Add EncodedImageCallback::OnEncodedImage().

webrtc/video/payload_router.cc

 /*
  *  Copyright (c) 2015 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 119 matching lines @@
     rtp_modules_[i]->SetSendingStatus(active_);
     rtp_modules_[i]->SetSendingMediaStatus(active_);
   }
   // Disable inactive modules.
   for (size_t i = num_sending_modules_; i < rtp_modules_.size(); ++i) {
     rtp_modules_[i]->SetSendingStatus(false);
     rtp_modules_[i]->SetSendingMediaStatus(false);
   }
 }
 
-int32_t PayloadRouter::Encoded(const EncodedImage& encoded_image,
-                               const CodecSpecificInfo* codec_specific_info,
-                               const RTPFragmentationHeader* fragmentation) {
+EncodedImageCallback::Result PayloadRouter::OnEncodedImage(
+    const EncodedImage& encoded_image,
+    const CodecSpecificInfo* codec_specific_info,
+    const RTPFragmentationHeader* fragmentation) {
   rtc::CritScope lock(&crit_);
   RTC_DCHECK(!rtp_modules_.empty());
   if (!active_ || num_sending_modules_ == 0)
-    return -1;
+    return Result(Result::ERROR_SEND_FAILED);
 
-  int stream_idx = 0;
+  int stream_index = 0;
 
   RTPVideoHeader rtp_video_header;
   memset(&rtp_video_header, 0, sizeof(RTPVideoHeader));
   if (codec_specific_info)
     CopyCodecSpecific(codec_specific_info, &rtp_video_header);
   rtp_video_header.rotation = encoded_image.rotation_;
   rtp_video_header.playout_delay = encoded_image.playout_delay_;
 
   RTC_DCHECK_LT(rtp_video_header.simulcastIdx, rtp_modules_.size());
   // The simulcast index might actually be larger than the number of modules
   // in case the encoder was processing a frame during a codec reconfig.
   if (rtp_video_header.simulcastIdx >= num_sending_modules_)
-    return -1;
-  stream_idx = rtp_video_header.simulcastIdx;
+    return Result(Result::ERROR_SEND_FAILED);
+  stream_index = rtp_video_header.simulcastIdx;
 
-  return rtp_modules_[stream_idx]->SendOutgoingData(
-      encoded_image._frameType, payload_type_, encoded_image._timeStamp,
-      encoded_image.capture_time_ms_, encoded_image._buffer,
-      encoded_image._length, fragmentation, &rtp_video_header);
+  int send_result = rtp_modules_[stream_index]->SendOutgoingData(
+          encoded_image._frameType, payload_type_, encoded_image._timeStamp,
+          encoded_image.capture_time_ms_, encoded_image._buffer,
+          encoded_image._length, fragmentation, &rtp_video_header);
+
+  if (send_result < 0)
+    return Result(Result::ERROR_SEND_FAILED);
+
+  uint32_t frame_id =
+      rtp_modules_[stream_index]->StartTimestamp() + encoded_image._timeStamp;

[stefan-webrtc, 2016/07/18 16:51:27]

Would it make sense to instead have SendOutgoingData return the timestamp that
will be used?

[Sergey Ulanov, 2016/07/21 00:09:39]

Yes, but RtpRtcp::SendOutgoingData() is used in bunch of places outside of
webrtc, so it's hard to change it, so this seems to be the least bad solution.
Or I could add a second version SendOutgoingData() that would also return
timestamp, WDYT?

[stefan-webrtc, 2016/07/25 15:19:33]

Missed this comment, sorry.

I would prefer to add a secibd versuib if SendOutgoingData() and deprecate the
old.

[Sergey Ulanov, 2016/07/26 19:19:53]

Done

+  return Result(Result::OK, frame_id);
 }
 
 void PayloadRouter::SetTargetSendBitrate(uint32_t bitrate_bps) {
   rtc::CritScope lock(&crit_);
   RTC_DCHECK_LE(streams_.size(), rtp_modules_.size());
 
   // TODO(sprang): Rebase https://codereview.webrtc.org/1913073002/ on top of
   // this.
   int bitrate_remainder = bitrate_bps;
   for (size_t i = 0; i < streams_.size() && bitrate_remainder > 0; ++i) {
@@ skipping 13 matching lines @@
   rtc::CritScope lock(&crit_);
   for (size_t i = 0; i < num_sending_modules_; ++i) {
     size_t module_payload_length = rtp_modules_[i]->MaxDataPayloadLength();
     if (module_payload_length < min_payload_length)
       min_payload_length = module_payload_length;
   }
   return min_payload_length;
 }
 
 }  // namespace webrtc