alsa plugin progress

Marcel,

I've been tweaking the alsa plugin. How close was it when you had it?

For now, I am setting the socket to nonblocking and polling until the connect succeeds (I keep trying when errno is EAGAIN). I can connect and things seem to be working but there is no audio. It looks like there's a mismatch with audio settings but I'm not sure where that's happening.

Brad

DEBUG: _snd_pcm_a2dp_open: name pcm.headphone mode 0
DEBUG: _snd_pcm_a2dp_open: bdaddr/dest is 00:08:F4:46:03:1C
DEBUG: a2dp_alloc: init
DEBUG: a2dp_connect: a2dp 0x805ac80
DEBUG: connect_l2cap: Can't connect to 1C:03:46:F4:08:00 on psm 25. Resource temporarily unavailable(11)
DEBUG: connect_l2cap: Can't connect to 1C:03:46:F4:08:00 on psm 25. Resource temporarily unavailable(11)
DEBUG: connect_l2cap: Can't connect to 1C:03:46:F4:08:00 on psm 25. Resource temporarily unavailable(11)
DEBUG: connect_l2cap: Connected [imtu 672, omtu 672, flush_to 65535]
DEBUG: connect_stream: Sent the Stream End Point Discovery Command
DEBUG: connect_stream: retrying discover response read...
DEBUG: connect_stream: Got a Stream End Point Discovery Response
DEBUG: connect_stream: received 1 capabilities
DEBUG: process_seid: SEID = 1
DEBUG: process_seid: Requested Capabilities for SEID = 1
DEBUG: process_seid: Got capabilities response
DEBUG: process_seid: headset does not support this channel mode
DEBUG: process_seid: headset does not support this frequency
DEBUG: process_seid: headset does not support this allocation_method
DEBUG: process_seid: headset does not support this subbands setting
DEBUG: process_seid: Sent set configurations command
DEBUG: process_seid: Set configurations command rejected
DEBUG: process_seid: Sent open stream command
DEBUG: process_seid: Got open stream confirm
DEBUG: connect_l2cap: Can't connect to 1C:03:46:F4:08:00 on psm 25. Resource temporarily unavailable(11)
DEBUG: connect_l2cap: Connected [imtu 672, omtu 0, flush_to 65535]
DEBUG: connect_stream: Sent stream start
DEBUG: connect_stream: Got start stream confirm
DEBUG: a2dp_constraint: a2dp 0x805ac80
Playing WAVE '/usr/lib/openoffice2/share/gallery/sounds/pluck.wav' :
Signed 16 bit Little Endian, Rate 11025 Hz, Mono
Warning: rate is not accurate (requested = 11025Hz, got = 44100Hz)
please, try the plug plugin (-Dplug:pcm.headphone)
DEBUG: a2dp_params: a2dp 0x805ac80
DEBUG: a2dp_params: format S16_LE rate 44100 channels 1
DEBUG: a2dp_params: frame_bytes 2 period_bytes 8192 period_size 4096 buffer_size 8192
DEBUG: a2dp_prepare: a2dp 0x805ac80
DEBUG: a2dp_drain: a2dp 0x805ac80
DEBUG: a2dp_stop: a2dp 0x805ac80
DEBUG: a2dp_stop: a2dp 0x805ac80
DEBUG: a2dp_close: a2dp 0x805ac80

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Hi Brad,

it was working somehow and sometimes I totally messed it up. The ALSA plugins are really designed for sound cards and for some options I actually have no idea how to translate them correctly into A2DP. Maybe you should simply play with different options and then might ask the gurus from the ALSA mailing list.

Regards

Marcel

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Marcel,

I've had this nagging feeling I need to reorganize the btsco project. I think if you make me a devel on the sourceforge "sbc" project, that could be the place for sbc with no unnecessary requirements. If you're willing to, I'd like to strip everything down to the 64-bit fixed-point implementation (with no assembly). 32 bits would be even better but I want to get going :)

A clean standalone libsbc would allow me to take a whack at gstreamer integration. I will come back to alsa-lib work once I have the feeling again that I'm making some sort of progress on all the other to-do items.

fwiw, gstreamer looks very well engineered. The sbc codec would be one plugin, A2DP would be another, and they even have a place for controller plugins like AVRCP.

Brad

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ha!

i just fixed those logical shifts to be arithmetic (by casting the value to signed first) and after tweaking some of the shifting amounts, encoding now works in 32 bits... on to the decoder... :)

brad

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