Add EncodedImageCallback::OnEncodedImage().

webrtc/modules/rtp_rtcp/source/rtp_sender.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 430 matching lines @@
 bool RTPSender::ActivateCVORtpHeaderExtension() {
   if (!video_rotation_active_) {
     rtc::CritScope lock(&send_critsect_);
     if (rtp_header_extension_map_.SetActive(kRtpExtensionVideoRotation, true)) {
       video_rotation_active_ = true;
     }
   }
   return video_rotation_active_;
 }
 
 int32_t RTPSender::SendOutgoingData(FrameType frame_type,

[stefan-webrtc, 2016/07/28 08:51:17]

As you are anyway introducing this new method, maybe you can change it to return
bool instead of int32_t?

Or maybe even return the new Result type from here too?

[Sergey Ulanov, 2016/07/28 20:52:46]

Done.
That would require moving Result struct somewhere where it can be shared with
this class.

[stefan-webrtc, 2016/07/29 07:13:37]

Sure, maybe module_common_types.h?

[Sergey Ulanov, 2016/08/01 07:22:13]

Another issue is that we need to keep the current version of
RtpRtcp::SendOutgoingData() and it's not possible to overload it with new return
type, without changing args. Are there any other parameters that can be updated?

                                     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) {
+                                    const RTPVideoHeader* rtp_hdr,
+                                    uint32_t* frame_id_out) {
   uint32_t ssrc;
   uint16_t sequence_number;
   {
     // Drop this packet if we're not sending media packets.
     rtc::CritScope lock(&send_critsect_);
     ssrc = ssrc_;
     sequence_number = sequence_number_;
     if (!sending_media_) {
       return 0;
     }
@@ skipping 36 matching lines @@
         rtp_header_extension_map_.SetActive(kRtpExtensionPlayoutDelay,
                                             playout_delay_active_);
       }
     }
 
     ret_val = video_->SendVideo(
         video_type, frame_type, payload_type, capture_timestamp,
         capture_time_ms, payload_data, payload_size, fragmentation, rtp_hdr);
   }
 
+  if (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.
+    *frame_id_out = timestamp_;
+  }
+
   rtc::CritScope cs(&statistics_crit_);
   // Note: This is currently only counting for video.
   if (frame_type == kVideoFrameKey) {
     ++frame_counts_.key_frames;
   } else if (frame_type == kVideoFrameDelta) {
     ++frame_counts_.delta_frames;
   }
   if (frame_count_observer_) {
     frame_count_observer_->FrameCountUpdated(frame_counts_, ssrc);
   }
@@ skipping 1359 matching lines @@
   rtc::CritScope lock(&send_critsect_);
 
   RtpState state;
   state.sequence_number = sequence_number_rtx_;
   state.start_timestamp = start_timestamp_;
 
   return state;
 }
 
 }  // namespace webrtc