Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker.
This change adds the plumbing of RtpPacketInfo from ChannelReceive::OnRtpPacket() to ChannelReceive::GetAudioFrameWithInfo() for audio. It is a step towards replacing the non-spec compliant ContributingSources that updates itself at packet-receive time, with the spec-compliant SourceTracker that will update itself at frame-delivery-to-track time.
Bug: webrtc:10668
Change-Id: I03385d6865bbc7bfbef7634f88de820a934f787a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139890
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28434}
diff --git a/api/audio/audio_frame.cc b/api/audio/audio_frame.cc
index 1e706b9..4c07aaf 100644
--- a/api/audio/audio_frame.cc
+++ b/api/audio/audio_frame.cc
@@ -39,6 +39,7 @@
speech_type_ = kUndefined;
vad_activity_ = kVadUnknown;
profile_timestamp_ms_ = 0;
+ packet_infos_ = RtpPacketInfos();
}
void AudioFrame::UpdateFrame(uint32_t timestamp,
@@ -72,6 +73,7 @@
timestamp_ = src.timestamp_;
elapsed_time_ms_ = src.elapsed_time_ms_;
ntp_time_ms_ = src.ntp_time_ms_;
+ packet_infos_ = src.packet_infos_;
muted_ = src.muted();
samples_per_channel_ = src.samples_per_channel_;
sample_rate_hz_ = src.sample_rate_hz_;