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 { |