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}

modules/audio_coding/neteq/packet.h

@@ -16,6 +16,7 @@
 #include <memory>
 
 #include "api/audio_codecs/audio_decoder.h"
+#include "api/rtp_packet_info.h"
 #include "modules/audio_coding/neteq/tick_timer.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/checks.h"
@@ -72,6 +73,7 @@ struct Packet {
   // Datagram excluding RTP header and header extension.
   rtc::Buffer payload;
   Priority priority;
+  RtpPacketInfo packet_info;
   std::unique_ptr<TickTimer::Stopwatch> waiting_time;
   std::unique_ptr<AudioDecoder::EncodedAudioFrame> frame;