April 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [ANNOUNCE] Audio Mini Summit 2014, Oct. 14, Düsseldorf, Germany
by Takashi Iwai 24 Apr '15

24 Apr '15

Hi, like previous years, we're going to hold a meeting to discuss lowlevel audio on Linux. This time it's along with Linux Plumbers conference, but for avoiding the conflicts with other LPC mini conferences, it'll be held on Tuesday October 14, at the same venue as LPC, Congress Center Düsseldorf, Germany. If anyone is interested, feel free to sign up in the attendee list below. It's a public accessible Google doc, so you can edit freely. http://goo.gl/VbXLPW (or https://docs.google.com/spreadsheets/d/1wBAiAyZSnMjVFJbxdXE5JoVdKGIBIWt-4tu…) The topics that have been currently raised are found in the URL below. If you have any more topics to discuss there, just put your favorite one in the URL below as well. http://goo.gl/JSasQ9 (or https://docs.google.com/spreadsheets/d/1cXzON65QYNrTXi31AIQHffUZ9NWwJuNEJAC…) Thanks! Takashi

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[alsa-devel] [RFC v2] ASoC: Topology File format
by Liam Girdwood 24 Apr '15

24 Apr '15

Here's V2 of the file format the userspace topology tool will read to generate the binary output file. The output file uses the topology UAPI structures to represent the objects in the text file. This is still WiP and is based on the UCM file format from Jaroslav. Changes from V1. o Added BE's and Codec <-> codec links o Added enumerated and binary controls o Added private data o Moved object names from name field into object top level (like section names). o Control ops binding changed to string+number # Firmware Topology Configuration for Broadwell # TLV scale used by both global and stream volumes SectionTLV."hsw_vol_tlv" { DBScale [ min "-9000" step "300" mute "1" ] } # Values for Equalizer 1 SectionText."EQU1" { Values [ "100Hz", "200Hz", "1kHz" ] } # Private data for Equalizer 1 SectionData."EQU1" { file "/path/to/file" }; # Values for Input Muxes 1,2,3 SectionText."In1" { Values [ "Line", "Mic", ] } # Controls from default pipeline SectionControl."Default Pipeline" { # All controls in this section belong to this Index group/pipeline Index "1" # Master DSP volume Mixer."Master Playback Volume" [ # Channel register and shift for Front Left/Right Channel."FL" [ reg "0" shift "0" ] Channel."FR" [ reg "0" shift "8" ] # max control value and whether value is inverted max "31" invert "false" # control uses bespoke driver get/put/info ID 0 ops "DRV0" # uses TLV data above tlv_array "hsw_vol_tlv" ] # Offload 0 volume Mixer."Media0 Playback Volume" [ # Channel register and shift for Front Left/Right Channel."FL" [ reg "1" shift "0" ] Channel."FR" [ reg "1" shift "8" ] # max control value and whether value is inverted max "31" invert "false" # control uses bespoke driver get/put/info ID 0 ops "DRV0" # uses TLV data above tlv_array "hsw_vol_tlv" ] # Enum controls Enum."Equ1" [ # Texts for enum texts "EQU1" # Channel register and shift for Front Left/Right Channel."FL" [ reg "2" shift "0" ] Channel."FR" [ reg "1" shift "8" ] # control uses bespoke driver get/put/info ID 1 ops "DRV1" # Private data for this control data "EQU1" ] Enum."Input Select" [ # Texts for enum texts "In1" # Channel register and shift for Front Left/Right Channel."FL" [ reg "2" shift "0" ] Channel."FR" [ reg "1" shift "8" ] # control uses bespoke driver get/put/info ID 1 ops "DRV2" ] # Byte controls Bytes [ base "0" num_regs "16" mask "0xff" ] } # DAPM Widgets for main pipeline SectionWidget."Pipeline1" { # All controls in this section belong to this Index group/pipeline Index "1" # SSP AIF input AIF_IN."SSP0 CODEC IN" [ no_pm "true" ] # SSP AIF output AIF_OUT."SSP0 CODEC OUT" [ no_pm "true" ] # Virtual playback mixer MIXER."Playback VMixer" [ no_pm "true" ] # Test input mux MUX."Capture Select" [ no_pm "true" enum "Input Select" ] } # DAPM Routes SectionGraph."Pipeline 1 Graph" { Index "1" Routes [ "Playback VMixer, , System Playback" "Playback VMixer, , Offload0 Playback" "Playback VMixer, , Offload1 Playback" "SSP0 CODEC OUT, , Playback VMixer" "Analog Capture, , SSP0 CODEC IN" ] } # PCM Configurations supported by FW SectionPCMConfig."PCM 48k Stereo 24bit" { Playback [ format "S24_LE" rate "48000" channels "2" tdm_slot "0xf" ] Capture [ format "S24_LE" rate "48000" channels "2" tdm_slot "0xf" ] } SectionPCMConfig."PCM 48k Stereo 16bit" { Playback [ format "S16_LE" rate "48000" channels "2" tdm_slot "0xf" ] Capture [ format "S16_LE" rate "48000" channels "2" tdm_slot "0xf" ] } SectionPCMConfig."PCM 48k 2P/4C 16bit" { Playback [ format "S16_LE" rate "48000" channels "2" tdm_slot "0xf" ] Capture [ format "S16_LE" rate "48000" channels "4" tdm_slot "0xf" ] } SectionPCMConfig."PCM 8k Mono 16bit" { Playback [ format "S16_LE" rate "8000" channels "1" ] Capture [ format "S16_LE" rate "8000" channels "1" ] } # PCM capabilities supported by FW SectionPCMCapabilities."System Playback" { Capabilities [ formats "S24_LE, S16_LE" rate_min "48000" rate_max "48000" channels_min "2" channels_max "2" ] } SectionPCMCapabilities."Analog Capture" { Capabilities [ formats "S24_LE, S16_LE" rate_min "48000" rate_max "48000" channels_min "2" channels_max "4" ] } SectionPCMCapabilities."Offload Pin Playback" { Capabilities [ formats "S24_LE, S16_LE" rate_min "8000" rate_max "192000" channels_min "2" channels_max "2" ] } SectionPCMCapabilities."Loopback Capture" { Capabilities [ formats "S16_LE" rate_min "48000" rate_max "48000" channels_min "2" channels_max "2" ] } SectionPCMCapabilities."BT" { Capabilities [ formats "S16_LE" rate_min "8000" rate_max "8000" channels_min "1" channels_max "1" ] } # PCM devices exported by Firmware SectionPCM."System Pin" { Index "1" # used for binding to the PCM ID "0" Playback [ Capabilities "System Playback" Config "PCM 48k Stereo 24bit" Config "PCM 48k Stereo 16bit" ] Capture [ Capabilities "Analog Capture" Config "PCM 48k Stereo 24bit" Config "PCM 48k Stereo 16bit" Config "PCM 48k 2P/4C 16bit" ] } SectionPCM."Offload Pin 0" { Index "1" # used for binding to the PCM ID "1" Playback [ Capabilities "Offload Pin Playback" Config "PCM 48k Stereo 24bit" Config "PCM 48k Stereo 16bit" ] } SectionPCM."Offload Pin 1" { Index "1" # used for binding to the PCM ID "2" Playback [ Capabilities "Offload Pin Playback" Config "PCM 48k Stereo 24bit" Config "PCM 48k Stereo 16bit" ] } SectionPCM."Loopback Pin" { Index "1" # used for binding to the PCM ID "3" Capture [ Capabilities "Loopback Capture" Config "PCM 48k Stereo 16bit" ] } # Backend DAI links SectionBE."SSP0-Codec" { Index "1" # used for binding to the BE DAI link ID "0" # BE DAI link capabilities and supported configs Playback [ Capabilities "System Playback" Config "PCM 48k Stereo 24bit" Config "PCM 48k Stereo 16bit" ] Capture [ Capabilities "Analog Capture" Config "PCM 48k Stereo 24bit" Config "PCM 48k Stereo 16bit" Config "PCM 48k 2P/4C 16bit" ] } SectionBE."SSP1-BT" { Index "1" # used for binding to the BE DAI link ID "1" # BE DAI link capabilities and supported configs Playback [ Capabilities "BT" Config "PCM 8k Mono 16bit" ] Capture [ Capabilities "BT" Config "PCM 8k Mono 16bit" ] } # Codec <-> Codec style links SectionCC."FM-Codec" { Index "1" # used for binding to the CC link ID "0" # CC DAI link capabilities and supported configs Playback [ Capabilities "System Playback" Config "PCM 48k Stereo 16bit" ] Capture [ Capabilities "Analog Capture" Config "PCM 48k Stereo 16bit" ] }

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[alsa-devel] [PATCH 0/8] Proc file cleanups
by Takashi Iwai 23 Apr '15

23 Apr '15

Hi, here is a series of patches for cleaning up the ALSA proc file management code. The biggest change is the use of seq_file for text reads, but all are internal and the API is kept intact. The rest are mostly small cleanups and refactoring. The patches are found in topic/proc branch of sound git tree. Takashi === Takashi Iwai (8): ALSA: core: Use seq_file for text proc file reads ALSA: core: Fix possible memory leaks at error path in info.c ALSA: core: Remove child proc file elements recursively ALSA: core: Manage asound root directory with snd_info_entry ALSA: core: Remove superfluous exit calls for proc entries ALSA: core: Don't ignore errors at creating proc files ALSA: core: Build conditionally and remove superfluous ifdefs ALSA: core: Clean up OSS proc file management include/sound/core.h | 4 - include/sound/info.h | 29 +- sound/core/Makefile | 8 +- sound/core/info.c | 779 ++++++++++++++++++---------------------------- sound/core/info_oss.c | 29 +- sound/core/init.c | 33 +- sound/core/seq/Makefile | 3 +- sound/core/seq/seq_info.c | 19 +- sound/core/sound.c | 24 +- sound/core/sound_oss.c | 30 +- 10 files changed, 373 insertions(+), 585 deletions(-) -- 2.3.5

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Re: [alsa-devel] snd-bebob and PrismSound Orpheus fails with snd_pcm_hw_params_set_channels
by Mike Ood 23 Apr '15

23 Apr '15

Clemens wrote: > You should not write an .asoundrc. > You should replace "hw" with "plughw" in the device name. Somewhat different messages, but still aborts immediately. (There is a relay click sound from Orpheus, as it always does before playing music, so it is getting slightly better..) % ls -l .asound* -rw-rw-r-- 1 user1 user1 65 Apr 23 21:48 .asoundrc-nogo % grep asound /usr/share/alsa/alsa.conf "/etc/asound.conf" "~/.asoundrc" % sudo ls -l /etc/asound.conf ls: cannot access /etc/asound.conf: No such file or directory % aplay -D plughw:2,0 test.wav Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo aplay: set_params:1297: Unable to install hw params: ACCESS: RW_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 32 CHANNELS: 2 RATE: 44100 PERIOD_TIME: (46439 46440) PERIOD_SIZE: 2048 PERIOD_BYTES: 8192 PERIODS: 4 BUFFER_TIME: (185759 185760) BUFFER_SIZE: 8192 BUFFER_BYTES: 32768 TICK_TIME: 0 % aplay -D plughw:2 test.wav Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo aplay: set_params:1297: Unable to install hw params: ACCESS: RW_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 32 CHANNELS: 2 RATE: 44100 PERIOD_TIME: (46439 46440) PERIOD_SIZE: 2048 PERIOD_BYTES: 8192 PERIODS: 4 BUFFER_TIME: (185759 185760) BUFFER_SIZE: 8192 BUFFER_BYTES: 32768 TICK_TIME: 0 % aplay -D default:2,0 test.wav ALSA lib conf.c:4705:(snd_config_expand) Unknown parameters 2,0 ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM default:2,0 aplay: main:722: audio open error: Invalid argument % aplay -D default:2 test.wav ALSA lib conf.c:4705:(snd_config_expand) Unknown parameters 2 ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM default:2 aplay: main:722: audio open error: Invalid argument Regards, Mike Ood

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[alsa-devel] alsa-driver-build on kernel 3.13
by Raymond Yau 23 Apr '15

23 Apr '15

Do alsa-driver-build still support on kernel 3.13 ? Some of the modules have been built but not installed on Ubuntu 14.04 LTS [ 7.547726] snd_hda_codec: Unknown symbol snd_hdac_get_connections (err 0) [ 7.547733] snd_hda_codec: Unknown symbol snd_hdac_power_up_pm (err 0) [ 7.547749] snd_hda_codec: Unknown symbol snd_hdac_regmap_read_raw (err 0) [ 7.547762] snd_hda_codec: Unknown symbol snd_hdac_power_down (err 0) [ 7.547768] snd_hda_codec: Unknown symbol snd_hdac_make_cmd (err 0) [ 7.547783] snd_hda_codec: Unknown symbol snd_hdac_regmap_init (err 0) [ 7.547797] snd_hda_codec: Unknown symbol snd_hdac_refresh_widgets (err 0) [ 7.547824] snd_hda_codec: Unknown symbol snd_hdac_bus_exit (err 0) [ 7.547832] snd_hda_codec: Unknown symbol _snd_hdac_read_parm (err 0) [ 7.547845] snd_hda_codec: Unknown symbol snd_hdac_bus_exec_verb_unlocked (err 0) [ 7.547849] snd_hda_codec: Unknown symbol snd_hdac_override_parm (err 0) [ 7.547857] snd_hda_codec: Unknown symbol snd_hdac_device_register (err 0) [ 7.547862] snd_hda_codec: Unknown symbol snd_hdac_regmap_update_raw (err 0) [ 7.547878] snd_hda_codec: Unknown symbol snd_hdac_bus_init (err 0) [ 7.547886] snd_hda_codec: Unknown symbol snd_hdac_read_parm_uncached (err 0) [ 7.547892] snd_hda_codec: Unknown symbol snd_hdac_device_unregister (err 0) [ 7.547896] snd_hda_codec: Unknown symbol snd_hdac_device_init (err 0) [ 7.547908] snd_hda_codec: Unknown symbol snd_hdac_power_down_pm (err 0) [ 7.547924] snd_hda_codec: Unknown symbol snd_hdac_regmap_write_raw (err 0) [ 7.547933] snd_hda_codec: Unknown symbol snd_hdac_regmap_exit (err 0) [ 7.547943] snd_hda_codec: Unknown symbol snd_hda_bus_type (err 0) [ 7.547949] snd_hda_codec: Unknown symbol snd_hdac_device_exit (err 0) [ 7.547956] snd_hda_codec: Unknown symbol snd_hdac_get_sub_nodes (err 0) [ 7.547964] snd_hda_codec: Unknown symbol snd_hdac_exec_verb (err 0) [ 7.547973] snd_hda_codec: Unknown symbol snd_array_new (err 0) [ 7.547981] snd_hda_codec: Unknown symbol snd_array_free (err 0) [ 7.547994] snd_hda_codec: Unknown symbol snd_hdac_power_up (err 0) [ 7.548197] snd_hda_codec: Unknown symbol snd_hdac_get_connections (err 0) [ 7.548202] snd_hda_codec: Unknown symbol snd_hdac_power_up_pm (err 0) [ 7.548214] snd_hda_codec: Unknown symbol snd_hdac_regmap_read_raw (err 0) [ 7.548225] snd_hda_codec: Unknown symbol snd_hdac_power_down (err 0) [ 7.548231] snd_hda_codec: Unknown symbol snd_hdac_make_cmd (err 0) [ 7.548244] snd_hda_codec: Unknown symbol snd_hdac_regmap_init (err 0) [ 7.548256] snd_hda_codec: Unknown symbol snd_hdac_refresh_widgets (err 0) [ 7.548281] snd_hda_codec: Unknown symbol snd_hdac_bus_exit (err 0) [ 7.548287] snd_hda_codec: Unknown symbol _snd_hdac_read_parm (err 0) [ 7.548298] snd_hda_codec: Unknown symbol snd_hdac_bus_exec_verb_unlocked (err 0) [ 7.548302] snd_hda_codec: Unknown symbol snd_hdac_override_parm (err 0) [ 7.548310] snd_hda_codec: Unknown symbol snd_hdac_device_register (err 0) [ 7.548315] snd_hda_codec: Unknown symbol snd_hdac_regmap_update_raw (err 0) [ 7.548330] snd_hda_codec: Unknown symbol snd_hdac_bus_init (err 0) [ 7.548338] snd_hda_codec: Unknown symbol snd_hdac_read_parm_uncached (err 0) [ 7.548343] snd_hda_codec: Unknown symbol snd_hdac_device_unregister (err 0) [ 7.548347] snd_hda_codec: Unknown symbol snd_hdac_device_init (err 0) [ 7.548357] snd_hda_codec: Unknown symbol snd_hdac_power_down_pm (err 0) [ 7.548372] snd_hda_codec: Unknown symbol snd_hdac_regmap_write_raw (err 0) [ 7.548380] snd_hda_codec: Unknown symbol snd_hdac_regmap_exit (err 0) [ 7.548389] snd_hda_codec: Unknown symbol snd_hda_bus_type (err 0) [ 7.548395] snd_hda_codec: Unknown symbol snd_hdac_device_exit (err 0) [ 7.548402] snd_hda_codec: Unknown symbol snd_hdac_get_sub_nodes (err 0) [ 7.548410] snd_hda_codec: Unknown symbol snd_hdac_exec_verb (err 0) [ 7.548418] snd_hda_codec: Unknown symbol snd_array_new (err 0) [ 7.548426] snd_hda_codec: Unknown symbol snd_array_free (err 0) [ 7.548438] snd_hda_codec: Unknown symbol snd_hdac_power_up (err 0)

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Re: [alsa-devel] snd-bebob and PrismSound Orpheus fails with snd_pcm_hw_params_set_channels
by Mike Ood 23 Apr '15

23 Apr '15

Clemens wrote: > You are trying to play a two-channel file ... > ... on a device that supports only ten channels. This is what I was vaguely afraid of... > An with "hw", you have disabled any automatic format conversion. > Try "plughw:2" or "default:2". Could you be kind enough to suggest how I should write .asoundrc? I only know the obvious one for jack: .asoundrc ---------------- pcm.jackplug { type plug slave { pcm "jack" } } pcm.jack { type jack playback_ports { 0 alsa_pcm:playback_1 1 alsa_pcm:playback_2 } capture_ports { 0 alsa_pcm:capture_1 1 alsa_pcm:capture_2 } } ---------------- Regards, Mike Ood

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[alsa-devel] snd-bebob and PrismSound Orpheus fails with snd_pcm_hw_params_set_channels
by Mike Ood 23 Apr '15

23 Apr '15

Hi All, I have been using my PrismSound Orpheus successfully with juju+ffado+jackd as: % jackd -R -dfirewire -r44100 -p4096 -n3 & % aplay -D pcm.jack test.wav or % alsaplayer -i text -o jack -d "system:playback_1,system:playback_2" test.wav on Ubuntu 14.04 with kernel 3.13 or so. *** Today I clean installed ubuntu 14.10 wishing to use snd-bebob, but unfortunately % sudo aplay -D hw:2,0 --channels=2 test.wav does not work with the following messages: Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo aplay: set_params:1239: Channels count non available aplay.c: 1237 err = snd_pcm_hw_params_set_channels(handle, params, hwparams.channels); 1238 if (err < 0) { 1239 error(_("Channels count non available")); 1240 prg_exit(EXIT_FAILURE); 1241 } My short capture program, which works fine with USB audio devices, does not work either at the same place. myprog.cpp: /* set the count of channels */ err = snd_pcm_hw_params_set_channels(handle, params, channels); if (err < 0) { cout << "Sound::asound_set_hwparams: Channels count " << channels << " is not available for playbacks." << snd_strerror(err) << endl; return err; } Does anybody have any suggestions for me? Mike ODD *** (for your information) *** % uname -a Linux host 3.16.0-34-generic #47-Ubuntu SMP Fri Apr 10 18:02:58 UTC 2015 x86_64 x86_64 x86_64 GNU/Linux % ls -l /dev/fw* crw------- 1 root root 250, 0 Apr 23 13:57 /dev/fw0 crw-rw----+ 1 root video 250, 1 Apr 23 13:57 /dev/fw1 % lsmod | grep snd_bebob snd_bebob 40777 0 snd_firewire_lib 29081 1 snd_bebob snd_hwdep 17698 3 snd_usb_audio,snd_bebob,snd_hda_codec snd_pcm 104102 8 snd_firewire_lib,snd_usb_audio,snd_bebob,snd_hda_codec_hdmi,snd_hda_codec,snd_hda_intel,snd_hda_controller snd_rawmidi 30876 4 snd_firewire_lib,snd_bebob,snd_usbmidi_lib,snd_seq_midi snd 87611 31 snd_hda_codec_realtek,snd_usb_audio,snd_bebob,snd_hwdep,snd_timer,snd_hda_codec_hdmi,snd_pcm,snd_seq,snd_rawmidi,snd_hda_codec_generic,snd_usbmidi_lib,snd_hda_codec,snd_hda_intel,snd_seq_device firewire_core 68671 3 snd_firewire_lib,snd_bebob,firewire_ohci % ls -l /proc/asound total 0 dr-xr-xr-x 6 root root 0 Apr 23 21:02 card0 dr-xr-xr-x 3 root root 0 Apr 23 21:02 card1 dr-xr-xr-x 5 root root 0 Apr 23 21:02 card2 dr-xr-xr-x 4 root root 0 Apr 23 21:02 card3 -r--r--r-- 1 root root 0 Apr 23 21:02 cards lrwxrwxrwx 1 root root 5 Apr 23 21:02 CODEC -> card3 -r--r--r-- 1 root root 0 Apr 23 21:02 devices lrwxrwxrwx 1 root root 5 Apr 23 21:02 HDMI -> card1 -r--r--r-- 1 root root 0 Apr 23 21:02 hwdep lrwxrwxrwx 1 root root 5 Apr 23 21:02 MID -> card0 -r--r--r-- 1 root root 0 Apr 23 21:02 modules lrwxrwxrwx 1 root root 5 Apr 23 21:02 Orpheus -> card2 dr-xr-xr-x 2 root root 0 Apr 23 21:02 oss -r--r--r-- 1 root root 0 Apr 23 21:02 pcm dr-xr-xr-x 2 root root 0 Apr 23 21:02 seq -r--r--r-- 1 root root 0 Apr 23 21:02 timers -r--r--r-- 1 root root 0 Apr 23 21:02 version % ls -l /proc/asound/card2 total 0 dr-xr-xr-x 2 root root 0 Apr 23 21:02 firewire -r--r--r-- 1 root root 0 Apr 23 21:02 id -r--r--r-- 1 root root 0 Apr 23 21:02 midi0 dr-xr-xr-x 3 root root 0 Apr 23 21:02 pcm0c dr-xr-xr-x 3 root root 0 Apr 23 21:02 pcm0p % cat /proc/asound/card2/id Orpheus % ls -l /proc/asound/card2/firewire total 0 -r--r--r-- 1 root root 0 Apr 23 21:03 clock -r--r--r-- 1 root root 0 Apr 23 21:03 firmware -r--r--r-- 1 root root 0 Apr 23 21:03 formation % cat /proc/asound/card2/firewire/clock Sampling rate: 44100 Clock Source: External (MSU-dest: 7) % cat /proc/asound/card2/firewire/firmware Manufacturer: bridgeCo Protocol Ver: 3 Build Ver: 0 GUID: 0x00119800010001F9 Model ID: 0x13 Model Rev: 0 Firmware Date: 20100205 Firmware Time: 105129 Firmware ID: 0x10048 Firmware Ver: 6572 Base Addr: 0x400C0080 Max Size: 1519984 Loader Date: 20070118 Loader Time: 101440 % cat /proc/asound/card2/firewire/formation Output Stream from device: Rate PCM MIDI 32000 0 0 44100 10 1 48000 10 1 88200 10 1 96000 10 1 176400 10 1 192000 10 1 Input Stream to device: Rate PCM MIDI 32000 0 0 44100 12 1 48000 12 1 88200 12 1 96000 12 1 176400 12 1 192000 12 1 % ls -l /proc/asound/card2/pcm0c total 0 -r--r--r-- 1 root root 0 Apr 23 21:05 info dr-xr-xr-x 2 root root 0 Apr 23 21:05 sub0 % ls -l /proc/asound/card2/pcm0p total 0 -r--r--r-- 1 root root 0 Apr 23 21:06 info dr-xr-xr-x 2 root root 0 Apr 23 21:06 sub0 % cat /proc/asound/card2/pcm0c/info card: 2 device: 0 subdevice: 0 stream: CAPTURE id: BeBoB name: Orpheus PCM subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 1 % cat /proc/asound/card2/pcm0p/info card: 2 device: 0 subdevice: 0 stream: PLAYBACK id: BeBoB name: Orpheus PCM subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 1 *** (end of information) ***

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[alsa-devel] [PATCH] ASoC: Intel: fix the makefile for atom code
by Vinod Koul 23 Apr '15

23 Apr '15

The tom code should be using SND_SST_MFLD_PLATFORM and not the baytrail one. So fix it now Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/Makefile | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index cd9aee9871a3..3853ec2ddbc7 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -4,7 +4,7 @@ obj-$(CONFIG_SND_SOC_INTEL_SST) += common/ # Platform Support obj-$(CONFIG_SND_SOC_INTEL_HASWELL) += haswell/ obj-$(CONFIG_SND_SOC_INTEL_BAYTRAIL) += baytrail/ -obj-$(CONFIG_SND_SOC_INTEL_BAYTRAIL) += atom/ +obj-$(CONFIG_SND_SST_MFLD_PLATFORM) += atom/ # Machine support obj-$(CONFIG_SND_SOC_INTEL_SST) += boards/ -- 1.7.9.5

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[alsa-devel] [RFC] File format for topology data
by Liam Girdwood 23 Apr '15

23 Apr '15

I've pasted below the topology data for the broadwell firmware using the text file format being used by UCM. Using the UCM format allows to to share parser code and allows us to also have UCM define topology data. I've not yet completed the byte or enum controls but the rest of the data is there for comments. I've also included the recent channel changes suggested in the UAPI header comments. # Firmware Topology Configuration for Broadwell # TLV scale used by both global and stream volumes SectionTLV."hsw_vol_tlv" { DBScale [ min "-9000" step "300" mute "1" ] } # Controls from default pipeline SectionControl."Default Pipeline" { Index "1" # Master DSP volume Mixer [ name "Master Playback Volume" Channel.0 [ map "Left" reg "0" shift "0" ] Channel.1 [ map "Right" reg "0" shift "8" ] max "31" invert "false" get "0" put "0" tlv_array "hsw_vol_tlv" ] # Offload 0 volume Mixer [ name "Media0 Playback Volume" Channel.0 [ map "Left" reg "1" shift "0" ] Channel.1 [ map "Right" reg "1" shift "8" ] max "31" invert "false" get "1" put "1" tlv_array "hsw_vol_tlv" ] # Enum controls Enum [ ... ] # Byte controls Bytes [ ... ] } # DAPM Widgets for main pipeline SectionWidget."Pipeline1" { Index "1" AIF_IN [ name "SSP0 CODEC IN" stream_name "" slot "" reg "0" shift "0" invert "false" ] AIF_OUT [ name "SSP0 CODEC OUT" stream_name "" slot "" reg "0" shift "0" invert "false" ] ... } # DAPM Routes SectionGraph."Pipeline 1 Graph" { Index 1 Routes [ "Playback VMixer, , System Playback" "Playback VMixer, , Offload0 Playback" "Playback VMixer, , Offload1 Playback" "SSP0 CODEC OUT, , Playback VMixer" "Analog Capture, , SSP0 CODEC IN" ] } # PCM Configurations supported by FW SectionPCMConfig."PCM 48k Stereo 24bit" { Playback [ format "S24_LE" rate "48000" channels "2" tdm_slot "0xf" ] Capture [ format "S24_LE" rate "48000" channels "2" tdm_slot "0xf" ] } SectionPCMConfig."PCM 48k Stereo 16bit" { Playback [ format "S16_LE" rate "48000" channels "2" tdm_slot "0xf" ] Capture [ format "S16_LE" rate "48000" channels "2" tdm_slot "0xf" ] } SectionPCMConfig."PCM 48k 2P/4C 16bit" { Playback [ format "S16_LE" rate "48000" channels "2" tdm_slot "0xf" ] Capture [ format "S16_LE" rate "48000" channels "4" tdm_slot "0xf" ] } # PCM capabilities supported by FW SectionPCMCapabilities."System Playback" { Capabilities [ formats "S24_LE, S16_LE" rate_min "48000" rate_max "48000" channels_min "2" channels_max "2" ] } SectionPCMCapabilities."Analog Capture" { Capabilities [ formats "S24_LE, S16_LE" rate_min "48000" rate_max "48000" channels_min "2" channels_max "4" ] } SectionPCMCapabilities."Offload Pin Playback" { Capabilities [ formats "S24_LE, S16_LE" rate_min "8000" rate_max "192000" channels_min "2" channels_max "2" ] } SectionPCMCapabilities."Loopback Capture" { Capabilities [ formats "S16_LE" rate_min "48000" rate_max "48000" channels_min "2" channels_max "2" ] } # PCM devices exported by Firmware SectionPCM."System Pin" { Index 1 Playback [ Capabilities "System Playback" Config "PCM 48k Stereo 24bit" Config "PCM 48k Stereo 16bit" ] Capture [ Capabilities "Analog Capture" Config "PCM 48k Stereo 24bit" Config "PCM 48k Stereo 16bit" Config "PCM 48k 2P/4C 16bit" ] } SectionPCM."Offload Pin 0" { Index 1 Playback [ Capabilities "Offload Pin Playback" Config "PCM 48k Stereo 24bit" Config "PCM 48k Stereo 16bit" ] } SectionPCM."Offload Pin 1" { Index 1 Playback [ Capabilities "Offload Pin Playback" Config "PCM 48k Stereo 24bit" Config "PCM 48k Stereo 16bit" ] } SectionPCM."Loopback Pin" { Index 1 Capture [ Capabilities "Loopback Capture" Config "PCM 48k Stereo 16bit" ] }

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[alsa-devel] [PATCH] ALSA: hda - add AZX_DCAPS_I915_POWERWELL to Baytrail
by mengdong.lin＠intel.com 23 Apr '15

23 Apr '15

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch addes AZX_DCAPS_I915_POWERWELL to BYT (Baytrail). Like Braswell and Skylake, the HDMI codec on Bytrail is also in the shared power well with GPU. This power well must be turned on before we reset link to probe the codec, to avoid communication failure with the codec. The side effect is that this power is always ON in S0 because the BYT HDMI codec does not support EPSS or D3ClkStop and so the controller doesn't enter D3 at runtime, and the HDMI codec and analog codec share a single physical HD-A link and so we cannot reset the HD-A link freely when we re-enable the power to use the HDMI codec. Next step is to test if an AGP reset or double AGP reset on BYT HDMI codec is okay to bring the HDMI codec back to a functional state after restoring the power. If okay, we can bind the power on/off with the HDMI codec PM without interrupting the analog audio. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e1c2105..34040d2 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -297,6 +297,9 @@ enum { AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_POWERWELL |\ AZX_DCAPS_SNOOP_TYPE(SCH)) +#define AZX_DCAPS_INTEL_BAYTRAIL \ + (AZX_DCAPS_INTEL_PCH_NOPM | AZX_DCAPS_I915_POWERWELL) + #define AZX_DCAPS_INTEL_BRASWELL \ (AZX_DCAPS_INTEL_PCH | AZX_DCAPS_I915_POWERWELL) @@ -1992,7 +1995,7 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH_NOPM }, /* BayTrail */ { PCI_DEVICE(0x8086, 0x0f04), - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_BAYTRAIL }, /* Braswell */ { PCI_DEVICE(0x8086, 0x2284), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_BRASWELL }, -- 1.9.1

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