Add audio streams to CallTest and a first A/V call test.
Add audio send and receive streams to CallTest and call the necessary voice engine APIs for the streams to be usable. Verifies the implementation by adding a simple test which monitors outgoing packets and checks that both audio and video is being sent with transport sequence numbers.
Audio streams are using a fake audio device with file input.
The CallTest implementation is to a big degree based on call_perf_tests.cc and should in the future replace a lot of that code.
R=pbos@webrtc.org
TBR=kjellander@webrtc.org
BUG=webrtc:5263
Review URL: https://codereview.webrtc.org/1542653002 .
Cr-Commit-Position: refs/heads/master@{#11171}
diff --git a/webrtc/call/bitrate_estimator_tests.cc b/webrtc/call/bitrate_estimator_tests.cc
index e371270..4b24bbd 100644
--- a/webrtc/call/bitrate_estimator_tests.cc
+++ b/webrtc/call/bitrate_estimator_tests.cc
@@ -120,17 +120,18 @@
receive_transport_->SetReceiver(sender_call_->Receiver());
video_send_config_ = VideoSendStream::Config(send_transport_.get());
- video_send_config_.rtp.ssrcs.push_back(kSendSsrcs[0]);
+ video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[0]);
// Encoders will be set separately per stream.
video_send_config_.encoder_settings.encoder = nullptr;
video_send_config_.encoder_settings.payload_name = "FAKE";
- video_send_config_.encoder_settings.payload_type = kFakeSendPayloadType;
+ video_send_config_.encoder_settings.payload_type =
+ kFakeVideoSendPayloadType;
video_encoder_config_.streams = test::CreateVideoStreams(1);
receive_config_ = VideoReceiveStream::Config(receive_transport_.get());
// receive_config_.decoders will be set by every stream separately.
receive_config_.rtp.remote_ssrc = video_send_config_.rtp.ssrcs[0];
- receive_config_.rtp.local_ssrc = kReceiverLocalSsrc;
+ receive_config_.rtp.local_ssrc = kReceiverLocalVideoSsrc;
receive_config_.rtp.remb = true;
receive_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kTOffset, kTOFExtensionId));
diff --git a/webrtc/call/call_perf_tests.cc b/webrtc/call/call_perf_tests.cc
index faefc42..79f1ff6 100644
--- a/webrtc/call/call_perf_tests.cc
+++ b/webrtc/call/call_perf_tests.cc
@@ -284,7 +284,7 @@
test::FakeDecoder fake_decoder;
- CreateSendConfig(1, &sync_send_transport);
+ CreateSendConfig(1, 0, &sync_send_transport);
CreateMatchingReceiveConfigs(&sync_receive_transport);
AudioSendStream::Config audio_send_config(&audio_send_transport);
@@ -318,9 +318,9 @@
if (create_audio_first) {
audio_receive_stream =
receiver_call_->CreateAudioReceiveStream(audio_recv_config);
- CreateStreams();
+ CreateVideoStreams();
} else {
- CreateStreams();
+ CreateVideoStreams();
audio_receive_stream =
receiver_call_->CreateAudioReceiveStream(audio_recv_config);
}
diff --git a/webrtc/call/packet_injection_tests.cc b/webrtc/call/packet_injection_tests.cc
index 315fc7b..277cd3e 100644
--- a/webrtc/call/packet_injection_tests.cc
+++ b/webrtc/call/packet_injection_tests.cc
@@ -40,7 +40,7 @@
CreateReceiverCall(Call::Config());
test::NullTransport null_transport;
- CreateSendConfig(1, &null_transport);
+ CreateSendConfig(1, 0, &null_transport);
CreateMatchingReceiveConfigs(&null_transport);
video_receive_configs_[0].decoders[0].payload_type = payload_type;
switch (codec_type) {
@@ -51,11 +51,11 @@
video_receive_configs_[0].decoders[0].payload_name = "H264";
break;
}
- CreateStreams();
+ CreateVideoStreams();
RTPHeader header;
EXPECT_TRUE(rtp_header_parser_->Parse(packet, length, &header));
- EXPECT_EQ(kSendSsrcs[0], header.ssrc)
+ EXPECT_EQ(kVideoSendSsrcs[0], header.ssrc)
<< "Packet should have configured SSRC to not be dropped early.";
EXPECT_EQ(payload_type, header.payloadType);
Start();
diff --git a/webrtc/test/call_test.cc b/webrtc/test/call_test.cc
index bbc1224..83fd844 100644
--- a/webrtc/test/call_test.cc
+++ b/webrtc/test/call_test.cc
@@ -7,8 +7,15 @@
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "webrtc/base/checks.h"
+#include "webrtc/common.h"
+#include "webrtc/config.h"
#include "webrtc/test/call_test.h"
#include "webrtc/test/encoder_settings.h"
+#include "webrtc/test/testsupport/fileutils.h"
+#include "webrtc/voice_engine/include/voe_base.h"
+#include "webrtc/voice_engine/include/voe_codec.h"
+#include "webrtc/voice_engine/include/voe_network.h"
namespace webrtc {
namespace test {
@@ -20,17 +27,40 @@
CallTest::CallTest()
: clock_(Clock::GetRealTimeClock()),
video_send_config_(nullptr),
- video_send_stream_(NULL),
- fake_encoder_(clock_) {}
+ video_send_stream_(nullptr),
+ audio_send_config_(nullptr),
+ audio_send_stream_(nullptr),
+ fake_encoder_(clock_),
+ num_video_streams_(0),
+ num_audio_streams_(0),
+ fake_send_audio_device_(nullptr),
+ fake_recv_audio_device_(nullptr) {}
CallTest::~CallTest() {
}
void CallTest::RunBaseTest(BaseTest* test,
const FakeNetworkPipe::Config& config) {
- CreateSenderCall(test->GetSenderCallConfig());
- if (test->ShouldCreateReceivers())
- CreateReceiverCall(test->GetReceiverCallConfig());
+ num_video_streams_ = test->GetNumVideoStreams();
+ num_audio_streams_ = test->GetNumAudioStreams();
+ RTC_DCHECK(num_video_streams_ > 0 || num_audio_streams_ > 0);
+ Call::Config send_config(test->GetSenderCallConfig());
+ if (num_audio_streams_ > 0) {
+ CreateVoiceEngines();
+ AudioState::Config audio_state_config;
+ audio_state_config.voice_engine = voe_send_.voice_engine;
+ send_config.audio_state = AudioState::Create(audio_state_config);
+ }
+ CreateSenderCall(send_config);
+ if (test->ShouldCreateReceivers()) {
+ Call::Config recv_config(test->GetReceiverCallConfig());
+ if (num_audio_streams_ > 0) {
+ AudioState::Config audio_state_config;
+ audio_state_config.voice_engine = voe_recv_.voice_engine;
+ recv_config.audio_state = AudioState::Create(audio_state_config);
+ }
+ CreateReceiverCall(recv_config);
+ }
send_transport_.reset(new PacketTransport(
sender_call_.get(), test, test::PacketTransport::kSender, config));
receive_transport_.reset(new PacketTransport(
@@ -47,14 +77,29 @@
receive_transport_->SetReceiver(nullptr);
}
- CreateSendConfig(test->GetNumStreams(), send_transport_.get());
+ CreateSendConfig(num_video_streams_, num_audio_streams_,
+ send_transport_.get());
if (test->ShouldCreateReceivers()) {
CreateMatchingReceiveConfigs(receive_transport_.get());
}
- test->ModifyVideoConfigs(&video_send_config_, &video_receive_configs_,
- &video_encoder_config_);
- CreateStreams();
- test->OnVideoStreamsCreated(video_send_stream_, video_receive_streams_);
+ if (num_audio_streams_ > 0)
+ SetupVoiceEngineTransports(send_transport_.get(), receive_transport_.get());
+
+ if (num_video_streams_ > 0) {
+ test->ModifyVideoConfigs(&video_send_config_, &video_receive_configs_,
+ &video_encoder_config_);
+ }
+ if (num_audio_streams_ > 0)
+ test->ModifyAudioConfigs(&audio_send_config_, &audio_receive_configs_);
+
+ if (num_video_streams_ > 0) {
+ CreateVideoStreams();
+ test->OnVideoStreamsCreated(video_send_stream_, video_receive_streams_);
+ }
+ if (num_audio_streams_ > 0) {
+ CreateAudioStreams();
+ test->OnAudioStreamsCreated(audio_send_stream_, audio_receive_streams_);
+ }
CreateFrameGeneratorCapturer();
test->OnFrameGeneratorCapturerCreated(frame_generator_capturer_.get());
@@ -66,12 +111,28 @@
Stop();
DestroyStreams();
+ DestroyCalls();
+ if (num_audio_streams_ > 0)
+ DestroyVoiceEngines();
}
void CallTest::Start() {
- video_send_stream_->Start();
- for (size_t i = 0; i < video_receive_streams_.size(); ++i)
- video_receive_streams_[i]->Start();
+ if (video_send_stream_)
+ video_send_stream_->Start();
+ for (VideoReceiveStream* video_recv_stream : video_receive_streams_)
+ video_recv_stream->Start();
+ if (audio_send_stream_) {
+ fake_send_audio_device_->Start();
+ audio_send_stream_->Start();
+ EXPECT_EQ(0, voe_send_.base->StartSend(voe_send_.channel_id));
+ }
+ for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_)
+ audio_recv_stream->Start();
+ if (!audio_receive_streams_.empty()) {
+ fake_recv_audio_device_->Start();
+ EXPECT_EQ(0, voe_recv_.base->StartPlayout(voe_recv_.channel_id));
+ EXPECT_EQ(0, voe_recv_.base->StartReceive(voe_recv_.channel_id));
+ }
if (frame_generator_capturer_.get() != NULL)
frame_generator_capturer_->Start();
}
@@ -79,9 +140,22 @@
void CallTest::Stop() {
if (frame_generator_capturer_.get() != NULL)
frame_generator_capturer_->Stop();
- for (size_t i = 0; i < video_receive_streams_.size(); ++i)
- video_receive_streams_[i]->Stop();
- video_send_stream_->Stop();
+ if (!audio_receive_streams_.empty()) {
+ fake_recv_audio_device_->Stop();
+ EXPECT_EQ(0, voe_recv_.base->StopReceive(voe_recv_.channel_id));
+ EXPECT_EQ(0, voe_recv_.base->StopPlayout(voe_recv_.channel_id));
+ }
+ for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_)
+ audio_recv_stream->Stop();
+ if (audio_send_stream_) {
+ fake_send_audio_device_->Stop();
+ EXPECT_EQ(0, voe_send_.base->StopSend(voe_send_.channel_id));
+ audio_send_stream_->Stop();
+ }
+ for (VideoReceiveStream* video_recv_stream : video_receive_streams_)
+ video_recv_stream->Stop();
+ if (video_send_stream_)
+ video_send_stream_->Stop();
}
void CallTest::CreateCalls(const Call::Config& sender_config,
@@ -99,44 +173,63 @@
}
void CallTest::DestroyCalls() {
- sender_call_.reset(nullptr);
- receiver_call_.reset(nullptr);
+ sender_call_.reset();
+ receiver_call_.reset();
}
-void CallTest::CreateSendConfig(size_t num_streams,
+void CallTest::CreateSendConfig(size_t num_video_streams,
+ size_t num_audio_streams,
Transport* send_transport) {
- assert(num_streams <= kNumSsrcs);
+ RTC_DCHECK(num_video_streams <= kNumSsrcs);
+ RTC_DCHECK_LE(num_audio_streams, 1u);
+ RTC_DCHECK(num_audio_streams == 0 || voe_send_.channel_id >= 0);
video_send_config_ = VideoSendStream::Config(send_transport);
video_send_config_.encoder_settings.encoder = &fake_encoder_;
video_send_config_.encoder_settings.payload_name = "FAKE";
- video_send_config_.encoder_settings.payload_type = kFakeSendPayloadType;
+ video_send_config_.encoder_settings.payload_type = kFakeVideoSendPayloadType;
video_send_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId));
- video_encoder_config_.streams = test::CreateVideoStreams(num_streams);
- for (size_t i = 0; i < num_streams; ++i)
- video_send_config_.rtp.ssrcs.push_back(kSendSsrcs[i]);
+ video_encoder_config_.streams = test::CreateVideoStreams(num_video_streams);
+ for (size_t i = 0; i < num_video_streams; ++i)
+ video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[i]);
video_send_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kVideoRotation, kVideoRotationRtpExtensionId));
+
+ if (num_audio_streams > 0) {
+ audio_send_config_ = AudioSendStream::Config(send_transport);
+ audio_send_config_.voe_channel_id = voe_send_.channel_id;
+ audio_send_config_.rtp.ssrc = kAudioSendSsrc;
+ }
}
-void CallTest::CreateMatchingReceiveConfigs(
- Transport* rtcp_send_transport) {
- assert(!video_send_config_.rtp.ssrcs.empty());
- assert(video_receive_configs_.empty());
- assert(allocated_decoders_.empty());
- VideoReceiveStream::Config config(rtcp_send_transport);
- config.rtp.remb = true;
- config.rtp.local_ssrc = kReceiverLocalSsrc;
+void CallTest::CreateMatchingReceiveConfigs(Transport* rtcp_send_transport) {
+ RTC_DCHECK(!video_send_config_.rtp.ssrcs.empty());
+ RTC_DCHECK(video_receive_configs_.empty());
+ RTC_DCHECK(allocated_decoders_.empty());
+ RTC_DCHECK(num_audio_streams_ == 0 || voe_send_.channel_id >= 0);
+ VideoReceiveStream::Config video_config(rtcp_send_transport);
+ video_config.rtp.remb = true;
+ video_config.rtp.local_ssrc = kReceiverLocalVideoSsrc;
for (const RtpExtension& extension : video_send_config_.rtp.extensions)
- config.rtp.extensions.push_back(extension);
+ video_config.rtp.extensions.push_back(extension);
for (size_t i = 0; i < video_send_config_.rtp.ssrcs.size(); ++i) {
VideoReceiveStream::Decoder decoder =
test::CreateMatchingDecoder(video_send_config_.encoder_settings);
allocated_decoders_.push_back(decoder.decoder);
- config.decoders.clear();
- config.decoders.push_back(decoder);
- config.rtp.remote_ssrc = video_send_config_.rtp.ssrcs[i];
- video_receive_configs_.push_back(config);
+ video_config.decoders.clear();
+ video_config.decoders.push_back(decoder);
+ video_config.rtp.remote_ssrc = video_send_config_.rtp.ssrcs[i];
+ video_receive_configs_.push_back(video_config);
+ }
+
+ RTC_DCHECK(num_audio_streams_ <= 1);
+ if (num_audio_streams_ == 1) {
+ AudioReceiveStream::Config audio_config;
+ audio_config.rtp.local_ssrc = kReceiverLocalAudioSsrc;
+ audio_config.rtcp_send_transport = rtcp_send_transport;
+ audio_config.voe_channel_id = voe_recv_.channel_id;
+ audio_config.rtp.remote_ssrc = audio_send_config_.rtp.ssrc;
+ audio_receive_configs_.push_back(audio_config);
}
}
@@ -147,41 +240,131 @@
stream.max_framerate, clock_));
}
-void CallTest::CreateStreams() {
- assert(video_send_stream_ == NULL);
- assert(video_receive_streams_.empty());
+void CallTest::CreateFakeAudioDevices() {
+ fake_send_audio_device_.reset(new FakeAudioDevice(
+ clock_, test::ResourcePath("voice_engine/audio_long16", "pcm")));
+ fake_recv_audio_device_.reset(new FakeAudioDevice(
+ clock_, test::ResourcePath("voice_engine/audio_long16", "pcm")));
+}
+
+void CallTest::CreateVideoStreams() {
+ RTC_DCHECK(video_send_stream_ == nullptr);
+ RTC_DCHECK(video_receive_streams_.empty());
+ RTC_DCHECK(audio_send_stream_ == nullptr);
+ RTC_DCHECK(audio_receive_streams_.empty());
video_send_stream_ = sender_call_->CreateVideoSendStream(
video_send_config_, video_encoder_config_);
-
for (size_t i = 0; i < video_receive_configs_.size(); ++i) {
video_receive_streams_.push_back(
receiver_call_->CreateVideoReceiveStream(video_receive_configs_[i]));
}
}
+void CallTest::CreateAudioStreams() {
+ audio_send_stream_ = sender_call_->CreateAudioSendStream(audio_send_config_);
+ for (size_t i = 0; i < audio_receive_configs_.size(); ++i) {
+ audio_receive_streams_.push_back(
+ receiver_call_->CreateAudioReceiveStream(audio_receive_configs_[i]));
+ }
+ CodecInst isac = {kAudioSendPayloadType, "ISAC", 16000, 480, 1, 32000};
+ EXPECT_EQ(0, voe_send_.codec->SetSendCodec(voe_send_.channel_id, isac));
+}
+
void CallTest::DestroyStreams() {
- if (video_send_stream_ != NULL)
+ if (video_send_stream_)
sender_call_->DestroyVideoSendStream(video_send_stream_);
- video_send_stream_ = NULL;
- for (size_t i = 0; i < video_receive_streams_.size(); ++i)
- receiver_call_->DestroyVideoReceiveStream(video_receive_streams_[i]);
+ video_send_stream_ = nullptr;
+ for (VideoReceiveStream* video_recv_stream : video_receive_streams_)
+ receiver_call_->DestroyVideoReceiveStream(video_recv_stream);
+
+ if (audio_send_stream_)
+ sender_call_->DestroyAudioSendStream(audio_send_stream_);
+ audio_send_stream_ = nullptr;
+ for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_)
+ receiver_call_->DestroyAudioReceiveStream(audio_recv_stream);
video_receive_streams_.clear();
+
allocated_decoders_.clear();
}
+void CallTest::CreateVoiceEngines() {
+ CreateFakeAudioDevices();
+ voe_send_.voice_engine = VoiceEngine::Create();
+ voe_send_.base = VoEBase::GetInterface(voe_send_.voice_engine);
+ voe_send_.network = VoENetwork::GetInterface(voe_send_.voice_engine);
+ voe_send_.codec = VoECodec::GetInterface(voe_send_.voice_engine);
+ EXPECT_EQ(0, voe_send_.base->Init(fake_send_audio_device_.get(), nullptr));
+ Config voe_config;
+ voe_config.Set<VoicePacing>(new VoicePacing(true));
+ voe_send_.channel_id = voe_send_.base->CreateChannel(voe_config);
+ EXPECT_GE(voe_send_.channel_id, 0);
+
+ voe_recv_.voice_engine = VoiceEngine::Create();
+ voe_recv_.base = VoEBase::GetInterface(voe_recv_.voice_engine);
+ voe_recv_.network = VoENetwork::GetInterface(voe_recv_.voice_engine);
+ voe_recv_.codec = VoECodec::GetInterface(voe_recv_.voice_engine);
+ EXPECT_EQ(0, voe_recv_.base->Init(fake_recv_audio_device_.get(), nullptr));
+ voe_recv_.channel_id = voe_recv_.base->CreateChannel();
+ EXPECT_GE(voe_recv_.channel_id, 0);
+}
+
+void CallTest::SetupVoiceEngineTransports(PacketTransport* send_transport,
+ PacketTransport* recv_transport) {
+ voe_send_.transport_adapter.reset(
+ new internal::TransportAdapter(send_transport));
+ voe_send_.transport_adapter->Enable();
+ EXPECT_EQ(0, voe_send_.network->RegisterExternalTransport(
+ voe_send_.channel_id, *voe_send_.transport_adapter.get()));
+
+ voe_recv_.transport_adapter.reset(
+ new internal::TransportAdapter(recv_transport));
+ voe_recv_.transport_adapter->Enable();
+ EXPECT_EQ(0, voe_recv_.network->RegisterExternalTransport(
+ voe_recv_.channel_id, *voe_recv_.transport_adapter.get()));
+}
+
+void CallTest::DestroyVoiceEngines() {
+ voe_recv_.base->DeleteChannel(voe_recv_.channel_id);
+ voe_recv_.channel_id = -1;
+ voe_recv_.base->Release();
+ voe_recv_.base = nullptr;
+ voe_recv_.network->Release();
+ voe_recv_.network = nullptr;
+ voe_recv_.codec->Release();
+ voe_recv_.codec = nullptr;
+
+ voe_send_.base->DeleteChannel(voe_send_.channel_id);
+ voe_send_.channel_id = -1;
+ voe_send_.base->Release();
+ voe_send_.base = nullptr;
+ voe_send_.network->Release();
+ voe_send_.network = nullptr;
+ voe_send_.codec->Release();
+ voe_send_.codec = nullptr;
+
+ VoiceEngine::Delete(voe_send_.voice_engine);
+ voe_send_.voice_engine = nullptr;
+ VoiceEngine::Delete(voe_recv_.voice_engine);
+ voe_recv_.voice_engine = nullptr;
+}
+
const int CallTest::kDefaultTimeoutMs = 30 * 1000;
const int CallTest::kLongTimeoutMs = 120 * 1000;
-const uint8_t CallTest::kSendPayloadType = 100;
-const uint8_t CallTest::kFakeSendPayloadType = 125;
+const uint8_t CallTest::kVideoSendPayloadType = 100;
+const uint8_t CallTest::kFakeVideoSendPayloadType = 125;
const uint8_t CallTest::kSendRtxPayloadType = 98;
const uint8_t CallTest::kRedPayloadType = 118;
const uint8_t CallTest::kRtxRedPayloadType = 99;
const uint8_t CallTest::kUlpfecPayloadType = 119;
+const uint8_t CallTest::kAudioSendPayloadType = 103;
const uint32_t CallTest::kSendRtxSsrcs[kNumSsrcs] = {0xBADCAFD, 0xBADCAFE,
0xBADCAFF};
-const uint32_t CallTest::kSendSsrcs[kNumSsrcs] = {0xC0FFED, 0xC0FFEE, 0xC0FFEF};
-const uint32_t CallTest::kReceiverLocalSsrc = 0x123456;
+const uint32_t CallTest::kVideoSendSsrcs[kNumSsrcs] = {0xC0FFED, 0xC0FFEE,
+ 0xC0FFEF};
+const uint32_t CallTest::kAudioSendSsrc = 0xDEADBEEF;
+const uint32_t CallTest::kReceiverLocalVideoSsrc = 0x123456;
+const uint32_t CallTest::kReceiverLocalAudioSsrc = 0x1234567;
const int CallTest::kNackRtpHistoryMs = 1000;
BaseTest::BaseTest(unsigned int timeout_ms) : RtpRtcpObserver(timeout_ms) {
@@ -204,10 +387,14 @@
void BaseTest::OnTransportsCreated(PacketTransport* send_transport,
PacketTransport* receive_transport) {}
-size_t BaseTest::GetNumStreams() const {
+size_t BaseTest::GetNumVideoStreams() const {
return 1;
}
+size_t BaseTest::GetNumAudioStreams() const {
+ return 0;
+}
+
void BaseTest::ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
@@ -217,6 +404,14 @@
VideoSendStream* send_stream,
const std::vector<VideoReceiveStream*>& receive_streams) {}
+void BaseTest::ModifyAudioConfigs(
+ AudioSendStream::Config* send_config,
+ std::vector<AudioReceiveStream::Config>* receive_configs) {}
+
+void BaseTest::OnAudioStreamsCreated(
+ AudioSendStream* send_stream,
+ const std::vector<AudioReceiveStream*>& receive_streams) {}
+
void BaseTest::OnFrameGeneratorCapturerCreated(
FrameGeneratorCapturer* frame_generator_capturer) {
}
diff --git a/webrtc/test/call_test.h b/webrtc/test/call_test.h
index 32820ed..46fbe7f 100644
--- a/webrtc/test/call_test.h
+++ b/webrtc/test/call_test.h
@@ -7,19 +7,26 @@
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_TEST_COMMON_CALL_TEST_H_
-#define WEBRTC_TEST_COMMON_CALL_TEST_H_
+#ifndef WEBRTC_TEST_CALL_TEST_H_
+#define WEBRTC_TEST_CALL_TEST_H_
#include <vector>
#include "webrtc/call.h"
+#include "webrtc/call/transport_adapter.h"
#include "webrtc/system_wrappers/include/scoped_vector.h"
+#include "webrtc/test/fake_audio_device.h"
#include "webrtc/test/fake_decoder.h"
#include "webrtc/test/fake_encoder.h"
#include "webrtc/test/frame_generator_capturer.h"
#include "webrtc/test/rtp_rtcp_observer.h"
namespace webrtc {
+
+class VoEBase;
+class VoECodec;
+class VoENetwork;
+
namespace test {
class BaseTest;
@@ -27,24 +34,30 @@
class CallTest : public ::testing::Test {
public:
CallTest();
- ~CallTest();
+ virtual ~CallTest();
static const size_t kNumSsrcs = 3;
static const int kDefaultTimeoutMs;
static const int kLongTimeoutMs;
- static const uint8_t kSendPayloadType;
+ static const uint8_t kVideoSendPayloadType;
static const uint8_t kSendRtxPayloadType;
- static const uint8_t kFakeSendPayloadType;
+ static const uint8_t kFakeVideoSendPayloadType;
static const uint8_t kRedPayloadType;
static const uint8_t kRtxRedPayloadType;
static const uint8_t kUlpfecPayloadType;
+ static const uint8_t kAudioSendPayloadType;
static const uint32_t kSendRtxSsrcs[kNumSsrcs];
- static const uint32_t kSendSsrcs[kNumSsrcs];
- static const uint32_t kReceiverLocalSsrc;
+ static const uint32_t kVideoSendSsrcs[kNumSsrcs];
+ static const uint32_t kAudioSendSsrc;
+ static const uint32_t kReceiverLocalVideoSsrc;
+ static const uint32_t kReceiverLocalAudioSsrc;
static const int kNackRtpHistoryMs;
protected:
+ // 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.
void RunBaseTest(BaseTest* test, const FakeNetworkPipe::Config& config);
void CreateCalls(const Call::Config& sender_config,
@@ -53,12 +66,16 @@
void CreateReceiverCall(const Call::Config& config);
void DestroyCalls();
- void CreateSendConfig(size_t num_streams, Transport* send_transport);
+ void CreateSendConfig(size_t num_video_streams,
+ size_t num_audio_streams,
+ Transport* send_transport);
void CreateMatchingReceiveConfigs(Transport* rtcp_send_transport);
void CreateFrameGeneratorCapturer();
+ void CreateFakeAudioDevices();
- void CreateStreams();
+ void CreateVideoStreams();
+ void CreateAudioStreams();
void Start();
void Stop();
void DestroyStreams();
@@ -70,15 +87,54 @@
VideoSendStream::Config video_send_config_;
VideoEncoderConfig video_encoder_config_;
VideoSendStream* video_send_stream_;
+ AudioSendStream::Config audio_send_config_;
+ AudioSendStream* audio_send_stream_;
rtc::scoped_ptr<Call> receiver_call_;
rtc::scoped_ptr<PacketTransport> receive_transport_;
std::vector<VideoReceiveStream::Config> video_receive_configs_;
std::vector<VideoReceiveStream*> video_receive_streams_;
+ std::vector<AudioReceiveStream::Config> audio_receive_configs_;
+ std::vector<AudioReceiveStream*> audio_receive_streams_;
rtc::scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer_;
test::FakeEncoder fake_encoder_;
ScopedVector<VideoDecoder> allocated_decoders_;
+ size_t num_video_streams_;
+ size_t num_audio_streams_;
+
+ 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
+ // necessary while voice engine is being refactored to the new stream API.
+ struct VoiceEngineState {
+ VoiceEngineState()
+ : voice_engine(nullptr),
+ base(nullptr),
+ network(nullptr),
+ codec(nullptr),
+ channel_id(-1),
+ transport_adapter(nullptr) {}
+
+ VoiceEngine* voice_engine;
+ VoEBase* base;
+ VoENetwork* network;
+ VoECodec* codec;
+ int channel_id;
+ rtc::scoped_ptr<internal::TransportAdapter> transport_adapter;
+ };
+
+ void CreateVoiceEngines();
+ void SetupVoiceEngineTransports(PacketTransport* send_transport,
+ PacketTransport* recv_transport);
+ void DestroyVoiceEngines();
+
+ VoiceEngineState voe_send_;
+ VoiceEngineState voe_recv_;
+
+ // The audio devices must outlive the voice engines.
+ rtc::scoped_ptr<test::FakeAudioDevice> fake_send_audio_device_;
+ rtc::scoped_ptr<test::FakeAudioDevice> fake_recv_audio_device_;
};
class BaseTest : public RtpRtcpObserver {
@@ -89,7 +145,8 @@
virtual void PerformTest() = 0;
virtual bool ShouldCreateReceivers() const = 0;
- virtual size_t GetNumStreams() const;
+ virtual size_t GetNumVideoStreams() const;
+ virtual size_t GetNumAudioStreams() const;
virtual Call::Config GetSenderCallConfig();
virtual Call::Config GetReceiverCallConfig();
@@ -105,6 +162,13 @@
VideoSendStream* send_stream,
const std::vector<VideoReceiveStream*>& receive_streams);
+ virtual void ModifyAudioConfigs(
+ AudioSendStream::Config* send_config,
+ std::vector<AudioReceiveStream::Config>* receive_configs);
+ virtual void OnAudioStreamsCreated(
+ AudioSendStream* send_stream,
+ const std::vector<AudioReceiveStream*>& receive_streams);
+
virtual void OnFrameGeneratorCapturerCreated(
FrameGeneratorCapturer* frame_generator_capturer);
};
@@ -126,4 +190,4 @@
} // namespace test
} // namespace webrtc
-#endif // WEBRTC_TEST_COMMON_CALL_TEST_H_
+#endif // WEBRTC_TEST_CALL_TEST_H_
diff --git a/webrtc/video/end_to_end_tests.cc b/webrtc/video/end_to_end_tests.cc
index f654dbb..3c774ab 100644
--- a/webrtc/video/end_to_end_tests.cc
+++ b/webrtc/video/end_to_end_tests.cc
@@ -31,7 +31,6 @@
#include "webrtc/test/call_test.h"
#include "webrtc/test/direct_transport.h"
#include "webrtc/test/encoder_settings.h"
-#include "webrtc/test/fake_audio_device.h"
#include "webrtc/test/fake_decoder.h"
#include "webrtc/test/fake_encoder.h"
#include "webrtc/test/frame_generator.h"
@@ -86,10 +85,10 @@
CreateCalls(Call::Config(), Call::Config());
test::NullTransport transport;
- CreateSendConfig(1, &transport);
+ CreateSendConfig(1, 0, &transport);
CreateMatchingReceiveConfigs(&transport);
- CreateStreams();
+ CreateVideoStreams();
video_receive_streams_[0]->Start();
video_receive_streams_[0]->Start();
@@ -101,10 +100,10 @@
CreateCalls(Call::Config(), Call::Config());
test::NullTransport transport;
- CreateSendConfig(1, &transport);
+ CreateSendConfig(1, 0, &transport);
CreateMatchingReceiveConfigs(&transport);
- CreateStreams();
+ CreateVideoStreams();
video_receive_streams_[0]->Stop();
video_receive_streams_[0]->Stop();
@@ -158,14 +157,14 @@
sender_transport.SetReceiver(receiver_call_->Receiver());
receiver_transport.SetReceiver(sender_call_->Receiver());
- CreateSendConfig(1, &sender_transport);
+ CreateSendConfig(1, 0, &sender_transport);
CreateMatchingReceiveConfigs(&receiver_transport);
TestFrameCallback pre_render_callback;
video_receive_configs_[0].pre_render_callback = &pre_render_callback;
video_receive_configs_[0].renderer = &renderer;
- CreateStreams();
+ CreateVideoStreams();
Start();
// Create frames that are smaller than the send width/height, this is done to
@@ -210,11 +209,11 @@
sender_transport.SetReceiver(receiver_call_->Receiver());
receiver_transport.SetReceiver(sender_call_->Receiver());
- CreateSendConfig(1, &sender_transport);
+ CreateSendConfig(1, 0, &sender_transport);
CreateMatchingReceiveConfigs(&receiver_transport);
video_receive_configs_[0].renderer = &renderer;
- CreateStreams();
+ CreateVideoStreams();
Start();
rtc::scoped_ptr<test::FrameGenerator> frame_generator(
@@ -308,7 +307,7 @@
(*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
send_config->encoder_settings.encoder = &fake_encoder_;
send_config->encoder_settings.payload_name = "H264";
- send_config->encoder_settings.payload_type = kFakeSendPayloadType;
+ send_config->encoder_settings.payload_type = kFakeVideoSendPayloadType;
encoder_config->streams[0].min_bitrate_bps = 50000;
encoder_config->streams[0].target_bitrate_bps =
encoder_config->streams[0].max_bitrate_bps = 2000000;
@@ -353,7 +352,7 @@
ssrc |= static_cast<uint32_t>(packet[5]) << 16;
ssrc |= static_cast<uint32_t>(packet[6]) << 8;
ssrc |= static_cast<uint32_t>(packet[7]) << 0;
- EXPECT_EQ(kReceiverLocalSsrc, ssrc);
+ EXPECT_EQ(kReceiverLocalVideoSsrc, ssrc);
observation_complete_.Set();
return SEND_PACKET;
@@ -474,10 +473,10 @@
if (header.payloadType == kRedPayloadType) {
encapsulated_payload_type =
static_cast<int>(packet[header.headerLength]);
- if (encapsulated_payload_type != kFakeSendPayloadType)
+ if (encapsulated_payload_type != kFakeVideoSendPayloadType)
EXPECT_EQ(kUlpfecPayloadType, encapsulated_payload_type);
} else {
- EXPECT_EQ(kFakeSendPayloadType, header.payloadType);
+ EXPECT_EQ(kFakeVideoSendPayloadType, header.payloadType);
}
if (protected_sequence_numbers_.count(header.sequenceNumber) != 0) {
@@ -501,7 +500,7 @@
return DROP_PACKET;
break;
case kDropNextMediaPacket:
- if (encapsulated_payload_type == kFakeSendPayloadType) {
+ if (encapsulated_payload_type == kFakeVideoSendPayloadType) {
protected_sequence_numbers_.insert(header.sequenceNumber);
protected_timestamps_.insert(header.timestamp);
state_ = kDropEveryOtherPacketUntilFec;
@@ -580,10 +579,10 @@
if (header.payloadType == kRedPayloadType) {
encapsulated_payload_type =
static_cast<int>(packet[header.headerLength]);
- if (encapsulated_payload_type != kFakeSendPayloadType)
+ if (encapsulated_payload_type != kFakeVideoSendPayloadType)
EXPECT_EQ(kUlpfecPayloadType, encapsulated_payload_type);
} else {
- EXPECT_EQ(kFakeSendPayloadType, header.payloadType);
+ EXPECT_EQ(kFakeVideoSendPayloadType, header.payloadType);
}
if (has_last_sequence_number_ &&
@@ -698,7 +697,7 @@
explicit RetransmissionObserver(bool use_rtx, bool use_red)
: EndToEndTest(kDefaultTimeoutMs),
payload_type_(GetPayloadType(false, use_red)),
- retransmission_ssrc_(use_rtx ? kSendRtxSsrcs[0] : kSendSsrcs[0]),
+ retransmission_ssrc_(use_rtx ? kSendRtxSsrcs[0] : kVideoSendSsrcs[0]),
retransmission_payload_type_(GetPayloadType(use_rtx, use_red)),
marker_bits_observed_(0),
num_packets_observed_(0),
@@ -726,7 +725,7 @@
return SEND_PACKET;
}
- EXPECT_EQ(kSendSsrcs[0], header.ssrc);
+ EXPECT_EQ(kVideoSendSsrcs[0], header.ssrc);
EXPECT_EQ(payload_type_, header.payloadType);
// Found the final packet of the frame to inflict loss to, drop this and
@@ -765,9 +764,9 @@
if (retransmission_ssrc_ == kSendRtxSsrcs[0]) {
send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[0]);
send_config->rtp.rtx.payload_type = kSendRtxPayloadType;
- (*receive_configs)[0].rtp.rtx[kFakeSendPayloadType].ssrc =
+ (*receive_configs)[0].rtp.rtx[kFakeVideoSendPayloadType].ssrc =
kSendRtxSsrcs[0];
- (*receive_configs)[0].rtp.rtx[kFakeSendPayloadType].payload_type =
+ (*receive_configs)[0].rtp.rtx[kFakeVideoSendPayloadType].payload_type =
kSendRtxPayloadType;
}
}
@@ -779,7 +778,7 @@
int GetPayloadType(bool use_rtx, bool use_red) {
return use_rtx ? kSendRtxPayloadType
- : (use_red ? kRedPayloadType : kFakeSendPayloadType);
+ : (use_red ? kRedPayloadType : kFakeVideoSendPayloadType);
}
rtc::CriticalSection crit_;
@@ -876,7 +875,7 @@
sender_transport.SetReceiver(receiver_call_->Receiver());
receiver_transport.SetReceiver(sender_call_->Receiver());
- CreateSendConfig(1, &sender_transport);
+ CreateSendConfig(1, 0, &sender_transport);
rtc::scoped_ptr<VideoEncoder> encoder(
VideoEncoder::Create(VideoEncoder::kVp8));
video_send_config_.encoder_settings.encoder = encoder.get();
@@ -890,7 +889,7 @@
video_receive_configs_[0].pre_render_callback = &pre_render_callback;
video_receive_configs_[0].renderer = &renderer;
- CreateStreams();
+ CreateVideoStreams();
Start();
// Create frames that are smaller than the send width/height, this is done to
@@ -1050,10 +1049,10 @@
send_transport.SetReceiver(&input_observer);
receive_transport.SetReceiver(sender_call_->Receiver());
- CreateSendConfig(1, &send_transport);
+ CreateSendConfig(1, 0, &send_transport);
CreateMatchingReceiveConfigs(&receive_transport);
- CreateStreams();
+ CreateVideoStreams();
CreateFrameGeneratorCapturer();
Start();
@@ -1227,7 +1226,7 @@
VideoReceiveStream::Config receive_config(receiver_transport.get());
receive_config.rtp.remote_ssrc = ssrc;
- receive_config.rtp.local_ssrc = test::CallTest::kReceiverLocalSsrc;
+ receive_config.rtp.local_ssrc = test::CallTest::kReceiverLocalVideoSsrc;
VideoReceiveStream::Decoder decoder =
test::CreateMatchingDecoder(send_config.encoder_settings);
allocated_decoders.push_back(decoder.decoder);
@@ -1659,12 +1658,12 @@
sender_transport.SetReceiver(receiver_call_->Receiver());
receiver_transport.SetReceiver(sender_call_->Receiver());
- CreateSendConfig(1, &sender_transport);
+ CreateSendConfig(1, 0, &sender_transport);
CreateMatchingReceiveConfigs(&receiver_transport);
video_send_config_.post_encode_callback = &post_encode_observer;
video_receive_configs_[0].pre_decode_callback = &pre_decode_observer;
- CreateStreams();
+ CreateVideoStreams();
Start();
rtc::scoped_ptr<test::FrameGenerator> frame_generator(
@@ -1705,13 +1704,13 @@
while (packet_type != RTCPUtility::RTCPPacketTypes::kInvalid) {
if (packet_type == RTCPUtility::RTCPPacketTypes::kPsfbRemb) {
const RTCPUtility::RTCPPacket& packet = parser.Packet();
- EXPECT_EQ(packet.PSFBAPP.SenderSSRC, kReceiverLocalSsrc);
+ EXPECT_EQ(packet.PSFBAPP.SenderSSRC, kReceiverLocalVideoSsrc);
received_psfb = true;
} else if (packet_type == RTCPUtility::RTCPPacketTypes::kPsfbRembItem) {
const RTCPUtility::RTCPPacket& packet = parser.Packet();
EXPECT_GT(packet.REMBItem.BitRate, 0u);
EXPECT_EQ(packet.REMBItem.NumberOfSSRCs, 1u);
- EXPECT_EQ(packet.REMBItem.SSRCs[0], kSendSsrcs[0]);
+ EXPECT_EQ(packet.REMBItem.SSRCs[0], kVideoSendSsrcs[0]);
received_remb = true;
}
packet_type = parser.Iterate();
@@ -1825,8 +1824,7 @@
receive_stream_nack_packets +=
stats.rtcp_packet_type_counts.nack_packets;
}
- if (send_stream_nack_packets >= 1 &&
- receive_stream_nack_packets >= 1) {
+ if (send_stream_nack_packets >= 1 && receive_stream_nack_packets >= 1) {
// NACK packet sent on receive stream and received on sent stream.
if (MinMetricRunTimePassed())
observation_complete_.Set();
@@ -1940,9 +1938,9 @@
if (use_rtx_) {
send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[0]);
send_config->rtp.rtx.payload_type = kSendRtxPayloadType;
- (*receive_configs)[0].rtp.rtx[kFakeSendPayloadType].ssrc =
+ (*receive_configs)[0].rtp.rtx[kFakeVideoSendPayloadType].ssrc =
kSendRtxSsrcs[0];
- (*receive_configs)[0].rtp.rtx[kFakeSendPayloadType].payload_type =
+ (*receive_configs)[0].rtp.rtx[kFakeVideoSendPayloadType].payload_type =
kSendRtxPayloadType;
}
encoder_config->content_type =
@@ -2236,7 +2234,7 @@
return SEND_PACKET;
}
- size_t GetNumStreams() const override { return num_ssrcs_; }
+ size_t GetNumVideoStreams() const override { return num_ssrcs_; }
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
@@ -2287,7 +2285,7 @@
VideoSendStream* send_stream_;
VideoEncoderConfig video_encoder_config_all_streams_;
- } test(kSendSsrcs, num_ssrcs, send_single_ssrc_first);
+ } test(kVideoSendSsrcs, num_ssrcs, send_single_ssrc_first);
RunBaseTest(&test, FakeNetworkPipe::Config());
}
@@ -2443,9 +2441,9 @@
stats.rtcp_packet_type_counts.unique_nack_requests != 0;
assert(stats.current_payload_type == -1 ||
- stats.current_payload_type == kFakeSendPayloadType);
+ stats.current_payload_type == kFakeVideoSendPayloadType);
receive_stats_filled_["IncomingPayloadType"] |=
- stats.current_payload_type == kFakeSendPayloadType;
+ stats.current_payload_type == kFakeVideoSendPayloadType;
}
return AllStatsFilled(receive_stats_filled_);
@@ -2552,7 +2550,7 @@
}
}
- size_t GetNumStreams() const override { return kNumSsrcs; }
+ size_t GetNumVideoStreams() const override { return kNumSsrcs; }
void OnVideoStreamsCreated(
VideoSendStream* send_stream,
@@ -2713,7 +2711,7 @@
return SEND_PACKET;
}
- size_t GetNumStreams() const override { return kNumSsrcs; }
+ size_t GetNumVideoStreams() const override { return kNumSsrcs; }
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
@@ -2759,7 +2757,7 @@
: test::RtpRtcpObserver(kDefaultTimeoutMs),
ssrcs_to_observe_(kNumSsrcs) {
for (size_t i = 0; i < kNumSsrcs; ++i) {
- configured_ssrcs_[kSendSsrcs[i]] = true;
+ configured_ssrcs_[kVideoSendSsrcs[i]] = true;
if (use_rtx)
configured_ssrcs_[kSendRtxSsrcs[i]] = true;
}
@@ -2852,7 +2850,7 @@
send_transport.SetReceiver(receiver_call_->Receiver());
receive_transport.SetReceiver(sender_call_->Receiver());
- CreateSendConfig(kNumSsrcs, &send_transport);
+ CreateSendConfig(kNumSsrcs, 0, &send_transport);
if (use_rtx) {
for (size_t i = 0; i < kNumSsrcs; ++i) {
@@ -2883,7 +2881,7 @@
CreateMatchingReceiveConfigs(&receive_transport);
- CreateStreams();
+ CreateVideoStreams();
CreateFrameGeneratorCapturer();
Start();
@@ -3129,10 +3127,10 @@
sender_transport.SetReceiver(receiver_call_->Receiver());
receiver_transport.SetReceiver(sender_call_->Receiver());
- CreateSendConfig(1, &sender_transport);
+ CreateSendConfig(1, 0, &sender_transport);
CreateMatchingReceiveConfigs(&receiver_transport);
- CreateStreams();
+ CreateVideoStreams();
CreateFrameGeneratorCapturer();
Start();
@@ -3170,10 +3168,10 @@
sender_call_->SignalNetworkState(kNetworkDown);
UnusedTransport transport;
- CreateSendConfig(1, &transport);
+ CreateSendConfig(1, 0, &transport);
UnusedEncoder unused_encoder;
video_send_config_.encoder_settings.encoder = &unused_encoder;
- CreateStreams();
+ CreateVideoStreams();
CreateFrameGeneratorCapturer();
Start();
@@ -3189,10 +3187,10 @@
test::DirectTransport sender_transport(sender_call_.get());
sender_transport.SetReceiver(receiver_call_->Receiver());
- CreateSendConfig(1, &sender_transport);
+ CreateSendConfig(1, 0, &sender_transport);
UnusedTransport transport;
CreateMatchingReceiveConfigs(&transport);
- CreateStreams();
+ CreateVideoStreams();
CreateFrameGeneratorCapturer();
Start();
@@ -3249,4 +3247,76 @@
VerifyEmptyFecConfig(default_receive_config.rtp.fec);
}
+TEST_F(EndToEndTest, TransportSeqNumOnAudioAndVideo) {
+ static const int kExtensionId = 8;
+ class TransportSequenceNumberTest : public test::EndToEndTest {
+ public:
+ TransportSequenceNumberTest()
+ : EndToEndTest(kDefaultTimeoutMs),
+ video_observed_(false),
+ audio_observed_(false) {
+ parser_->RegisterRtpHeaderExtension(kRtpExtensionTransportSequenceNumber,
+ kExtensionId);
+ }
+
+ size_t GetNumVideoStreams() const override { return 1; }
+ size_t GetNumAudioStreams() const override { return 1; }
+
+ void ModifyVideoConfigs(
+ VideoSendStream::Config* send_config,
+ std::vector<VideoReceiveStream::Config>* receive_configs,
+ VideoEncoderConfig* encoder_config) override {
+ send_config->rtp.extensions.clear();
+ send_config->rtp.extensions.push_back(
+ RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId));
+ (*receive_configs)[0].rtp.extensions = send_config->rtp.extensions;
+ }
+
+ void ModifyAudioConfigs(
+ AudioSendStream::Config* send_config,
+ std::vector<AudioReceiveStream::Config>* receive_configs) override {
+ send_config->rtp.extensions.clear();
+ send_config->rtp.extensions.push_back(
+ RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId));
+ (*receive_configs)[0].rtp.extensions.clear();
+ (*receive_configs)[0].rtp.extensions = send_config->rtp.extensions;
+ }
+
+ Action OnSendRtp(const uint8_t* packet, size_t length) override {
+ RTPHeader header;
+ EXPECT_TRUE(parser_->Parse(packet, length, &header));
+ EXPECT_TRUE(header.extension.hasTransportSequenceNumber);
+ // Unwrap packet id and verify uniqueness.
+ int64_t packet_id =
+ unwrapper_.Unwrap(header.extension.transportSequenceNumber);
+ EXPECT_TRUE(received_packet_ids_.insert(packet_id).second);
+
+ if (header.ssrc == kVideoSendSsrcs[0])
+ video_observed_ = true;
+ if (header.ssrc == kAudioSendSsrc)
+ audio_observed_ = true;
+ if (audio_observed_ && video_observed_ &&
+ received_packet_ids_.size() == 50) {
+ size_t packet_id_range =
+ *received_packet_ids_.rbegin() - *received_packet_ids_.begin() + 1;
+ EXPECT_EQ(received_packet_ids_.size(), packet_id_range);
+ observation_complete_.Set();
+ }
+ return SEND_PACKET;
+ }
+
+ void PerformTest() override {
+ EXPECT_TRUE(Wait()) << "Timed out while waiting for audio and video "
+ "packets with transport sequence number.";
+ }
+
+ private:
+ bool video_observed_;
+ bool audio_observed_;
+ SequenceNumberUnwrapper unwrapper_;
+ std::set<int64_t> received_packet_ids_;
+ } test;
+
+ RunBaseTest(&test, FakeNetworkPipe::Config());
+}
} // namespace webrtc
diff --git a/webrtc/video/video_quality_test.cc b/webrtc/video/video_quality_test.cc
index 5b23643..08ae0a9 100644
--- a/webrtc/video/video_quality_test.cc
+++ b/webrtc/video/video_quality_test.cc
@@ -781,7 +781,7 @@
trace_to_stderr_.reset(new test::TraceToStderr);
size_t num_streams = params_.ss.streams.size();
- CreateSendConfig(num_streams, send_transport);
+ CreateSendConfig(num_streams, 0, send_transport);
int payload_type;
if (params_.common.codec == "VP8") {
@@ -964,7 +964,7 @@
disable_quality_check ? -1.1 : params_.analyzer.avg_ssim_threshold,
params_.analyzer.test_durations_secs * params_.common.fps,
graph_data_output_file, graph_title,
- kSendSsrcs[params_.ss.selected_stream]);
+ kVideoSendSsrcs[params_.ss.selected_stream]);
analyzer.SetReceiver(receiver_call_->Receiver());
send_transport.SetReceiver(&analyzer);
@@ -979,7 +979,7 @@
if (params_.screenshare.enabled)
SetupScreenshare();
- CreateStreams();
+ CreateVideoStreams();
analyzer.input_ = video_send_stream_->Input();
analyzer.send_stream_ = video_send_stream_;
diff --git a/webrtc/video/video_send_stream_tests.cc b/webrtc/video/video_send_stream_tests.cc
index 24ee296..f0bac12 100644
--- a/webrtc/video/video_send_stream_tests.cc
+++ b/webrtc/video/video_send_stream_tests.cc
@@ -68,8 +68,8 @@
CreateSenderCall(call_config);
test::NullTransport transport;
- CreateSendConfig(1, &transport);
- CreateStreams();
+ CreateSendConfig(1, 0, &transport);
+ CreateVideoStreams();
video_send_stream_->Start();
video_send_stream_->Start();
DestroyStreams();
@@ -80,8 +80,8 @@
CreateSenderCall(call_config);
test::NullTransport transport;
- CreateSendConfig(1, &transport);
- CreateStreams();
+ CreateSendConfig(1, 0, &transport);
+ CreateVideoStreams();
video_send_stream_->Stop();
video_send_stream_->Stop();
DestroyStreams();
@@ -327,14 +327,14 @@
if (send_count_++ % 2 != 0) {
// Receive statistics reporting having lost 50% of the packets.
FakeReceiveStatistics lossy_receive_stats(
- VideoSendStreamTest::kSendSsrcs[0], header.sequenceNumber,
+ VideoSendStreamTest::kVideoSendSsrcs[0], header.sequenceNumber,
send_count_ / 2, 127);
RTCPSender rtcp_sender(false, Clock::GetRealTimeClock(),
&lossy_receive_stats, nullptr,
transport_adapter_.get());
rtcp_sender.SetRTCPStatus(RtcpMode::kReducedSize);
- rtcp_sender.SetRemoteSSRC(VideoSendStreamTest::kSendSsrcs[0]);
+ rtcp_sender.SetRemoteSSRC(VideoSendStreamTest::kVideoSendSsrcs[0]);
RTCPSender::FeedbackState feedback_state;
@@ -345,11 +345,12 @@
if (header.payloadType == VideoSendStreamTest::kRedPayloadType) {
encapsulated_payload_type = static_cast<int>(packet[header.headerLength]);
if (encapsulated_payload_type !=
- VideoSendStreamTest::kFakeSendPayloadType)
+ VideoSendStreamTest::kFakeVideoSendPayloadType)
EXPECT_EQ(VideoSendStreamTest::kUlpfecPayloadType,
encapsulated_payload_type);
} else {
- EXPECT_EQ(VideoSendStreamTest::kFakeSendPayloadType, header.payloadType);
+ EXPECT_EQ(VideoSendStreamTest::kFakeVideoSendPayloadType,
+ header.payloadType);
}
if (header_extensions_enabled_) {
@@ -459,7 +460,7 @@
nullptr, transport_adapter_.get());
rtcp_sender.SetRTCPStatus(RtcpMode::kReducedSize);
- rtcp_sender.SetRemoteSSRC(kSendSsrcs[0]);
+ rtcp_sender.SetRemoteSSRC(kVideoSendSsrcs[0]);
RTCPSender::FeedbackState feedback_state;
@@ -471,8 +472,8 @@
uint16_t sequence_number = header.sequenceNumber;
if (header.ssrc == retransmit_ssrc_ &&
- retransmit_ssrc_ != kSendSsrcs[0]) {
- // Not kSendSsrcs[0], assume correct RTX packet. Extract sequence
+ retransmit_ssrc_ != kVideoSendSsrcs[0]) {
+ // Not kVideoSendSsrcs[0], assume correct RTX packet. Extract sequence
// number.
const uint8_t* rtx_header = packet + header.headerLength;
sequence_number = (rtx_header[0] << 8) + rtx_header[1];
@@ -496,7 +497,7 @@
transport_adapter_->Enable();
send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
send_config->rtp.rtx.payload_type = retransmit_payload_type_;
- if (retransmit_ssrc_ != kSendSsrcs[0])
+ if (retransmit_ssrc_ != kVideoSendSsrcs[0])
send_config->rtp.rtx.ssrcs.push_back(retransmit_ssrc_);
}
@@ -516,7 +517,7 @@
TEST_F(VideoSendStreamTest, RetransmitsNack) {
// Normal NACKs should use the send SSRC.
- TestNackRetransmission(kSendSsrcs[0], kFakeSendPayloadType);
+ TestNackRetransmission(kVideoSendSsrcs[0], kFakeVideoSendPayloadType);
}
TEST_F(VideoSendStreamTest, RetransmitsNackOverRtx) {
@@ -641,13 +642,13 @@
if (packet_count_++ % 2 != 0) {
// Receive statistics reporting having lost 50% of the packets.
FakeReceiveStatistics lossy_receive_stats(
- kSendSsrcs[0], header.sequenceNumber, packet_count_ / 2, 127);
+ kVideoSendSsrcs[0], header.sequenceNumber, packet_count_ / 2, 127);
RTCPSender rtcp_sender(false, Clock::GetRealTimeClock(),
&lossy_receive_stats, nullptr,
transport_adapter_.get());
rtcp_sender.SetRTCPStatus(RtcpMode::kReducedSize);
- rtcp_sender.SetRemoteSSRC(kSendSsrcs[0]);
+ rtcp_sender.SetRemoteSSRC(kVideoSendSsrcs[0]);
RTCPSender::FeedbackState feedback_state;
@@ -864,13 +865,13 @@
virtual void SendRtcpFeedback(int remb_value)
EXCLUSIVE_LOCKS_REQUIRED(crit_) {
- FakeReceiveStatistics receive_stats(
- kSendSsrcs[0], last_sequence_number_, rtp_count_, 0);
+ FakeReceiveStatistics receive_stats(kVideoSendSsrcs[0],
+ last_sequence_number_, rtp_count_, 0);
RTCPSender rtcp_sender(false, clock_, &receive_stats, nullptr,
transport_adapter_.get());
rtcp_sender.SetRTCPStatus(RtcpMode::kReducedSize);
- rtcp_sender.SetRemoteSSRC(kSendSsrcs[0]);
+ rtcp_sender.SetRemoteSSRC(kVideoSendSsrcs[0]);
if (remb_value > 0) {
rtcp_sender.SetREMBStatus(true);
rtcp_sender.SetREMBData(remb_value, std::vector<uint32_t>());
@@ -921,12 +922,12 @@
kVideoMutedThresholdMs)
observation_complete_.Set();
// Receive statistics reporting having lost 50% of the packets.
- FakeReceiveStatistics receive_stats(kSendSsrcs[0], 1, 1, 0);
+ FakeReceiveStatistics receive_stats(kVideoSendSsrcs[0], 1, 1, 0);
RTCPSender rtcp_sender(false, Clock::GetRealTimeClock(), &receive_stats,
nullptr, transport_adapter_.get());
rtcp_sender.SetRTCPStatus(RtcpMode::kReducedSize);
- rtcp_sender.SetRemoteSSRC(kSendSsrcs[0]);
+ rtcp_sender.SetRemoteSSRC(kVideoSendSsrcs[0]);
RTCPSender::FeedbackState feedback_state;
@@ -942,7 +943,7 @@
transport_adapter_->Enable();
}
- size_t GetNumStreams() const override { return 3; }
+ size_t GetNumVideoStreams() const override { return 3; }
virtual void OnFrameGeneratorCapturerCreated(
test::FrameGeneratorCapturer* frame_generator_capturer) {
@@ -1085,7 +1086,7 @@
CreateSenderCall(Call::Config());
test::NullTransport transport;
- CreateSendConfig(1, &transport);
+ CreateSendConfig(1, 0, &transport);
Call::Config::BitrateConfig bitrate_config;
bitrate_config.start_bitrate_bps =
@@ -1095,7 +1096,7 @@
StartBitrateObserver encoder;
video_send_config_.encoder_settings.encoder = &encoder;
- CreateStreams();
+ CreateVideoStreams();
EXPECT_EQ(video_encoder_config_.streams[0].max_bitrate_bps / 1000,
encoder.GetStartBitrateKbps());
@@ -1145,10 +1146,10 @@
CreateSenderCall(Call::Config());
test::NullTransport transport;
- CreateSendConfig(1, &transport);
+ CreateSendConfig(1, 0, &transport);
FrameObserver observer;
video_send_config_.pre_encode_callback = &observer;
- CreateStreams();
+ CreateVideoStreams();
// Prepare five input frames. Send ordinary VideoFrame and texture frames
// alternatively.
@@ -1819,7 +1820,7 @@
EXPECT_EQ(kNumStreams, encoder_config->streams.size());
}
- size_t GetNumStreams() const override { return kNumStreams; }
+ size_t GetNumVideoStreams() const override { return kNumStreams; }
void PerformTest() override {
EXPECT_TRUE(Wait())
@@ -1827,12 +1828,12 @@
VideoSendStream::Stats stats = send_stream_->GetStats();
for (size_t i = 0; i < kNumStreams; ++i) {
- ASSERT_TRUE(stats.substreams.find(kSendSsrcs[i]) !=
+ ASSERT_TRUE(stats.substreams.find(kVideoSendSsrcs[i]) !=
stats.substreams.end())
- << "No stats for SSRC: " << kSendSsrcs[i]
+ << "No stats for SSRC: " << kVideoSendSsrcs[i]
<< ", stats should exist as soon as frames have been encoded.";
VideoSendStream::StreamStats ssrc_stats =
- stats.substreams[kSendSsrcs[i]];
+ stats.substreams[kVideoSendSsrcs[i]];
EXPECT_EQ(kEncodedResolution[i].width, ssrc_stats.width);
EXPECT_EQ(kEncodedResolution[i].height, ssrc_stats.height);
}
diff --git a/webrtc/video_engine_tests.isolate b/webrtc/video_engine_tests.isolate
index 5aa9623..f2f961f 100644
--- a/webrtc/video_engine_tests.isolate
+++ b/webrtc/video_engine_tests.isolate
@@ -11,6 +11,7 @@
'variables': {
'files': [
'<(DEPTH)/resources/foreman_cif_short.yuv',
+ '<(DEPTH)/resources/voice_engine/audio_long16.pcm',
],
},
}],