Add EncodedImageCallback::OnEncodedImage().

webrtc/modules/rtp_rtcp/source/rtp_sender_video.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 61 matching lines @@
   return payload;
 }
 
 void RTPSenderVideo::SendVideoPacket(uint8_t* data_buffer,
                                      size_t payload_length,
                                      size_t rtp_header_length,
                                      uint16_t seq_num,
                                      uint32_t capture_timestamp,
                                      int64_t capture_time_ms,
                                      StorageType storage) {
-  if (rtp_sender_->SendToNetwork(data_buffer, payload_length, rtp_header_length,
-                                 capture_time_ms, storage,
-                                 RtpPacketSender::kLowPriority) == 0) {
-    rtc::CritScope cs(&stats_crit_);
-    video_bitrate_.Update(payload_length + rtp_header_length,
-                          clock_->TimeInMilliseconds());
-    TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
-                         "Video::PacketNormal", "timestamp", capture_timestamp,
-                         "seqnum", seq_num);
-  } else {
+  if (!rtp_sender_->SendToNetwork(data_buffer, payload_length,
+                                  rtp_header_length, capture_time_ms, storage,
+                                  RtpPacketSender::kLowPriority)) {
     LOG(LS_WARNING) << "Failed to send video packet " << seq_num;
+    return;
   }
+  rtc::CritScope cs(&stats_crit_);
+  video_bitrate_.Update(payload_length + rtp_header_length,
+                        clock_->TimeInMilliseconds());
+  TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
+                       "Video::PacketNormal", "timestamp", capture_timestamp,
+                       "seqnum", seq_num);
 }
 
 void RTPSenderVideo::SendVideoPacketAsRed(uint8_t* data_buffer,
                                           size_t payload_length,
                                           size_t rtp_header_length,
                                           uint16_t media_seq_num,
                                           uint32_t capture_timestamp,
                                           int64_t capture_time_ms,
                                           StorageType media_packet_storage,
                                           bool protect) {
@@ skipping 18 matching lines @@
           red_payload_type_, fec_payload_type_, next_fec_sequence_number,
           rtp_header_length);
       RTC_DCHECK_EQ(num_fec_packets, fec_packets.size());
       if (retransmission_settings_ & kRetransmitFECPackets)
         fec_storage = kAllowRetransmission;
     }
   }
   if (rtp_sender_->SendToNetwork(
           red_packet->data(), red_packet->length() - rtp_header_length,
           rtp_header_length, capture_time_ms, media_packet_storage,
-          RtpPacketSender::kLowPriority) == 0) {
+          RtpPacketSender::kLowPriority)) {
     rtc::CritScope cs(&stats_crit_);
     video_bitrate_.Update(red_packet->length(), clock_->TimeInMilliseconds());
     TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
                          "Video::PacketRed", "timestamp", capture_timestamp,
                          "seqnum", media_seq_num);
   } else {
     LOG(LS_WARNING) << "Failed to send RED packet " << media_seq_num;
   }
   for (RedPacket* fec_packet : fec_packets) {
     if (rtp_sender_->SendToNetwork(
             fec_packet->data(), fec_packet->length() - rtp_header_length,
             rtp_header_length, capture_time_ms, fec_storage,
-            RtpPacketSender::kLowPriority) == 0) {
+            RtpPacketSender::kLowPriority)) {
       rtc::CritScope cs(&stats_crit_);
       fec_bitrate_.Update(fec_packet->length(), clock_->TimeInMilliseconds());
       TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
                            "Video::PacketFec", "timestamp", capture_timestamp,
                            "seqnum", next_fec_sequence_number);
     } else {
       LOG(LS_WARNING) << "Failed to send FEC packet "
                       << next_fec_sequence_number;
     }
     delete fec_packet;
@@ skipping 43 matching lines @@
                                       const FecProtectionParams* key_params) {
   rtc::CritScope cs(&crit_);
   RTC_DCHECK(delta_params);
   RTC_DCHECK(key_params);
   if (fec_enabled_) {
     delta_fec_params_ = *delta_params;
     key_fec_params_ = *key_params;
   }
 }
 
-int32_t RTPSenderVideo::SendVideo(RtpVideoCodecTypes video_type,
-                                  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* video_header) {
-  if (payload_size == 0) {
-    return -1;
-  }
+bool RTPSenderVideo::SendVideo(RtpVideoCodecTypes video_type,
+                               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* video_header) {
+  if (payload_size == 0)
+    return false;
 
   std::unique_ptr<RtpPacketizer> packetizer(RtpPacketizer::Create(
       video_type, rtp_sender_->MaxDataPayloadLength(),
       video_header ? &(video_header->codecHeader) : nullptr, frame_type));
 
   StorageType storage;
   int red_payload_type;
   bool first_frame = first_frame_sent_();
   {
     rtc::CritScope cs(&crit_);
@@ skipping 24 matching lines @@
   packetizer->SetPayloadData(data, payload_bytes_to_send, frag);
 
   bool first = true;
   bool last = false;
   while (!last) {
     uint8_t dataBuffer[IP_PACKET_SIZE] = {0};
     size_t payload_bytes_in_packet = 0;
 
     if (!packetizer->NextPacket(&dataBuffer[rtp_header_length],
                                 &payload_bytes_in_packet, &last)) {
-      return -1;
+      return false;
     }
 
     // Write RTP header.
     int32_t header_length = rtp_sender_->BuildRtpHeader(
         dataBuffer, payload_type, last, capture_timestamp, capture_time_ms);
     if (header_length <= 0)
-      return -1;
+      return false;
 
     // According to
     // http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/
     // ts_126114v120700p.pdf Section 7.4.5:
     // The MTSI client shall add the payload bytes as defined in this clause
     // onto the last RTP packet in each group of packets which make up a key
     // frame (I-frame or IDR frame in H.264 (AVC), or an IRAP picture in H.265
     // (HEVC)). The MTSI client may also add the payload bytes onto the last RTP
     // packet in each group of packets which make up another type of frame
     // (e.g. a P-Frame) only if the current value is different from the previous
@@ skipping 34 matching lines @@
       if (last) {
         LOG(LS_INFO)
             << "Sent last RTP packet of the first video frame (pre-pacer)";
       }
     }
     first = false;
   }
 
   TRACE_EVENT_ASYNC_END1("webrtc", "Video", capture_time_ms, "timestamp",
                          rtp_sender_->Timestamp());
-  return 0;
+  return true;
 }
 
 uint32_t RTPSenderVideo::VideoBitrateSent() const {
   rtc::CritScope cs(&stats_crit_);
   return video_bitrate_.Rate(clock_->TimeInMilliseconds()).value_or(0);
 }
 
 uint32_t RTPSenderVideo::FecOverheadRate() const {
   rtc::CritScope cs(&stats_crit_);
   return fec_bitrate_.Rate(clock_->TimeInMilliseconds()).value_or(0);
 }
 
 int RTPSenderVideo::SelectiveRetransmissions() const {
   rtc::CritScope cs(&crit_);
   return retransmission_settings_;
 }
 
 void RTPSenderVideo::SetSelectiveRetransmissions(uint8_t settings) {
   rtc::CritScope cs(&crit_);
   retransmission_settings_ = settings;
 }
 
 }  // namespace webrtc