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}
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@ -23,6 +23,7 @@
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#include "modules/audio_coding/neteq/include/neteq.h"
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#include "modules/audio_coding/neteq/tools/audio_sink.h"
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#include "modules/audio_coding/neteq/tools/neteq_input.h"
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#include "system_wrappers/include/clock.h"
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namespace webrtc {
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namespace test {
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@ -106,6 +107,7 @@ class NetEqTest : public NetEqSimulator {
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private:
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void RegisterDecoders(const DecoderMap& codecs);
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SimulatedClock clock_;
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absl::optional<Action> next_action_;
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absl::optional<int> last_packet_time_ms_;
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std::unique_ptr<NetEq> neteq_;
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