3e8ef940fe
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}
NetEQ RTP Play tool
Testing of the command line arguments
The command line tool neteq_rtpplay can be tested by running neteq_rtpplay_test.sh, which is not use on try bots, but it can be used before submitting any CLs that may break the behavior of the command line arguments of neteq_rtpplay.
Run neteq_rtpplay_test.sh as follows from the src/ folder:
src$ ./modules/audio_coding/neteq/tools/neteq_rtpplay_test.sh \
out/Default/neteq_rtpplay \
resources/audio_coding/neteq_opus.rtp \
resources/short_mixed_mono_48.pcm
You can replace the RTP and PCM files with any other compatible files.
If you get an error using the files indicated above, try running gclient sync.
Requirements: awk and md5sum.