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webrtc/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 Remove dependency on rtc_base_approved from most targets 2022-04-25 12:15:30 +00:00
voip Break apart AudioCodingModule and AcmReceiver 2023-02-01 16:09:26 +00:00
audio_level.cc Migrate audio/ to use webrtc::Mutex 2020-07-06 14:21:38 +00:00
audio_level.h Migrate audio/ to use webrtc::Mutex 2020-07-06 14:21:38 +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_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_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_send_stream_tests.cc CallTest: migrate timeouts to TimeDelta. 2022-08-16 12:06:54 +00:00
audio_send_stream_unittest.cc pc: Add asynchronous RtpSender::SetParameters() call 2022-11-15 15:31:40 +00:00
audio_state.cc Rewrite AudioState null poller to use TaskQueueBase interface 2022-08-16 13:16:24 +00:00
audio_state.h Rewrite AudioState null poller to use TaskQueueBase interface 2022-08-16 13:16:24 +00:00
audio_state_unittest.cc Migrate remaining webrtc usage of TaskQueueBase to absl::AnyInvocable 2022-07-20 08:15:08 +00:00
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.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_receive_frame_transformer_delegate.cc Add GetContributionSources to TransformableIncomingAudioFrame 2022-10-11 12:52:21 +00:00
channel_receive_frame_transformer_delegate.h Use backticks not vertical bars to denote variables in comments for /audio 2021-07-27 15:36:40 +00:00
channel_receive_frame_transformer_delegate_unittest.cc Move rtc::make_ref_counted to api/ 2022-06-15 09:47:38 +00:00
channel_receive_unittest.cc Create unit test for the population of capture_start_ntp_time 2023-02-06 14:00:39 +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
channel_send_frame_transformer_delegate.cc Update audio code to not use implicit T* --> scoped_refptr<T> conversion 2022-01-13 15:49:49 +00:00
channel_send_frame_transformer_delegate.h Use backticks not vertical bars to denote variables in comments for /audio 2021-07-27 15:36:40 +00:00
channel_send_frame_transformer_delegate_unittest.cc Move rtc::make_ref_counted to api/ 2022-06-15 09:47:38 +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
conversion.h Make header files self contained. 2022-10-08 08:38:36 +00:00
DEPS pc: Add asynchronous RtpSender::SetParameters() call 2022-11-15 15:31:40 +00:00
mock_voe_channel_proxy.h Implement RTCOutboundRtpStreamStats.targetBitrate for audio. 2021-11-12 09:24:34 +00:00
OWNERS Add alessiob@webrtc.org in audio/OWNERS 2022-09-09 07:33:11 +00:00
remix_resample.cc Reland "Rename FATAL() into RTC_FATAL()." 2020-11-18 20:49:08 +00:00
remix_resample.h Use backticks not vertical bars to denote variables in comments for /audio 2021-07-27 15:36:40 +00:00
remix_resample_unittest.cc Clarify and extend test support for certain sample rates in audio processing 2022-08-03 14:26:36 +00:00