February 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] tlv320aic3x regcache issue
by Jack Mitchell 26 Feb '14

26 Feb '14

Hi, I have come across an issue where the regcache mechanism used in the tlv320aic3x driver is causing problems. When switching from 44.1khz to 48khz some of the registers are not written due to the default register cache holding the same value as being written. However, the codec requires these registers to be written in order to trigger the change, this has been proven by changing the register default to something different, therefore forcing a write at the correct time and it triggers the change as expected. What would be the best way to force these registers to be written? Is there some sort of force write flag, or a way to turn off the cache just during this portion of the configuration? Any direction in the proper way of doing this would be much appreciated. Cheers, Jack -- Jack Mitchell (jack(a)embed.me.uk) Embedded Systems Engineer Cambridgeshire, UK http://www.embed.me.uk --

1 0

[alsa-devel] [PATCH 00/48] Rewrite kernel messages with standard printk helpers
by Takashi Iwai 26 Feb '14

26 Feb '14

Hi, this is another long series of patches for trivial cleanups. This time, they replace the printk and snd_print*() calls in ALSA PCI and USB drivers with dev_err() and co. This will give nicer and more consistent outputs. The only drawback by these replacements is that alsa-info.sh won't be able to catch the dmesg lines because it doesn't contain any longer the word "ALSA", but only show the module name like "snd_intel8x0". I'm going to fix alsa-info.sh on git soon later, but if you guys think "oh no it's disaster!", let me know. We need to reconsider a better way to put the known markers (like explicitly putting "ALSA" at the beginning.) Takashi

1 49

[alsa-devel] [PATCH 2/2] ASoC: cirrus: clps711x: Add bindings documentation for the CLPS711X DAI
by Alexander Shiyan 26 Feb '14

26 Feb '14

This patch adds the devicetree documentation for the Cirrus Logic CLPS711X Digital Audio Interface (DAI). Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> --- .../bindings/sound/cirrus,clps711x-dai.txt | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cirrus,clps711x-dai.txt diff --git a/Documentation/devicetree/bindings/sound/cirrus,clps711x-dai.txt b/Documentation/devicetree/bindings/sound/cirrus,clps711x-dai.txt new file mode 100644 index 0000000..718599e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cirrus,clps711x-dai.txt @@ -0,0 +1,20 @@ +* Cirrus Logic CLPS711X Digital Audio Interface (DAI) + +Required properties: +- compatible: Should be "cirrus,clps711x-dai". +- reg: Should contain registers location and length. +- clocks: A clock specifier for the CPU PLL clock. +- clock-names: The CPU PLL clock should be named "pll". +- interrupts: Should contain DAI interrupt number. +- syscon: phandle to syscon device node which control DAI (SYSCON3). + +Example: + dai: dai@80002000 { + #sound-dai-cells = <0>; + compatible = "cirrus,clps711x-dai"; + reg = <0x80002000 0x604>; + clocks = <&clks 3>; + clock-names = "pll"; + interrupts = <32>; + syscon = <&syscon3>; + }; -- 1.8.3.2

1 0

[alsa-devel] [PATCH] ALSA: rme96: Convert to the new pm_ops
by Takashi Iwai 26 Feb '14

26 Feb '14

This driver slipped from the last rewrite. Just convert to the new standard pm ops. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/rme96.c | 34 ++++++++++++++++++---------------- 1 file changed, 18 insertions(+), 16 deletions(-) diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index 00d46116e04a..f5014a8e69ad 100644 --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -240,7 +240,7 @@ struct rme96 { u8 rev; /* card revision number */ -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP u32 playback_pointer; u32 capture_pointer; void *playback_suspend_buffer; @@ -1570,7 +1570,7 @@ snd_rme96_free(void *private_data) pci_release_regions(rme96->pci); rme96->port = 0; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP vfree(rme96->playback_suspend_buffer); vfree(rme96->capture_suspend_buffer); #endif @@ -2372,13 +2372,12 @@ snd_rme96_create_switches(struct snd_card *card, * Card initialisation */ -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP -static int -snd_rme96_suspend(struct pci_dev *pci, - pm_message_t state) +static int rme96_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct rme96 *rme96 = card->private_data; snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); @@ -2407,10 +2406,10 @@ snd_rme96_suspend(struct pci_dev *pci, return 0; } -static int -snd_rme96_resume(struct pci_dev *pci) +static int rme96_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct rme96 *rme96 = card->private_data; pci_restore_state(pci); @@ -2451,7 +2450,11 @@ snd_rme96_resume(struct pci_dev *pci) return 0; } -#endif +static SIMPLE_DEV_PM_OPS(rme96_pm, rme96_suspend, rme96_resume); +#define RME96_PM_OPS &rme96_pm +#else +#define RME96_PM_OPS NULL +#endif /* CONFIG_PM_SLEEP */ static void snd_rme96_card_free(struct snd_card *card) { @@ -2488,7 +2491,7 @@ snd_rme96_probe(struct pci_dev *pci, return err; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP rme96->playback_suspend_buffer = vmalloc(RME96_BUFFER_SIZE); if (!rme96->playback_suspend_buffer) { snd_printk(KERN_ERR @@ -2547,10 +2550,9 @@ static struct pci_driver rme96_driver = { .id_table = snd_rme96_ids, .probe = snd_rme96_probe, .remove = snd_rme96_remove, -#ifdef CONFIG_PM - .suspend = snd_rme96_suspend, - .resume = snd_rme96_resume, -#endif + .driver = { + .pm = RME96_PM_OPS, + }, }; module_pci_driver(rme96_driver); -- 1.9.0

1 0

[alsa-devel] [alsa-plugins] [PATCH 1/2] Allow to specify a client name as parameters
by Corfu, ValentinX 26 Feb '14

26 Feb '14

Hi Takashi, I reworked the patches and hope to be the final ones. J You will find attached at each of the emails. Kind Regards, Valentin Corfu Intel GmbH Dornacher Strasse 1 85622 Feldkirchen/Muenchen, Deutschland Sitz der Gesellschaft: Feldkirchen bei Muenchen Geschaeftsfuehrer: Christian Lamprechter, Hannes Schwaderer, Douglas Lusk Registergericht: Muenchen HRB 47456 Ust.-IdNr./VAT Registration No.: DE129385895 Citibank Frankfurt a.M. (BLZ 502 109 00) 600119052

2 1

[alsa-devel] [PATCH - speaker-test 1/1] speaker-test: add --force-frequency option to allow hz outside range
by erik＠caspar.alsa-project.org 26 Feb '14

26 Feb '14

From: Erik Ackermann <domohawk(a)gmail.com> Signed-off-by: Erik Ackermann <domohawk(a)gmail.com> diff --git a/speaker-test/speaker-test.c b/speaker-test/speaker-test.c index 261f399..22927f2 100644 --- a/speaker-test/speaker-test.c +++ b/speaker-test/speaker-test.c @@ -106,6 +106,7 @@ static snd_pcm_uframes_t period_size; static const char *given_test_wav_file = NULL; static char *wav_file_dir = SOUNDSDIR; static int debug = 0; +static int force_frequency = 0; static int in_aborting = 0; static snd_pcm_t *pcm_handle = NULL; @@ -1015,6 +1016,7 @@ static void help(void) "-w,--wavfile Use the given WAV file as a test sound\n" "-W,--wavdir Specify the directory containing WAV files\n" "-m,--chmap Specify the channel map to override\n" + "-X,--force-frequency force frequencies outside the 30-5000hz range\n" "\n")); printf(_("Recognized sample formats are:")); for (fmt = supported_formats; *fmt >= 0; fmt++) { @@ -1057,6 +1059,7 @@ int main(int argc, char *argv[]) { {"wavfile", 1, NULL, 'w'}, {"wavdir", 1, NULL, 'W'}, {"debug", 0, NULL, 'd'}, + {"force-frequncy", 1, NULL, 'X'}, #ifdef CONFIG_SUPPORT_CHMAP {"chmap", 1, NULL, 'm'}, #endif @@ -1078,7 +1081,7 @@ int main(int argc, char *argv[]) { while (1) { int c; - if ((c = getopt_long(argc, argv, "hD:r:c:f:F:b:p:P:t:l:s:w:W:d" + if ((c = getopt_long(argc, argv, "hD:r:c:f:F:b:p:P:t:l:s:w:W:d:X" #ifdef CONFIG_SUPPORT_CHMAP "m:" #endif @@ -1114,8 +1117,6 @@ int main(int argc, char *argv[]) { break; case 'f': freq = atof(optarg); - freq = freq < 30.0 ? 30.0 : freq; - freq = freq > 5000.0 ? 5000.0 : freq; break; case 'b': buffer_time = atoi(optarg); @@ -1173,6 +1174,9 @@ int main(int argc, char *argv[]) { case 'd': debug = 1; break; + case 'X': + force_frequency = 1; + break; #ifdef CONFIG_SUPPORT_CHMAP case 'm': chmap = optarg; @@ -1190,6 +1194,11 @@ int main(int argc, char *argv[]) { exit(EXIT_SUCCESS); } + if (!force_frequency) { + freq = freq < 30.0 ? 30.0 : freq; + freq = freq > 5000.0 ? 5000.0 : freq; + } + if (test_type == TEST_WAV) format = SND_PCM_FORMAT_S16_LE; /* fixed format */ -- 1.8.3.2

4 7

[alsa-devel] MAYA44 USB audio device no inputs control (again)
by Leo Combes 26 Feb '14

26 Feb '14

Hi, I have bought a Maya 44 USB, with 4 output and 4 input channels. http://www.esi-audio.com/products/maya44usb/ The soundcard works fine in playback, but in recording has some problems, specificly there is no way to control the volume of the recording on inputs. It seems to be default setted to 50%. Alsamixer, in playback section shows 3 controls, "speaker", "line" and "line1". But, if I choose capture, it shows "This sound device does not have any capture controls". It seems like there are controls, and they work fine, running on Windows. I searched the problem on Google and I found a lot of users with the same problem. The most interesting thread that I found, is one on this mailist, sadly after some messages, user didn't catch up with the conversation: http://mailman.alsa-project.org/pipermail/alsa-devel/2011-November/046242.h… I have a hardware mixer connected to 2 inputs of the Maya. If I try to recording with Audacity or any other software, the volume of recording is never greater than -6dB. If I try increase the volume in the mixer to get louder recording, the only thing I get is a completely distorted sound not louder than -6dB, what tells me that Maya inputs have a presetted volume at 50%. Besides, if I increase the volume in the mixer, Audicity's vumeter confirm that it never go further than -6dB. Following the thread that the user left unfinished, here goes the information that I get with my system, to see if could find a solution. leo-ThinkPad leo # aplay -L null Discard all samples (playback) or generate zero samples (capture) pulse PulseAudio Sound Server sysdefault:CARD=MID HDA Intel MID, CX20585 Analog Default Audio Device front:CARD=MID,DEV=0 HDA Intel MID, CX20585 Analog Front speakers surround40:CARD=MID,DEV=0 HDA Intel MID, CX20585 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=MID,DEV=0 HDA Intel MID, CX20585 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=MID,DEV=0 HDA Intel MID, CX20585 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=MID,DEV=0 HDA Intel MID, CX20585 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=MID,DEV=0 HDA Intel MID, CX20585 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers dmix:CARD=MID,DEV=0 HDA Intel MID, CX20585 Analog Direct sample mixing device dsnoop:CARD=MID,DEV=0 HDA Intel MID, CX20585 Analog Direct sample snooping device hw:CARD=MID,DEV=0 HDA Intel MID, CX20585 Analog Direct hardware device without any conversions plughw:CARD=MID,DEV=0 HDA Intel MID, CX20585 Analog Hardware device with all software conversions hdmi:CARD=NVidia,DEV=0 HDA NVidia, HDMI 0 HDMI Audio Output hdmi:CARD=NVidia,DEV=1 HDA NVidia, HDMI 0 HDMI Audio Output hdmi:CARD=NVidia,DEV=2 HDA NVidia, HDMI 0 HDMI Audio Output hdmi:CARD=NVidia,DEV=3 HDA NVidia, HDMI 0 HDMI Audio Output dmix:CARD=NVidia,DEV=3 HDA NVidia, HDMI 0 Direct sample mixing device dmix:CARD=NVidia,DEV=7 HDA NVidia, HDMI 0 Direct sample mixing device dmix:CARD=NVidia,DEV=8 HDA NVidia, HDMI 0 Direct sample mixing device dmix:CARD=NVidia,DEV=9 HDA NVidia, HDMI 0 Direct sample mixing device dsnoop:CARD=NVidia,DEV=3 HDA NVidia, HDMI 0 Direct sample snooping device dsnoop:CARD=NVidia,DEV=7 HDA NVidia, HDMI 0 Direct sample snooping device dsnoop:CARD=NVidia,DEV=8 HDA NVidia, HDMI 0 Direct sample snooping device dsnoop:CARD=NVidia,DEV=9 HDA NVidia, HDMI 0 Direct sample snooping device hw:CARD=NVidia,DEV=3 HDA NVidia, HDMI 0 Direct hardware device without any conversions hw:CARD=NVidia,DEV=7 HDA NVidia, HDMI 0 Direct hardware device without any conversions hw:CARD=NVidia,DEV=8 HDA NVidia, HDMI 0 Direct hardware device without any conversions hw:CARD=NVidia,DEV=9 HDA NVidia, HDMI 0 Direct hardware device without any conversions plughw:CARD=NVidia,DEV=3 HDA NVidia, HDMI 0 Hardware device with all software conversions plughw:CARD=NVidia,DEV=7 HDA NVidia, HDMI 0 Hardware device with all software conversions plughw:CARD=NVidia,DEV=8 HDA NVidia, HDMI 0 Hardware device with all software conversions plughw:CARD=NVidia,DEV=9 HDA NVidia, HDMI 0 Hardware device with all software conversions sysdefault:CARD=USB MAYA44 USB+, USB Audio Default Audio Device front:CARD=USB,DEV=0 MAYA44 USB+, USB Audio Front speakers surround40:CARD=USB,DEV=0 MAYA44 USB+, USB Audio 4.0 Surround output to Front and Rear speakers surround41:CARD=USB,DEV=0 MAYA44 USB+, USB Audio 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=USB,DEV=0 MAYA44 USB+, USB Audio 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=USB,DEV=0 MAYA44 USB+, USB Audio 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=USB,DEV=0 MAYA44 USB+, USB Audio 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=USB,DEV=0 MAYA44 USB+, USB Audio IEC958 (S/PDIF) Digital Audio Output dmix:CARD=USB,DEV=0 MAYA44 USB+, USB Audio Direct sample mixing device dsnoop:CARD=USB,DEV=0 MAYA44 USB+, USB Audio Direct sample snooping device hw:CARD=USB,DEV=0 MAYA44 USB+, USB Audio Direct hardware device without any conversions plughw:CARD=USB,DEV=0 MAYA44 USB+, USB Audio Hardware device with all software conversions leo-ThinkPad leo # lsmod | grep snd snd_hda_codec_hdmi 41684 4 snd_usb_audio 155972 3 snd_hda_codec_conexant 57181 1 snd_usbmidi_lib 29576 1 snd_usb_audio snd_hda_intel 57183 2 snd_hda_codec 194881 3 snd_hda_codec_hdmi,snd_hda_codec_conexant,snd_hda_intel snd_hwdep 13613 2 snd_usb_audio,snd_hda_codec snd_pcm 107140 6 snd_usb_audio,snd_hda_codec_hdmi,snd_hda_codec,snd_hda_intel snd_page_alloc 18798 2 snd_pcm,snd_hda_intel snd_seq_midi 13324 0 snd_seq_midi_event 14899 1 snd_seq_midi snd_rawmidi 30465 2 snd_usbmidi_lib,snd_seq_midi snd_seq 66061 2 snd_seq_midi_event,snd_seq_midi snd_seq_device 14497 3 snd_seq,snd_rawmidi,snd_seq_midi snd_timer 30038 2 snd_pcm,snd_seq snd 73802 22 snd_usb_audio,snd_hwdep,snd_timer,snd_hda_codec_hdmi,snd_hda_codec_conexant,snd_pcm,snd_seq,snd_rawmidi,snd_usbmidi_lib,snd_hda_codec,snd_hda_intel,thinkpad_acpi,snd_seq_device,snd_seq_midi soundcore 12680 1 snd leo-ThinkPad leo # amixer contents numid=2,iface=MIXER,name='Line Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=4,iface=MIXER,name='Line Playback Switch',index=1 ; type=BOOLEAN,access=rw------,values=1 : values=off numid=3,iface=MIXER,name='Line Playback Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=255,step=0 : values=0,0 | dBminmax-min=0.00dB,max=0.99dB numid=5,iface=MIXER,name='Line Playback Volume',index=1 ; type=INTEGER,access=rw---R--,values=2,min=0,max=255,step=0 : values=0,0 | dBminmax-min=0.00dB,max=0.99dB numid=6,iface=MIXER,name='Speaker Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=7,iface=MIXER,name='Speaker Playback Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=255,step=0 : values=255,255 | dBminmax-min=0.00dB,max=0.99dB numid=1,iface=PCM,name='Playback Channel Map' ; type=INTEGER,access=r----R--,values=4,min=0,max=36,step=0 : values=3,4,0,0 | | TLV size error (257, 8, 0)! leo-ThinkPad leo # amixer -c2 Simple mixer control 'Speaker',0 Capabilities: pvolume pswitch pswitch-joined penum Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [0.99dB] [on] Front Right: Playback 255 [100%] [0.99dB] [on] Simple mixer control 'Line',0 Capabilities: pvolume pswitch pswitch-joined penum Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 0 [0%] [0.00dB] [off] Front Right: Playback 0 [0%] [0.00dB] [off] Simple mixer control 'Line',1 Capabilities: pvolume pswitch pswitch-joined penum Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 0 [0%] [0.00dB] [off] Front Right: Playback 0 [0%] [0.00dB] [off] Thanks in advance!

1 0

[alsa-devel] [PATCH 1/1] ALSA: compress: no need to set metadata before next track
by Richard Fitzgerald 26 Feb '14

26 Feb '14

The metadata is mainly for MP3 gapless playback, since the MP3 audio stream does not contain enough information to enable gapless. Other audio formats do not necessarily require any additional metadata so we should allow next_track to be called without any metadata. Signed-off-by: Zhao Weijia <weijia.zhao(a)capelabs.com> Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/core/compress_offload.c | 6 ------ 1 files changed, 0 insertions(+), 6 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 7a20897..5a351c7 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -744,12 +744,6 @@ static int snd_compr_next_track(struct snd_compr_stream *stream) if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) return -EPERM; - /* you can signal next track isf this is intended to be a gapless stream - * and current track metadata is set - */ - if (stream->metadata_set == false) - return -EPERM; - retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_NEXT_TRACK); if (retval != 0) return retval; -- 1.7.2.5

1 0

[alsa-devel] [PATCH] ASoC: Intel: Baytrail: Fix implicit declaration of function 'memcpy_fromio'
by Jarkko Nikula 26 Feb '14

26 Feb '14

Some kernel configurations can cause following build error: sound/soc/intel/sst-baytrail-ipc.c: In function ‘sst_byt_get_dsp_position’: sound/soc/intel/sst-baytrail-ipc.c:744:2: error: implicit declaration of function ‘memcpy_fromio’ [-Werror=implicit-function-declaration] memcpy_fromio(&fw_tstamp, ^ cc1: some warnings being treated as errors Fix this by including <asm/io.h> explicitly. Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- Also sst-dsp., sst-baytrail-dsp.c and sst-haswell-dsp.c are using memcpy_fromio/_toio but there <asm/io.h> seems to come via <linux/dma-mapping.h> or <linux/io.h> so I didn't touch them. --- sound/soc/intel/sst-baytrail-ipc.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/sst-baytrail-ipc.c b/sound/soc/intel/sst-baytrail-ipc.c index 8c91a68b90bc..8b067e34094e 100644 --- a/sound/soc/intel/sst-baytrail-ipc.c +++ b/sound/soc/intel/sst-baytrail-ipc.c @@ -27,6 +27,7 @@ #include <linux/kthread.h> #include <linux/firmware.h> #include <asm/div64.h> +#include <asm/io.h> #include "sst-baytrail-ipc.h" #include "sst-dsp.h" -- 1.8.5.3

3 3

[alsa-devel] [alsa-plugins] [PATCH 2/2] Allow number of channels
by Corfu, ValentinX 26 Feb '14

26 Feb '14

Intel GmbH Dornacher Strasse 1 85622 Feldkirchen/Muenchen, Deutschland Sitz der Gesellschaft: Feldkirchen bei Muenchen Geschaeftsfuehrer: Christian Lamprechter, Hannes Schwaderer, Douglas Lusk Registergericht: Muenchen HRB 47456 Ust.-IdNr./VAT Registration No.: DE129385895 Citibank Frankfurt a.M. (BLZ 502 109 00) 600119052

1 0