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/acm2/acm_receiver.cc

@@ -34,7 +34,9 @@ namespace acm2 {
 
 AcmReceiver::AcmReceiver(const AudioCodingModule::Config& config)
     : last_audio_buffer_(new int16_t[AudioFrame::kMaxDataSizeSamples]),
-      neteq_(NetEq::Create(config.neteq_config, config.decoder_factory)),
+      neteq_(NetEq::Create(config.neteq_config,
+                           config.clock,
+                           config.decoder_factory)),
       clock_(config.clock),
       resampled_last_output_frame_(true) {
   RTC_DCHECK(clock_);