| Index: webrtc/media/engine/webrtcvideoengine2.cc
|
| diff --git a/webrtc/media/engine/webrtcvideoengine2.cc b/webrtc/media/engine/webrtcvideoengine2.cc
|
| index ec6b033d2f7a3862be69a9c25390d646968cfd2c..521bbdb49ee3fe06acea3dd6c9c3668cc6e6c31c 100644
|
| --- a/webrtc/media/engine/webrtcvideoengine2.cc
|
| +++ b/webrtc/media/engine/webrtcvideoengine2.cc
|
| @@ -1514,7 +1514,6 @@ WebRtcVideoChannel2::WebRtcVideoSendStream::WebRtcVideoSendStream(
|
| pending_encoder_reconfiguration_(false),
|
| allocated_encoder_(nullptr, webrtc::kVideoCodecUnknown, false),
|
| sending_(false),
|
| - first_frame_timestamp_ms_(0),
|
| last_frame_timestamp_ms_(0) {
|
| parameters_.config.rtp.max_packet_size = kVideoMtu;
|
| parameters_.conference_mode = send_params.conference_mode;
|
| @@ -1573,12 +1572,15 @@ void WebRtcVideoChannel2::WebRtcVideoSendStream::OnFrame(
|
| }
|
|
|
| int64_t frame_delta_ms = frame.GetTimeStamp() / rtc::kNumNanosecsPerMillisec;
|
| +
|
| // frame->GetTimeStamp() is essentially a delta, align to webrtc time
|
| - if (first_frame_timestamp_ms_ == 0) {
|
| - first_frame_timestamp_ms_ = rtc::Time() - frame_delta_ms;
|
| + if (!first_frame_timestamp_ms_) {
|
| + first_frame_timestamp_ms_ =
|
| + rtc::Optional<int64_t>(rtc::Time() - frame_delta_ms);
|
| }
|
|
|
| - last_frame_timestamp_ms_ = first_frame_timestamp_ms_ + frame_delta_ms;
|
| + last_frame_timestamp_ms_ = *first_frame_timestamp_ms_ + frame_delta_ms;
|
| +
|
| video_frame.set_render_time_ms(last_frame_timestamp_ms_);
|
| // Reconfigure codec if necessary.
|
| SetDimensions(video_frame.width(), video_frame.height());
|
| @@ -1608,7 +1610,7 @@ void WebRtcVideoChannel2::WebRtcVideoSendStream::SetSource(
|
|
|
| // Reset timestamps to realign new incoming frames to a webrtc timestamp. A
|
| // new capturer may have a different timestamp delta than the previous one.
|
| - first_frame_timestamp_ms_ = 0;
|
| + first_frame_timestamp_ms_ = rtc::Optional<int64_t>();
|
|
|
| if (source == NULL) {
|
| if (stream_ != NULL) {
|
|
|