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,
-                                    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) {
+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) {
   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;
-    }
+    if (!sending_media_)
+      return true;
   }
   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 -1;
+    return false;
   }
 
-  int32_t ret_val;
+  bool result;
   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);
 
-    ret_val = audio_->SendAudio(frame_type, payload_type, capture_timestamp,
-                                payload_data, payload_size, fragmentation);
+    result = 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 0;
+      return true;
 
-    if (rtp_hdr) {
-      playout_delay_oracle_.UpdateRequest(ssrc, rtp_hdr->playout_delay,
+    if (rtp_header) {
+      playout_delay_oracle_.UpdateRequest(ssrc, rtp_header->playout_delay,
                                           sequence_number);
     }
 
     // Update the active/inactive status of playout delay extension based
     // on what the oracle indicates.
     {
       rtc::CritScope lock(&send_critsect_);
       if (playout_delay_active_ != playout_delay_oracle_.send_playout_delay()) {
         playout_delay_active_ = playout_delay_oracle_.send_playout_delay();
         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);
+    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_;
   }
 
   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);
   }
 
-  return ret_val;
+  return result;
 }
 
 size_t RTPSender::TrySendRedundantPayloads(size_t bytes_to_send,
                                            int probe_cluster_id) {
   {
     rtc::CritScope lock(&send_critsect_);
     if (!sending_media_)
       return 0;
     if ((rtx_ & kRtxRedundantPayloads) == 0)
       return 0;
@@ skipping 401 matching lines @@
   if (audio_configured_ || bytes == 0)
     return 0;
   size_t bytes_sent = TrySendRedundantPayloads(bytes, probe_cluster_id);
   if (bytes_sent < bytes)
     bytes_sent +=
         SendPadData(bytes - bytes_sent, false, 0, 0, probe_cluster_id);
   return bytes_sent;
 }
 
 // TODO(pwestin): send in the RtpHeaderParser to avoid parsing it again.
-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) {
+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) {
   size_t length = payload_length + rtp_header_length;
   RtpUtility::RtpHeaderParser rtp_parser(buffer, length);
 
   RTPHeader rtp_header;
   rtp_parser.Parse(&rtp_header);
 
   int64_t now_ms = clock_->TimeInMilliseconds();
 
   // |capture_time_ms| <= 0 is considered invalid.
   // TODO(holmer): This should be changed all over Video Engine so that negative
   // time is consider invalid, while 0 is considered a valid time.
   if (capture_time_ms > 0) {
     UpdateTransmissionTimeOffset(buffer, length, rtp_header,
                                  now_ms - capture_time_ms);
   }
 
   UpdateAbsoluteSendTime(buffer, length, rtp_header, now_ms);
 
   // Used for NACK and to spread out the transmission of packets.
   if (packet_history_.PutRTPPacket(buffer, length, capture_time_ms, storage) !=
       0) {
-    return -1;
+    return false;
   }
 
   if (paced_sender_) {
     // Correct offset between implementations of millisecond time stamps in
     // TickTime and Clock.
     int64_t corrected_time_ms = capture_time_ms + clock_delta_ms_;
     paced_sender_->InsertPacket(priority, rtp_header.ssrc,
                                 rtp_header.sequenceNumber, corrected_time_ms,
                                 payload_length, false);
     if (last_capture_time_ms_sent_ == 0 ||
         corrected_time_ms > last_capture_time_ms_sent_) {
       last_capture_time_ms_sent_ = corrected_time_ms;
       TRACE_EVENT_ASYNC_BEGIN1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
                                "PacedSend", corrected_time_ms,
                                "capture_time_ms", corrected_time_ms);
     }
-    return 0;
+    return true;
   }
 
   PacketOptions options;
   if (UpdateTransportSequenceNumber(buffer, length, rtp_header,
                                     &options.packet_id)) {
     if (transport_feedback_observer_)
       transport_feedback_observer_->AddPacket(options.packet_id, length,
                                               PacketInfo::kNotAProbe);
   }
   UpdateDelayStatistics(capture_time_ms, now_ms);
   UpdateOnSendPacket(options.packet_id, capture_time_ms, rtp_header.ssrc);
 
   bool sent = SendPacketToNetwork(buffer, length, options);
 
   // Mark the packet as sent in the history even if send failed. Dropping a
   // packet here should be treated as any other packet drop so we should be
   // ready for a retransmission.
   packet_history_.SetSent(rtp_header.sequenceNumber);
 
   if (!sent)
-    return -1;
+    return false;
 
   {
     rtc::CritScope lock(&send_critsect_);
     media_has_been_sent_ = true;
   }
   UpdateRtpStats(buffer, length, rtp_header, false, false);
-  return 0;
+  return true;
 }
 
 void RTPSender::UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms) {
   if (!send_side_delay_observer_ || capture_time_ms <= 0)
     return;
 
   uint32_t ssrc;
   int avg_delay_ms = 0;
   int max_delay_ms = 0;
   {
@@ skipping 863 matching lines @@
   rtc::CritScope lock(&send_critsect_);
 
   RtpState state;
   state.sequence_number = sequence_number_rtx_;
   state.start_timestamp = start_timestamp_;
 
   return state;
 }
 
 }  // namespace webrtc