Add audio streams to CallTest and a first A/V call test.

webrtc/test/call_test.h

 /*
  *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  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_
 
 #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;
 
 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:
   void RunBaseTest(BaseTest* test, const FakeNetworkPipe::Config& config);
 
   void CreateCalls(const Call::Config& sender_config,
                    const Call::Config& receiver_config);
   void CreateSenderCall(const Call::Config& config);
   void CreateReceiverCall(const Call::Config& config);
   void DestroyCalls();
 
-  void CreateSendConfig(size_t num_streams, Transport* send_transport);
-  void CreateMatchingReceiveConfigs(Transport* rtcp_send_transport);
+  void CreateSendConfig(size_t num_video_streams,
+                        size_t num_audio_streams,
+                        Transport* send_transport);
+  void CreateMatchingReceiveConfigs(bool create_audio_recv_stream,

[pbos-webrtc, 2016/01/07 14:23:37]

Store how many was created in CreateSendConfig and remove this parameter.

[stefan-webrtc, 2016/01/07 15:16:44]

Done.

+                                    Transport* rtcp_send_transport);
 
   void CreateFrameGeneratorCapturer();
+  void CreateFakeAudioDevices();
 
-  void CreateStreams();
+  void CreateVideoStreams();
+  void CreateAudioStreams();
   void Start();
   void Stop();
   void DestroyStreams();
 
   Clock* const clock_;
 
   rtc::scoped_ptr<Call> sender_call_;
   rtc::scoped_ptr<PacketTransport> send_transport_;
   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_;
+
+ private:
+  // TODO(holmer): Remove once voice engine is fully refactored to the new API.

[pbos-webrtc, 2016/01/07 14:23:37]

VoiceEngine

[stefan-webrtc, 2016/01/07 15:16:44]

Done.

+  // 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.
+  void CreateVoiceEngines();
+  void SetupVoiceEngineTransports(PacketTransport* send_transport,
+                                  PacketTransport* recv_transport);
+  void DestroyVoiceEngines();
+
+  int voe_send_channel_id_;
+  int voe_recv_channel_id_;
+
+  VoiceEngine* send_voice_engine_;

[pbos-webrtc, 2016/01/07 14:23:37]

Split into send/recv member "blocks", maybe a struct as you're suggesting to me
right now in person.

[stefan-webrtc, 2016/01/07 15:16:44]

Done.

+  VoiceEngine* recv_voice_engine_;
+  VoEBase* voe_send_base_;
+  VoENetwork* voe_send_network_;
+  VoECodec* voe_send_codec_;
+  VoEBase* voe_recv_base_;
+  VoENetwork* voe_recv_network_;
+  VoECodec* voe_recv_codec_;
+  rtc::scoped_ptr<internal::TransportAdapter> voe_send_transport_adapter_;
+  rtc::scoped_ptr<internal::TransportAdapter> voe_recv_transport_adapter_;
+
+  // 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 {
  public:
   explicit BaseTest(unsigned int timeout_ms);
   virtual ~BaseTest();
 
   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();
   virtual void OnCallsCreated(Call* sender_call, Call* receiver_call);
   virtual void OnTransportsCreated(PacketTransport* send_transport,
                                    PacketTransport* receive_transport);
 
   virtual void ModifyVideoConfigs(
       VideoSendStream::Config* send_config,
       std::vector<VideoReceiveStream::Config>* receive_configs,
       VideoEncoderConfig* encoder_config);
   virtual void OnVideoStreamsCreated(
       VideoSendStream* send_stream,
       const std::vector<VideoReceiveStream*>& receive_streams);
 
+  virtual void ModifyAudioConfigs(
+      AudioSendStream::Config* send_config,
+      std::vector<AudioReceiveStream::Config>* receive_configsg);

[pbos-webrtc, 2016/01/07 14:23:37]

configsgsgsgsgsgsgsg

[stefan-webrtc, 2016/01/07 15:16:44]

Done.

+  virtual void OnAudioStreamsCreated(
+      AudioSendStream* send_stream,
+      const std::vector<AudioReceiveStream*>& receive_streams);
+
   virtual void OnFrameGeneratorCapturerCreated(
       FrameGeneratorCapturer* frame_generator_capturer);
 };
 
 class SendTest : public BaseTest {
  public:
   explicit SendTest(unsigned int timeout_ms);
 
   bool ShouldCreateReceivers() const override;
 };
 
 class EndToEndTest : public BaseTest {
  public:
   explicit EndToEndTest(unsigned int timeout_ms);
 
   bool ShouldCreateReceivers() const override;
 };
 
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_TEST_COMMON_CALL_TEST_H_