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webrtc_m130/audio
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Harald Alvestrand 95d12adf37 Create unit test for the population of capture_start_ntp_time 
This verifies that receiving two RTCP SR packets is enough to get
a defined capture start time stat.

Bug: webrtc:13931
Change-Id: Ib5f7c2954eab6500917f25c44f523d3aedae5e94
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291520
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39261}
2023-02-06 14:00:39 +00:00
..
test
Delete unused Audio Bwe integration test.
2023-01-26 09:31:44 +00:00
utility
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voip
Break apart AudioCodingModule and AcmReceiver
2023-02-01 16:09:26 +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
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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
Create unit test for the population of capture_start_ntp_time
2023-02-06 14:00:39 +00:00
channel_receive_frame_transformer_delegate_unittest.cc
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channel_receive_frame_transformer_delegate.cc
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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
Break apart AudioCodingModule and AcmReceiver
2023-02-01 16:09:26 +00:00
channel_receive.h
audio: make packets lost a signed integer
2022-11-01 11:46:49 +00:00
channel_send_frame_transformer_delegate_unittest.cc
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channel_send_frame_transformer_delegate.cc
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channel_send_frame_transformer_delegate.h
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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
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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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