Reland of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #1 id:1 of https://codereview.webrtc.org/2784543002/ )
Reason for revert:
Intend to fix perf failures and reland.
Original issue's description:
> Revert of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #4 id:60001 of https://codereview.webrtc.org/2774463003/ )
>
> Reason for revert:
> Reverting since this seems to break multiple WebRTC Perf buildbots
>
> Original issue's description:
> > Don't hardcode MediaType::ANY in FakeNetworkPipe.
> >
> > Instead let each test set the appropriate media type. This simplifies
> > demuxing in Call and later in RtpTransportController.
> >
> > BUG=webrtc:7135
> >
> > Review-Url: https://codereview.webrtc.org/2774463003
> > Cr-Commit-Position: refs/heads/master@{#17418}
> > Committed: https://chromium.googlesource.com/external/webrtc/+/9c47b00e24da2941eb095df5a4459c6d98a8a88d
>
> TBR=stefan@webrtc.org,deadbeef@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,sprang@webrtc.org,nisse@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7135
>
> Review-Url: https://codereview.webrtc.org/2784543002
> Cr-Commit-Position: refs/heads/master@{#17427}
> Committed: https://chromium.googlesource.com/external/webrtc/+/3a3bd5061089da5327fc549337a8430054d66057
TBR=stefan@webrtc.org,deadbeef@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,sprang@webrtc.org,lliuu@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/2783853002
Cr-Commit-Position: refs/heads/master@{#17459}
diff --git a/webrtc/audio/test/low_bandwidth_audio_test.cc b/webrtc/audio/test/low_bandwidth_audio_test.cc
index c1b1a02..50d7e67 100644
--- a/webrtc/audio/test/low_bandwidth_audio_test.cc
+++ b/webrtc/audio/test/low_bandwidth_audio_test.cc
@@ -80,12 +80,14 @@
Call* sender_call) {
return new test::PacketTransport(
sender_call, this, test::PacketTransport::kSender,
+ MediaType::AUDIO,
GetNetworkPipeConfig());
}
test::PacketTransport* AudioQualityTest::CreateReceiveTransport() {
return new test::PacketTransport(nullptr, this,
- test::PacketTransport::kReceiver, GetNetworkPipeConfig());
+ test::PacketTransport::kReceiver, MediaType::AUDIO,
+ GetNetworkPipeConfig());
}
void AudioQualityTest::ModifyAudioConfigs(
diff --git a/webrtc/call/bitrate_estimator_tests.cc b/webrtc/call/bitrate_estimator_tests.cc
index bb4960f..00a3b9e 100644
--- a/webrtc/call/bitrate_estimator_tests.cc
+++ b/webrtc/call/bitrate_estimator_tests.cc
@@ -108,9 +108,11 @@
receiver_call_.reset(Call::Create(config));
sender_call_.reset(Call::Create(config));
- send_transport_.reset(new test::DirectTransport(sender_call_.get()));
+ send_transport_.reset(
+ new test::DirectTransport(sender_call_.get(), MediaType::VIDEO));
send_transport_->SetReceiver(receiver_call_->Receiver());
- receive_transport_.reset(new test::DirectTransport(receiver_call_.get()));
+ receive_transport_.reset(
+ new test::DirectTransport(receiver_call_.get(), MediaType::VIDEO));
receive_transport_->SetReceiver(sender_call_->Receiver());
video_send_config_ = VideoSendStream::Config(send_transport_.get());
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index 94acc9f..396d3c9 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -1221,6 +1221,8 @@
const PacketTime& packet_time) {
TRACE_EVENT0("webrtc", "Call::DeliverRtp");
+ RTC_DCHECK(media_type == MediaType::AUDIO || media_type == MediaType::VIDEO);
+
ReadLockScoped read_lock(*receive_crit_);
// TODO(nisse): We should parse the RTP header only here, and pass
// on parsed_packet to the receive streams.
@@ -1234,7 +1236,7 @@
uint32_t ssrc = parsed_packet->Ssrc();
- if (media_type == MediaType::ANY || media_type == MediaType::AUDIO) {
+ if (media_type == MediaType::AUDIO) {
auto it = audio_receive_ssrcs_.find(ssrc);
if (it != audio_receive_ssrcs_.end()) {
received_bytes_per_second_counter_.Add(static_cast<int>(length));
@@ -1244,7 +1246,7 @@
return DELIVERY_OK;
}
}
- if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) {
+ if (media_type == MediaType::VIDEO) {
auto it = video_receive_ssrcs_.find(ssrc);
if (it != video_receive_ssrcs_.end()) {
received_bytes_per_second_counter_.Add(static_cast<int>(length));
@@ -1260,7 +1262,7 @@
return DELIVERY_OK;
}
}
- if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) {
+ if (media_type == MediaType::VIDEO) {
received_bytes_per_second_counter_.Add(static_cast<int>(length));
// TODO(brandtr): Update here when FlexFEC supports protecting audio.
received_video_bytes_per_second_counter_.Add(static_cast<int>(length));
@@ -1320,10 +1322,7 @@
return;
}
// For audio, we only support send side BWE.
- // TODO(nisse): Tests passes MediaType::ANY, see
- // FakeNetworkPipe::Process. We need to treat that as video. Tests
- // should be fixed to use the same MediaType as the production code.
- if (media_type != MediaType::AUDIO ||
+ if (media_type == MediaType::VIDEO ||
(use_send_side_bwe && header.extension.hasTransportSequenceNumber)) {
receive_side_cc_.OnReceivedPacket(
packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(),
diff --git a/webrtc/call/call_perf_tests.cc b/webrtc/call/call_perf_tests.cc
index bd5ab0f..d62bec1 100644
--- a/webrtc/call/call_perf_tests.cc
+++ b/webrtc/call/call_perf_tests.cc
@@ -193,6 +193,7 @@
audio_net_config.loss_percent = 5;
test::PacketTransport audio_send_transport(sender_call_.get(), &observer,
test::PacketTransport::kSender,
+ MediaType::AUDIO,
audio_net_config);
MediaTypePacketReceiver audio_receiver(receiver_call_->Receiver(),
MediaType::AUDIO);
@@ -200,6 +201,7 @@
test::PacketTransport video_send_transport(sender_call_.get(), &observer,
test::PacketTransport::kSender,
+ MediaType::VIDEO,
FakeNetworkPipe::Config());
MediaTypePacketReceiver video_receiver(receiver_call_->Receiver(),
MediaType::VIDEO);
@@ -207,6 +209,7 @@
test::PacketTransport receive_transport(
receiver_call_.get(), &observer, test::PacketTransport::kReceiver,
+ MediaType::VIDEO,
FakeNetworkPipe::Config());
receive_transport.SetReceiver(sender_call_->Receiver());
@@ -343,12 +346,14 @@
private:
test::PacketTransport* CreateSendTransport(Call* sender_call) override {
return new test::PacketTransport(
- sender_call, this, test::PacketTransport::kSender, net_config_);
+ sender_call, this, test::PacketTransport::kSender, MediaType::VIDEO,
+ net_config_);
}
test::PacketTransport* CreateReceiveTransport() override {
return new test::PacketTransport(
- nullptr, this, test::PacketTransport::kReceiver, net_config_);
+ nullptr, this, test::PacketTransport::kReceiver, MediaType::VIDEO,
+ net_config_);
}
void OnFrame(const VideoFrame& video_frame) override {
diff --git a/webrtc/call/rampup_tests.cc b/webrtc/call/rampup_tests.cc
index 292edc6..0a2a738 100644
--- a/webrtc/call/rampup_tests.cc
+++ b/webrtc/call/rampup_tests.cc
@@ -89,13 +89,34 @@
send_stream_ = send_stream;
}
+MediaType RampUpTester::SelectMediaType() {
+ if (num_video_streams_ > 0) {
+ if (num_audio_streams_ > 0) {
+ // Rely on call to set media type from payload type.
+ return MediaType::ANY;
+ } else {
+ return MediaType::VIDEO;
+ }
+ } else {
+ return MediaType::AUDIO;
+ }
+}
+
test::PacketTransport* RampUpTester::CreateSendTransport(Call* sender_call) {
send_transport_ = new test::PacketTransport(sender_call, this,
test::PacketTransport::kSender,
+ SelectMediaType(),
forward_transport_config_);
return send_transport_;
}
+test::PacketTransport* RampUpTester::CreateReceiveTransport() {
+ return new test::PacketTransport(nullptr, this,
+ test::PacketTransport::kReceiver,
+ SelectMediaType(),
+ FakeNetworkPipe::Config());
+}
+
size_t RampUpTester::GetNumVideoStreams() const {
return num_video_streams_;
}
diff --git a/webrtc/call/rampup_tests.h b/webrtc/call/rampup_tests.h
index 6710dd1..5b24536 100644
--- a/webrtc/call/rampup_tests.h
+++ b/webrtc/call/rampup_tests.h
@@ -84,7 +84,9 @@
void OnVideoStreamsCreated(
VideoSendStream* send_stream,
const std::vector<VideoReceiveStream*>& receive_streams) override;
+ MediaType SelectMediaType();
test::PacketTransport* CreateSendTransport(Call* sender_call) override;
+ test::PacketTransport* CreateReceiveTransport() override;
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
diff --git a/webrtc/media/engine/fakewebrtccall.cc b/webrtc/media/engine/fakewebrtccall.cc
index 9a05ae6..f442a3c 100644
--- a/webrtc/media/engine/fakewebrtccall.cc
+++ b/webrtc/media/engine/fakewebrtccall.cc
@@ -520,19 +520,20 @@
size_t length,
const webrtc::PacketTime& packet_time) {
EXPECT_GE(length, 12u);
+ RTC_DCHECK(media_type == webrtc::MediaType::AUDIO ||
+ media_type == webrtc::MediaType::VIDEO);
+
uint32_t ssrc;
if (!GetRtpSsrc(packet, length, &ssrc))
return DELIVERY_PACKET_ERROR;
- if (media_type == webrtc::MediaType::ANY ||
- media_type == webrtc::MediaType::VIDEO) {
+ if (media_type == webrtc::MediaType::VIDEO) {
for (auto receiver : video_receive_streams_) {
if (receiver->GetConfig().rtp.remote_ssrc == ssrc)
return DELIVERY_OK;
}
}
- if (media_type == webrtc::MediaType::ANY ||
- media_type == webrtc::MediaType::AUDIO) {
+ if (media_type == webrtc::MediaType::AUDIO) {
for (auto receiver : audio_receive_streams_) {
if (receiver->GetConfig().rtp.remote_ssrc == ssrc) {
receiver->DeliverRtp(packet, length, packet_time);
diff --git a/webrtc/test/call_test.cc b/webrtc/test/call_test.cc
index 3a13eee..75595ab 100644
--- a/webrtc/test/call_test.cc
+++ b/webrtc/test/call_test.cc
@@ -26,6 +26,29 @@
const int kVideoRotationRtpExtensionId = 4;
}
+void CallTest::PayloadDemuxer::SetReceiver(PacketReceiver* receiver) {
+ receiver_ = receiver;
+}
+
+PacketReceiver::DeliveryStatus CallTest::PayloadDemuxer::DeliverPacket(
+ MediaType media_type,
+ const uint8_t* packet,
+ size_t length,
+ const PacketTime& packet_time) {
+ if (media_type == MediaType::ANY) {
+ // This simplistic demux logic will not make much sense for RTCP
+ // packets, but it seems that doesn't matter.
+ RTC_CHECK_GE(length, 2);
+ uint8_t pt = packet[1] & 0x7f;
+ if (pt == kFakeVideoSendPayloadType || pt == kFlexfecPayloadType) {
+ media_type = MediaType::VIDEO;
+ } else {
+ media_type = MediaType::AUDIO;
+ }
+ }
+ return receiver_->DeliverPacket(media_type, packet, length, packet_time);
+}
+
CallTest::CallTest()
: clock_(Clock::GetRealTimeClock()),
video_send_config_(nullptr),
@@ -75,8 +98,20 @@
send_transport_.reset(test->CreateSendTransport(sender_call_.get()));
if (test->ShouldCreateReceivers()) {
- send_transport_->SetReceiver(receiver_call_->Receiver());
- receive_transport_->SetReceiver(sender_call_->Receiver());
+ // For tests using only video or only audio, we rely on each test
+ // configuring the underlying FakeNetworkPipe with the right media
+ // type. But for tests sending both video and audio over the same
+ // FakeNetworkPipe, we need to "demux", i.e., setting the
+ // MediaType based on RTP payload type.
+ if (num_video_streams_ > 0 && num_audio_streams_ > 0) {
+ receive_demuxer_.SetReceiver(receiver_call_->Receiver());
+ send_transport_->SetReceiver(&receive_demuxer_);
+ send_demuxer_.SetReceiver(sender_call_->Receiver());
+ receive_transport_->SetReceiver(&send_demuxer_);
+ } else {
+ send_transport_->SetReceiver(receiver_call_->Receiver());
+ receive_transport_->SetReceiver(sender_call_->Receiver());
+ }
if (num_video_streams_ > 0)
receiver_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
if (num_audio_streams_ > 0)
@@ -459,11 +494,13 @@
test::PacketTransport* BaseTest::CreateSendTransport(Call* sender_call) {
return new PacketTransport(sender_call, this, test::PacketTransport::kSender,
+ MediaType::VIDEO,
FakeNetworkPipe::Config());
}
test::PacketTransport* BaseTest::CreateReceiveTransport() {
return new PacketTransport(nullptr, this, test::PacketTransport::kReceiver,
+ MediaType::VIDEO,
FakeNetworkPipe::Config());
}
diff --git a/webrtc/test/call_test.h b/webrtc/test/call_test.h
index 78fbbfb..b4f6658 100644
--- a/webrtc/test/call_test.h
+++ b/webrtc/test/call_test.h
@@ -59,6 +59,23 @@
static const int kNackRtpHistoryMs;
protected:
+ // Needed for tests sending both audio and video on the same
+ // FakeNetworkPipe. We then need to set correct MediaType based on
+ // packet payload type, before passing the packet on to Call.
+ class PayloadDemuxer : public PacketReceiver {
+ public:
+ PayloadDemuxer() = default;
+
+ void SetReceiver(PacketReceiver* receiver);
+ DeliveryStatus DeliverPacket(MediaType media_type,
+ const uint8_t* packet,
+ size_t length,
+ const PacketTime& packet_time) override;
+
+ private:
+ PacketReceiver* receiver_ = nullptr;
+ };
+
// RunBaseTest overwrites the audio_state and the voice_engine of the send and
// receive Call configs to simplify test code and avoid having old VoiceEngine
// APIs in the tests.
@@ -124,6 +141,9 @@
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
test::FakeVideoRenderer fake_renderer_;
+ PayloadDemuxer receive_demuxer_;
+ PayloadDemuxer send_demuxer_;
+
private:
// TODO(holmer): Remove once VoiceEngine is fully refactored to the new API.
// These methods are used to set up legacy voice engines and channels which is
@@ -172,6 +192,7 @@
virtual Call::Config GetReceiverCallConfig();
virtual void OnCallsCreated(Call* sender_call, Call* receiver_call);
+ // The default implementation creates MediaType::VIDEO transports.
virtual test::PacketTransport* CreateSendTransport(Call* sender_call);
virtual test::PacketTransport* CreateReceiveTransport();
diff --git a/webrtc/test/direct_transport.cc b/webrtc/test/direct_transport.cc
index 922874d..90531ea 100644
--- a/webrtc/test/direct_transport.cc
+++ b/webrtc/test/direct_transport.cc
@@ -15,17 +15,17 @@
namespace webrtc {
namespace test {
-DirectTransport::DirectTransport(Call* send_call)
- : DirectTransport(FakeNetworkPipe::Config(), send_call) {}
+DirectTransport::DirectTransport(Call* send_call, MediaType media_type)
+ : DirectTransport(FakeNetworkPipe::Config(), send_call, media_type) {}
DirectTransport::DirectTransport(const FakeNetworkPipe::Config& config,
- Call* send_call)
+ Call* send_call, MediaType media_type)
: send_call_(send_call),
packet_event_(false, false),
thread_(NetworkProcess, this, "NetworkProcess"),
clock_(Clock::GetRealTimeClock()),
shutting_down_(false),
- fake_network_(clock_, config) {
+ fake_network_(clock_, config, media_type) {
thread_.Start();
if (send_call_) {
send_call_->SignalChannelNetworkState(MediaType::AUDIO, kNetworkUp);
diff --git a/webrtc/test/direct_transport.h b/webrtc/test/direct_transport.h
index eb552c5..20a6857 100644
--- a/webrtc/test/direct_transport.h
+++ b/webrtc/test/direct_transport.h
@@ -18,11 +18,11 @@
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/event.h"
#include "webrtc/base/platform_thread.h"
+#include "webrtc/call/call.h"
#include "webrtc/test/fake_network_pipe.h"
namespace webrtc {
-class Call;
class Clock;
class PacketReceiver;
@@ -30,8 +30,17 @@
class DirectTransport : public Transport {
public:
- explicit DirectTransport(Call* send_call);
- DirectTransport(const FakeNetworkPipe::Config& config, Call* send_call);
+ DirectTransport(Call* send_call, MediaType media_type);
+ DirectTransport(const FakeNetworkPipe::Config& config, Call* send_call,
+ MediaType media_type);
+ // These deprecated variants always use MediaType::VIDEO.
+ RTC_DEPRECATED explicit DirectTransport(Call* send_call)
+ : DirectTransport(send_call, MediaType::VIDEO) {}
+
+ RTC_DEPRECATED DirectTransport(const FakeNetworkPipe::Config& config,
+ Call* send_call)
+ : DirectTransport(config, send_call, MediaType::VIDEO) {}
+
~DirectTransport();
void SetConfig(const FakeNetworkPipe::Config& config);
diff --git a/webrtc/test/fake_network_pipe.cc b/webrtc/test/fake_network_pipe.cc
index 8f2fee6..1034167 100644
--- a/webrtc/test/fake_network_pipe.cc
+++ b/webrtc/test/fake_network_pipe.cc
@@ -28,13 +28,16 @@
}
FakeNetworkPipe::FakeNetworkPipe(Clock* clock,
- const FakeNetworkPipe::Config& config)
- : FakeNetworkPipe(clock, config, 1) {}
+ const FakeNetworkPipe::Config& config,
+ MediaType media_type)
+ : FakeNetworkPipe(clock, config, media_type, 1) {}
FakeNetworkPipe::FakeNetworkPipe(Clock* clock,
const FakeNetworkPipe::Config& config,
+ MediaType media_type,
uint64_t seed)
: clock_(clock),
+ media_type_(media_type),
packet_receiver_(NULL),
random_(seed),
config_(),
@@ -199,7 +202,7 @@
while (!packets_to_deliver.empty()) {
NetworkPacket* packet = packets_to_deliver.front();
packets_to_deliver.pop();
- packet_receiver_->DeliverPacket(MediaType::ANY, packet->data(),
+ packet_receiver_->DeliverPacket(media_type_, packet->data(),
packet->data_length(), PacketTime());
delete packet;
}
diff --git a/webrtc/test/fake_network_pipe.h b/webrtc/test/fake_network_pipe.h
index 0c73e85..adf76c3 100644
--- a/webrtc/test/fake_network_pipe.h
+++ b/webrtc/test/fake_network_pipe.h
@@ -25,6 +25,7 @@
class Clock;
class PacketReceiver;
+enum class MediaType;
class NetworkPacket {
public:
@@ -82,9 +83,10 @@
int avg_burst_loss_length = -1;
};
- FakeNetworkPipe(Clock* clock, const FakeNetworkPipe::Config& config);
+ FakeNetworkPipe(Clock* clock, const FakeNetworkPipe::Config& config,
+ MediaType media_type);
FakeNetworkPipe(Clock* clock,
- const FakeNetworkPipe::Config& config,
+ const FakeNetworkPipe::Config& config, MediaType media_type,
uint64_t seed);
~FakeNetworkPipe();
@@ -110,6 +112,7 @@
private:
Clock* const clock_;
+ const MediaType media_type_;
rtc::CriticalSection lock_;
PacketReceiver* packet_receiver_;
std::queue<NetworkPacket*> capacity_link_;
diff --git a/webrtc/test/fake_network_pipe_unittest.cc b/webrtc/test/fake_network_pipe_unittest.cc
index aa27399..c19b001 100644
--- a/webrtc/test/fake_network_pipe_unittest.cc
+++ b/webrtc/test/fake_network_pipe_unittest.cc
@@ -96,7 +96,7 @@
config.queue_length_packets = 20;
config.link_capacity_kbps = 80;
std::unique_ptr<FakeNetworkPipe> pipe(
- new FakeNetworkPipe(&fake_clock_, config));
+ new FakeNetworkPipe(&fake_clock_, config, MediaType::VIDEO));
pipe->SetReceiver(receiver_.get());
// Add 10 packets of 1000 bytes, = 80 kb, and verify it takes one second to
@@ -110,22 +110,22 @@
kPacketSize);
// Time haven't increased yet, so we souldn't get any packets.
- EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(0);
+ EXPECT_CALL(*receiver_, DeliverPacket(MediaType::VIDEO, _, _, _)).Times(0);
pipe->Process();
// Advance enough time to release one packet.
fake_clock_.AdvanceTimeMilliseconds(kPacketTimeMs);
- EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(1);
+ EXPECT_CALL(*receiver_, DeliverPacket(MediaType::VIDEO, _, _, _)).Times(1);
pipe->Process();
// Release all but one packet
fake_clock_.AdvanceTimeMilliseconds(9 * kPacketTimeMs - 1);
- EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(8);
+ EXPECT_CALL(*receiver_, DeliverPacket(MediaType::VIDEO, _, _, _)).Times(8);
pipe->Process();
// And the last one.
fake_clock_.AdvanceTimeMilliseconds(1);
- EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(1);
+ EXPECT_CALL(*receiver_, DeliverPacket(MediaType::VIDEO, _, _, _)).Times(1);
pipe->Process();
}
@@ -136,7 +136,7 @@
config.queue_delay_ms = 100;
config.link_capacity_kbps = 80;
std::unique_ptr<FakeNetworkPipe> pipe(
- new FakeNetworkPipe(&fake_clock_, config));
+ new FakeNetworkPipe(&fake_clock_, config, MediaType::AUDIO));
pipe->SetReceiver(receiver_.get());
const int kNumPackets = 2;
@@ -149,17 +149,17 @@
// Increase more than kPacketTimeMs, but not more than the extra delay.
fake_clock_.AdvanceTimeMilliseconds(kPacketTimeMs);
- EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(0);
+ EXPECT_CALL(*receiver_, DeliverPacket(MediaType::AUDIO, _, _, _)).Times(0);
pipe->Process();
// Advance the network delay to get the first packet.
fake_clock_.AdvanceTimeMilliseconds(config.queue_delay_ms);
- EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(1);
+ EXPECT_CALL(*receiver_, DeliverPacket(MediaType::AUDIO, _, _, _)).Times(1);
pipe->Process();
// Advance one more kPacketTimeMs to get the last packet.
fake_clock_.AdvanceTimeMilliseconds(kPacketTimeMs);
- EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(1);
+ EXPECT_CALL(*receiver_, DeliverPacket(MediaType::AUDIO, _, _, _)).Times(1);
pipe->Process();
}
@@ -170,7 +170,7 @@
config.queue_length_packets = 2;
config.link_capacity_kbps = 80;
std::unique_ptr<FakeNetworkPipe> pipe(
- new FakeNetworkPipe(&fake_clock_, config));
+ new FakeNetworkPipe(&fake_clock_, config, MediaType::VIDEO));
pipe->SetReceiver(receiver_.get());
const int kPacketSize = 1000;
@@ -183,7 +183,7 @@
// Increase time enough to deliver all three packets, verify only two are
// delivered.
fake_clock_.AdvanceTimeMilliseconds(3 * kPacketTimeMs);
- EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(2);
+ EXPECT_CALL(*receiver_, DeliverPacket(MediaType::VIDEO, _, _, _)).Times(2);
pipe->Process();
}
@@ -194,7 +194,7 @@
config.queue_delay_ms = 20;
config.link_capacity_kbps = 80;
std::unique_ptr<FakeNetworkPipe> pipe(
- new FakeNetworkPipe(&fake_clock_, config));
+ new FakeNetworkPipe(&fake_clock_, config, MediaType::VIDEO));
pipe->SetReceiver(receiver_.get());
const int kPacketSize = 1000;
@@ -206,7 +206,7 @@
fake_clock_.AdvanceTimeMilliseconds(3 * kPacketTimeMs +
config.queue_delay_ms);
- EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(2);
+ EXPECT_CALL(*receiver_, DeliverPacket(MediaType::VIDEO, _, _, _)).Times(2);
pipe->Process();
// Packet 1: kPacketTimeMs + config.queue_delay_ms,
@@ -224,7 +224,7 @@
config.queue_length_packets = 20;
config.link_capacity_kbps = 80;
std::unique_ptr<FakeNetworkPipe> pipe(
- new FakeNetworkPipe(&fake_clock_, config));
+ new FakeNetworkPipe(&fake_clock_, config, MediaType::VIDEO));
pipe->SetReceiver(receiver_.get());
// Add 10 packets of 1000 bytes, = 80 kb, and verify it takes one second to
@@ -237,13 +237,13 @@
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(MediaType::VIDEO, _, _, _)).Times(0);
pipe->Process();
// Advance time in steps to release one packet at a time.
for (int i = 0; i < kNumPackets; ++i) {
fake_clock_.AdvanceTimeMilliseconds(packet_time_ms);
- EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(1);
+ EXPECT_CALL(*receiver_, DeliverPacket(MediaType::VIDEO, _, _, _)).Times(1);
pipe->Process();
}
@@ -259,20 +259,20 @@
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(MediaType::VIDEO, _, _, _)).Times(0);
pipe->Process();
// Advance time in steps to release one packet at a time.
for (int i = 0; i < kNumPackets; ++i) {
fake_clock_.AdvanceTimeMilliseconds(packet_time_ms);
- EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(1);
+ EXPECT_CALL(*receiver_, DeliverPacket(MediaType::VIDEO, _, _, _)).Times(1);
pipe->Process();
}
// Check that all the packets were sent.
EXPECT_EQ(static_cast<size_t>(2 * kNumPackets), pipe->sent_packets());
fake_clock_.AdvanceTimeMilliseconds(pipe->TimeUntilNextProcess());
- EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(0);
+ EXPECT_CALL(*receiver_, DeliverPacket(MediaType::VIDEO, _, _, _)).Times(0);
pipe->Process();
}
@@ -283,7 +283,7 @@
config.queue_length_packets = 20;
config.link_capacity_kbps = 80;
std::unique_ptr<FakeNetworkPipe> pipe(
- new FakeNetworkPipe(&fake_clock_, config));
+ new FakeNetworkPipe(&fake_clock_, config, MediaType::AUDIO));
pipe->SetReceiver(receiver_.get());
// Add 10 packets of 1000 bytes, = 80 kb.
@@ -306,27 +306,27 @@
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(MediaType::AUDIO, _, _, _)).Times(0);
pipe->Process();
// Advance time in steps to release one packet at a time.
for (int i = 0; i < kNumPackets; ++i) {
fake_clock_.AdvanceTimeMilliseconds(packet_time_1_ms);
- EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(1);
+ EXPECT_CALL(*receiver_, DeliverPacket(MediaType::AUDIO, _, _, _)).Times(1);
pipe->Process();
}
// Advance time in steps to release one packet at a time.
for (int i = 0; i < kNumPackets; ++i) {
fake_clock_.AdvanceTimeMilliseconds(packet_time_2_ms);
- EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(1);
+ EXPECT_CALL(*receiver_, DeliverPacket(MediaType::AUDIO, _, _, _)).Times(1);
pipe->Process();
}
// Check that all the packets were sent.
EXPECT_EQ(static_cast<size_t>(2 * kNumPackets), pipe->sent_packets());
fake_clock_.AdvanceTimeMilliseconds(pipe->TimeUntilNextProcess());
- EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(0);
+ EXPECT_CALL(*receiver_, DeliverPacket(MediaType::AUDIO, _, _, _)).Times(0);
pipe->Process();
}
@@ -338,7 +338,7 @@
config.queue_delay_ms = 100;
config.delay_standard_deviation_ms = 10;
std::unique_ptr<FakeNetworkPipe> pipe(
- new FakeNetworkPipe(&fake_clock_, config));
+ new FakeNetworkPipe(&fake_clock_, config, MediaType::VIDEO));
ReorderTestReceiver* receiver = new ReorderTestReceiver();
receiver_.reset(receiver);
pipe->SetReceiver(receiver_.get());
@@ -390,7 +390,7 @@
config.loss_percent = kLossPercent;
config.avg_burst_loss_length = kAvgBurstLength;
std::unique_ptr<FakeNetworkPipe> pipe(
- new FakeNetworkPipe(&fake_clock_, config));
+ new FakeNetworkPipe(&fake_clock_, config, MediaType::VIDEO));
ReorderTestReceiver* receiver = new ReorderTestReceiver();
receiver_.reset(receiver);
pipe->SetReceiver(receiver_.get());
diff --git a/webrtc/test/layer_filtering_transport.cc b/webrtc/test/layer_filtering_transport.cc
index 2680553..5ae06b7 100644
--- a/webrtc/test/layer_filtering_transport.cc
+++ b/webrtc/test/layer_filtering_transport.cc
@@ -27,7 +27,7 @@
uint8_t vp9_video_payload_type,
int selected_tl,
int selected_sl)
- : test::DirectTransport(config, send_call),
+ : test::DirectTransport(config, send_call, MediaType::VIDEO),
vp8_video_payload_type_(vp8_video_payload_type),
vp9_video_payload_type_(vp9_video_payload_type),
selected_tl_(selected_tl),
diff --git a/webrtc/test/rtp_rtcp_observer.h b/webrtc/test/rtp_rtcp_observer.h
index 8e1375f..b268e4b 100644
--- a/webrtc/test/rtp_rtcp_observer.h
+++ b/webrtc/test/rtp_rtcp_observer.h
@@ -94,8 +94,9 @@
PacketTransport(Call* send_call,
RtpRtcpObserver* observer,
TransportType transport_type,
+ MediaType media_type,
const FakeNetworkPipe::Config& configuration)
- : test::DirectTransport(configuration, send_call),
+ : test::DirectTransport(configuration, send_call, media_type),
observer_(observer),
transport_type_(transport_type) {}
diff --git a/webrtc/video/end_to_end_tests.cc b/webrtc/video/end_to_end_tests.cc
index 6f030c3..12b0b52 100644
--- a/webrtc/video/end_to_end_tests.cc
+++ b/webrtc/video/end_to_end_tests.cc
@@ -210,8 +210,10 @@
CreateCalls(Call::Config(&event_log_), Call::Config(&event_log_));
- test::DirectTransport sender_transport(sender_call_.get());
- test::DirectTransport receiver_transport(receiver_call_.get());
+ test::DirectTransport sender_transport(
+ sender_call_.get(), MediaType::VIDEO);
+ test::DirectTransport receiver_transport(
+ receiver_call_.get(), MediaType::VIDEO);
sender_transport.SetReceiver(receiver_call_->Receiver());
receiver_transport.SetReceiver(sender_call_->Receiver());
@@ -257,8 +259,10 @@
CreateCalls(Call::Config(&event_log_), Call::Config(&event_log_));
- test::DirectTransport sender_transport(sender_call_.get());
- test::DirectTransport receiver_transport(receiver_call_.get());
+ test::DirectTransport sender_transport(
+ sender_call_.get(), MediaType::VIDEO);
+ test::DirectTransport receiver_transport(
+ receiver_call_.get(), MediaType::VIDEO);
sender_transport.SetReceiver(receiver_call_->Receiver());
receiver_transport.SetReceiver(sender_call_->Receiver());
@@ -529,6 +533,7 @@
test::PacketTransport* CreateReceiveTransport() override {
test::PacketTransport* receive_transport = new test::PacketTransport(
nullptr, this, test::PacketTransport::kReceiver,
+ MediaType::AUDIO,
FakeNetworkPipe::Config());
receive_transport_ = receive_transport;
return receive_transport;
@@ -790,7 +795,8 @@
FakeNetworkPipe::Config config;
config.queue_delay_ms = kNetworkDelayMs;
return new test::PacketTransport(sender_call, this,
- test::PacketTransport::kSender, config);
+ test::PacketTransport::kSender,
+ MediaType::VIDEO, config);
}
void OnFrame(const VideoFrame& video_frame) override {
@@ -969,7 +975,8 @@
FakeNetworkPipe::Config config;
config.queue_delay_ms = kNetworkDelayMs;
return new test::PacketTransport(sender_call, this,
- test::PacketTransport::kSender, config);
+ test::PacketTransport::kSender,
+ MediaType::VIDEO, config);
}
// TODO(holmer): Investigate why we don't send FEC packets when the bitrate
@@ -1311,8 +1318,10 @@
CreateCalls(Call::Config(&event_log_), Call::Config(&event_log_));
- test::DirectTransport send_transport(sender_call_.get());
- test::DirectTransport receive_transport(receiver_call_.get());
+ test::DirectTransport send_transport(
+ sender_call_.get(), MediaType::VIDEO);
+ test::DirectTransport receive_transport(
+ receiver_call_.get(), MediaType::VIDEO);
PacketInputObserver input_observer(receiver_call_->Receiver());
send_transport.SetReceiver(&input_observer);
receive_transport.SetReceiver(sender_call_->Receiver());
@@ -1539,10 +1548,10 @@
VideoReceiveStream::Config* receive_config) {
}
virtual test::DirectTransport* CreateSendTransport(Call* sender_call) {
- return new test::DirectTransport(sender_call);
+ return new test::DirectTransport(sender_call, MediaType::VIDEO);
}
virtual test::DirectTransport* CreateReceiveTransport(Call* receiver_call) {
- return new test::DirectTransport(receiver_call);
+ return new test::DirectTransport(receiver_call, MediaType::VIDEO);
}
};
@@ -1621,7 +1630,7 @@
RtpExtensionHeaderObserver(Call* sender_call,
const uint32_t& first_media_ssrc,
const std::map<uint32_t, uint32_t>& ssrc_map)
- : DirectTransport(sender_call),
+ : DirectTransport(sender_call, MediaType::VIDEO),
done_(false, false),
parser_(RtpHeaderParser::Create()),
first_media_ssrc_(first_media_ssrc),
@@ -1937,8 +1946,10 @@
CreateCalls(Call::Config(&event_log_), Call::Config(&event_log_));
- test::DirectTransport sender_transport(sender_call_.get());
- test::DirectTransport receiver_transport(receiver_call_.get());
+ test::DirectTransport sender_transport(
+ sender_call_.get(), MediaType::VIDEO);
+ test::DirectTransport receiver_transport(
+ receiver_call_.get(), MediaType::VIDEO);
sender_transport.SetReceiver(receiver_call_->Receiver());
receiver_transport.SetReceiver(sender_call_->Receiver());
@@ -2102,6 +2113,7 @@
test::PacketTransport* CreateReceiveTransport() override {
receive_transport_ = new test::PacketTransport(
nullptr, this, test::PacketTransport::kReceiver,
+ MediaType::VIDEO,
FakeNetworkPipe::Config());
return receive_transport_;
}
@@ -3065,7 +3077,8 @@
FakeNetworkPipe::Config network_config;
network_config.loss_percent = 5;
return new test::PacketTransport(
- sender_call, this, test::PacketTransport::kSender, network_config);
+ sender_call, this, test::PacketTransport::kSender, MediaType::VIDEO,
+ network_config);
}
Call::Config GetSenderCallConfig() override {
@@ -3619,9 +3632,11 @@
test::PacketTransport send_transport(sender_call_.get(), &observer,
test::PacketTransport::kSender,
+ MediaType::VIDEO,
FakeNetworkPipe::Config());
test::PacketTransport receive_transport(nullptr, &observer,
test::PacketTransport::kReceiver,
+ MediaType::VIDEO,
FakeNetworkPipe::Config());
send_transport.SetReceiver(receiver_call_->Receiver());
receive_transport.SetReceiver(sender_call_->Receiver());
@@ -3914,9 +3929,11 @@
FakeNetworkPipe::Config config;
config.queue_delay_ms = kSendDelayMs;
- test::DirectTransport sender_transport(config, sender_call_.get());
+ test::DirectTransport sender_transport(
+ config, sender_call_.get(), MediaType::VIDEO);
config.queue_delay_ms = kReceiveDelayMs;
- test::DirectTransport receiver_transport(config, receiver_call_.get());
+ test::DirectTransport receiver_transport(
+ config, receiver_call_.get(), MediaType::VIDEO);
sender_transport.SetReceiver(receiver_call_->Receiver());
receiver_transport.SetReceiver(sender_call_->Receiver());
@@ -3978,7 +3995,7 @@
CreateCalls(config, config);
receiver_call_->SignalChannelNetworkState(network_to_bring_up, kNetworkUp);
- test::DirectTransport sender_transport(sender_call_.get());
+ test::DirectTransport sender_transport(sender_call_.get(), MediaType::VIDEO);
sender_transport.SetReceiver(receiver_call_->Receiver());
CreateSendConfig(1, 0, 0, &sender_transport);
CreateMatchingReceiveConfigs(transport);
diff --git a/webrtc/video/replay.cc b/webrtc/video/replay.cc
index 2188d3e..6347dc9 100644
--- a/webrtc/video/replay.cc
+++ b/webrtc/video/replay.cc
@@ -283,8 +283,8 @@
if (!rtp_reader->NextPacket(&packet))
break;
++num_packets;
- switch (call->Receiver()->DeliverPacket(webrtc::MediaType::ANY, packet.data,
- packet.length, PacketTime())) {
+ switch (call->Receiver()->DeliverPacket(
+ webrtc::MediaType::VIDEO, packet.data, packet.length, PacketTime())) {
case PacketReceiver::DELIVERY_OK:
break;
case PacketReceiver::DELIVERY_UNKNOWN_SSRC: {
diff --git a/webrtc/video/video_quality_test.cc b/webrtc/video/video_quality_test.cc
index 4c3c731..96a4882 100644
--- a/webrtc/video/video_quality_test.cc
+++ b/webrtc/video/video_quality_test.cc
@@ -1590,7 +1590,8 @@
test::LayerFilteringTransport send_transport(
params_.pipe, sender_call_.get(), kPayloadTypeVP8, kPayloadTypeVP9,
params_.video.selected_tl, params_.ss.selected_sl);
- test::DirectTransport recv_transport(params_.pipe, receiver_call_.get());
+ test::DirectTransport recv_transport(
+ params_.pipe, receiver_call_.get(), MediaType::VIDEO);
std::string graph_title = params_.analyzer.graph_title;
if (graph_title.empty())
diff --git a/webrtc/video/video_send_stream_tests.cc b/webrtc/video/video_send_stream_tests.cc
index cf09b3a..18bb9b8 100644
--- a/webrtc/video/video_send_stream_tests.cc
+++ b/webrtc/video/video_send_stream_tests.cc
@@ -439,7 +439,8 @@
config.loss_percent = 5;
config.queue_delay_ms = kNetworkDelayMs;
return new test::PacketTransport(sender_call, this,
- test::PacketTransport::kSender, config);
+ test::PacketTransport::kSender,
+ MediaType::VIDEO, config);
}
void ModifyVideoConfigs(
@@ -594,7 +595,8 @@
config.loss_percent = 5;
config.queue_delay_ms = kNetworkDelayMs;
return new test::PacketTransport(sender_call, this,
- test::PacketTransport::kSender, config);
+ test::PacketTransport::kSender,
+ MediaType::VIDEO, config);
}
void ModifyVideoConfigs(
@@ -1263,7 +1265,8 @@
config.link_capacity_kbps = kCapacityKbps;
config.queue_delay_ms = kNetworkDelayMs;
return new test::PacketTransport(sender_call, this,
- test::PacketTransport::kSender, config);
+ test::PacketTransport::kSender,
+ MediaType::VIDEO, config);
}
void ModifyVideoConfigs(
