Prepare the AudioSendStream to be hooked up to send-side BWE.

This CL contains three changes as a preparation for adding audio send streams
to the send-side BWE:
1. Audio packets are passed through the pacer with high priority. This
is needed to be able to set transport sequence numbers on the packets.
2. A feedback observer is passed to the audio stream's rtcp receiver so
that the BWE can get notified of any BWE feedback being received on the
audio feedback channel.
3. Support for the transport sequence number header extension is added
to audio send streams.

BUG=webrtc:5263,webrtc:5307
R=mflodman@webrtc.org, solenberg@webrtc.org

Review URL: https://codereview.webrtc.org/1479023002 .

Cr-Commit-Position: refs/heads/master@{#10909}

webrtc/modules/rtp_rtcp/source/rtp_sender.cc

@@ -469,7 +469,8 @@ int32_t RTPSender::CheckPayloadType(int8_t payload_type,
   std::map<int8_t, RtpUtility::Payload*>::iterator it =
       payload_type_map_.find(payload_type);
   if (it == payload_type_map_.end()) {
-    LOG(LS_WARNING) << "Payload type " << payload_type << " not registered.";
+    LOG(LS_WARNING) << "Payload type " << static_cast<int>(payload_type)
+                    << " not registered.";
     return -1;
   }
   SetSendPayloadType(payload_type);
@@ -512,7 +513,8 @@ int32_t RTPSender::SendOutgoingData(FrameType frame_type,
   }
   RtpVideoCodecTypes video_type = kRtpVideoGeneric;
   if (CheckPayloadType(payload_type, &video_type) != 0) {
-    LOG(LS_ERROR) << "Don't send data with unknown payload type.";
+    LOG(LS_ERROR) << "Don't send data with unknown payload type: "
+                  << static_cast<int>(payload_type) << ".";
     return -1;
   }
 
@@ -725,7 +727,7 @@ int32_t RTPSender::ReSendPacket(uint16_t packet_id, int64_t min_resend_time) {
     // TickTime.
     int64_t corrected_capture_tims_ms = capture_time_ms + clock_delta_ms_;
     paced_sender_->InsertPacket(
-        RtpPacketSender::kHighPriority, header.ssrc, header.sequenceNumber,
+        RtpPacketSender::kNormalPriority, header.ssrc, header.sequenceNumber,
         corrected_capture_tims_ms, length - header.headerLength, true);
 
     return length;
@@ -1003,7 +1005,7 @@ bool RTPSender::IsFecPacket(const uint8_t* buffer,
 }
 
 size_t RTPSender::TimeToSendPadding(size_t bytes) {
-  if (bytes == 0)
+  if (audio_configured_ || bytes == 0)
     return 0;
   {
     CriticalSectionScoped cs(send_critsect_.get());

webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc

@@ -16,6 +16,7 @@
 #include "webrtc/base/trace_event.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
 
 namespace webrtc {
 
@@ -368,7 +369,8 @@ int32_t RTPSenderAudio::SendAudio(
                            _rtpSender->Timestamp(), "seqnum",
                            _rtpSender->SequenceNumber());
     return _rtpSender->SendToNetwork(dataBuffer, payloadSize, rtpHeaderLength,
-                                     -1, kAllowRetransmission,
+                                     TickTime::MillisecondTimestamp(),
+                                     kAllowRetransmission,
                                      RtpPacketSender::kHighPriority);
   }
 
@@ -476,9 +478,9 @@ RTPSenderAudio::SendTelephoneEventPacket(bool ended,
                              "Audio::SendTelephoneEvent", "timestamp",
                              dtmfTimeStamp, "seqnum",
                              _rtpSender->SequenceNumber());
-        retVal = _rtpSender->SendToNetwork(dtmfbuffer, 4, 12, -1,
-                                           kAllowRetransmission,
-                                           RtpPacketSender::kHighPriority);
+        retVal = _rtpSender->SendToNetwork(
+            dtmfbuffer, 4, 12, TickTime::MillisecondTimestamp(),
+            kAllowRetransmission, RtpPacketSender::kHighPriority);
         sendCount--;
 
     }while (sendCount > 0 && retVal == 0);

webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc

@@ -104,7 +104,7 @@ void RTPSenderVideo::SendVideoPacket(uint8_t* data_buffer,
                                      StorageType storage) {
   if (_rtpSender.SendToNetwork(data_buffer, payload_length, rtp_header_length,
                                capture_time_ms, storage,
-                               RtpPacketSender::kNormalPriority) == 0) {
+                               RtpPacketSender::kLowPriority) == 0) {
     _videoBitrate.Update(payload_length + rtp_header_length);
     TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
                          "Video::PacketNormal", "timestamp", capture_timestamp,
@@ -150,7 +150,7 @@ void RTPSenderVideo::SendVideoPacketAsRed(uint8_t* data_buffer,
   if (_rtpSender.SendToNetwork(
           red_packet->data(), red_packet->length() - rtp_header_length,
           rtp_header_length, capture_time_ms, media_packet_storage,
-          RtpPacketSender::kNormalPriority) == 0) {
+          RtpPacketSender::kLowPriority) == 0) {
     _videoBitrate.Update(red_packet->length());
     TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
                          "Video::PacketRed", "timestamp", capture_timestamp,
@@ -162,7 +162,7 @@ void RTPSenderVideo::SendVideoPacketAsRed(uint8_t* data_buffer,
     if (_rtpSender.SendToNetwork(
             fec_packet->data(), fec_packet->length() - rtp_header_length,
             rtp_header_length, capture_time_ms, fec_storage,
-            RtpPacketSender::kNormalPriority) == 0) {
+            RtpPacketSender::kLowPriority) == 0) {
       _fecOverheadRate.Update(fec_packet->length());
       TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
                            "Video::PacketFec", "timestamp", capture_timestamp,