| Index: webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
| diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
| index 9d134afbccc81a934c0146af9ffb116395dc1ac1..7e953ec4e07aecab05374ff3abe7d95b446a289c 100644
|
| --- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
| +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
| @@ -713,13 +713,11 @@ int32_t RTPSender::ReSendPacket(uint16_t packet_id, int64_t min_resend_time) {
|
| // Convert from TickTime to Clock since capture_time_ms is based on
|
| // TickTime.
|
| int64_t corrected_capture_tims_ms = capture_time_ms + clock_delta_ms_;
|
| - if (!paced_sender_->SendPacket(
|
| - RtpPacketSender::kHighPriority, header.ssrc, header.sequenceNumber,
|
| - corrected_capture_tims_ms, length - header.headerLength, true)) {
|
| - // We can't send the packet right now.
|
| - // We will be called when it is time.
|
| - return length;
|
| - }
|
| + paced_sender_->InsertPacket(
|
| + RtpPacketSender::kHighPriority, header.ssrc, header.sequenceNumber,
|
| + corrected_capture_tims_ms, length - header.headerLength, true);
|
| +
|
| + return length;
|
| }
|
| int rtx = kRtxOff;
|
| {
|
| @@ -1038,24 +1036,21 @@ int32_t RTPSender::SendToNetwork(uint8_t* buffer,
|
| return -1;
|
| }
|
|
|
| - if (paced_sender_ && storage != kDontStore) {
|
| + if (paced_sender_) {
|
| // Correct offset between implementations of millisecond time stamps in
|
| // TickTime and Clock.
|
| int64_t corrected_time_ms = capture_time_ms + clock_delta_ms_;
|
| - if (!paced_sender_->SendPacket(priority, rtp_header.ssrc,
|
| - rtp_header.sequenceNumber, corrected_time_ms,
|
| - payload_length, false)) {
|
| - if (last_capture_time_ms_sent_ == 0 ||
|
| - corrected_time_ms > last_capture_time_ms_sent_) {
|
| - last_capture_time_ms_sent_ = corrected_time_ms;
|
| - TRACE_EVENT_ASYNC_BEGIN1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
|
| - "PacedSend", corrected_time_ms,
|
| - "capture_time_ms", corrected_time_ms);
|
| - }
|
| - // We can't send the packet right now.
|
| - // We will be called when it is time.
|
| - return 0;
|
| + paced_sender_->InsertPacket(priority, rtp_header.ssrc,
|
| + rtp_header.sequenceNumber, corrected_time_ms,
|
| + payload_length, false);
|
| + if (last_capture_time_ms_sent_ == 0 ||
|
| + corrected_time_ms > last_capture_time_ms_sent_) {
|
| + last_capture_time_ms_sent_ = corrected_time_ms;
|
| + TRACE_EVENT_ASYNC_BEGIN1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
|
| + "PacedSend", corrected_time_ms,
|
| + "capture_time_ms", corrected_time_ms);
|
| }
|
| + return 0;
|
| }
|
| if (capture_time_ms > 0) {
|
| UpdateDelayStatistics(capture_time_ms, now_ms);
|
| @@ -1064,12 +1059,11 @@ int32_t RTPSender::SendToNetwork(uint8_t* buffer,
|
| size_t length = payload_length + rtp_header_length;
|
| bool sent = SendPacketToNetwork(buffer, length, PacketOptions());
|
|
|
| - if (storage != kDontStore) {
|
| - // Mark the packet as sent in the history even if send failed. Dropping a
|
| - // packet here should be treated as any other packet drop so we should be
|
| - // ready for a retransmission.
|
| - packet_history_.SetSent(rtp_header.sequenceNumber);
|
| - }
|
| + // Mark the packet as sent in the history even if send failed. Dropping a
|
| + // packet here should be treated as any other packet drop so we should be
|
| + // ready for a retransmission.
|
| + packet_history_.SetSent(rtp_header.sequenceNumber);
|
| +
|
| if (!sent)
|
| return -1;
|
|
|
| @@ -1785,13 +1779,6 @@ uint32_t RTPSender::MaxConfiguredBitrateVideo() const {
|
| return video_->MaxConfiguredBitrateVideo();
|
| }
|
|
|
| -int32_t RTPSender::SendRTPIntraRequest() {
|
| - if (audio_configured_) {
|
| - return -1;
|
| - }
|
| - return video_->SendRTPIntraRequest();
|
| -}
|
| -
|
| void RTPSender::SetGenericFECStatus(bool enable,
|
| uint8_t payload_type_red,
|
| uint8_t payload_type_fec) {
|
|
|