webrtc/audio/audio_send_stream.cc - Issue 2503383002: Expose RtpCodecParameters to VoiceMediaInfo stats.

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Unified Diff: webrtc/audio/audio_send_stream.cc

Issue 2503383002: Expose RtpCodecParameters to VoiceMediaInfo stats. (Closed)

Patch Set: Using uint32_t in RtpCodecParametersMap instead of int Created 4 years, 1 month 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 a96eaf83f83a2928226651630e24a1b596c36587..def0ae6007f4b9ab6e9d688dc11219c699972e21 100644

--- a/webrtc/audio/audio_send_stream.cc

+++ b/webrtc/audio/audio_send_stream.cc

@@ -169,8 +169,9 @@ webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const {

webrtc::CodecInst codec_inst = {0};

if (codec->GetSendCodec(config_.voe_channel_id, codec_inst) != -1) {

- RTC_DCHECK_NE(codec_inst.pltype, -1);

+ RTC_DCHECK_GE(codec_inst.pltype, 0);

stats.codec_name = codec_inst.plname;

+ stats.codec_payload_type = rtc::Optional<uint32_t>(codec_inst.pltype);

// Get data from the last remote RTCP report.

for (const auto& block : channel_proxy_->GetRemoteRTCPReportBlocks()) {