> >http://pastebin.com/NnDvQ2kU > :~$ aplay --dump-hw-params -D surround40:CARD=Live /usr/share/sounds/alsa/Front_Left.wav appropriate number of channels is not available WAVE '/usr/share/sounds/alsa/Front_Left.wav' : Signed 16 bit Little Endian, frequency 48000 Hz, Mono HW Params of device "surround40:CARD=Live": -------------------- ACCESS: MMAP_COMPLEX RW_INTERLEAVED RW_NONINTERLEAVED FORMAT: U8 S16_LE SUBFORMAT: STD SAMPLE_BITS: [8 16] FRAME_BITS: [32 64] CHANNELS: 4 RATE: [4000 96000] PERIOD_TIME: (166 8192000] PERIOD_SIZE: [16 32768] PERIOD_BYTES: [64 262144] PERIODS: [1 1024] BUFFER_TIME: (666 8192000] BUFFER_SIZE: [64 32768] BUFFER_BYTES: [256 262144] TICK_TIME: ALL -------------------- Both hw and front (mono and stereo profile) support MMAP_INTERLEAVED and RW_INTERLEAVED BUT surround40 and surround51 support MMAP_COMPLEX or RW_INTERLEAVED 44100Hz is supported but disabled when pulseaudio use no resample Seem bug or limitaion of multi plugin when the slaves does not support RW_NONINTERLEAVED Do pulseaudio expect all profiles of the sink or card use same access type ? Seem no way to force pulseaudio to try RW_INTERLEAVED first since udev-detect only has tsched flag http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/module-udev-detect.c What is the purpose of bool *use_mmap in pa_alsa_set_hw_params when user are not recommended to use module-alsa-sink indefault.pa int pa_alsa_set_hw_params( snd_pcm_t *pcm_handle, pa_sample_spec *ss, snd_pcm_uframes_t *period_size, snd_pcm_uframes_t *buffer_size, snd_pcm_uframes_t tsched_size, bool *use_mmap, bool *use_tsched, bool require_exact_channel_number) { http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/module-alsa-card.c "mmap=<enable memory mapping?> " http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/module-alsa-sink.c "mmap=<enable memory mapping?> " http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/module-alsa-source.c "mmap=<enable memory mapping?> " > > >Do pactl can select the capture source? > > I don't know what specific command shold I use. > pactl --help You have to post output of pactl list Capture source is not a mixer element but a hctl control control.64 { iface MIXER name 'Capture Source' value.0 Mic value.1 Mic comment { access 'read write' type ENUMERATED count 2 item.0 Mic item.1 CD item.2 Video item.3 Aux item.4 Line item.5 Mix item.6 'Mix Mono' item.7 Phone } } Those mixer elements with " Capture exclusive group: 0" are belong to "Capture Source" control cswitch-exclusive mean only one of the source can be switch on within capture exclusive group Simple mixer control 'Line',0 Capabilities: pvolume pswitch pswitch-joined cswitch cswitch-exclusive Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Front Left: Playback 31 [100%] [12.00dB] [on] Capture [off] Front Right: Playback 31 [100%] [12.00dB] [on] Capture [off] To select line in as capture source, you need to set the line capture switch on those mixer elements with "pswitch-joined" have input to ac97 analog mixer you have to switch on line playback volume and switch if you want tv card output sound through line in of ac97 mixer to the front line out Pcm playback volume/switch with pswitch-joined is also an input to ac97 analog mixer Simple mixer control 'PCM',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 23 [74%] [0.00dB] [on] Front Right: Playback 23 [74%] [0.00dB] [on] You need to select stereo mix if you want to capture from mix of line in, mic, pcm.... and change those source playback volume and switch http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mixer/paths Is "pactl set-source-port" the correct command to set capture source control if pulseaudio does not save and restore the selected source port of the default sink for those sound cards with ac97 codec http://cgit.freedesktop.org/pulseaudio/pulseaudio/commit/src/modules/alsa/alsa-mixer.c?id=300a5e3ed70064c296e09bc4e40531f3257154c5 > > Which devices do you want to use? Those Multichannel's I had when I disabled pulse. http://git.alsa-project.org/?p=alsa-lib.git;a=blob;f=src/conf/cards/EMU10K1.conf;hb=HEAD You have to ask the author of snd-emu10k1 as I only know how to use devices in the above conf https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/Documentation/sound/alsa/emu10k1-jack.txt >> card 0: Live [SB Live! Value [CT4832]], device 2: emu10k1 efx [Multichannel Capture/PT Playback] >> Subdevices: 8/8 >> Subdevice #0: subdevice #0 >> Subdevice #1: subdevice #1 >> Subdevice #2: subdevice #2 >> Subdevice #3: subdevice #3 >> Subdevice #4: subdevice #4 >> Subdevice #5: subdevice #5 >> Subdevice #6: subdevice #6 >> Subdevice #7: subdevice #7 >> >> card 0: Live [SB Live! Value [CT4832]], device 3: emu10k1 [Multichannel Playback] >> Subdevices: 1/1 >> Subdevice #0: subdevice #0 >> >> ARECORD >> >> **** List of CAPTURE Hardware Devices **** >> >> card 0: Live [SB Live! Value [CT4832]], device 1: emu10k1 mic [Mic Capture] >> Subdevices: 1/1 >> Subdevice #0: subdevice #0 >> >> card 0: Live [SB Live! Value [CT4832]], device 2: emu10k1 efx [Multichannel Capture/PT Playback] >> Subdevices: 1/1 >> Subdevice #0: subdevice #0 >> >> > Hmmm. Now I use surround 4.0 >http://i.imgur.com/UZsnWqs.png > > but whatever is needed to accomplish it. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.freedesktop.org/archives/pulseaudio-discuss/attachments/20141124/dfd0f14b/attachment.html>
