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webrtc_m130/audio
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Jakob Ivarsson 01ab084f47 Add minimum overhead to configured priorty bitrate instead of maximum. 
This makes an assumption that if we have variable frame length then we
will increase payload bitrate up to priority bitrate before adapting the
frame length.

Bug: webrtc:11001
Change-Id: Iec51d5ccce053d55ccd30a9e4712765227e10852
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169852
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30713}
2020-03-06 15:44:21 +00:00
..
test
Use newer version of TimeDelta and TimeStamp factories in webrtc
2020-02-10 12:21:17 +00:00
utility
Reformat GN files.
2020-01-21 12:13:11 +00:00
audio_level.cc
Format almost everything.
2019-07-08 13:45:15 +00:00
audio_level.h
[getStats] Implement "media-source" audio levels, fixing Chrome bug.
2019-07-04 08:13:45 +00:00
audio_receive_stream_unittest.cc
Delete media transport integration.
2019-11-26 19:19:36 +00:00
audio_receive_stream.cc
Add periodic logging of sync delays.
2020-02-11 09:43:49 +00:00
audio_receive_stream.h
Add periodic logging of sync delays.
2020-02-11 09:43:49 +00:00
audio_send_stream_tests.cc
Migrate several call tests from legacy RtpHeaderParser to RtpPacket parsing.
2020-01-07 13:02:52 +00:00
audio_send_stream_unittest.cc
Make sure that the audio stream is allocated with the correct overhead.
2020-03-06 14:49:37 +00:00
audio_send_stream.cc
Add minimum overhead to configured priorty bitrate instead of maximum.
2020-03-06 15:44:21 +00:00
audio_send_stream.h
Ready to support of absolute capture timestamp header extension.
2020-03-06 13:16:29 +00:00
audio_state_unittest.cc
Format almost everything.
2019-07-08 13:45:15 +00:00
audio_state.cc
Revert "Inlines NullAudioPoller functionality into AudioState class."
2020-01-30 18:14:11 +00:00
audio_state.h
Revert "Inlines NullAudioPoller functionality into AudioState class."
2020-01-30 18:14:11 +00:00
audio_transport_impl.cc
Refactoring AudioSender api out of AudioSendStream for more abstraction to reuse AudioTransportImpl for voip api
2020-01-13 18:31:30 +00:00
audio_transport_impl.h
Refactoring AudioSender api out of AudioSendStream for more abstraction to reuse AudioTransportImpl for voip api
2020-01-13 18:31:30 +00:00
BUILD.gn
Refactoring mock_transport to be used separately
2020-02-21 17:02:52 +00:00
channel_receive.cc
Only update the current time of a played out frame if a new frame is played out.
2020-02-25 12:48:32 +00:00
channel_receive.h
Delete media transport integration.
2019-11-26 19:19:36 +00:00
channel_send.cc
Reland "Send absolute capture time through audio coding module."
2020-01-27 13:18:27 +00:00
channel_send.h
Cleanup: Using RtpRtcp directly from AudioSendStream
2020-01-15 19:01:50 +00:00
conversion.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
DEPS
Move remaining traces of VoiceEngine
2018-01-17 13:27:47 +00:00
mock_voe_channel_proxy.h
Cleanup: Using RtpRtcp directly from AudioSendStream
2020-01-15 19:01:50 +00:00
null_audio_poller.cc
Revert "Inlines NullAudioPoller functionality into AudioState class."
2020-01-30 18:14:11 +00:00
null_audio_poller.h
Revert "Inlines NullAudioPoller functionality into AudioState class."
2020-01-30 18:14:11 +00:00
OWNERS
Remove wildcard ownership for build files.
2020-02-19 14:05:46 +00:00
remix_resample_unittest.cc
Add RTC_ prefix to non-standard format specifier macro "PRIdNS"
2019-08-07 13:36:05 +00:00
remix_resample.cc
Reland "Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker.""
2019-07-24 16:47:13 +00:00
remix_resample.h
Remove dependencies on modules:module_api from AudioProcessing.
2018-04-12 22:05:27 +00:00