Wire up PacketTime to ReceiveStreams.
BUG=webrtc:4758
Review URL: https://codereview.webrtc.org/1333483002
Cr-Commit-Position: refs/heads/master@{#9892}
diff --git a/talk/media/webrtc/fakewebrtccall.cc b/talk/media/webrtc/fakewebrtccall.cc
index 742c970..a85bdb1 100644
--- a/talk/media/webrtc/fakewebrtccall.cc
+++ b/talk/media/webrtc/fakewebrtccall.cc
@@ -314,9 +314,11 @@
return this;
}
-FakeCall::DeliveryStatus FakeCall::DeliverPacket(webrtc::MediaType media_type,
- const uint8_t* packet,
- size_t length) {
+FakeCall::DeliveryStatus FakeCall::DeliverPacket(
+ webrtc::MediaType media_type,
+ const uint8_t* packet,
+ size_t length,
+ const webrtc::PacketTime& packet_time) {
EXPECT_GE(length, 12u);
uint32_t ssrc;
if (!GetRtpSsrc(packet, length, &ssrc))
diff --git a/talk/media/webrtc/fakewebrtccall.h b/talk/media/webrtc/fakewebrtccall.h
index 5f21782..422848d 100644
--- a/talk/media/webrtc/fakewebrtccall.h
+++ b/talk/media/webrtc/fakewebrtccall.h
@@ -58,7 +58,9 @@
bool DeliverRtcp(const uint8_t* packet, size_t length) override {
return true;
}
- bool DeliverRtp(const uint8_t* packet, size_t length) override {
+ bool DeliverRtp(const uint8_t* packet,
+ size_t length,
+ const webrtc::PacketTime& packet_time) override {
return true;
}
@@ -136,7 +138,9 @@
bool DeliverRtcp(const uint8_t* packet, size_t length) override {
return true;
}
- bool DeliverRtp(const uint8_t* packet, size_t length) override {
+ bool DeliverRtp(const uint8_t* packet,
+ size_t length,
+ const webrtc::PacketTime& packet_time) override {
return true;
}
@@ -187,7 +191,9 @@
webrtc::PacketReceiver* Receiver() override;
DeliveryStatus DeliverPacket(webrtc::MediaType media_type,
- const uint8_t* packet, size_t length) override;
+ const uint8_t* packet,
+ size_t length,
+ const webrtc::PacketTime& packet_time) override;
webrtc::Call::Stats GetStats() const override;
diff --git a/talk/media/webrtc/webrtcvideoengine2.cc b/talk/media/webrtc/webrtcvideoengine2.cc
index 923fee6..7d41817 100644
--- a/talk/media/webrtc/webrtcvideoengine2.cc
+++ b/talk/media/webrtc/webrtcvideoengine2.cc
@@ -1451,9 +1451,13 @@
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;
@@ -1493,9 +1497,10 @@
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;
}
@@ -1504,9 +1509,12 @@
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.";
}
}
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
index 93f4b97..bcf1738 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -2949,8 +2949,12 @@
// If hooked up to a "Call", forward packet there too.
if (call_) {
- call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO,
- reinterpret_cast<const uint8_t*>(packet->data()), packet->size());
+ const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp,
+ packet_time.not_before);
+ call_->Receiver()->DeliverPacket(
+ webrtc::MediaType::AUDIO,
+ reinterpret_cast<const uint8_t*>(packet->data()), packet->size(),
+ webrtc_packet_time);
}
// Pick which channel to send this packet to. If this packet doesn't match
@@ -2990,8 +2994,12 @@
// If hooked up to a "Call", forward packet there too.
if (call_) {
- call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO,
- reinterpret_cast<const uint8_t*>(packet->data()), packet->size());
+ const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp,
+ packet_time.not_before);
+ call_->Receiver()->DeliverPacket(
+ webrtc::MediaType::AUDIO,
+ reinterpret_cast<const uint8_t*>(packet->data()), packet->size(),
+ webrtc_packet_time);
}
// Sending channels need all RTCP packets with feedback information.
diff --git a/webrtc/call.h b/webrtc/call.h
index 160a918..e426cc5 100644
--- a/webrtc/call.h
+++ b/webrtc/call.h
@@ -45,7 +45,9 @@
virtual DeliveryStatus DeliverPacket(MediaType media_type,
const uint8_t* packet,
- size_t length) = 0;
+ size_t length,
+ const PacketTime& packet_time) = 0;
+
protected:
virtual ~PacketReceiver() {}
};
diff --git a/webrtc/stream.h b/webrtc/stream.h
index fd30571..5afab0f 100644
--- a/webrtc/stream.h
+++ b/webrtc/stream.h
@@ -42,7 +42,9 @@
class ReceiveStream : public Stream {
public:
// Called when a RTP packet is received.
- virtual bool DeliverRtp(const uint8_t* packet, size_t length) = 0;
+ virtual bool DeliverRtp(const uint8_t* packet,
+ size_t length,
+ const PacketTime& packet_time) = 0;
};
// Common base class for send streams.
diff --git a/webrtc/test/fake_network_pipe.cc b/webrtc/test/fake_network_pipe.cc
index 7a63fa8..b3c9ee9 100644
--- a/webrtc/test/fake_network_pipe.cc
+++ b/webrtc/test/fake_network_pipe.cc
@@ -200,7 +200,7 @@
NetworkPacket* packet = packets_to_deliver.front();
packets_to_deliver.pop();
packet_receiver_->DeliverPacket(MediaType::ANY, packet->data(),
- packet->data_length());
+ packet->data_length(), PacketTime());
delete packet;
}
}
diff --git a/webrtc/test/fake_network_pipe_unittest.cc b/webrtc/test/fake_network_pipe_unittest.cc
index 6557343..a753fc5 100644
--- a/webrtc/test/fake_network_pipe_unittest.cc
+++ b/webrtc/test/fake_network_pipe_unittest.cc
@@ -29,12 +29,13 @@
virtual ~MockReceiver() {}
void IncomingPacket(const uint8_t* data, size_t length) {
- DeliverPacket(MediaType::ANY, data, length);
+ DeliverPacket(MediaType::ANY, data, length, PacketTime());
delete [] data;
}
- MOCK_METHOD3(DeliverPacket,
- DeliveryStatus(MediaType, const uint8_t*, size_t));
+ MOCK_METHOD4(
+ DeliverPacket,
+ DeliveryStatus(MediaType, const uint8_t*, size_t, const PacketTime&));
};
class FakeNetworkPipeTest : public ::testing::Test {
@@ -42,7 +43,7 @@
virtual void SetUp() {
TickTime::UseFakeClock(12345);
receiver_.reset(new MockReceiver());
- ON_CALL(*receiver_, DeliverPacket(_, _, _))
+ ON_CALL(*receiver_, DeliverPacket(_, _, _, _))
.WillByDefault(Return(PacketReceiver::DELIVERY_OK));
}
@@ -84,26 +85,22 @@
kPacketSize);
// Time haven't increased yet, so we souldn't get any packets.
- EXPECT_CALL(*receiver_, DeliverPacket(_, _, _))
- .Times(0);
+ EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(0);
pipe->Process();
// Advance enough time to release one packet.
TickTime::AdvanceFakeClock(kPacketTimeMs);
- EXPECT_CALL(*receiver_, DeliverPacket(_, _, _))
- .Times(1);
+ EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(1);
pipe->Process();
// Release all but one packet
TickTime::AdvanceFakeClock(9 * kPacketTimeMs - 1);
- EXPECT_CALL(*receiver_, DeliverPacket(_, _, _))
- .Times(8);
+ EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(8);
pipe->Process();
// And the last one.
TickTime::AdvanceFakeClock(1);
- EXPECT_CALL(*receiver_, DeliverPacket(_, _, _))
- .Times(1);
+ EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(1);
pipe->Process();
}
@@ -126,20 +123,17 @@
// Increase more than kPacketTimeMs, but not more than the extra delay.
TickTime::AdvanceFakeClock(kPacketTimeMs);
- EXPECT_CALL(*receiver_, DeliverPacket(_, _, _))
- .Times(0);
+ EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(0);
pipe->Process();
// Advance the network delay to get the first packet.
TickTime::AdvanceFakeClock(config.queue_delay_ms);
- EXPECT_CALL(*receiver_, DeliverPacket(_, _, _))
- .Times(1);
+ EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(1);
pipe->Process();
// Advance one more kPacketTimeMs to get the last packet.
TickTime::AdvanceFakeClock(kPacketTimeMs);
- EXPECT_CALL(*receiver_, DeliverPacket(_, _, _))
- .Times(1);
+ EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(1);
pipe->Process();
}
@@ -162,8 +156,7 @@
// Increase time enough to deliver all three packets, verify only two are
// delivered.
TickTime::AdvanceFakeClock(3 * kPacketTimeMs);
- EXPECT_CALL(*receiver_, DeliverPacket(_, _, _))
- .Times(2);
+ EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(2);
pipe->Process();
}
@@ -184,8 +177,7 @@
SendPackets(pipe.get(), 3, kPacketSize);
TickTime::AdvanceFakeClock(3 * kPacketTimeMs + config.queue_delay_ms);
- EXPECT_CALL(*receiver_, DeliverPacket(_, _, _))
- .Times(2);
+ EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(2);
pipe->Process();
// Packet 1: kPacketTimeMs + config.queue_delay_ms,
@@ -215,13 +207,13 @@
int packet_time_ms = PacketTimeMs(config.link_capacity_kbps, kPacketSize);
// Time hasn't increased yet, so we souldn't get any packets.
- EXPECT_CALL(*receiver_, DeliverPacket(_, _, _)).Times(0);
+ EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(0);
pipe->Process();
// Advance time in steps to release one packet at a time.
for (int i = 0; i < kNumPackets; ++i) {
TickTime::AdvanceFakeClock(packet_time_ms);
- EXPECT_CALL(*receiver_, DeliverPacket(_, _, _)).Times(1);
+ EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(1);
pipe->Process();
}
@@ -237,20 +229,20 @@
packet_time_ms = PacketTimeMs(config.link_capacity_kbps, kPacketSize);
// Time hasn't increased yet, so we souldn't get any packets.
- EXPECT_CALL(*receiver_, DeliverPacket(_, _, _)).Times(0);
+ EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(0);
pipe->Process();
// Advance time in steps to release one packet at a time.
for (int i = 0; i < kNumPackets; ++i) {
TickTime::AdvanceFakeClock(packet_time_ms);
- EXPECT_CALL(*receiver_, DeliverPacket(_, _, _)).Times(1);
+ EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(1);
pipe->Process();
}
// Check that all the packets were sent.
EXPECT_EQ(static_cast<size_t>(2 * kNumPackets), pipe->sent_packets());
TickTime::AdvanceFakeClock(pipe->TimeUntilNextProcess());
- EXPECT_CALL(*receiver_, DeliverPacket(_, _, _)).Times(0);
+ EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(0);
pipe->Process();
}
@@ -283,27 +275,27 @@
int packet_time_2_ms = PacketTimeMs(config.link_capacity_kbps, kPacketSize);
// Time hasn't increased yet, so we souldn't get any packets.
- EXPECT_CALL(*receiver_, DeliverPacket(_, _, _)).Times(0);
+ EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(0);
pipe->Process();
// Advance time in steps to release one packet at a time.
for (int i = 0; i < kNumPackets; ++i) {
TickTime::AdvanceFakeClock(packet_time_1_ms);
- EXPECT_CALL(*receiver_, DeliverPacket(_, _, _)).Times(1);
+ EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(1);
pipe->Process();
}
// Advance time in steps to release one packet at a time.
for (int i = 0; i < kNumPackets; ++i) {
TickTime::AdvanceFakeClock(packet_time_2_ms);
- EXPECT_CALL(*receiver_, DeliverPacket(_, _, _)).Times(1);
+ EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(1);
pipe->Process();
}
// Check that all the packets were sent.
EXPECT_EQ(static_cast<size_t>(2 * kNumPackets), pipe->sent_packets());
TickTime::AdvanceFakeClock(pipe->TimeUntilNextProcess());
- EXPECT_CALL(*receiver_, DeliverPacket(_, _, _)).Times(0);
+ EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(0);
pipe->Process();
}
} // namespace webrtc
diff --git a/webrtc/video/audio_receive_stream.cc b/webrtc/video/audio_receive_stream.cc
index 6731ea4..9b40002 100644
--- a/webrtc/video/audio_receive_stream.cc
+++ b/webrtc/video/audio_receive_stream.cc
@@ -87,7 +87,9 @@
return false;
}
-bool AudioReceiveStream::DeliverRtp(const uint8_t* packet, size_t length) {
+bool AudioReceiveStream::DeliverRtp(const uint8_t* packet,
+ size_t length,
+ const PacketTime& packet_time) {
RTPHeader header;
if (!rtp_header_parser_->Parse(packet, length, &header)) {
@@ -99,6 +101,8 @@
if (config_.combined_audio_video_bwe &&
header.extension.hasAbsoluteSendTime) {
int64_t arrival_time_ms = TickTime::MillisecondTimestamp();
+ if (packet_time.timestamp >= 0)
+ arrival_time_ms = packet_time.timestamp;
size_t payload_size = length - header.headerLength;
remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_size,
header, false);
diff --git a/webrtc/video/audio_receive_stream.h b/webrtc/video/audio_receive_stream.h
index 08d929a..c9ac04a 100644
--- a/webrtc/video/audio_receive_stream.h
+++ b/webrtc/video/audio_receive_stream.h
@@ -31,7 +31,9 @@
void Stop() override;
void SignalNetworkState(NetworkState state) override;
bool DeliverRtcp(const uint8_t* packet, size_t length) override;
- bool DeliverRtp(const uint8_t* packet, size_t length) override;
+ bool DeliverRtp(const uint8_t* packet,
+ size_t length,
+ const PacketTime& packet_time) override;
// webrtc::AudioReceiveStream implementation.
webrtc::AudioReceiveStream::Stats GetStats() const override;
diff --git a/webrtc/video/call.cc b/webrtc/video/call.cc
index 6c58fc1..f3a5db4 100644
--- a/webrtc/video/call.cc
+++ b/webrtc/video/call.cc
@@ -69,8 +69,10 @@
Stats GetStats() const override;
- DeliveryStatus DeliverPacket(MediaType media_type, const uint8_t* packet,
- size_t length) override;
+ DeliveryStatus DeliverPacket(MediaType media_type,
+ const uint8_t* packet,
+ size_t length,
+ const PacketTime& packet_time) override;
void SetBitrateConfig(
const webrtc::Call::Config::BitrateConfig& bitrate_config) override;
@@ -79,8 +81,10 @@
private:
DeliveryStatus DeliverRtcp(MediaType media_type, const uint8_t* packet,
size_t length);
- DeliveryStatus DeliverRtp(MediaType media_type, const uint8_t* packet,
- size_t length);
+ DeliveryStatus DeliverRtp(MediaType media_type,
+ const uint8_t* packet,
+ size_t length,
+ const PacketTime& packet_time);
void SetBitrateControllerConfig(
const webrtc::Call::Config::BitrateConfig& bitrate_config);
@@ -475,7 +479,8 @@
PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
const uint8_t* packet,
- size_t length) {
+ size_t length,
+ const PacketTime& packet_time) {
// Minimum RTP header size.
if (length < 12)
return DELIVERY_PACKET_ERROR;
@@ -486,27 +491,31 @@
if (media_type == MediaType::ANY || media_type == MediaType::AUDIO) {
auto it = audio_receive_ssrcs_.find(ssrc);
if (it != audio_receive_ssrcs_.end()) {
- return it->second->DeliverRtp(packet, length) ? DELIVERY_OK
- : DELIVERY_PACKET_ERROR;
+ return it->second->DeliverRtp(packet, length, packet_time)
+ ? DELIVERY_OK
+ : DELIVERY_PACKET_ERROR;
}
}
if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) {
auto it = video_receive_ssrcs_.find(ssrc);
if (it != video_receive_ssrcs_.end()) {
- return it->second->DeliverRtp(packet, length) ? DELIVERY_OK
- : DELIVERY_PACKET_ERROR;
+ return it->second->DeliverRtp(packet, length, packet_time)
+ ? DELIVERY_OK
+ : DELIVERY_PACKET_ERROR;
}
}
return DELIVERY_UNKNOWN_SSRC;
}
-PacketReceiver::DeliveryStatus Call::DeliverPacket(MediaType media_type,
- const uint8_t* packet,
- size_t length) {
+PacketReceiver::DeliveryStatus Call::DeliverPacket(
+ MediaType media_type,
+ const uint8_t* packet,
+ size_t length,
+ const PacketTime& packet_time) {
if (RtpHeaderParser::IsRtcp(packet, length))
return DeliverRtcp(media_type, packet, length);
- return DeliverRtp(media_type, packet, length);
+ return DeliverRtp(media_type, packet, length, packet_time);
}
} // namespace internal
diff --git a/webrtc/video/call_perf_tests.cc b/webrtc/video/call_perf_tests.cc
index 592b68e..a301452 100644
--- a/webrtc/video/call_perf_tests.cc
+++ b/webrtc/video/call_perf_tests.cc
@@ -197,8 +197,10 @@
: channel_(channel),
voe_network_(voe_network),
parser_(RtpHeaderParser::Create()) {}
- DeliveryStatus DeliverPacket(MediaType media_type, const uint8_t* packet,
- size_t length) override {
+ DeliveryStatus DeliverPacket(MediaType media_type,
+ const uint8_t* packet,
+ size_t length,
+ const PacketTime& packet_time) override {
EXPECT_TRUE(media_type == MediaType::ANY ||
media_type == MediaType::AUDIO);
int ret;
@@ -540,8 +542,10 @@
test::RtpRtcpObserver::SetReceivers(this, receive_transport_receiver);
}
- DeliveryStatus DeliverPacket(MediaType media_type, const uint8_t* packet,
- size_t length) override {
+ DeliveryStatus DeliverPacket(MediaType media_type,
+ const uint8_t* packet,
+ size_t length,
+ const PacketTime& packet_time) override {
VideoSendStream::Stats stats = send_stream_->GetStats();
if (stats.substreams.size() > 0) {
DCHECK_EQ(1u, stats.substreams.size());
@@ -575,8 +579,8 @@
observation_complete_->Set();
}
}
- return send_transport_receiver_->DeliverPacket(media_type, packet,
- length);
+ return send_transport_receiver_->DeliverPacket(media_type, packet, length,
+ packet_time);
}
void OnStreamsCreated(
diff --git a/webrtc/video/end_to_end_tests.cc b/webrtc/video/end_to_end_tests.cc
index 9f62ec8..a71c2e0 100644
--- a/webrtc/video/end_to_end_tests.cc
+++ b/webrtc/video/end_to_end_tests.cc
@@ -993,13 +993,16 @@
}
private:
- DeliveryStatus DeliverPacket(MediaType media_type, const uint8_t* packet,
- size_t length) override {
+ DeliveryStatus DeliverPacket(MediaType media_type,
+ const uint8_t* packet,
+ size_t length,
+ const PacketTime& packet_time) override {
if (RtpHeaderParser::IsRtcp(packet, length)) {
- return receiver_->DeliverPacket(media_type, packet, length);
+ return receiver_->DeliverPacket(media_type, packet, length,
+ packet_time);
} else {
DeliveryStatus delivery_status =
- receiver_->DeliverPacket(media_type, packet, length);
+ receiver_->DeliverPacket(media_type, packet, length, packet_time);
EXPECT_EQ(DELIVERY_UNKNOWN_SSRC, delivery_status);
delivered_packet_->Set();
return delivery_status;
@@ -1552,8 +1555,10 @@
receiver_call_(nullptr),
has_seen_pacer_delay_(false) {}
- DeliveryStatus DeliverPacket(MediaType media_type, const uint8_t* packet,
- size_t length) override {
+ DeliveryStatus DeliverPacket(MediaType media_type,
+ const uint8_t* packet,
+ size_t length,
+ const PacketTime& packet_time) override {
Call::Stats sender_stats = sender_call_->GetStats();
Call::Stats receiver_stats = receiver_call_->GetStats();
if (!has_seen_pacer_delay_)
@@ -1563,7 +1568,7 @@
observation_complete_->Set();
}
return receiver_call_->Receiver()->DeliverPacket(media_type, packet,
- length);
+ length, packet_time);
}
void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
@@ -1719,15 +1724,17 @@
return SEND_PACKET;
}
- DeliveryStatus DeliverPacket(MediaType media_type, const uint8_t* packet,
- size_t length) override {
+ DeliveryStatus DeliverPacket(MediaType media_type,
+ const uint8_t* packet,
+ size_t length,
+ const PacketTime& packet_time) override {
// GetStats calls GetSendChannelRtcpStatistics
// (via VideoSendStream::GetRtt) which updates ReportBlockStats used by
// WebRTC.Video.SentPacketsLostInPercent.
// TODO(asapersson): Remove dependency on calling GetStats.
sender_call_->GetStats();
return receiver_call_->Receiver()->DeliverPacket(media_type, packet,
- length);
+ length, packet_time);
}
bool MinMetricRunTimePassed() {
diff --git a/webrtc/video/full_stack.cc b/webrtc/video/full_stack.cc
index 45c28ad..1fee087 100644
--- a/webrtc/video/full_stack.cc
+++ b/webrtc/video/full_stack.cc
@@ -109,8 +109,10 @@
virtual void SetReceiver(PacketReceiver* receiver) { receiver_ = receiver; }
- DeliveryStatus DeliverPacket(MediaType media_type, const uint8_t* packet,
- size_t length) override {
+ DeliveryStatus DeliverPacket(MediaType media_type,
+ const uint8_t* packet,
+ size_t length,
+ const PacketTime& packet_time) override {
rtc::scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
RTPHeader header;
parser->Parse(packet, length, &header);
@@ -120,7 +122,7 @@
Clock::GetRealTimeClock()->CurrentNtpInMilliseconds();
}
- return receiver_->DeliverPacket(media_type, packet, length);
+ return receiver_->DeliverPacket(media_type, packet, length, packet_time);
}
void IncomingCapturedFrame(const VideoFrame& video_frame) override {
diff --git a/webrtc/video/packet_injection_tests.cc b/webrtc/video/packet_injection_tests.cc
index 133935c..18ca058 100644
--- a/webrtc/video/packet_injection_tests.cc
+++ b/webrtc/video/packet_injection_tests.cc
@@ -61,7 +61,7 @@
Start();
EXPECT_EQ(PacketReceiver::DELIVERY_PACKET_ERROR,
receiver_call_->Receiver()->DeliverPacket(MediaType::VIDEO, packet,
- length));
+ length, PacketTime()));
Stop();
DestroyStreams();
diff --git a/webrtc/video/rampup_tests.cc b/webrtc/video/rampup_tests.cc
index 92b55bf..fb533cb 100644
--- a/webrtc/video/rampup_tests.cc
+++ b/webrtc/video/rampup_tests.cc
@@ -265,7 +265,10 @@
}
PacketReceiver::DeliveryStatus LowRateStreamObserver::DeliverPacket(
- MediaType media_type, const uint8_t* packet, size_t length) {
+ MediaType media_type,
+ const uint8_t* packet,
+ size_t length,
+ const PacketTime& packet_time) {
rtc::CritScope lock(&crit_);
RTPHeader header;
EXPECT_TRUE(rtp_parser_->Parse(packet, length, &header));
diff --git a/webrtc/video/rampup_tests.h b/webrtc/video/rampup_tests.h
index ae9e9d9..56c5e75 100644
--- a/webrtc/video/rampup_tests.h
+++ b/webrtc/video/rampup_tests.h
@@ -102,8 +102,10 @@
bool SendRtp(const uint8_t* data, size_t length) override;
- DeliveryStatus DeliverPacket(MediaType media_type, const uint8_t* packet,
- size_t length) override;
+ DeliveryStatus DeliverPacket(MediaType media_type,
+ const uint8_t* packet,
+ size_t length,
+ const PacketTime& packet_time) override;
bool SendRtcp(const uint8_t* packet, size_t length) override;
diff --git a/webrtc/video/replay.cc b/webrtc/video/replay.cc
index 2a8a0a8..6f0703b 100644
--- a/webrtc/video/replay.cc
+++ b/webrtc/video/replay.cc
@@ -285,7 +285,7 @@
break;
++num_packets;
switch (call->Receiver()->DeliverPacket(webrtc::MediaType::ANY, packet.data,
- packet.length)) {
+ packet.length, PacketTime())) {
case PacketReceiver::DELIVERY_OK:
break;
case PacketReceiver::DELIVERY_UNKNOWN_SSRC: {
diff --git a/webrtc/video/video_receive_stream.cc b/webrtc/video/video_receive_stream.cc
index 5737ca9..9f0e26f 100644
--- a/webrtc/video/video_receive_stream.cc
+++ b/webrtc/video/video_receive_stream.cc
@@ -290,8 +290,10 @@
return vie_channel_->ReceivedRTCPPacket(packet, length) == 0;
}
-bool VideoReceiveStream::DeliverRtp(const uint8_t* packet, size_t length) {
- return vie_channel_->ReceivedRTPPacket(packet, length, PacketTime()) == 0;
+bool VideoReceiveStream::DeliverRtp(const uint8_t* packet,
+ size_t length,
+ const PacketTime& packet_time) {
+ return vie_channel_->ReceivedRTPPacket(packet, length, packet_time) == 0;
}
void VideoReceiveStream::FrameCallback(VideoFrame* video_frame) {
diff --git a/webrtc/video/video_receive_stream.h b/webrtc/video/video_receive_stream.h
index 47a4d60..1574238 100644
--- a/webrtc/video/video_receive_stream.h
+++ b/webrtc/video/video_receive_stream.h
@@ -49,7 +49,9 @@
void Stop() override;
void SignalNetworkState(NetworkState state) override;
bool DeliverRtcp(const uint8_t* packet, size_t length) override;
- bool DeliverRtp(const uint8_t* packet, size_t length) override;
+ bool DeliverRtp(const uint8_t* packet,
+ size_t length,
+ const PacketTime& packet_time) override;
// webrtc::VideoReceiveStream implementation.
webrtc::VideoReceiveStream::Stats GetStats() const override;
diff --git a/webrtc/video/video_send_stream_tests.cc b/webrtc/video/video_send_stream_tests.cc
index 0890a10..c558099 100644
--- a/webrtc/video/video_send_stream_tests.cc
+++ b/webrtc/video/video_send_stream_tests.cc
@@ -957,8 +957,10 @@
}
private:
- DeliveryStatus DeliverPacket(MediaType media_type, const uint8_t* packet,
- size_t length) override {
+ DeliveryStatus DeliverPacket(MediaType media_type,
+ const uint8_t* packet,
+ size_t length,
+ const PacketTime& packet_time) override {
EXPECT_TRUE(media_type == MediaType::ANY ||
media_type == MediaType::VIDEO);
if (RtpHeaderParser::IsRtcp(packet, length))
