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Unified Diff: webrtc/audio/audio_receive_stream.cc

Issue 2111813002: Revert of Move RtcEventLog object from inside VoiceEngine to Call. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 6 months ago
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Index: webrtc/audio/audio_receive_stream.cc
diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc
index 0a2bc2b4c6a432b899159ea2b9f7396010ac2a26..a684003326426fb792a896816b4cb2550ad63d89 100644
--- a/webrtc/audio/audio_receive_stream.cc
+++ b/webrtc/audio/audio_receive_stream.cc
@@ -81,8 +81,7 @@
AudioReceiveStream::AudioReceiveStream(
CongestionController* congestion_controller,
const webrtc::AudioReceiveStream::Config& config,
- const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
- webrtc::RtcEventLog* event_log)
+ const rtc::scoped_refptr<webrtc::AudioState>& audio_state)
: config_(config),
audio_state_(audio_state),
rtp_header_parser_(RtpHeaderParser::Create()) {
@@ -94,7 +93,6 @@
VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id);
- channel_proxy_->SetRtcEventLog(event_log);
channel_proxy_->SetLocalSSRC(config.rtp.local_ssrc);
// TODO(solenberg): Config NACK history window (which is a packet count),
// using the actual packet size for the configured codec.
@@ -146,7 +144,6 @@
LOG(LS_INFO) << "~AudioReceiveStream: " << config_.ToString();
channel_proxy_->DeRegisterExternalTransport();
channel_proxy_->ResetCongestionControlObjects();
- channel_proxy_->SetRtcEventLog(nullptr);
if (remote_bitrate_estimator_) {
remote_bitrate_estimator_->RemoveStream(config_.rtp.remote_ssrc);
}
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