Return audio stats regarless if we have a codec.
Bug: b/331602608 Change-Id: I2d12a3ed83645fe1e7cbd8950fd86d5ba2d7c94d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361743 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42964}
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@ -257,13 +257,15 @@ webrtc::AudioReceiveStreamInterface::Stats AudioReceiveStreamImpl::GetStats(
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webrtc::CallReceiveStatistics call_stats =
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channel_receive_->GetRTCPStatistics();
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// TODO(solenberg): Don't return here if we can't get the codec - return the
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// stats we *can* get.
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auto receive_codec = channel_receive_->GetReceiveCodec();
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if (!receive_codec) {
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return stats;
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if (receive_codec) {
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stats.codec_name = receive_codec->second.name;
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stats.codec_payload_type = receive_codec->first;
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int clockrate_khz = receive_codec->second.clockrate_hz / 1000;
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if (clockrate_khz > 0) {
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stats.jitter_ms = call_stats.jitterSamples / clockrate_khz;
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}
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}
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stats.payload_bytes_received = call_stats.payload_bytes_received;
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stats.header_and_padding_bytes_received =
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call_stats.header_and_padding_bytes_received;
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@ -272,12 +274,6 @@ webrtc::AudioReceiveStreamInterface::Stats AudioReceiveStreamImpl::GetStats(
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stats.nacks_sent = call_stats.nacks_sent;
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stats.capture_start_ntp_time_ms = call_stats.capture_start_ntp_time_ms_;
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stats.last_packet_received = call_stats.last_packet_received;
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stats.codec_name = receive_codec->second.name;
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stats.codec_payload_type = receive_codec->first;
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int clockrate_khz = receive_codec->second.clockrate_hz / 1000;
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if (clockrate_khz > 0) {
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stats.jitter_ms = call_stats.jitterSamples / clockrate_khz;
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}
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stats.delay_estimate_ms = channel_receive_->GetDelayEstimate();
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stats.audio_level = channel_receive_->GetSpeechOutputLevelFullRange();
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stats.total_output_energy = channel_receive_->GetTotalOutputEnergy();
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