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/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_