Commit Graph

1028 Commits

Author SHA1 Message Date
Martin Pulec
3e0d06f259 Audio: fixed segv when rtp_init fails 2015-09-28 11:55:37 +02:00
Martin Pulec
2c28c13721 Control: allow SSRC change of the RTP stream 2015-09-26 12:29:57 +02:00
Martin Pulec
6ce04bf63c Change RTP dest by replacing session
This solves problem when migrating between IPv4 to IPv6 socket/address.
2015-09-16 17:41:46 +02:00
Martin Pulec
c57b75d2b3 Audio_frame2: fix 2015-09-02 17:01:33 +02:00
Martin Pulec
a390d0ab89 Audio codec: use new module API 2015-09-02 17:01:07 +02:00
Martin Pulec
637e6b051f Fixed typo 2015-09-01 13:13:03 +02:00
Martin Pulec
072ea8cbb5 Register modules with macro 2015-08-31 11:11:05 +02:00
Martin Pulec
d6997d830d Audio: use UDP recv buffer 256 kB
Seems to be sufficient for 32 channels while small enough. Moreover, we
have deterministic buffer size.
2015-08-31 11:11:05 +02:00
Martin Pulec
95c51e0507 audio_frame2: optimize allocation a bit
+ preallocate some big frame in audio decoder
2015-08-31 11:11:04 +02:00
Martin Pulec
54d45ba489 Audio: move audio types in separate header/src 2015-08-31 11:11:04 +02:00
Martin Pulec
7967ea635f ALSA playback: provide supported audio format 2015-08-31 11:11:04 +02:00
Martin Pulec
343089e7b8 Audio decoder: return if we cannot find out output format 2015-08-31 11:11:04 +02:00
Martin Pulec
100683c983 Warning about some potentialy broken modules 2015-08-31 11:11:04 +02:00
Martin Pulec
a9f6386c13 Get audio output format from display 2015-08-31 11:11:04 +02:00
Martin Pulec
75d5f162a0 Audio: start after full initialization 2015-08-31 11:11:04 +02:00
Martin Pulec
445812f0b7 JACK: remove resampling and bps change
These stuff should be now done by decoder.
2015-08-31 11:11:04 +02:00
Martin Pulec
289f765858 Audio dec: obtain supported fmts from playback dev 2015-08-31 11:11:04 +02:00
Martin Pulec
ba4071f3b8 Audio decoder: use audio_frame2::resample()
Use audio_frame2::resample instead some obsolete ad hoc code.
2015-08-31 11:11:03 +02:00
Martin Pulec
9672fcc20c audio_codec_decompress: return obj instead of pointer 2015-08-31 11:11:03 +02:00
Martin Pulec
7abec1d5e5 Compile fixes (MSW and OS X) 2015-08-28 14:31:03 +02:00
Martin Pulec
2cc6aab0e2 Added possibility to send message synchronously
+ in capabilities list, given bitrate is computed according to the
  detected capture format (provided that '-t' argument is given)
2015-08-25 17:05:23 +02:00
Martin Pulec
b92d842dcd Audio send/decoder: compute dBFS in a separate thr
It is a bit time comsuming so it may break frames timing.
2015-08-19 09:53:10 +02:00
Martin Pulec
2e9b0c22a8 ALSA play: fixed a leak 2015-08-19 09:53:10 +02:00
Martin Pulec
7de85ac2d9 ALSA cap: Init num of channels after rate and fmt
With Hammerfall DSP, snd_pcm_hw_params_set_channels_first() and then
setting exact rate and format generated incorrect config.
2015-08-19 09:53:10 +02:00
Martin Pulec
df8e0c2f01 Audio play mods: uses now new module loading API 2015-08-19 09:53:10 +02:00
Martin Pulec
999846a543 Audio cap mods: uses now new module loading API 2015-08-19 09:53:10 +02:00
Martin Pulec
d69d1b622d ALSA: minor improvents
Fixed support for non-interleaved access (currently mono only).
2015-08-19 09:53:10 +02:00
Martin Pulec
e833784756 Let audio capture driver decide about sample rate
Do not enforce 48 kHz anymore, modules can pick also another one which
fits better (eg. device cannot do such a sample rate).
2015-08-19 09:53:10 +02:00
Martin Pulec
8e6bbfd7f7 ALSA cap.: select bps automatically 2015-08-19 09:53:10 +02:00
Martin Pulec
d9969bcf20 Audio capture drivers: use capture format from cmd
Note: proposed audio format is not mandatory so drivers may use
different format if appropriate.
2015-08-19 09:53:10 +02:00
Martin Pulec
0846a6a163 Audio testcard: rewritten a bit 2015-08-19 09:53:10 +02:00
Martin Pulec
0fc4ba2942 Audio codec: choose sample rate automatically 2015-08-19 09:53:10 +02:00
Martin Pulec
899ca0b56f Audio sender: don't auto resample to 48000
+ some rework - audio_frame2 has now methods for changing bps and
resampling
2015-08-19 09:53:10 +02:00
Martin Pulec
7ca27e5d7d Libavcodec audio: support for further sample fmts
+ fixed AAC (bitrate == 0 caused malfunctioning)
2015-07-20 16:03:27 +02:00
Martin Pulec
477216b900 Some messages now use logger. 2015-07-10 16:42:27 +02:00
Martin Pulec
5165648e4c Added keyboard control (OS X and Linux)
Allows changing volume and mute audio from UG terminal right now.
2015-07-10 16:42:26 +02:00
Martin Pulec
c1a6855dd1 Audio: init sent/recv modules only if used 2015-07-10 16:42:26 +02:00
Martin Pulec
ce5edcc935 Modified some periodical stat messages. 2015-07-10 16:42:26 +02:00
Martin Pulec
e47737c4df Added dummy audio playback 2015-06-30 10:45:41 +02:00
Martin Pulec
80c6054719 Poratudio playback: missing newline 2015-06-29 15:07:42 +02:00
Martin Pulec
90dd0153b5 CoreAudio: handle underflows differently
Do not try to stop CoreAudio from callback (potentially unsafe). Instead,
silence underrun warning after some period.

+ silenced some deprecation compile warning (OS X API change) by using newer
functions
2015-06-29 12:06:11 +02:00
Martin Pulec
fc6dc4ba5e CoreAudio playback: do not write samples when overrun 2015-06-26 15:10:55 +02:00
Martin Pulec
719685c266 Portaudio playback: clear if there are no data
+ be more verbose about underruns
2015-06-26 15:01:47 +02:00
Martin Pulec
10ff4f22e1 CoreAudio playback: pause after 2 secs of inactivity 2015-06-25 17:22:34 +02:00
Martin Pulec
3c0ae774af ALSA cap.: support for 8-bit audio 2015-06-24 17:10:44 +02:00
Martin Pulec
bbf609ab90 ALSA capture: allow configuration of various opts 2015-06-24 16:29:31 +02:00
Martin Pulec
1a4c53e76c Portaudio: ring-buffer -> 1 sec
+ warn when there is more than half of the buffer full
2015-06-17 16:30:09 +02:00
Martin Pulec
84861d683c Pbuf: use high_resolution_timer instead of timespec 2015-06-17 14:27:49 +02:00
Martin Pulec
ae354bd67a Flush audio UDP buffers after reinit 2015-06-17 11:49:08 +02:00
Martin Pulec
22bd5bd80d Respect explicit network port mapping from cmdline 2015-06-11 15:50:59 +02:00