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webrtc/audio
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Niels Möller 5304a32a94 Delete StreamStatistician::IsRetransmitOfOldPacket 
Return value always passed as the |retransmitted| argument to
ReceiveStatistics::IncomingPacket. The implementation of this method,
StreamStatisticianImpl::IncomingPacket, can call its own
IsRetransmitOfOldPacket, which is demoted to a private method.

Bug: webrtc:7135
Change-Id: I904db676738689c7a1db4caa588f70e64e3c357d
Reviewed-on: https://webrtc-review.googlesource.com/95649
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24494}
2018-08-30 11:00:13 +00:00
..
test Mark DirectTransport subclasses ctors as deprecated and switch from them 2018-08-20 12:05:05 +00:00
utility Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
audio_level.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_level.h Delete AudioMonitor and related code. 2018-01-30 09:48:29 +00:00
audio_receive_stream.cc Refactor voe::Channel to not use RtpReceiver. 2018-08-16 10:18:20 +00:00
audio_receive_stream.h Replace rtc::Optional with absl::optional in audio, call and video 2018-06-15 12:09:49 +00:00
audio_receive_stream_unittest.cc Refactor voe::Channel to not use RtpReceiver. 2018-08-16 10:18:20 +00:00
audio_send_stream.cc Delete class voe::RtcEventLogProxy 2018-08-15 09:59:15 +00:00
audio_send_stream.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_send_stream_tests.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_send_stream_unittest.cc Makes treatment of received reports of packets lost signed. 2018-08-15 14:27:23 +00:00
audio_state.cc Remove clang:find_bad_constructs suppression from call:call. 2018-08-29 11:57:00 +00:00
audio_state.h Remove clang:find_bad_constructs suppression from call:call. 2018-08-29 11:57:00 +00:00
audio_state_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_transport_impl.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_transport_impl.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
BUILD.gn Mark DirectTransport subclasses ctors as deprecated and switch from them 2018-08-20 12:05:05 +00:00
channel.cc Delete StreamStatistician::IsRetransmitOfOldPacket 2018-08-30 11:00:13 +00:00
channel.h Delete StreamStatistician::IsRetransmitOfOldPacket 2018-08-30 11:00:13 +00:00
channel_proxy.cc Refactor voe::Channel to not use RtpReceiver. 2018-08-16 10:18:20 +00:00
channel_proxy.h Refactor voe::Channel to not use RtpReceiver. 2018-08-16 10:18:20 +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 Refactor voe::Channel to not use RtpReceiver. 2018-08-16 10:18:20 +00:00
null_audio_poller.cc Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API (II) 2017-11-01 11:04:26 +00:00
null_audio_poller.h Remove clang:find_bad_constructs suppression from call:call. 2018-08-29 11:57:00 +00:00
OWNERS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
remix_resample.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
remix_resample.h Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +00:00
remix_resample_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
time_interval.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
time_interval.h Replace rtc::Optional with absl::optional in audio, call and video 2018-06-15 12:09:49 +00:00
time_interval_unittest.cc Replacing rtc::TimeDelta with webrtc::TimeDelta. 2018-05-08 13:22:53 +00:00
transport_feedback_packet_loss_tracker.cc Remove clang:find_bad_constructs suppression from call:call. 2018-08-29 11:57:00 +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
transport_feedback_packet_loss_tracker_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00