Add a PacketOptions struct to webrtc::Transport.
This allows us to pass packet meta data, such as transport sequence
number, to libjingle and further down to the socket implementation. A
similar struct already exist in libjingle, see rtc::PacketOptions in asyncpacketsocket.h.
BUG=4173
Review URL: https://codereview.webrtc.org/1376673004
Cr-Commit-Position: refs/heads/master@{#10144}
diff --git a/talk/media/webrtc/webrtcvideoengine2.cc b/talk/media/webrtc/webrtcvideoengine2.cc
index 7b29001..2f06340 100644
--- a/talk/media/webrtc/webrtcvideoengine2.cc
+++ b/talk/media/webrtc/webrtcvideoengine2.cc
@@ -1658,7 +1658,9 @@
}
}
-bool WebRtcVideoChannel2::SendRtp(const uint8_t* data, size_t len) {
+bool WebRtcVideoChannel2::SendRtp(const uint8_t* data,
+ size_t len,
+ const webrtc::PacketOptions& options) {
rtc::Buffer packet(data, len, kMaxRtpPacketLen);
return MediaChannel::SendPacket(&packet);
}
diff --git a/talk/media/webrtc/webrtcvideoengine2.h b/talk/media/webrtc/webrtcvideoengine2.h
index 6b36642..0fc28d1 100644
--- a/talk/media/webrtc/webrtcvideoengine2.h
+++ b/talk/media/webrtc/webrtcvideoengine2.h
@@ -464,7 +464,9 @@
void Construct(webrtc::Call* call, WebRtcVideoEngine2* engine);
void SetDefaultOptions();
- bool SendRtp(const uint8_t* data, size_t len) override;
+ bool SendRtp(const uint8_t* data,
+ size_t len,
+ const webrtc::PacketOptions& options) override;
bool SendRtcp(const uint8_t* data, size_t len) override;
void StartAllSendStreams();
diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h
index 7950024..69e88e4 100644
--- a/talk/media/webrtc/webrtcvoiceengine.h
+++ b/talk/media/webrtc/webrtcvoiceengine.h
@@ -223,7 +223,9 @@
bool GetStats(VoiceMediaInfo* info) override;
// implements Transport interface
- bool SendRtp(const uint8_t* data, size_t len) override {
+ bool SendRtp(const uint8_t* data,
+ size_t len,
+ const webrtc::PacketOptions& options) override {
rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len,
kMaxRtpPacketLen);
return VoiceMediaChannel::SendPacket(&packet);
diff --git a/webrtc/call/transport_adapter.cc b/webrtc/call/transport_adapter.cc
index e57591a..5e59b7b 100644
--- a/webrtc/call/transport_adapter.cc
+++ b/webrtc/call/transport_adapter.cc
@@ -20,11 +20,13 @@
RTC_DCHECK(nullptr != transport);
}
-bool TransportAdapter::SendRtp(const uint8_t* packet, size_t length) {
+bool TransportAdapter::SendRtp(const uint8_t* packet,
+ size_t length,
+ const PacketOptions& options) {
if (enabled_.Value() == 0)
return false;
- return transport_->SendRtp(packet, length);
+ return transport_->SendRtp(packet, length, options);
}
bool TransportAdapter::SendRtcp(const uint8_t* packet, size_t length) {
diff --git a/webrtc/call/transport_adapter.h b/webrtc/call/transport_adapter.h
index 07ea716..72f9baa 100644
--- a/webrtc/call/transport_adapter.h
+++ b/webrtc/call/transport_adapter.h
@@ -21,7 +21,9 @@
public:
explicit TransportAdapter(Transport* transport);
- bool SendRtp(const uint8_t* packet, size_t length) override;
+ bool SendRtp(const uint8_t* packet,
+ size_t length,
+ const PacketOptions& options) override;
bool SendRtcp(const uint8_t* packet, size_t length) override;
void Enable();
diff --git a/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc b/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
index b243385..483ee13 100644
--- a/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
@@ -96,7 +96,9 @@
packet_loss_ = 0;
}
- bool SendRtp(const uint8_t* data, size_t len) override {
+ bool SendRtp(const uint8_t* data,
+ size_t len,
+ const PacketOptions& options) override {
count_++;
const unsigned char* ptr = static_cast<const unsigned char*>(data);
uint32_t ssrc = (ptr[8] << 24) + (ptr[9] << 16) + (ptr[10] << 8) + ptr[11];
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc
index 554e6ad..7a7645f 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc
@@ -32,7 +32,9 @@
rtcp_receiver_(rtcp_receiver) {
}
- bool SendRtp(const uint8_t* /*data*/, size_t /*len*/) override {
+ bool SendRtp(const uint8_t* /*data*/,
+ size_t /*len*/,
+ const PacketOptions& options) override {
return false;
}
bool SendRtcp(const uint8_t* packet, size_t packetLength) override {
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
index cdbc47d..1581845 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
@@ -39,7 +39,9 @@
void SetRTCPReceiver(RTCPReceiver* rtcp_receiver) {
rtcp_receiver_ = rtcp_receiver;
}
- bool SendRtp(const uint8_t* /*data*/, size_t /*len*/) override {
+ bool SendRtp(const uint8_t* /*data*/,
+ size_t /*len*/,
+ const PacketOptions& options) override {
ADD_FAILURE(); // FAIL() gives a compile error.
return false;
}
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
index 0da2101..ba42c8d 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
@@ -199,7 +199,9 @@
public:
TestTransport() {}
- bool SendRtp(const uint8_t* /*data*/, size_t /*len*/) override {
+ bool SendRtp(const uint8_t* /*data*/,
+ size_t /*len*/,
+ const PacketOptions& options) override {
return false;
}
bool SendRtcp(const uint8_t* data, size_t len) override {
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
index f6103b3..03b91a9 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
@@ -61,7 +61,9 @@
clock_ = clock;
delay_ms_ = delay_ms;
}
- bool SendRtp(const uint8_t* data, size_t len) override {
+ bool SendRtp(const uint8_t* data,
+ size_t len,
+ const PacketOptions& options) override {
RTPHeader header;
rtc::scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
EXPECT_TRUE(parser->Parse(static_cast<const uint8_t*>(data), len, &header));
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index ae292a0..252ffb2 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -662,18 +662,18 @@
UpdateAbsoluteSendTime(padding_packet, length, rtp_header, now_ms);
- uint16_t transport_seq = 0;
+ PacketOptions options;
if (using_transport_seq) {
- transport_seq =
+ options.packet_id =
UpdateTransportSequenceNumber(padding_packet, length, rtp_header);
}
- if (!SendPacketToNetwork(padding_packet, length))
+ if (!SendPacketToNetwork(padding_packet, length, options))
break;
if (using_transport_seq && transport_feedback_observer_) {
- transport_feedback_observer_->OnPacketSent(
- PacketInfo(0, now_ms, transport_seq, length, true));
+ transport_feedback_observer_->OnPacketSent(PacketInfo(
+ 0, now_ms, options.packet_id, length, true));
}
bytes_sent += padding_bytes_in_packet;
@@ -732,11 +732,14 @@
return static_cast<int32_t>(length);
}
-bool RTPSender::SendPacketToNetwork(const uint8_t *packet, size_t size) {
+bool RTPSender::SendPacketToNetwork(const uint8_t* packet,
+ size_t size,
+ const PacketOptions& options) {
int bytes_sent = -1;
if (transport_) {
- bytes_sent =
- transport_->SendRtp(packet, size) ? static_cast<int>(size) : -1;
+ bytes_sent = transport_->SendRtp(packet, size, options)
+ ? static_cast<int>(size)
+ : -1;
}
TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
"RTPSender::SendPacketToNetwork", "size", size, "sent",
@@ -916,25 +919,25 @@
diff_ms);
UpdateAbsoluteSendTime(buffer_to_send_ptr, length, rtp_header, now_ms);
- uint16_t transport_seq = 0;
// TODO(sprang): Potentially too much overhead in IsRegistered()?
bool using_transport_seq = rtp_header_extension_map_.IsRegistered(
kRtpExtensionTransportSequenceNumber) &&
transport_sequence_number_allocator_ &&
!is_retransmit;
+ PacketOptions options;
if (using_transport_seq) {
- transport_seq =
+ options.packet_id =
UpdateTransportSequenceNumber(buffer_to_send_ptr, length, rtp_header);
}
- bool ret = SendPacketToNetwork(buffer_to_send_ptr, length);
+ bool ret = SendPacketToNetwork(buffer_to_send_ptr, length, options);
if (ret) {
CriticalSectionScoped lock(send_critsect_.get());
media_has_been_sent_ = true;
}
if (using_transport_seq && transport_feedback_observer_) {
transport_feedback_observer_->OnPacketSent(
- PacketInfo(0, now_ms, transport_seq, length, true));
+ PacketInfo(0, now_ms, options.packet_id, length, true));
}
UpdateRtpStats(buffer_to_send_ptr, length, rtp_header, send_over_rtx,
is_retransmit);
@@ -1058,7 +1061,7 @@
}
size_t length = payload_length + rtp_header_length;
- bool sent = SendPacketToNetwork(buffer, length);
+ bool sent = SendPacketToNetwork(buffer, length, PacketOptions());
if (storage != kDontStore) {
// Mark the packet as sent in the history even if send failed. Dropping a
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.h b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
index 564300f..57516f8 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
@@ -350,7 +350,9 @@
void BuildRtxPacket(uint8_t* buffer, size_t* length,
uint8_t* buffer_rtx);
- bool SendPacketToNetwork(const uint8_t *packet, size_t size);
+ bool SendPacketToNetwork(const uint8_t* packet,
+ size_t size,
+ const PacketOptions& options);
void UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms);
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index 0403ce8..305ea13 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -78,7 +78,9 @@
~LoopbackTransportTest() {
STLDeleteContainerPointers(sent_packets_.begin(), sent_packets_.end());
}
- bool SendRtp(const uint8_t *data, size_t len) override {
+ bool SendRtp(const uint8_t* data,
+ size_t len,
+ const PacketOptions& options) override {
packets_sent_++;
rtc::Buffer* buffer =
new rtc::Buffer(reinterpret_cast<const uint8_t*>(data), len);
@@ -864,25 +866,26 @@
// Send 10 packets of increasing size.
for (size_t i = 0; i < kNumPayloadSizes; ++i) {
int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
- EXPECT_CALL(transport, SendRtp(_, _)).WillOnce(testing::Return(true));
+ EXPECT_CALL(transport, SendRtp(_, _, _)).WillOnce(testing::Return(true));
SendPacket(capture_time_ms, kPayloadSizes[i]);
rtp_sender_->TimeToSendPacket(seq_num++, capture_time_ms, false);
fake_clock_.AdvanceTimeMilliseconds(33);
}
// The amount of padding to send it too small to send a payload packet.
- EXPECT_CALL(transport, SendRtp(_, kMaxPaddingSize + rtp_header_len))
+ EXPECT_CALL(transport, SendRtp(_, kMaxPaddingSize + rtp_header_len, _))
.WillOnce(testing::Return(true));
EXPECT_EQ(kMaxPaddingSize, rtp_sender_->TimeToSendPadding(49));
EXPECT_CALL(transport,
- SendRtp(_, kPayloadSizes[0] + rtp_header_len + kRtxHeaderSize))
+ SendRtp(_, kPayloadSizes[0] + rtp_header_len + kRtxHeaderSize, _))
.WillOnce(testing::Return(true));
EXPECT_EQ(kPayloadSizes[0], rtp_sender_->TimeToSendPadding(500));
EXPECT_CALL(transport, SendRtp(_, kPayloadSizes[kNumPayloadSizes - 1] +
- rtp_header_len + kRtxHeaderSize))
+ rtp_header_len + kRtxHeaderSize,
+ _))
.WillOnce(testing::Return(true));
- EXPECT_CALL(transport, SendRtp(_, kMaxPaddingSize + rtp_header_len))
+ EXPECT_CALL(transport, SendRtp(_, kMaxPaddingSize + rtp_header_len, _))
.WillOnce(testing::Return(true));
EXPECT_EQ(kPayloadSizes[kNumPayloadSizes - 1] + kMaxPaddingSize,
rtp_sender_->TimeToSendPadding(999));
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
index 361f45d..0270e55 100644
--- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
@@ -31,7 +31,9 @@
packet_loss_ = n;
}
-bool LoopBackTransport::SendRtp(const uint8_t* data, size_t len) {
+bool LoopBackTransport::SendRtp(const uint8_t* data,
+ size_t len,
+ const PacketOptions& options) {
count_++;
if (packet_loss_ > 0) {
if ((count_ % packet_loss_) == 0) {
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api.h b/webrtc/modules/rtp_rtcp/test/testAPI/test_api.h
index 8397e63..73334a8 100644
--- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api.h
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api.h
@@ -36,7 +36,9 @@
RtpReceiver* receiver,
ReceiveStatistics* receive_statistics);
void DropEveryNthPacket(int n);
- bool SendRtp(const uint8_t* data, size_t len) override;
+ bool SendRtp(const uint8_t* data,
+ size_t len,
+ const PacketOptions& options) override;
bool SendRtcp(const uint8_t* data, size_t len) override;
private:
diff --git a/webrtc/test/channel_transport/udp_transport_impl.cc b/webrtc/test/channel_transport/udp_transport_impl.cc
index f66f4ef..a530ec6 100644
--- a/webrtc/test/channel_transport/udp_transport_impl.cc
+++ b/webrtc/test/channel_transport/udp_transport_impl.cc
@@ -1931,7 +1931,9 @@
return -1;
}
-bool UdpTransportImpl::SendRtp(const uint8_t* data, size_t length) {
+bool UdpTransportImpl::SendRtp(const uint8_t* data,
+ size_t length,
+ const PacketOptions& packet_options) {
WEBRTC_TRACE(kTraceStream, kTraceTransport, _id, "%s", __FUNCTION__);
CriticalSectionScoped cs(_crit);
diff --git a/webrtc/test/channel_transport/udp_transport_impl.h b/webrtc/test/channel_transport/udp_transport_impl.h
index 80562f3..f80ee02 100644
--- a/webrtc/test/channel_transport/udp_transport_impl.h
+++ b/webrtc/test/channel_transport/udp_transport_impl.h
@@ -116,7 +116,9 @@
size_t length,
uint16_t rtcpPort) override;
// Transport functions
- bool SendRtp(const uint8_t* data, size_t length) override;
+ bool SendRtp(const uint8_t* data,
+ size_t length,
+ const PacketOptions& packet_options) override;
bool SendRtcp(const uint8_t* data, size_t length) override;
// UdpTransport functions continue.
diff --git a/webrtc/test/direct_transport.cc b/webrtc/test/direct_transport.cc
index d0f797c..1f92e92 100644
--- a/webrtc/test/direct_transport.cc
+++ b/webrtc/test/direct_transport.cc
@@ -57,7 +57,9 @@
fake_network_.SetReceiver(receiver);
}
-bool DirectTransport::SendRtp(const uint8_t* data, size_t length) {
+bool DirectTransport::SendRtp(const uint8_t* data,
+ size_t length,
+ const PacketOptions& options) {
fake_network_.SendPacket(data, length);
packet_event_->Set();
return true;
diff --git a/webrtc/test/direct_transport.h b/webrtc/test/direct_transport.h
index a0aaecb..1c6f937 100644
--- a/webrtc/test/direct_transport.h
+++ b/webrtc/test/direct_transport.h
@@ -39,7 +39,9 @@
virtual void StopSending();
virtual void SetReceiver(PacketReceiver* receiver);
- bool SendRtp(const uint8_t* data, size_t length) override;
+ bool SendRtp(const uint8_t* data,
+ size_t length,
+ const PacketOptions& options) override;
bool SendRtcp(const uint8_t* data, size_t length) override;
private:
diff --git a/webrtc/test/layer_filtering_transport.cc b/webrtc/test/layer_filtering_transport.cc
index 5ad3f8c..d6f3591 100644
--- a/webrtc/test/layer_filtering_transport.cc
+++ b/webrtc/test/layer_filtering_transport.cc
@@ -33,10 +33,12 @@
current_seq_num_(10000) {
} // TODO(ivica): random seq num?
-bool LayerFilteringTransport::SendRtp(const uint8_t* packet, size_t length) {
+bool LayerFilteringTransport::SendRtp(const uint8_t* packet,
+ size_t length,
+ const PacketOptions& options) {
if (tl_discard_threshold_ == 0 && sl_discard_threshold_ == 0) {
// Nothing to change, forward the packet immediately.
- return test::DirectTransport::SendRtp(packet, length);
+ return test::DirectTransport::SendRtp(packet, length, options);
}
bool set_marker_bit = false;
@@ -93,7 +95,7 @@
++current_seq_num_; // Increase only if packet not discarded.
- return test::DirectTransport::SendRtp(temp_buffer, length);
+ return test::DirectTransport::SendRtp(temp_buffer, length, options);
}
} // namespace test
diff --git a/webrtc/test/layer_filtering_transport.h b/webrtc/test/layer_filtering_transport.h
index 96a2cba..68994ae 100644
--- a/webrtc/test/layer_filtering_transport.h
+++ b/webrtc/test/layer_filtering_transport.h
@@ -24,7 +24,9 @@
uint8_t vp9_video_payload_type,
uint8_t tl_discard_threshold,
uint8_t sl_discard_threshold);
- bool SendRtp(const uint8_t* data, size_t length) override;
+ bool SendRtp(const uint8_t* data,
+ size_t length,
+ const PacketOptions& options) override;
private:
// Used to distinguish between VP8 and VP9.
diff --git a/webrtc/test/mock_transport.h b/webrtc/test/mock_transport.h
index ca08809..4937134 100644
--- a/webrtc/test/mock_transport.h
+++ b/webrtc/test/mock_transport.h
@@ -18,7 +18,10 @@
class MockTransport : public Transport {
public:
- MOCK_METHOD2(SendRtp, bool(const uint8_t* data, size_t len));
+ MOCK_METHOD3(SendRtp,
+ bool(const uint8_t* data,
+ size_t len,
+ const PacketOptions& options));
MOCK_METHOD2(SendRtcp, bool(const uint8_t* data, size_t len));
};
} // namespace webrtc
diff --git a/webrtc/test/null_transport.cc b/webrtc/test/null_transport.cc
index 3cba638..7fa36d1 100644
--- a/webrtc/test/null_transport.cc
+++ b/webrtc/test/null_transport.cc
@@ -12,7 +12,9 @@
namespace webrtc {
namespace test {
-bool NullTransport::SendRtp(const uint8_t* packet, size_t length) {
+bool NullTransport::SendRtp(const uint8_t* packet,
+ size_t length,
+ const PacketOptions& options) {
return true;
}
diff --git a/webrtc/test/null_transport.h b/webrtc/test/null_transport.h
index a60d8cb..f4b7046 100644
--- a/webrtc/test/null_transport.h
+++ b/webrtc/test/null_transport.h
@@ -19,7 +19,9 @@
namespace test {
class NullTransport : public Transport {
public:
- bool SendRtp(const uint8_t* packet, size_t length) override;
+ bool SendRtp(const uint8_t* packet,
+ size_t length,
+ const PacketOptions& options) override;
bool SendRtcp(const uint8_t* packet, size_t length) override;
};
} // namespace test
diff --git a/webrtc/test/rtp_rtcp_observer.h b/webrtc/test/rtp_rtcp_observer.h
index 24f7256..97aa7b6 100644
--- a/webrtc/test/rtp_rtcp_observer.h
+++ b/webrtc/test/rtp_rtcp_observer.h
@@ -126,7 +126,9 @@
on_rtcp_(on_rtcp) {}
private:
- bool SendRtp(const uint8_t* packet, size_t length) override {
+ bool SendRtp(const uint8_t* packet,
+ size_t length,
+ const PacketOptions& options) override {
EXPECT_FALSE(RtpHeaderParser::IsRtcp(packet, length));
Action action;
{
@@ -138,7 +140,7 @@
// Drop packet silently.
return true;
case SEND_PACKET:
- return test::DirectTransport::SendRtp(packet, length);
+ return test::DirectTransport::SendRtp(packet, length, options);
}
return true; // Will never happen, makes compiler happy.
}
diff --git a/webrtc/transport.h b/webrtc/transport.h
index 86b3e2e..7b62f65 100644
--- a/webrtc/transport.h
+++ b/webrtc/transport.h
@@ -17,9 +17,17 @@
namespace webrtc {
+struct PacketOptions {
+ // A 16 bits positive id. Negative ids are invalid and should be interpreted
+ // as packet_id not being set.
+ int packet_id = -1;
+};
+
class Transport {
public:
- virtual bool SendRtp(const uint8_t* packet, size_t length) = 0;
+ virtual bool SendRtp(const uint8_t* packet,
+ size_t length,
+ const PacketOptions& options) = 0;
virtual bool SendRtcp(const uint8_t* packet, size_t length) = 0;
protected:
diff --git a/webrtc/video/end_to_end_tests.cc b/webrtc/video/end_to_end_tests.cc
index c729f2a..305dcb0 100644
--- a/webrtc/video/end_to_end_tests.cc
+++ b/webrtc/video/end_to_end_tests.cc
@@ -61,7 +61,9 @@
protected:
class UnusedTransport : public Transport {
private:
- bool SendRtp(const uint8_t* packet, size_t length) override {
+ bool SendRtp(const uint8_t* packet,
+ size_t length,
+ const PacketOptions& options) override {
ADD_FAILURE() << "Unexpected RTP sent.";
return false;
}
@@ -1351,13 +1353,17 @@
}
virtual ~RtpExtensionHeaderObserver() {}
- bool SendRtp(const uint8_t* data, size_t length) override {
+ bool SendRtp(const uint8_t* data,
+ size_t length,
+ const PacketOptions& options) override {
if (IsDone())
return false;
RTPHeader header;
EXPECT_TRUE(parser_->Parse(data, length, &header));
if (header.extension.hasTransportSequenceNumber) {
+ EXPECT_EQ(options.packet_id,
+ header.extension.transportSequenceNumber);
if (!streams_observed_.empty()) {
EXPECT_EQ(static_cast<uint16_t>(last_seq_ + 1),
header.extension.transportSequenceNumber);
@@ -1377,7 +1383,7 @@
if (IsDone())
done_->Set();
}
- return test::DirectTransport::SendRtp(data, length);
+ return test::DirectTransport::SendRtp(data, length, options);
}
bool IsDone() {
diff --git a/webrtc/video/video_quality_test.cc b/webrtc/video/video_quality_test.cc
index 488532d..c452e11 100644
--- a/webrtc/video/video_quality_test.cc
+++ b/webrtc/video/video_quality_test.cc
@@ -138,7 +138,9 @@
input_->IncomingCapturedFrame(video_frame);
}
- bool SendRtp(const uint8_t* packet, size_t length) override {
+ bool SendRtp(const uint8_t* packet,
+ size_t length,
+ const PacketOptions& options) override {
rtc::scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
RTPHeader header;
parser->Parse(packet, length, &header);
@@ -156,7 +158,7 @@
length - (header.headerLength + header.paddingLength);
}
- return transport_->SendRtp(packet, length);
+ return transport_->SendRtp(packet, length, options);
}
bool SendRtcp(const uint8_t* packet, size_t length) override {
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index 3392d85..e4dc320 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -214,9 +214,9 @@
return 0;
}
-bool
-Channel::SendRtp(const uint8_t *data, size_t len)
-{
+bool Channel::SendRtp(const uint8_t* data,
+ size_t len,
+ const PacketOptions& options) {
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::SendPacket(channel=%d, len=%" PRIuS ")", len);
@@ -233,7 +233,7 @@
uint8_t* bufferToSendPtr = (uint8_t*)data;
size_t bufferLength = len;
- if (!_transportPtr->SendRtp(bufferToSendPtr, bufferLength)) {
+ if (!_transportPtr->SendRtp(bufferToSendPtr, bufferLength, options)) {
std::string transport_name =
_externalTransport ? "external transport" : "WebRtc sockets";
WEBRTC_TRACE(kTraceError, kTraceVoice,
diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h
index e7e6d76..a550738 100644
--- a/webrtc/voice_engine/channel.h
+++ b/webrtc/voice_engine/channel.h
@@ -378,7 +378,9 @@
uint8_t volume) override;
// From Transport (called by the RTP/RTCP module)
- bool SendRtp(const uint8_t* data, size_t len) override;
+ bool SendRtp(const uint8_t* data,
+ size_t len,
+ const PacketOptions& packet_options) override;
bool SendRtcp(const uint8_t* data, size_t len) override;
// From MixerParticipant
diff --git a/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc b/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc
index 30ef265..581a768 100644
--- a/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc
+++ b/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc
@@ -108,7 +108,9 @@
EXPECT_TRUE(webrtc::VoiceEngine::Delete(local_voe_));
}
-bool ConferenceTransport::SendRtp(const uint8_t* data, size_t len) {
+bool ConferenceTransport::SendRtp(const uint8_t* data,
+ size_t len,
+ const webrtc::PacketOptions& options) {
StorePacket(Packet::Rtp, data, len);
return true;
}
diff --git a/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h b/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h
index 2a770b2..602f07f 100644
--- a/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h
+++ b/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h
@@ -98,7 +98,9 @@
bool GetReceiverStatistics(unsigned int id, webrtc::CallStatistics* stats);
// Inherit from class webrtc::Transport.
- bool SendRtp(const uint8_t *data, size_t len) override;
+ bool SendRtp(const uint8_t* data,
+ size_t len,
+ const webrtc::PacketOptions& options) override;
bool SendRtcp(const uint8_t *data, size_t len) override;
private:
diff --git a/webrtc/voice_engine/test/auto_test/fixtures/after_initialization_fixture.h b/webrtc/voice_engine/test/auto_test/fixtures/after_initialization_fixture.h
index 422e550..c3bd9e8 100644
--- a/webrtc/voice_engine/test/auto_test/fixtures/after_initialization_fixture.h
+++ b/webrtc/voice_engine/test/auto_test/fixtures/after_initialization_fixture.h
@@ -41,7 +41,9 @@
~LoopBackTransport() { thread_->Stop(); }
- bool SendRtp(const uint8_t* data, size_t len) override {
+ bool SendRtp(const uint8_t* data,
+ size_t len,
+ const webrtc::PacketOptions& options) override {
StorePacket(Packet::Rtp, data, len);
return true;
}
diff --git a/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc b/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc
index d4700a9..bc6ce56 100644
--- a/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc
+++ b/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc
@@ -28,7 +28,9 @@
audio_level_id_(-1),
absolute_sender_time_id_(-1) {}
- bool SendRtp(const uint8_t* data, size_t len) override {
+ bool SendRtp(const uint8_t* data,
+ size_t len,
+ const webrtc::PacketOptions& options) override {
webrtc::RTPHeader header;
if (parser_->Parse(reinterpret_cast<const uint8_t*>(data), len, &header)) {
bool ok = true;
