Preferred Sync Reference' ; type=ENUMERATED,access=rw------,values=1,items=3 ; Item #0 'Word Clock' ; Item #1 'MADI' ; Item #2 'Sync In' : values=2 numid=10,iface=MIXER,name='SYNC IN SyncCheck' ; type=ENUMERATED,access=r-------,values=1,items=4 ; Item #0 'No Lock' ; Item #1 'Lock' ; Item #2 'Sync' ; Item #3 'N/A' : values=0 numid=3,iface=MIXER,name='System Clock Mode' ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'Master' ; Item #1 'AutoSync' : values=1 numid=6,iface=MIXER,name='System Sample Rate' ; type=INTEGER,access=r-------,values=1,min=27000,max=207000,step=1 : values=47999 'Preferred Sync Reference' is SyncIn, 'SYNC IN SyncCheck' says 'No Lock', meaning no signal at all, but the 'System Sample Rate' coming from the master card is detected with 1Hz offset. When putting the master to 32000Hz, 31999Hz are shown and so on. [root@localhost ~]# cat /proc/asound/card0/hdspm RME MADI (Card #1) Rev.d2 Status2first3bits: 1 HW Serial: 0x0000019196d4 IRQ: 16 Registers bus: 0xfe9f0000 VM: 0xf8920000 --- System --- IRQ Pending: Audio=0, MIDI0=0, MIDI1=0, IRQcount=0 HW pointer: id = 0, rawptr = 0 (0->16320) estimated= 0 (bytes) MIDI FIFO: Out1=0x0, Out2=0x0, In1=0x0, In2=0x0 MIDIoverMADI FIFO: In=0x0, Out=0x0 Register: ctrl1=0x10344ca, ctrl2=0x0, status1=0xc30004, status2=0xc0020361 No TCO module detected. --- Settings --- Size (Latency): 2048 samples (2 periods of 8192 bytes) Line out: on ClearTrackMarker = off, Transmit in 64 Channel Mode, Auto Input off AutoSync Reference: AutoSync Preferred Sync Reference: TCO System Clock Frequency: 48000 --- Status: Inputs MADI=NoLock, WordClock=NoLock AutoSync: Reference= Sync In, Freq=0 (MADI = 3, Word = 3) Input: Coax, Mode=56 channels Now the proc-output reports 'No TCO module detected', which is correct, as no expansion modules where plugged to the cards. But it also says 'Preferred Sync Reference: TCO' and 'Freq=0', where amixer said SyncIn and 47999? Output of the master card is below. Regards, Andre [root@localhost ~]# amixer -c1 contents numid=5,iface=MIXER,name='AutoSync Reference' ; type=ENUMERATED,access=r-------,values=1,items=5 ; Item #0 'Word Clock' ; Item #1 'MADI' ; Item #2 'TCO' ; Item #3 'Sync In' ; Item #4 'None' : values=4 numid=15,iface=MIXER,name='Input Select' ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'optical' ; Item #1 'coaxial' : values=0 numid=2,iface=MIXER,name='Internal Clock' ; type=ENUMERATED,access=rw------,values=1,items=9 ; Item #0 '32 kHz' ; Item #1 '44.1 kHz' ; Item #2 '48 kHz' ; Item #3 '64 kHz' ; Item #4 '88.2 kHz' ; Item #5 '96 kHz' ; Item #6 '128 kHz' ; Item #7 '176.4 kHz' ; Item #8 '192 kHz' : values=2 numid=4,iface=MIXER,name='Preferred Sync Reference' ; type=ENUMERATED,access=rw------,values=1,items=3 ; Item #0 'Word Clock' ; Item #1 'MADI' ; Item #2 'Sync In' : values=0 numid=10,iface=MIXER,name='SYNC IN SyncCheck' ; type=ENUMERATED,access=r-------,values=1,items=4 ; Item #0 'No Lock' ; Item #1 'Lock' ; Item #2 'Sync' ; Item #3 'N/A' : values=0 numid=3,iface=MIXER,name='System Clock Mode' ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'Master' ; Item #1 'AutoSync' : values=0 numid=6,iface=MIXER,name='System Sample Rate' ; type=INTEGER,access=r-------,values=1,min=27000,max=207000,step=1 : values=48000 [root@localhost ~]# cat /proc/asound/card1/hdspm RME MADI (Card #2) Rev.d2 Status2first3bits: 1 HW Serial: 0x000001b1c606 IRQ: 17 Registers bus: 0xfeaf0000 VM: 0xf9280000 --- System --- IRQ Pending: Audio=0, MIDI0=0, MIDI1=0, IRQcount=0 HW pointer: id = 0, rawptr = 0 (0->16320) estimated= 0 (bytes) MIDI FIFO: Out1=0x0, Out2=0x0, In1=0x0, In2=0x0 MIDIoverMADI FIFO: In=0x0, Out=0x0 Register: ctrl1=0x10004da, ctrl2=0x0, status1=0x2800000, status2=0xc0020701 No TCO module detected. --- Settings --- Size (Latency): 2048 samples (2 periods of 8192 bytes) Line out: on ClearTrackMarker = off, Transmit in 64 Channel Mode, Auto Input off AutoSync Reference: Master Preferred Sync Reference: Word Clock System Clock Frequency: 48000 --- Status: Inputs MADI=NoLock, WordClock=NoLock AutoSync: Reference= Input not valid, Freq=0 (MADI = 10, Word = 0) Input: Optical, Mode=56 channels

2 2

Re: [alsa-devel] [PATCH 2.6.39-rc3] hda_intel: ALSA HD Audio patch for Intel Panther Point DeviceIDs
by Takashi Iwai 09 Oct '12

09 Oct '12

At Wed, 20 Apr 2011 10:59:57 -0700, Seth Heasley wrote: > > This patch adds the HD Audio Controller DeviceIDs for the Intel Panther Point PCH. > > Signed-off-by: Seth Heasley <seth.heasley(a)intel.com> Applied now. Thanks. Takashi > --- > --- linux-2.6.39-rc3/sound/pci/hda/hda_intel.c.orig 2011-04-11 17:21:51.000000000 -0700 > +++ linux-2.6.39-rc3/sound/pci/hda/hda_intel.c 2011-04-14 11:44:40.000000000 -0700 > @@ -126,6 +126,7 @@ > "{Intel, ICH10}," > "{Intel, PCH}," > "{Intel, CPT}," > + "{Intel, PPT}," > "{Intel, PBG}," > "{Intel, SCH}," > "{ATI, SB450}," > @@ -2759,6 +2760,8 @@ > { PCI_DEVICE(0x8086, 0x1c20), .driver_data = AZX_DRIVER_PCH }, > /* PBG */ > { PCI_DEVICE(0x8086, 0x1d20), .driver_data = AZX_DRIVER_PCH }, > + /* Panther Point */ > + { PCI_DEVICE(0x8086, 0x1e20), .driver_data = AZX_DRIVER_PCH }, > /* SCH */ > { PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH }, > /* Generic Intel */ >

3 2

[alsa-devel] [PATCH] snd-usb-us122l: Fix missing NULL checks
by Karsten Wiese 27 Apr '12

27 Apr '12

Fix missing NULL checks in usb_stream_hwdep_poll() and usb_stream_hwdep_ioctl(). Wake up poll waiters before returning from usb_stream_hwdep_ioctl(). Signed-off-by: Karsten Wiese <fzu(a)wemgehoertderstaat.de> Cc: stable(a)kernel.org --- sound/usb/usx2y/us122l.c | 41 ++++++++++++++++++++--------------------- 1 files changed, 20 insertions(+), 21 deletions(-) diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c index 91bb296..7abda8d 100644 --- a/sound/usb/usx2y/us122l.c +++ b/sound/usb/usx2y/us122l.c @@ -269,29 +269,26 @@ static unsigned int usb_stream_hwdep_poll(struct snd_hwdep *hw, struct file *file, poll_table *wait) { struct us122l *us122l = hw->private_data; - struct usb_stream *s = us122l->sk.s; unsigned *polled; unsigned int mask; poll_wait(file, &us122l->sk.sleep, wait); - switch (s->state) { - case usb_stream_ready: - if (us122l->first == file) - polled = &s->periods_polled; - else - polled = &us122l->second_periods_polled; - if (*polled != s->periods_done) { - *polled = s->periods_done; - mask = POLLIN | POLLOUT | POLLWRNORM; - break; + mask = POLLIN | POLLOUT | POLLWRNORM | POLLERR; + if (mutex_trylock(&us122l->mutex)) { + struct usb_stream *s = us122l->sk.s; + if (s && s->state == usb_stream_ready) { + if (us122l->first == file) + polled = &s->periods_polled; + else + polled = &us122l->second_periods_polled; + if (*polled != s->periods_done) { + *polled = s->periods_done; + mask = POLLIN | POLLOUT | POLLWRNORM; + } else + mask = 0; } - /* Fall through */ - mask = 0; - break; - default: - mask = POLLIN | POLLOUT | POLLWRNORM | POLLERR; - break; + mutex_unlock(&us122l->mutex); } return mask; } @@ -377,6 +374,7 @@ static int usb_stream_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, { struct usb_stream_config *cfg; struct us122l *us122l = hw->private_data; + struct usb_stream *s; unsigned min_period_frames; int err = 0; bool high_speed; @@ -422,18 +420,18 @@ static int usb_stream_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, snd_power_wait(hw->card, SNDRV_CTL_POWER_D0); mutex_lock(&us122l->mutex); + s = us122l->sk.s; if (!us122l->master) us122l->master = file; else if (us122l->master != file) { - if (memcmp(cfg, &us122l->sk.s->cfg, sizeof(*cfg))) { + if (!s || memcmp(cfg, &s->cfg, sizeof(*cfg))) { err = -EIO; goto unlock; } us122l->slave = file; } - if (!us122l->sk.s || - memcmp(cfg, &us122l->sk.s->cfg, sizeof(*cfg)) || - us122l->sk.s->state == usb_stream_xrun) { + if (!s || memcmp(cfg, &s->cfg, sizeof(*cfg)) || + s->state == usb_stream_xrun) { us122l_stop(us122l); if (!us122l_start(us122l, cfg->sample_rate, cfg->period_frames)) err = -EIO; @@ -444,6 +442,7 @@ unlock: mutex_unlock(&us122l->mutex); free: kfree(cfg); + wake_up_all(&us122l->sk.sleep); return err; } -- 1.7.2.3

2 5

Re: [alsa-devel] Port C Out selector , Port E Out selector of ad1988 and ad1988_3stack_dac_nids
by Raymond Yau 10 Mar '12

10 Mar '12

2011/9/16 Takashi Iwai <tiwai(a)suse.de>: > At Fri, 16 Sep 2011 11:59:54 +0800, > Raymond Yau wrote: >> >> 2011/9/12 Takashi Iwai <tiwai(a)suse.de>: >> > At Wed, 31 Aug 2011 11:21:00 +0800, >> > Raymond Yau wrote: >> >> >> >> HDA - Add Headphone Playback Volume control for ad1988/ad1989 >> >> >> >> - use DAC0 instead of DAC1 for Port-A Headphone >> >> - assign 0x03 to spec->multiout.hp_nid except model="6stack-dig-fp" >> > >> > Thanks, applied now. >> > >> > >> > Takashi >> > >> >> while testing the headphone volume control , >> >> I discover that the "Port C Out selector" and "Port E Out Selector" >> (i.e. node 0x31 and 0x32 ) by commit in patch_anlog.c >> >> fd66e0d0591dd12eb0bea1e9f3aa194bb93cebbd >> >> >> >> * AD1988 >> * >> * Output pins and routes >> * >> * Pin Mix Sel DAC >> * port-A 0x11 (mute/hp) <- 0x22 <- 0x37 <- 03/04/06 >> * port-B 0x14 (mute/hp) <- 0x2b <- 0x30 <- 03/04/06 >> * port-C 0x15 (mute) <- 0x2c <- 0x31 <- 05/0a >> * port-D 0x12 (mute/hp) <- 0x29 <- 04 >> * port-E 0x17 (mute/hp) <- 0x26 <- 0x32 <- 05/0a >> >> >> >> The connection list of node 0x31 and 0x32 are different from datasheet >> >> >> Table 10. Audio Widget >> >> 31 Audio selector Selects DAC_2 and DAC_4 for PORT-C >> PORT-E Out Selector >> 32 Audio selector Selects DAC_2 and DAC_4 for PORT-E >> PORT-C In Selector >> >> >> >> Codec: Analog Devices AD1988 >> Address: 0 >> AFG Function Id: 0x1 (unsol 0) >> Vendor Id: 0x11d41988 >> Subsystem Id: 0x104381e1 >> Revision Id: 0x100400 >> >> >> Node 0x31 [Audio Selector] wcaps 0x300101: Stereo >> Connection: 2 >> 0x04* 0x0a >> Node 0x32 [Audio Selector] wcaps 0x300101: Stereo >> Connection: 2 >> 0x05* 0x04 >> >> >> >> is the following patch based on real hardware ? >> >> >> commit 8c7c7b8dd2828b42c1230c6b0235e7d1dcf57e5 >> >> hda-codec - Fix surrounds on 3stack mode of AD1988 > > I don't see this commit id in my tree... > > > Takashi > http://git.alsa-project.org/?p=alsa-kernel.git;a=commit;h=fd66e0d0591dd12eb… Thu, 17 Nov 2005 14:31:34 +0000 (15:31 +0100) Add AD1988 codec support to hda-codec driver. http://git.alsa-project.org/?p=alsa-kernel.git;a=commitdiff;h=f8c7c7b8dd282… Thu, 24 Nov 2005 15:17:20 +0000 (16:17 +0100)] Fixed the swapped surround/CLFE on 3stack mode of AD1988. hda-emu codecs/ad1988a -m "3stack" route -a 0x15 Out[0a] -x Sel[31] -- Mix[2c] -> Pin[15] Out[04] -> Sel[31] -- Mix[2c] -> Pin[15] route -a 0x17 Out[04] -> Sel[32] -- Mix[26] -> Pin[17] Out[05] -x Sel[32] -- Mix[26] -> Pin[17]

2 4

[alsa-devel] ALC892, Clevo P150HM: Input Source gets reset to "Mic" every resume
by Julian Sikorski 02 Mar '12

02 Mar '12

Hi, whenever I resume the machine, the Input Source would get changed back from "Internal Mic" to "Mic". This is a bit annoying, because pulseaudio is unaware of that change. As a result, everything looks fine in pavucontrol but no sound gets recorded. A quick look at alsamixer -c0 revealed the problem. My alsa info: http://www.alsa-project.org/db/?f=c5776b615a5f3b9ae624e4250fb0c2b69118ce26 This is unrelated to the other problems I have been reporting, since this happens on a mostly-stock Fedora 3.1.2 kernel (the patch I added deals with xhci, not alsa). Regards, Julian

3 8

[alsa-devel] [PATCH 1/2] ASoC: Convert kirkwood-t5325 to use snd_soc_register_card()
by Axel Lin 07 Feb '12

07 Feb '12

Use snd_soc_register_card() instead of creating a "soc-audio" platform device. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- I don't have this hardware, I'd appreciate if someone can test this patch. Axel arch/arm/mach-kirkwood/t5325-setup.c | 6 ++++ sound/soc/kirkwood/kirkwood-t5325.c | 47 +++++++++++++++++---------------- 2 files changed, 30 insertions(+), 23 deletions(-) diff --git a/arch/arm/mach-kirkwood/t5325-setup.c b/arch/arm/mach-kirkwood/t5325-setup.c index 966b2b3..f9d2a11 100644 --- a/arch/arm/mach-kirkwood/t5325-setup.c +++ b/arch/arm/mach-kirkwood/t5325-setup.c @@ -106,6 +106,11 @@ static struct platform_device hp_t5325_button_device = { } }; +static struct platform_device hp_t5325_audio_device = { + .name = "t5325-audio", + .id = -1, +}; + static unsigned int hp_t5325_mpp_config[] __initdata = { MPP0_NF_IO2, MPP1_SPI_MOSI, @@ -179,6 +184,7 @@ static void __init hp_t5325_init(void) kirkwood_sata_init(&hp_t5325_sata_data); kirkwood_ehci_init(); platform_device_register(&hp_t5325_button_device); + platform_device_register(&hp_t5325_audio_device); i2c_register_board_info(0, i2c_board_info, ARRAY_SIZE(i2c_board_info)); kirkwood_audio_init(); diff --git a/sound/soc/kirkwood/kirkwood-t5325.c b/sound/soc/kirkwood/kirkwood-t5325.c index 16b83e5..ce4f408 100644 --- a/sound/soc/kirkwood/kirkwood-t5325.c +++ b/sound/soc/kirkwood/kirkwood-t5325.c @@ -80,7 +80,6 @@ static struct snd_soc_dai_link t5325_dai[] = { }, }; - static struct snd_soc_card t5325 = { .name = "t5325", .owner = THIS_MODULE, @@ -93,38 +92,40 @@ static struct snd_soc_card t5325 = { .num_dapm_routes = ARRAY_SIZE(t5325_route), }; -static struct platform_device *t5325_snd_device; - -static int __init t5325_init(void) +static int __devinit t5325_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &t5325; int ret; - if (!machine_is_t5325()) - return 0; - - t5325_snd_device = platform_device_alloc("soc-audio", -1); - if (!t5325_snd_device) - return -ENOMEM; - - platform_set_drvdata(t5325_snd_device, - &t5325); - - ret = platform_device_add(t5325_snd_device); - if (ret) { - printk(KERN_ERR "%s: platform_device_add failed\n", __func__); - platform_device_put(t5325_snd_device); - } + card->dev = &pdev->dev; + ret = snd_soc_register_card(card); + if (ret) + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); return ret; } -module_init(t5325_init); -static void __exit t5325_exit(void) +static int __devexit t5325_remove(struct platform_device *pdev) { - platform_device_unregister(t5325_snd_device); + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + return 0; } -module_exit(t5325_exit); + +static struct platform_driver t5325_driver = { + .driver = { + .name = "t5325-audio", + .owner = THIS_MODULE, + }, + .probe = t5325_probe, + .remove = __devexit_p(t5325_remove), +}; + +module_platform_driver(t5325_driver); MODULE_AUTHOR("Arnaud Patard <arnaud.patard(a)rtp-net.org>"); MODULE_DESCRIPTION("ALSA SoC t5325 audio client"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:t5325-audio"); -- 1.7.5.4

2 2

[alsa-devel] [PATCH] ALSA: HDA: Remove unconnected PCM devices for Intel HDMI
by David Henningsson 01 Feb '12

01 Feb '12

Since I'm a little new in HDMI land and the patch is non-trivial, I wouldn't mind some comments on this one. Also thanks to Jaroslav who cleared a few things out about the logical and physical devices which helped me figure out what I should do about it :-) Anyway, here's the patch. Some newer chips have more than one HDMI output, but usually not all of them are exposed as physical jacks. Removing the unused PCM devices (as indicated by BIOS in the pin config default) will reduce user confusion as they currently have to choose between several HDMI devices, some of them not working anyway. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

4 14

[alsa-devel] [PATCH] ASoC: Convert blackfin directory to module_platform_driver
by Axel Lin 19 Jan '12

19 Jan '12

Factor out some boilerplate code. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/blackfin/bf5xx-ac97-pcm.c | 12 +----------- sound/soc/blackfin/bf5xx-ac97.c | 13 +------------ sound/soc/blackfin/bf5xx-i2s-pcm.c | 12 +----------- sound/soc/blackfin/bf5xx-i2s.c | 13 +------------ sound/soc/blackfin/bf5xx-tdm-pcm.c | 12 +----------- sound/soc/blackfin/bf5xx-tdm.c | 12 +----------- sound/soc/blackfin/bfin-eval-adau1373.c | 12 +----------- sound/soc/blackfin/bfin-eval-adau1701.c | 12 +----------- sound/soc/blackfin/bfin-eval-adav80x.c | 12 +----------- 9 files changed, 9 insertions(+), 101 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c index 56815c1..fcff583 100644 --- a/sound/soc/blackfin/bf5xx-ac97-pcm.c +++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c @@ -475,17 +475,7 @@ static struct platform_driver bf5xx_pcm_driver = { .remove = __devexit_p(bf5xx_soc_platform_remove), }; -static int __init snd_bf5xx_pcm_init(void) -{ - return platform_driver_register(&bf5xx_pcm_driver); -} -module_init(snd_bf5xx_pcm_init); - -static void __exit snd_bf5xx_pcm_exit(void) -{ - platform_driver_unregister(&bf5xx_pcm_driver); -} -module_exit(snd_bf5xx_pcm_exit); +module_platform_driver(bf5xx_pcm_driver); MODULE_AUTHOR("Cliff Cai"); MODULE_DESCRIPTION("ADI Blackfin AC97 PCM DMA module"); diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index 6d21625..f4e9dc4 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -375,18 +375,7 @@ static struct platform_driver asoc_bfin_ac97_driver = { .remove = __devexit_p(asoc_bfin_ac97_remove), }; -static int __init bfin_ac97_init(void) -{ - return platform_driver_register(&asoc_bfin_ac97_driver); -} -module_init(bfin_ac97_init); - -static void __exit bfin_ac97_exit(void) -{ - platform_driver_unregister(&asoc_bfin_ac97_driver); -} -module_exit(bfin_ac97_exit); - +module_platform_driver(asoc_bfin_ac97_driver); MODULE_AUTHOR("Roy Huang"); MODULE_DESCRIPTION("AC97 driver for ADI Blackfin"); diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c index 7565e15..6ec3d41 100644 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.c +++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c @@ -314,17 +314,7 @@ static struct platform_driver bfin_i2s_pcm_driver = { .remove = __devexit_p(bfin_i2s_soc_platform_remove), }; -static int __init snd_bfin_i2s_pcm_init(void) -{ - return platform_driver_register(&bfin_i2s_pcm_driver); -} -module_init(snd_bfin_i2s_pcm_init); - -static void __exit snd_bfin_i2s_pcm_exit(void) -{ - platform_driver_unregister(&bfin_i2s_pcm_driver); -} -module_exit(snd_bfin_i2s_pcm_exit); +module_platform_driver(bfin_i2s_pcm_driver); MODULE_AUTHOR("Cliff Cai"); MODULE_DESCRIPTION("ADI Blackfin I2S PCM DMA module"); diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c index 00cc3e0..a400633 100644 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ b/sound/soc/blackfin/bf5xx-i2s.c @@ -288,18 +288,7 @@ static struct platform_driver bfin_i2s_driver = { }, }; -static int __init bfin_i2s_init(void) -{ - return platform_driver_register(&bfin_i2s_driver); -} - -static void __exit bfin_i2s_exit(void) -{ - platform_driver_unregister(&bfin_i2s_driver); -} - -module_init(bfin_i2s_init); -module_exit(bfin_i2s_exit); +module_platform_driver(bfin_i2s_driver); /* Module information */ MODULE_AUTHOR("Cliff Cai"); diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.c b/sound/soc/blackfin/bf5xx-tdm-pcm.c index c95cc03..4406f9a 100644 --- a/sound/soc/blackfin/bf5xx-tdm-pcm.c +++ b/sound/soc/blackfin/bf5xx-tdm-pcm.c @@ -339,17 +339,7 @@ static struct platform_driver bfin_tdm_driver = { .remove = __devexit_p(bf5xx_soc_platform_remove), }; -static int __init snd_bfin_tdm_init(void) -{ - return platform_driver_register(&bfin_tdm_driver); -} -module_init(snd_bfin_tdm_init); - -static void __exit snd_bfin_tdm_exit(void) -{ - platform_driver_unregister(&bfin_tdm_driver); -} -module_exit(snd_bfin_tdm_exit); +module_platform_driver(bfin_tdm_driver); MODULE_AUTHOR("Barry Song"); MODULE_DESCRIPTION("ADI Blackfin TDM PCM DMA module"); diff --git a/sound/soc/blackfin/bf5xx-tdm.c b/sound/soc/blackfin/bf5xx-tdm.c index a822d1e..bf65e0e 100644 --- a/sound/soc/blackfin/bf5xx-tdm.c +++ b/sound/soc/blackfin/bf5xx-tdm.c @@ -314,17 +314,7 @@ static struct platform_driver bfin_tdm_driver = { }, }; -static int __init bfin_tdm_init(void) -{ - return platform_driver_register(&bfin_tdm_driver); -} -module_init(bfin_tdm_init); - -static void __exit bfin_tdm_exit(void) -{ - platform_driver_unregister(&bfin_tdm_driver); -} -module_exit(bfin_tdm_exit); +module_platform_driver(bfin_tdm_driver); /* Module information */ MODULE_AUTHOR("Barry Song"); diff --git a/sound/soc/blackfin/bfin-eval-adau1373.c b/sound/soc/blackfin/bfin-eval-adau1373.c index 8df2a3b..85ed39a 100644 --- a/sound/soc/blackfin/bfin-eval-adau1373.c +++ b/sound/soc/blackfin/bfin-eval-adau1373.c @@ -184,17 +184,7 @@ static struct platform_driver bfin_eval_adau1373_driver = { .remove = __devexit_p(bfin_eval_adau1373_remove), }; -static int __init bfin_eval_adau1373_init(void) -{ - return platform_driver_register(&bfin_eval_adau1373_driver); -} -module_init(bfin_eval_adau1373_init); - -static void __exit bfin_eval_adau1373_exit(void) -{ - platform_driver_unregister(&bfin_eval_adau1373_driver); -} -module_exit(bfin_eval_adau1373_exit); +module_platform_driver(bfin_eval_adau1373_driver); MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); MODULE_DESCRIPTION("ALSA SoC bfin adau1373 driver"); diff --git a/sound/soc/blackfin/bfin-eval-adau1701.c b/sound/soc/blackfin/bfin-eval-adau1701.c index e5550ac..1a88fe9 100644 --- a/sound/soc/blackfin/bfin-eval-adau1701.c +++ b/sound/soc/blackfin/bfin-eval-adau1701.c @@ -121,17 +121,7 @@ static struct platform_driver bfin_eval_adau1701_driver = { .remove = __devexit_p(bfin_eval_adau1701_remove), }; -static int __init bfin_eval_adau1701_init(void) -{ - return platform_driver_register(&bfin_eval_adau1701_driver); -} -module_init(bfin_eval_adau1701_init); - -static void __exit bfin_eval_adau1701_exit(void) -{ - platform_driver_unregister(&bfin_eval_adau1701_driver); -} -module_exit(bfin_eval_adau1701_exit); +module_platform_driver(bfin_eval_adau1701_driver); MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); MODULE_DESCRIPTION("ALSA SoC bfin ADAU1701 driver"); diff --git a/sound/soc/blackfin/bfin-eval-adav80x.c b/sound/soc/blackfin/bfin-eval-adav80x.c index 897cfa6..0bc995f 100644 --- a/sound/soc/blackfin/bfin-eval-adav80x.c +++ b/sound/soc/blackfin/bfin-eval-adav80x.c @@ -157,17 +157,7 @@ static struct platform_driver bfin_eval_adav80x_driver = { .id_table = bfin_eval_adav80x_ids, }; -static int __init bfin_eval_adav80x_init(void) -{ - return platform_driver_register(&bfin_eval_adav80x_driver); -} -module_init(bfin_eval_adav80x_init); - -static void __exit bfin_eval_adav80x_exit(void) -{ - platform_driver_unregister(&bfin_eval_adav80x_driver); -} -module_exit(bfin_eval_adav80x_exit); +module_platform_driver(bfin_eval_adav80x_driver); MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); MODULE_DESCRIPTION("ALSA SoC bfin adav80x driver"); -- 1.7.5.4

3 2

[alsa-devel] [PATCH] alsa: export compress headers
by Vinod Koul 16 Jan '12

16 Jan '12

Export compress_offload.h and compress_params.h for userland to use Signed-off-by: Vinod Koul <vinod.koul(a)linux.intel.com> --- include/sound/Kbuild | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/include/sound/Kbuild b/include/sound/Kbuild index 802947f..6df30ed 100644 --- a/include/sound/Kbuild +++ b/include/sound/Kbuild @@ -6,3 +6,5 @@ header-y += hdsp.h header-y += hdspm.h header-y += sb16_csp.h header-y += sfnt_info.h +header-y += compress_params.h +header-y += compress_offload.h -- 1.7.0.4

3 2

[alsa-devel] overrun in case of record
by Rajeev kumar 12 Jan '12

12 Jan '12

Hello Mark I am testing my audio driver on spear platform.During recording I am getting two issues. 1. Overrun issue 2. pcm_read error I am using arecord for capturing data. #arecord -f dat > t overrun!!! (at least 32.949 ms long) Overrun issue: The issue is only reproducible if we try to record for more than 40 sec.The strange part is that Instead of overrun condition generated, it will record complete data for 10sec.(This is verified with the number of bytes recorded). So could you please let me know whether this is an error condition or just a warning. pcm_read error: For this I have seen a bug is locked in the bugzila(*Bug 472469* <show_bug.cgi?id=472469> -arecord: input/output error ). and the status is closed saying insufficient data. Could you please let me know what is the meaning of this bug? Is it like, data is not available in the buffer and pcm_read from the user space fails? But why this situation will arise? Best Regards Rajeev

4 15

Jump to page:

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

Alsa-devel December 2011