mollyim/webrtc
1
0
Fork 0
mirror of https://github.com/mollyim/webrtc.git synced 2025-05-17 23:57:59 +01:00
Code Issues Projects Releases Packages Wiki Activity Actions
webrtc/audio
History
Tommi c848268ab1 Use SequenceChecker(SequenceChecker::kDetached) in a few places. 
This CL is partly a test to see if there's an impact on binary size:
- Not a big difference for binaries (decrease): -776b to -4Kb
- For libraries (libwebrtc.a) it actually increases the size: +40Kb

Secondarily this CL is basically to introduce this pattern to the
code base. In terms of LOC, this makes things slightly more compact.

From:

  class Foo {
   public:
     Foo() {
       checker_.Detach();
     }
   private:
    SequenceChecker checker_;
  };

To:

  class Foo {
   public:
     Foo() = default;
   private:
    SequenceChecker checker_{SequenceChecker::kDetached};
  };

Bug: none
Change-Id: I59fc34ccea10847e13455a349851ce9a0af458e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299020
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39664}
2023-03-24 07:44:18 +00:00
..
test De-flake NonSenderRttStats and make it faster to run on average. 2023-03-10 13:25:34 +00:00
utility Remove dependency on rtc_base_approved from most targets 2022-04-25 12:15:30 +00:00
voip Implement support for Chrome task origin tracing. #3.5/4 2023-03-01 11:11:37 +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 Use SequenceChecker(SequenceChecker::kDetached) in a few places. 2023-03-24 07:44:18 +00:00
audio_receive_stream.h Use SequenceChecker(SequenceChecker::kDetached) in a few places. 2023-03-24 07:44:18 +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 Use SequenceChecker(SequenceChecker::kDetached) in a few places. 2023-03-24 07:44:18 +00:00
audio_state.h Use SequenceChecker(SequenceChecker::kDetached) in a few places. 2023-03-24 07:44:18 +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_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.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_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 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_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