Index: webrtc/video/rtp_stream_receiver.cc |
diff --git a/webrtc/video/rtp_stream_receiver.cc b/webrtc/video/rtp_stream_receiver.cc |
index d49d0959f32fc826dd3da0cb26d7dda3fe1e0079..0025539fba20968b2af16cc707a60b588e47213a 100644 |
--- a/webrtc/video/rtp_stream_receiver.cc |
+++ b/webrtc/video/rtp_stream_receiver.cc |
@@ -26,6 +26,7 @@ |
#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h" |
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" |
#include "webrtc/modules/rtp_rtcp/include/ulpfec_receiver.h" |
+#include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h" |
#include "webrtc/modules/video_coding/frame_object.h" |
#include "webrtc/modules/video_coding/h264_sprop_parameter_sets.h" |
#include "webrtc/modules/video_coding/h264_sps_pps_tracker.h" |
@@ -316,9 +317,7 @@ void RtpStreamReceiver::OnIncomingSSRCChanged(const uint32_t ssrc) { |
rtp_rtcp_->SetRemoteSSRC(ssrc); |
} |
-bool RtpStreamReceiver::DeliverRtp(const uint8_t* rtp_packet, |
- size_t rtp_packet_length, |
- const PacketTime& packet_time) { |
+bool RtpStreamReceiver::OnRtpPacket(const RtpPacketReceived& packet) { |
{ |
rtc::CritScope lock(&receive_cs_); |
if (!receiving_) { |
@@ -327,16 +326,9 @@ bool RtpStreamReceiver::DeliverRtp(const uint8_t* rtp_packet, |
} |
RTPHeader header; |
- if (!rtp_header_parser_->Parse(rtp_packet, rtp_packet_length, |
- &header)) { |
- return false; |
- } |
- int64_t arrival_time_ms; |
+ packet.GetHeader(&header); |
+ |
int64_t now_ms = clock_->TimeInMilliseconds(); |
- if (packet_time.timestamp != -1) |
- arrival_time_ms = (packet_time.timestamp + 500) / 1000; |
- else |
- arrival_time_ms = now_ms; |
{ |
// Periodically log the RTP header of incoming packets. |
@@ -346,7 +338,7 @@ bool RtpStreamReceiver::DeliverRtp(const uint8_t* rtp_packet, |
ss << "Packet received on SSRC: " << header.ssrc << " with payload type: " |
brandtr
2017/02/09 14:22:17
Might as well remove the usage of |header| here, t
nisse-webrtc
2017/02/10 10:26:55
Done, except that (i) the ReceivePacket and Incomi
|
<< static_cast<int>(header.payloadType) << ", timestamp: " |
<< header.timestamp << ", sequence number: " << header.sequenceNumber |
- << ", arrival time: " << arrival_time_ms; |
+ << ", arrival time: " << packet.arrival_time_ms(); |
if (header.extension.hasTransmissionTimeOffset) |
ss << ", toffset: " << header.extension.transmissionTimeOffset; |
if (header.extension.hasAbsoluteSendTime) |
@@ -360,12 +352,13 @@ bool RtpStreamReceiver::DeliverRtp(const uint8_t* rtp_packet, |
bool in_order = IsPacketInOrder(header); |
rtp_payload_registry_.SetIncomingPayloadType(header); |
- bool ret = ReceivePacket(rtp_packet, rtp_packet_length, header, in_order); |
+ // TODO(nisse): Is .data() and .size() right? Strip headers or not? |
+ bool ret = ReceivePacket(packet.data(), packet.size(), header, in_order); |
brandtr
2017/02/09 14:22:17
I believe this is correct.
If it were incorrect,
nisse-webrtc
2017/02/10 10:26:55
We'll see what happens to the tests.
|
// Update receive statistics after ReceivePacket. |
// Receive statistics will be reset if the payload type changes (make sure |
// that the first packet is included in the stats). |
rtp_receive_statistics_->IncomingPacket( |
- header, rtp_packet_length, IsPacketRetransmitted(header, in_order)); |
+ header, packet.size(), IsPacketRetransmitted(header, in_order)); |
return ret; |
} |