Revert of Add EncodedImageCallback::OnEncodedImage().

webrtc/modules/rtp_rtcp/source/rtp_sender.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_sender.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index 58dbc3ebffb983e84f17c94ba2dbb599fd13e169..2fe80ae90f5c154099a459287e76ea575439316a 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -448,15 +448,14 @@
   return video_rotation_active_;
 }
 
-bool RTPSender::SendOutgoingData(FrameType frame_type,
-                                 int8_t payload_type,
-                                 uint32_t capture_timestamp,
-                                 int64_t capture_time_ms,
-                                 const uint8_t* payload_data,
-                                 size_t payload_size,
-                                 const RTPFragmentationHeader* fragmentation,
-                                 const RTPVideoHeader* rtp_header,
-                                 uint32_t* transport_frame_id_out) {
+int32_t RTPSender::SendOutgoingData(FrameType frame_type,
+                                    int8_t payload_type,
+                                    uint32_t capture_timestamp,
+                                    int64_t capture_time_ms,
+                                    const uint8_t* payload_data,
+                                    size_t payload_size,
+                                    const RTPFragmentationHeader* fragmentation,
+                                    const RTPVideoHeader* rtp_hdr) {
   uint32_t ssrc;
   uint16_t sequence_number;
   {
@@ -464,35 +463,36 @@
     rtc::CritScope lock(&send_critsect_);
     ssrc = ssrc_;
     sequence_number = sequence_number_;
-    if (!sending_media_)
-      return true;
+    if (!sending_media_) {
+      return 0;
+    }
   }
   RtpVideoCodecTypes video_type = kRtpVideoGeneric;
   if (CheckPayloadType(payload_type, &video_type) != 0) {
     LOG(LS_ERROR) << "Don't send data with unknown payload type: "
                   << static_cast<int>(payload_type) << ".";
-    return false;
-  }
-
-  bool result;
+    return -1;
+  }
+
+  int32_t ret_val;
   if (audio_configured_) {
     TRACE_EVENT_ASYNC_STEP1("webrtc", "Audio", capture_timestamp,
                             "Send", "type", FrameTypeToString(frame_type));
     assert(frame_type == kAudioFrameSpeech || frame_type == kAudioFrameCN ||
            frame_type == kEmptyFrame);
 
-    result = audio_->SendAudio(frame_type, payload_type, capture_timestamp,
-                               payload_data, payload_size, fragmentation);
+    ret_val = audio_->SendAudio(frame_type, payload_type, capture_timestamp,
+                                payload_data, payload_size, fragmentation);
   } else {
     TRACE_EVENT_ASYNC_STEP1("webrtc", "Video", capture_time_ms,
                             "Send", "type", FrameTypeToString(frame_type));
     assert(frame_type != kAudioFrameSpeech && frame_type != kAudioFrameCN);
 
     if (frame_type == kEmptyFrame)
-      return true;
-
-    if (rtp_header) {
-      playout_delay_oracle_.UpdateRequest(ssrc, rtp_header->playout_delay,
+      return 0;
+
+    if (rtp_hdr) {
+      playout_delay_oracle_.UpdateRequest(ssrc, rtp_hdr->playout_delay,
                                           sequence_number);
     }
 
@@ -507,16 +507,9 @@
       }
     }
 
-    result = video_->SendVideo(video_type, frame_type, payload_type,
-                               capture_timestamp, capture_time_ms, payload_data,
-                               payload_size, fragmentation, rtp_header);
-  }
-
-  if (transport_frame_id_out) {
-    rtc::CritScope lock(&send_critsect_);
-    // TODO(sergeyu): Move RTP timestamp calculation from BuildRTPheader() to
-    // SendOutgoingData() and pass it to SendVideo()/SendAudio() calls.
-    *transport_frame_id_out = timestamp_;
+    ret_val = video_->SendVideo(
+        video_type, frame_type, payload_type, capture_timestamp,
+        capture_time_ms, payload_data, payload_size, fragmentation, rtp_hdr);
   }
 
   rtc::CritScope cs(&statistics_crit_);
@@ -530,7 +523,7 @@
     frame_count_observer_->FrameCountUpdated(frame_counts_, ssrc);
   }
 
-  return result;
+  return ret_val;
 }
 
 size_t RTPSender::TrySendRedundantPayloads(size_t bytes_to_send,
@@ -952,12 +945,12 @@
 }
 
 // TODO(pwestin): send in the RtpHeaderParser to avoid parsing it again.
-bool RTPSender::SendToNetwork(uint8_t* buffer,
-                              size_t payload_length,
-                              size_t rtp_header_length,
-                              int64_t capture_time_ms,
-                              StorageType storage,
-                              RtpPacketSender::Priority priority) {
+int32_t RTPSender::SendToNetwork(uint8_t* buffer,
+                                 size_t payload_length,
+                                 size_t rtp_header_length,
+                                 int64_t capture_time_ms,
+                                 StorageType storage,
+                                 RtpPacketSender::Priority priority) {
   size_t length = payload_length + rtp_header_length;
   RtpUtility::RtpHeaderParser rtp_parser(buffer, length);
 
@@ -979,7 +972,7 @@
   // Used for NACK and to spread out the transmission of packets.
   if (packet_history_.PutRTPPacket(buffer, length, capture_time_ms, storage) !=
       0) {
-    return false;
+    return -1;
   }
 
   if (paced_sender_) {
@@ -996,7 +989,7 @@
                                "PacedSend", corrected_time_ms,
                                "capture_time_ms", corrected_time_ms);
     }
-    return true;
+    return 0;
   }
 
   PacketOptions options;
@@ -1017,14 +1010,14 @@
   packet_history_.SetSent(rtp_header.sequenceNumber);
 
   if (!sent)
-    return false;
+    return -1;
 
   {
     rtc::CritScope lock(&send_critsect_);
     media_has_been_sent_ = true;
   }
   UpdateRtpStats(buffer, length, rtp_header, false, false);
-  return true;
+  return 0;
 }
 
 void RTPSender::UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms) {