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platform-external-webrtc/audio
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Sebastian Jansson 0a6510ddf9 Removes rtp_transport checks in AudioSendStream 
There's already a DCHECK at construction time ensuring that it's set.

Bug: webrtC:9883
Change-Id: I9f41b77273bb859626546ab3534d483d9172ea5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155581
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29393}
2019-10-07 12:58:55 +00:00
..
test
Propagate task queue to create test::DirectTransport by TaskQueueBase interface
2019-09-30 03:23:07 +00:00
utility
Format almost everything.
2019-07-08 13:45:15 +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
Remove clock drift metric from NetEq.
2019-09-02 13:50:55 +00:00
audio_receive_stream.cc
Delete audio methods SignalNetworkState
2019-08-30 09:27:30 +00:00
audio_receive_stream.h
Delete audio methods SignalNetworkState
2019-08-30 09:27:30 +00:00
audio_send_stream_tests.cc
Reland "Delete test/constants.h"
2019-02-19 08:51:20 +00:00
audio_send_stream_unittest.cc
Adds trial to calculate audio overhead based on available data.
2019-10-02 13:42:15 +00:00
audio_send_stream.cc
Removes rtp_transport checks in AudioSendStream
2019-10-07 12:58:55 +00:00
audio_send_stream.h
Replaces static modifier functions in AudioSendStream.
2019-10-07 11:33:39 +00:00
audio_state_unittest.cc
Format almost everything.
2019-07-08 13:45:15 +00:00
audio_state.cc
Use std::make_unique instead of absl::make_unique.
2019-09-17 15:47:29 +00:00
audio_state.h
[getStats] Implement "media-source" audio levels, fixing Chrome bug.
2019-07-04 08:13:45 +00:00
audio_transport_impl.cc
[getStats] Implement "media-source" audio levels, fixing Chrome bug.
2019-07-04 08:13:45 +00:00
audio_transport_impl.h
[getStats] Implement "media-source" audio levels, fixing Chrome bug.
2019-07-04 08:13:45 +00:00
BUILD.gn
Removing AudioAllocationSettings moving functionality to AudioSendStream.
2019-10-03 10:52:16 +00:00
channel_receive.cc
Use std::make_unique instead of absl::make_unique.
2019-09-17 15:47:29 +00:00
channel_receive.h
Move MediaTransportInterface out of the libjingle_peerconnection_api target
2019-09-13 10:49:56 +00:00
channel_send.cc
Adds support for passing a vector of packets to the paced sender.
2019-10-04 08:56:11 +00:00
channel_send.h
Move MediaTransportInterface out of the libjingle_peerconnection_api target
2019-09-13 10:49:56 +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
Set local ssrc at construction of Rtp module
2019-08-21 12:44:09 +00:00
null_audio_poller.cc
Deprecating ThreadChecker specific interface.
2019-04-08 16:58:07 +00:00
null_audio_poller.h
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
2019-01-11 17:11:39 +00:00
OWNERS
…
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
transport_feedback_packet_loss_tracker_unittest.cc
Format almost everything.
2019-07-08 13:45:15 +00:00
transport_feedback_packet_loss_tracker.cc
Format almost everything.
2019-07-08 13:45:15 +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