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/include/neteq.h

@@ -31,6 +31,7 @@ namespace webrtc {
 // Forward declarations.
 class AudioFrame;
 class AudioDecoderFactory;
+class Clock;
 
 struct NetEqNetworkStatistics {
   uint16_t current_buffer_size_ms;    // Current jitter buffer size in ms.
@@ -149,6 +150,7 @@ class NetEq {
   // method.
   static NetEq* Create(
       const NetEq::Config& config,
+      Clock* clock,
       const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory);
 
   virtual ~NetEq() {}