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Unified Diff: webrtc/api/call/audio_send_stream.cc

Issue 2503383002: Expose RtpCodecParameters to VoiceMediaInfo stats. (Closed)
Patch Set: Addressed comments, using int Created 4 years, 1 month ago
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Index: webrtc/api/call/audio_send_stream.cc
diff --git a/webrtc/api/call/audio_send_stream.cc b/webrtc/api/call/audio_send_stream.cc
index 3ce35e7a4b865b6019a1f638ddc2d54e2ea1f0c9..b6190073c188746753c0a4730c39af0d70eb1392 100644
--- a/webrtc/api/call/audio_send_stream.cc
+++ b/webrtc/api/call/audio_send_stream.cc
@@ -30,6 +30,7 @@ std::string ToString(const webrtc::CodecInst& codec_inst) {
namespace webrtc {
AudioSendStream::Stats::Stats() = default;
+AudioSendStream::Stats::~Stats() = default;
AudioSendStream::Config::Config(Transport* send_transport)
: send_transport(send_transport) {}
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