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Revert "Reland "Return audio stats regarless if we have a codec.""

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This reverts commit 4334cdfc5c0619a5f06125ea1f039cb123ccf21e.

Reason for revert: Breaks downstream project.

Original change's description:
> Reland "Return audio stats regarless if we have a codec."
>
> This is a reland of commit 7fff587a096c6ef40f5601f47ef50b221b3a4abf
>
> Original change's description:
> > 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}
>
> Bug: b/331602608
> Change-Id: I95c89e7059005bc8dd8569ef41bfe9e863b4082f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361762
> Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42969}

Bug: b/331602608
Change-Id: Ifbe332a8749d024f603b75b6e787551dd6762dd6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/363001
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43059}
This commit is contained in:
Tomas Lundqvist 2024-09-19 13:53:17 +00:00 committed by WebRTC LUCI CQ
parent f566dee902
commit 64e8e64e80
1 changed files with 11 additions and 7 deletions
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18
audio/audio_receive_stream.cc
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@ -257,15 +257,13 @@ webrtc::AudioReceiveStreamInterface::Stats AudioReceiveStreamImpl::GetStats(
webrtc::CallReceiveStatistics call_stats = webrtc::CallReceiveStatistics call_stats =
channel_receive_->GetRTCPStatistics(); channel_receive_->GetRTCPStatistics();
// TODO(solenberg): Don't return here if we can't get the codec - return the
// stats we *can* get.
auto receive_codec = channel_receive_->GetReceiveCodec(); auto receive_codec = channel_receive_->GetReceiveCodec();
if (receive_codec) { if (!receive_codec) {
stats.codec_name = receive_codec->second.name; return stats;
stats.codec_payload_type = receive_codec->first;
int clockrate_khz = receive_codec->second.clockrate_hz / 1000;
if (clockrate_khz > 0) {
stats.jitter_ms = call_stats.jitterSamples / clockrate_khz;
}
} }
stats.payload_bytes_received = call_stats.payload_bytes_received; stats.payload_bytes_received = call_stats.payload_bytes_received;
stats.header_and_padding_bytes_received = stats.header_and_padding_bytes_received =
call_stats.header_and_padding_bytes_received; call_stats.header_and_padding_bytes_received;
@ -274,6 +272,12 @@ webrtc::AudioReceiveStreamInterface::Stats AudioReceiveStreamImpl::GetStats(
stats.nacks_sent = call_stats.nacks_sent; stats.nacks_sent = call_stats.nacks_sent;
stats.capture_start_ntp_time_ms = call_stats.capture_start_ntp_time_ms_; stats.capture_start_ntp_time_ms = call_stats.capture_start_ntp_time_ms_;
stats.last_packet_received = call_stats.last_packet_received; stats.last_packet_received = call_stats.last_packet_received;
stats.codec_name = receive_codec->second.name;
stats.codec_payload_type = receive_codec->first;
int clockrate_khz = receive_codec->second.clockrate_hz / 1000;
if (clockrate_khz > 0) {
stats.jitter_ms = call_stats.jitterSamples / clockrate_khz;
}
stats.delay_estimate_ms = channel_receive_->GetDelayEstimate(); stats.delay_estimate_ms = channel_receive_->GetDelayEstimate();
stats.audio_level = channel_receive_->GetSpeechOutputLevelFullRange(); stats.audio_level = channel_receive_->GetSpeechOutputLevelFullRange();
stats.total_output_energy = channel_receive_->GetTotalOutputEnergy(); stats.total_output_energy = channel_receive_->GetTotalOutputEnergy();