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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: rebase again 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 9b1e2fe5d3eaf40216c046d1ee48bb1e6de03e54..9aa2fc004c2295dc2c2c6bdb97996f6d0d6848c0 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();
@@ -74,6 +75,10 @@ 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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