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webrtc_m130/audio
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Philipp Hancke b3e5969658 stats: use uint64_t for RTCSentRtpStreamStats.packetsSent 
spec update from https://github.com/w3c/webrtc-stats/pull/744

BUG=webrtc:14989

Change-Id: I9d0adcf951501bc281054c77bb6bc03e47192523
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295505
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#39575}
2023-03-16 06:46:19 +00:00
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test
De-flake NonSenderRttStats and make it faster to run on average.
2023-03-10 13:25:34 +00:00
utility
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voip
Implement support for Chrome task origin tracing. #3.5/4
2023-03-01 11:11:37 +00:00
audio_level.cc
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audio_level.h
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audio_receive_stream_unittest.cc
Remove rtp header extension from config of Call audio and video receivers
2023-01-31 11:58:43 +00:00
audio_receive_stream.cc
Remove rtp header extension from config of Call audio and video receivers
2023-01-31 11:58:43 +00:00
audio_receive_stream.h
Remove rtp header extension from config of Call audio and video receivers
2023-01-31 11:58:43 +00:00
audio_send_stream_tests.cc
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audio_send_stream_unittest.cc
pc: Add asynchronous RtpSender::SetParameters() call
2022-11-15 15:31:40 +00:00
audio_send_stream.cc
Remove default enabled field trial WebRTC-SendSideBwe-WithOverhead
2022-11-30 20:19:36 +00:00
audio_send_stream.h
Remove default enabled field trial WebRTC-SendSideBwe-WithOverhead
2022-11-30 20:19:36 +00:00
audio_state_unittest.cc
Implement support for Chrome task origin tracing. #3.5/4
2023-03-01 11:11:37 +00:00
audio_state.cc
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audio_state.h
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audio_transport_impl.cc
Make capture timestamp optional in ADM.
2023-01-23 17:29:06 +00:00
audio_transport_impl.h
Make capture timestamp optional in ADM.
2023-01-23 17:29:06 +00:00
BUILD.gn
De-flake NonSenderRttStats and make it faster to run on average.
2023-03-10 13:25:34 +00:00
channel_receive_frame_transformer_delegate_unittest.cc
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channel_receive_frame_transformer_delegate.cc
Add GetContributionSources to TransformableIncomingAudioFrame
2022-10-11 12:52:21 +00:00
channel_receive_frame_transformer_delegate.h
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channel_receive_unittest.cc
Create unit test for the population of capture_start_ntp_time
2023-02-06 14:00:39 +00:00
channel_receive.cc
Delete RtpRtcpInterface::RemoteNtp as redundant to GetSenderReportStats
2023-02-28 13:55:27 +00:00
channel_receive.h
stats: use uint64_t for RTCSentRtpStreamStats.packetsSent
2023-03-16 06:46:19 +00:00
channel_send_frame_transformer_delegate_unittest.cc
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channel_send_frame_transformer_delegate.cc
Add a clone method to the audio frame transformer API.
2023-03-06 08:22:25 +00:00
channel_send_frame_transformer_delegate.h
Add a clone method to the audio frame transformer API.
2023-03-06 08:22:25 +00:00
channel_send_unittest.cc
Update RTP timestamp based on capture timestamp when audio send stream is resumed.
2023-01-27 15:46:32 +00:00
channel_send.cc
Break apart AudioCodingModule and AcmReceiver
2023-02-01 16:09:26 +00:00
channel_send.h
[Stats] Expose totalPacketSendDelay for audio as well.
2022-10-27 10:33:16 +00:00
conversion.h
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DEPS
pc: Add asynchronous RtpSender::SetParameters() call
2022-11-15 15:31:40 +00:00
mock_voe_channel_proxy.h
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OWNERS
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remix_resample_unittest.cc
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remix_resample.cc
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remix_resample.h
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