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Unified Diff: webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc

Issue 2203233002: Revert of Add EncodedImageCallback::OnEncodedImage(). (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 4 months ago
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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 {
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