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

Issue 1479023002: Prepare the AudioSendStream to be hooked up to send-side BWE. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Moved method. Created 5 years ago
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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 dc544fbe69fe01534aae27c20e843c65dd3c0802..a31491a288a712c26d605b59358e4a50b4183090 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -469,7 +469,8 @@ int32_t RTPSender::CheckPayloadType(int8_t payload_type,
std::map<int8_t, RtpUtility::Payload*>::iterator it =
payload_type_map_.find(payload_type);
if (it == payload_type_map_.end()) {
- LOG(LS_WARNING) << "Payload type " << payload_type << " not registered.";
+ LOG(LS_WARNING) << "Payload type " << static_cast<int>(payload_type)
+ << " not registered.";
return -1;
}
SetSendPayloadType(payload_type);
@@ -512,7 +513,8 @@ int32_t RTPSender::SendOutgoingData(FrameType frame_type,
}
RtpVideoCodecTypes video_type = kRtpVideoGeneric;
if (CheckPayloadType(payload_type, &video_type) != 0) {
- LOG(LS_ERROR) << "Don't send data with unknown payload type.";
+ LOG(LS_ERROR) << "Don't send data with unknown payload type: "
+ << static_cast<int>(payload_type) << ".";
return -1;
}
@@ -725,7 +727,7 @@ int32_t RTPSender::ReSendPacket(uint16_t packet_id, int64_t min_resend_time) {
// TickTime.
int64_t corrected_capture_tims_ms = capture_time_ms + clock_delta_ms_;
paced_sender_->InsertPacket(
- RtpPacketSender::kHighPriority, header.ssrc, header.sequenceNumber,
+ RtpPacketSender::kNormalPriority, header.ssrc, header.sequenceNumber,
corrected_capture_tims_ms, length - header.headerLength, true);
return length;
@@ -1005,6 +1007,8 @@ bool RTPSender::IsFecPacket(const uint8_t* buffer,
size_t RTPSender::TimeToSendPadding(size_t bytes) {
if (bytes == 0)
return 0;
+ if (audio_configured_)
mflodman 2015/12/04 15:25:19 Maybe do bytes == 0 || audio_configured_ to reduce
stefan-webrtc 2015/12/04 15:30:22 Done.
+ return 0;
{
CriticalSectionScoped cs(send_critsect_.get());
if (!sending_media_)

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