Add flag enabling more packets to be retransmittable.

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 214 matching lines @@
     uint32_t rate) {
   RTC_DCHECK_LT(strlen(payload_name), RTP_PAYLOAD_NAME_SIZE);
   rtc::CritScope lock(&send_critsect_);
 
   std::map<int8_t, RtpUtility::Payload*>::iterator it =
       payload_type_map_.find(payload_number);
 
   if (payload_type_map_.end() != it) {
     // We already use this payload type.
     RtpUtility::Payload* payload = it->second;
-    assert(payload);
+    RTC_DCHECK(payload);
 
     // Check if it's the same as we already have.
     if (RtpUtility::StringCompare(
             payload->name, payload_name, RTP_PAYLOAD_NAME_SIZE - 1)) {
       if (audio_configured_ && payload->audio &&
           payload->typeSpecific.Audio.frequency == frequency &&
           (payload->typeSpecific.Audio.rate == rate ||
            payload->typeSpecific.Audio.rate == 0 || rate == 0)) {
         payload->typeSpecific.Audio.rate = rate;
         // Ensure that we update the rate if new or old is zero.
@@ skipping 102 matching lines @@
   }
   std::map<int8_t, RtpUtility::Payload*>::iterator it =
       payload_type_map_.find(payload_type);
   if (it == payload_type_map_.end()) {
     LOG(LS_WARNING) << "Payload type " << static_cast<int>(payload_type)
                     << " not registered.";
     return -1;
   }
   SetSendPayloadType(payload_type);
   RtpUtility::Payload* payload = it->second;
-  assert(payload);
+  RTC_DCHECK(payload);
   if (!payload->audio && !audio_configured_) {
     video_->SetVideoCodecType(payload->typeSpecific.Video.videoCodecType);
     *video_type = payload->typeSpecific.Video.videoCodecType;
   }
   return 0;
 }
 
 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* transport_frame_id_out,
+                                 int64_t expected_retransmission_time_ms) {
   uint32_t ssrc;
   uint16_t sequence_number;
   uint32_t rtp_timestamp;
   {
     // Drop this packet if we're not sending media packets.
     rtc::CritScope lock(&send_critsect_);
     RTC_DCHECK(ssrc_);
 
     ssrc = *ssrc_;
     sequence_number = sequence_number_;
     rtp_timestamp = timestamp_offset_ + capture_timestamp;
     if (transport_frame_id_out)
       *transport_frame_id_out = rtp_timestamp;
     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 false;
   }
 
+  switch (frame_type) {
+    case kAudioFrameSpeech:
+    case kAudioFrameCN:
+      RTC_CHECK(audio_configured_);
+      break;
+    case kVideoFrameKey:
+    case kVideoFrameDelta:
+      RTC_CHECK(!audio_configured_);
+      break;
+    case kEmptyFrame:
+      break;
+    default:

[danilchap, 2017/08/29 17:31:19]

it is recommended not to have default to get compile-time error when a new enum
value will be added.

[sprang_webrtc, 2017/08/31 15:54:28]

Done.

+      FATAL() << "Unrecognized frame type.";
+  }
+
   bool result;
   if (audio_configured_) {
     TRACE_EVENT_ASYNC_STEP1("webrtc", "Audio", rtp_timestamp, "Send", "type",
                             FrameTypeToString(frame_type));
-    assert(frame_type == kAudioFrameSpeech || frame_type == kAudioFrameCN ||
-           frame_type == kEmptyFrame);
 
     result = audio_->SendAudio(frame_type, payload_type, rtp_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,
                                           sequence_number);
     }
 
     result = video_->SendVideo(video_type, frame_type, payload_type,
                                rtp_timestamp, capture_time_ms, payload_data,
-                               payload_size, fragmentation, rtp_header);
+                               payload_size, fragmentation, rtp_header,
+                               expected_retransmission_time_ms);
   }
 
   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_) {
@@ skipping 665 matching lines @@
 }
 
 rtc::Optional<uint32_t> RTPSender::FlexfecSsrc() const {
   if (video_) {
     return video_->FlexfecSsrc();
   }
   return rtc::Optional<uint32_t>();
 }
 
 void RTPSender::SetCsrcs(const std::vector<uint32_t>& csrcs) {
-  assert(csrcs.size() <= kRtpCsrcSize);
+  RTC_DCHECK_LE(csrcs.size(), kRtpCsrcSize);
   rtc::CritScope lock(&send_critsect_);
   csrcs_ = csrcs;
 }
 
 void RTPSender::SetSequenceNumber(uint16_t seq) {
   rtc::CritScope lock(&send_critsect_);
   sequence_number_forced_ = true;
   sequence_number_ = seq;
 }
 
@@ skipping 10 matching lines @@
     return -1;
   }
   return audio_->SendTelephoneEvent(key, time_ms, level);
 }
 
 int32_t RTPSender::SetAudioLevel(uint8_t level_d_bov) {
   return audio_->SetAudioLevel(level_d_bov);
 }
 
 RtpVideoCodecTypes RTPSender::VideoCodecType() const {
-  assert(!audio_configured_ && "Sender is an audio stream!");
+  RTC_DCHECK(!audio_configured_) << "Sender is an audio stream!";
   return video_->VideoCodecType();
 }
 
 void RTPSender::SetUlpfecConfig(int red_payload_type, int ulpfec_payload_type) {
   RTC_DCHECK(!audio_configured_);
   video_->SetUlpfecConfig(red_payload_type, ulpfec_payload_type);
 }
 
 bool RTPSender::SetFecParameters(const FecProtectionParams& delta_params,
                                  const FecProtectionParams& key_params) {
@@ skipping 145 matching lines @@
     rtc::CritScope lock(&send_critsect_);
     packet->SetTimestamp(last_rtp_timestamp_);
     packet->set_capture_time_ms(capture_time_ms_);
   }
   AssignSequenceNumber(packet.get());
   SendToNetwork(std::move(packet), StorageType::kDontRetransmit,
                 RtpPacketSender::Priority::kLowPriority);
 }
 
 }  // namespace webrtc