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Unified Diff: talk/media/webrtc/webrtcvideoengine2.cc

Issue 1333483002: Wire up PacketTime to ReceiveStreams. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years, 3 months ago
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Index: talk/media/webrtc/webrtcvideoengine2.cc
diff --git a/talk/media/webrtc/webrtcvideoengine2.cc b/talk/media/webrtc/webrtcvideoengine2.cc
index 5957329ae360eb4059ade164293eab73a8762aa9..7454b9eb19aa27a25c397368d1d95a9e9e8e5e78 100644
--- a/talk/media/webrtc/webrtcvideoengine2.cc
+++ b/talk/media/webrtc/webrtcvideoengine2.cc
@@ -1435,9 +1435,13 @@ bool WebRtcVideoChannel2::RequestIntraFrame() {
void WebRtcVideoChannel2::OnPacketReceived(
rtc::Buffer* packet,
const rtc::PacketTime& packet_time) {
+ const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp,
+ packet_time.not_before);
const webrtc::PacketReceiver::DeliveryStatus delivery_result =
- call_->Receiver()->DeliverPacket(webrtc::MediaType::VIDEO,
- reinterpret_cast<const uint8_t*>(packet->data()), packet->size());
+ call_->Receiver()->DeliverPacket(
+ webrtc::MediaType::VIDEO,
+ reinterpret_cast<const uint8_t*>(packet->data()), packet->size(),
+ webrtc_packet_time);
switch (delivery_result) {
case webrtc::PacketReceiver::DELIVERY_OK:
return;
@@ -1477,9 +1481,10 @@ void WebRtcVideoChannel2::OnPacketReceived(
break;
}
- if (call_->Receiver()->DeliverPacket(webrtc::MediaType::VIDEO,
- reinterpret_cast<const uint8_t*>(packet->data()), packet->size()) !=
- webrtc::PacketReceiver::DELIVERY_OK) {
+ if (call_->Receiver()->DeliverPacket(
+ webrtc::MediaType::VIDEO,
+ reinterpret_cast<const uint8_t*>(packet->data()), packet->size(),
+ webrtc_packet_time) != webrtc::PacketReceiver::DELIVERY_OK) {
LOG(LS_WARNING) << "Failed to deliver RTP packet on re-delivery.";
return;
}
@@ -1488,9 +1493,12 @@ void WebRtcVideoChannel2::OnPacketReceived(
void WebRtcVideoChannel2::OnRtcpReceived(
rtc::Buffer* packet,
const rtc::PacketTime& packet_time) {
- if (call_->Receiver()->DeliverPacket(webrtc::MediaType::VIDEO,
- reinterpret_cast<const uint8_t*>(packet->data()), packet->size()) !=
- webrtc::PacketReceiver::DELIVERY_OK) {
+ const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp,
+ packet_time.not_before);
+ if (call_->Receiver()->DeliverPacket(
+ webrtc::MediaType::VIDEO,
+ reinterpret_cast<const uint8_t*>(packet->data()), packet->size(),
+ webrtc_packet_time) != webrtc::PacketReceiver::DELIVERY_OK) {
LOG(LS_WARNING) << "Failed to deliver RTCP packet.";
}
}
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