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webrtc_m130/audio
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Oskar Sundbom 56ef305b67 Move event logging of config into AudioSendStream. 
It was previously logged in Call, but streams are not always created
with the full configuration, which caused header extensions to be
missing from the log.

Bug: webrtc:9885
Change-Id: I86c0424004c4629ebab0f6b155b83fb90e15b131
Reviewed-on: https://webrtc-review.googlesource.com/c/108601
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25483}
2018-11-02 12:04:24 +00:00
..
test
Use MediaTransportInterface, for audio streams.
2018-10-26 11:40:57 +00:00
utility
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
audio_level.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
audio_level.h
Delete AudioMonitor and related code.
2018-01-30 09:48:29 +00:00
audio_receive_stream_unittest.cc
Use MediaTransportInterface, for audio streams.
2018-10-26 11:40:57 +00:00
audio_receive_stream.cc
Use MediaTransportInterface, for audio streams.
2018-10-26 11:40:57 +00:00
audio_receive_stream.h
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
audio_send_stream_tests.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
audio_send_stream_unittest.cc
Propagate SDP negotiation of extmap-allow-mixed to RtpHeaderExtensionMap
2018-10-30 08:06:49 +00:00
audio_send_stream.cc
Move event logging of config into AudioSendStream.
2018-11-02 12:04:24 +00:00
audio_send_stream.h
Use MediaTransportInterface, for audio streams.
2018-10-26 11:40:57 +00:00
audio_state_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
audio_state.cc
Remove clang:find_bad_constructs suppression from call:call.
2018-08-29 11:57:00 +00:00
audio_state.h
Remove clang:find_bad_constructs suppression from call:call.
2018-08-29 11:57:00 +00:00
audio_transport_impl.cc
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
audio_transport_impl.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
BUILD.gn
Move event logging of config into AudioSendStream.
2018-11-02 12:04:24 +00:00
channel_receive_proxy.cc
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
channel_receive_proxy.h
Set ChannelReceive transport at construction time.
2018-10-05 10:56:40 +00:00
channel_receive.cc
Use MediaTransportInterface, for audio streams.
2018-10-26 11:40:57 +00:00
channel_receive.h
Use MediaTransportInterface, for audio streams.
2018-10-26 11:40:57 +00:00
channel_send_proxy.cc
Propagate SDP negotiation of extmap-allow-mixed to RtpHeaderExtensionMap
2018-10-30 08:06:49 +00:00
channel_send_proxy.h
Propagate SDP negotiation of extmap-allow-mixed to RtpHeaderExtensionMap
2018-10-30 08:06:49 +00:00
channel_send.cc
Revert "Use only first payload timestamp for RTCP SR generation for audio"
2018-10-30 15:47:59 +00:00
channel_send.h
Propagate SDP negotiation of extmap-allow-mixed to RtpHeaderExtensionMap
2018-10-30 08:06:49 +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
Propagate SDP negotiation of extmap-allow-mixed to RtpHeaderExtensionMap
2018-10-30 08:06:49 +00:00
null_audio_poller.cc
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
null_audio_poller.h
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
OWNERS
Moving src/webrtc into src/.
2017-09-15 04:25:06 +00:00
remix_resample_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
remix_resample.cc
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
remix_resample.h
Remove dependencies on modules:module_api from AudioProcessing.
2018-04-12 22:05:27 +00:00
time_interval_unittest.cc
Replacing rtc::TimeDelta with webrtc::TimeDelta.
2018-05-08 13:22:53 +00:00
time_interval.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
time_interval.h
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
transport_feedback_packet_loss_tracker_unittest.cc
Cleanup modules_common_types
2018-09-18 08:08:33 +00:00
transport_feedback_packet_loss_tracker.cc
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
transport_feedback_packet_loss_tracker.h
Remove clang:find_bad_constructs suppression from call:call.
2018-08-29 11:57:00 +00:00