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) { |