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Unified Diff: webrtc/video/video_send_stream.cc

Issue 1924793002: Remove webrtc/stream.h and unutilized inheritance. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: re-rebase Created 4 years, 8 months ago
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Index: webrtc/video/video_send_stream.cc
diff --git a/webrtc/video/video_send_stream.cc b/webrtc/video/video_send_stream.cc
index 40871af1211d7c5d94c73eea4f9fdf7cdd701995..cecc7e9b399645d511ec14c27c0977b5cdd18839 100644
--- a/webrtc/video/video_send_stream.cc
+++ b/webrtc/video/video_send_stream.cc
@@ -491,8 +491,25 @@ VideoSendStream::~VideoSendStream() {
}
}
-VideoCaptureInput* VideoSendStream::Input() {
- return &input_;
+void VideoSendStream::SignalNetworkState(NetworkState state) {
+ // When network goes up, enable RTCP status before setting transmission state.
+ // When it goes down, disable RTCP afterwards. This ensures that any packets
+ // sent due to the network state changed will not be dropped.
+ if (state == kNetworkUp) {
+ for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
+ rtp_rtcp->SetRTCPStatus(config_.rtp.rtcp_mode);
+ }
+ vie_encoder_.SetNetworkTransmissionState(state == kNetworkUp);
+ if (state == kNetworkDown) {
+ for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
+ rtp_rtcp->SetRTCPStatus(RtcpMode::kOff);
+ }
+}
+
+bool VideoSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
+ for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
+ rtp_rtcp->IncomingRtcpPacket(packet, length);
+ return true;
}
void VideoSendStream::Start() {
@@ -514,6 +531,10 @@ void VideoSendStream::Stop() {
payload_router_.set_active(false);
}
+VideoCaptureInput* VideoSendStream::Input() {
+ return &input_;
+}
+
bool VideoSendStream::EncoderThreadFunction(void* obj) {
static_cast<VideoSendStream*>(obj)->EncoderProcess();
// We're done, return false to abort.
@@ -580,12 +601,6 @@ void VideoSendStream::ReconfigureVideoEncoder(
encoder_wakeup_event_.Set();
}
-bool VideoSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
- for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
- rtp_rtcp->IncomingRtcpPacket(packet, length);
- return true;
-}
-
VideoSendStream::Stats VideoSendStream::GetStats() {
return stats_proxy_.GetStats();
}
@@ -714,21 +729,6 @@ std::map<uint32_t, RtpState> VideoSendStream::GetRtpStates() const {
return rtp_states;
}
-void VideoSendStream::SignalNetworkState(NetworkState state) {
- // When network goes up, enable RTCP status before setting transmission state.
- // When it goes down, disable RTCP afterwards. This ensures that any packets
- // sent due to the network state changed will not be dropped.
- if (state == kNetworkUp) {
- for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
- rtp_rtcp->SetRTCPStatus(config_.rtp.rtcp_mode);
- }
- vie_encoder_.SetNetworkTransmissionState(state == kNetworkUp);
- if (state == kNetworkDown) {
- for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
- rtp_rtcp->SetRTCPStatus(RtcpMode::kOff);
- }
-}
-
int VideoSendStream::GetPaddingNeededBps() const {
return vie_encoder_.GetPaddingNeededBps();
}
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