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Unified Diff: webrtc/audio/audio_send_stream.cc

Issue 1955363003: Configure VoE NACK through AudioSendStream::Config. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: misc Created 4 years, 6 months ago
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Index: webrtc/audio/audio_send_stream.cc
diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc
index e9659c67e8c48ecd3bf09369d27562322b1e5be3..ae5d8c1ed5f04db81c3d8a48c123de4e5f6b3454 100644
--- a/webrtc/audio/audio_send_stream.cc
+++ b/webrtc/audio/audio_send_stream.cc
@@ -39,6 +39,7 @@ std::string AudioSendStream::Config::Rtp::ToString() const {
}
}
ss << ']';
+ ss << ", nack: " << nack.ToString();
ss << ", c_name: " << c_name;
ss << '}';
return ss.str();
@@ -75,6 +76,10 @@ AudioSendStream::AudioSendStream(
channel_proxy_->SetRTCPStatus(true);
channel_proxy_->SetLocalSSRC(config.rtp.ssrc);
channel_proxy_->SetRTCP_CNAME(config.rtp.c_name);
+ // TODO(solenberg): Config NACK history window (which is a packet count),
+ // using the actual packet size for the configured codec.
+ channel_proxy_->SetNACKStatus(config_.rtp.nack.rtp_history_ms != 0,
+ config_.rtp.nack.rtp_history_ms / 20);
channel_proxy_->RegisterExternalTransport(config.send_transport);
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