| Index: webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
|
| diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
|
| index 5af1b4ab6f7d1436db16e9223f01875dd205d934..5364a9b831d57002bb49e6b2aa5d7fa9a39b33d2 100644
|
| --- a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
|
| +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
|
| @@ -79,18 +79,18 @@
|
| 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)) {
|
| + 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 {
|
| 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,
|
| @@ -206,17 +206,18 @@
|
| }
|
| }
|
|
|
| -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;
|
| +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;
|
| + }
|
|
|
| std::unique_ptr<RtpPacketizer> packetizer(RtpPacketizer::Create(
|
| video_type, rtp_sender_->MaxDataPayloadLength(),
|
| @@ -261,14 +262,14 @@
|
|
|
| if (!packetizer->NextPacket(&dataBuffer[rtp_header_length],
|
| &payload_bytes_in_packet, &last)) {
|
| - return false;
|
| + return -1;
|
| }
|
|
|
| // Write RTP header.
|
| int32_t header_length = rtp_sender_->BuildRtpHeader(
|
| dataBuffer, payload_type, last, capture_timestamp, capture_time_ms);
|
| if (header_length <= 0)
|
| - return false;
|
| + return -1;
|
|
|
| // According to
|
| // http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/
|
| @@ -323,7 +324,7 @@
|
|
|
| TRACE_EVENT_ASYNC_END1("webrtc", "Video", capture_time_ms, "timestamp",
|
| rtp_sender_->Timestamp());
|
| - return true;
|
| + return 0;
|
| }
|
|
|
| uint32_t RTPSenderVideo::VideoBitrateSent() const {
|
|
|