call/call_unittest.cc

/*
 *  Copyright (c) 2015 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.
 */

#include "call/call.h"

#include <list>
#include <map>
#include <memory>
#include <utility>

#include "absl/memory/memory.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/task_queue/default_task_queue_factory.h"
#include "api/test/mock_audio_mixer.h"
#include "api/transport/field_trial_based_config.h"
#include "audio/audio_receive_stream.h"
#include "audio/audio_send_stream.h"
#include "call/audio_state.h"
#include "modules/audio_device/include/mock_audio_device.h"
#include "modules/audio_processing/include/mock_audio_processing.h"
#include "modules/rtp_rtcp/include/rtp_rtcp.h"
#include "test/fake_encoder.h"
#include "test/gtest.h"
#include "test/mock_audio_decoder_factory.h"
#include "test/mock_transport.h"
#include "test/run_loop.h"

namespace {

struct CallHelper {
  explicit CallHelper(bool use_null_audio_processing) {
    task_queue_factory_ = webrtc::CreateDefaultTaskQueueFactory();
    webrtc::AudioState::Config audio_state_config;
    audio_state_config.audio_mixer =
        new rtc::RefCountedObject<webrtc::test::MockAudioMixer>();
    audio_state_config.audio_processing =
        use_null_audio_processing
            ? nullptr
            : new rtc::RefCountedObject<webrtc::test::MockAudioProcessing>();
    audio_state_config.audio_device_module =
        new rtc::RefCountedObject<webrtc::test::MockAudioDeviceModule>();
    webrtc::Call::Config config(&event_log_);
    config.audio_state = webrtc::AudioState::Create(audio_state_config);
    config.task_queue_factory = task_queue_factory_.get();
    config.trials = &field_trials_;
    call_.reset(webrtc::Call::Create(config));
  }

  webrtc::Call* operator->() { return call_.get(); }

 private:
  webrtc::test::RunLoop loop_;
  webrtc::RtcEventLogNull event_log_;
  webrtc::FieldTrialBasedConfig field_trials_;
  std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory_;
  std::unique_ptr<webrtc::Call> call_;
};
}  // namespace

namespace webrtc {

TEST(CallTest, ConstructDestruct) {
  for (bool use_null_audio_processing : {false, true}) {
    CallHelper call(use_null_audio_processing);
  }
}

TEST(CallTest, CreateDestroy_AudioSendStream) {
  for (bool use_null_audio_processing : {false, true}) {
    CallHelper call(use_null_audio_processing);
    MockTransport send_transport;
    AudioSendStream::Config config(&send_transport);
    config.rtp.ssrc = 42;
    AudioSendStream* stream = call->CreateAudioSendStream(config);
    EXPECT_NE(stream, nullptr);
    call->DestroyAudioSendStream(stream);
  }
}

TEST(CallTest, CreateDestroy_AudioReceiveStream) {
  for (bool use_null_audio_processing : {false, true}) {
    CallHelper call(use_null_audio_processing);
    AudioReceiveStream::Config config;
    MockTransport rtcp_send_transport;
    config.rtp.remote_ssrc = 42;
    config.rtcp_send_transport = &rtcp_send_transport;
    config.decoder_factory =
        new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>();
    AudioReceiveStream* stream = call->CreateAudioReceiveStream(config);
    EXPECT_NE(stream, nullptr);
    call->DestroyAudioReceiveStream(stream);
  }
}

TEST(CallTest, CreateDestroy_AudioSendStreams) {
  for (bool use_null_audio_processing : {false, true}) {
    CallHelper call(use_null_audio_processing);
    MockTransport send_transport;
    AudioSendStream::Config config(&send_transport);
    std::list<AudioSendStream*> streams;
    for (int i = 0; i < 2; ++i) {
      for (uint32_t ssrc = 0; ssrc < 1234567; ssrc += 34567) {
        config.rtp.ssrc = ssrc;
        AudioSendStream* stream = call->CreateAudioSendStream(config);
        EXPECT_NE(stream, nullptr);
        if (ssrc & 1) {
          streams.push_back(stream);
        } else {
          streams.push_front(stream);
        }
      }
      for (auto s : streams) {
        call->DestroyAudioSendStream(s);
      }
      streams.clear();
    }
  }
}

TEST(CallTest, CreateDestroy_AudioReceiveStreams) {
  for (bool use_null_audio_processing : {false, true}) {
    CallHelper call(use_null_audio_processing);
    AudioReceiveStream::Config config;
    MockTransport rtcp_send_transport;
    config.rtcp_send_transport = &rtcp_send_transport;
    config.decoder_factory =
        new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>();
    std::list<AudioReceiveStream*> streams;
    for (int i = 0; i < 2; ++i) {
      for (uint32_t ssrc = 0; ssrc < 1234567; ssrc += 34567) {
        config.rtp.remote_ssrc = ssrc;
        AudioReceiveStream* stream = call->CreateAudioReceiveStream(config);
        EXPECT_NE(stream, nullptr);
        if (ssrc & 1) {
          streams.push_back(stream);
        } else {
          streams.push_front(stream);
        }
      }
      for (auto s : streams) {
        call->DestroyAudioReceiveStream(s);
      }
      streams.clear();
    }
  }
}

TEST(CallTest, CreateDestroy_AssociateAudioSendReceiveStreams_RecvFirst) {
  for (bool use_null_audio_processing : {false, true}) {
    CallHelper call(use_null_audio_processing);
    AudioReceiveStream::Config recv_config;
    MockTransport rtcp_send_transport;
    recv_config.rtp.remote_ssrc = 42;
    recv_config.rtp.local_ssrc = 777;
    recv_config.rtcp_send_transport = &rtcp_send_transport;
    recv_config.decoder_factory =
        new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>();
    AudioReceiveStream* recv_stream =
        call->CreateAudioReceiveStream(recv_config);
    EXPECT_NE(recv_stream, nullptr);

    MockTransport send_transport;
    AudioSendStream::Config send_config(&send_transport);
    send_config.rtp.ssrc = 777;
    AudioSendStream* send_stream = call->CreateAudioSendStream(send_config);
    EXPECT_NE(send_stream, nullptr);

    internal::AudioReceiveStream* internal_recv_stream =
        static_cast<internal::AudioReceiveStream*>(recv_stream);
    EXPECT_EQ(send_stream,
              internal_recv_stream->GetAssociatedSendStreamForTesting());

    call->DestroyAudioSendStream(send_stream);
    EXPECT_EQ(nullptr,
              internal_recv_stream->GetAssociatedSendStreamForTesting());

    call->DestroyAudioReceiveStream(recv_stream);
  }
}

TEST(CallTest, CreateDestroy_AssociateAudioSendReceiveStreams_SendFirst) {
  for (bool use_null_audio_processing : {false, true}) {
    CallHelper call(use_null_audio_processing);
    MockTransport send_transport;
    AudioSendStream::Config send_config(&send_transport);
    send_config.rtp.ssrc = 777;
    AudioSendStream* send_stream = call->CreateAudioSendStream(send_config);
    EXPECT_NE(send_stream, nullptr);

    AudioReceiveStream::Config recv_config;
    MockTransport rtcp_send_transport;
    recv_config.rtp.remote_ssrc = 42;
    recv_config.rtp.local_ssrc = 777;
    recv_config.rtcp_send_transport = &rtcp_send_transport;
    recv_config.decoder_factory =
        new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>();
    AudioReceiveStream* recv_stream =
        call->CreateAudioReceiveStream(recv_config);
    EXPECT_NE(recv_stream, nullptr);

    internal::AudioReceiveStream* internal_recv_stream =
        static_cast<internal::AudioReceiveStream*>(recv_stream);
    EXPECT_EQ(send_stream,
              internal_recv_stream->GetAssociatedSendStreamForTesting());

    call->DestroyAudioReceiveStream(recv_stream);

    call->DestroyAudioSendStream(send_stream);
  }
}

TEST(CallTest, CreateDestroy_FlexfecReceiveStream) {
  for (bool use_null_audio_processing : {false, true}) {
    CallHelper call(use_null_audio_processing);
    MockTransport rtcp_send_transport;
    FlexfecReceiveStream::Config config(&rtcp_send_transport);
    config.payload_type = 118;
    config.remote_ssrc = 38837212;
    config.protected_media_ssrcs = {27273};

    FlexfecReceiveStream* stream = call->CreateFlexfecReceiveStream(config);
    EXPECT_NE(stream, nullptr);
    call->DestroyFlexfecReceiveStream(stream);
  }
}

TEST(CallTest, CreateDestroy_FlexfecReceiveStreams) {
  for (bool use_null_audio_processing : {false, true}) {
    CallHelper call(use_null_audio_processing);
    MockTransport rtcp_send_transport;
    FlexfecReceiveStream::Config config(&rtcp_send_transport);
    config.payload_type = 118;
    std::list<FlexfecReceiveStream*> streams;

    for (int i = 0; i < 2; ++i) {
      for (uint32_t ssrc = 0; ssrc < 1234567; ssrc += 34567) {
        config.remote_ssrc = ssrc;
        config.protected_media_ssrcs = {ssrc + 1};
        FlexfecReceiveStream* stream = call->CreateFlexfecReceiveStream(config);
        EXPECT_NE(stream, nullptr);
        if (ssrc & 1) {
          streams.push_back(stream);
        } else {
          streams.push_front(stream);
        }
      }
      for (auto s : streams) {
        call->DestroyFlexfecReceiveStream(s);
      }
      streams.clear();
    }
  }
}

TEST(CallTest, MultipleFlexfecReceiveStreamsProtectingSingleVideoStream) {
  for (bool use_null_audio_processing : {false, true}) {
    CallHelper call(use_null_audio_processing);
    MockTransport rtcp_send_transport;
    FlexfecReceiveStream::Config config(&rtcp_send_transport);
    config.payload_type = 118;
    config.protected_media_ssrcs = {1324234};
    FlexfecReceiveStream* stream;
    std::list<FlexfecReceiveStream*> streams;

    config.remote_ssrc = 838383;
    stream = call->CreateFlexfecReceiveStream(config);
    EXPECT_NE(stream, nullptr);
    streams.push_back(stream);

    config.remote_ssrc = 424993;
    stream = call->CreateFlexfecReceiveStream(config);
    EXPECT_NE(stream, nullptr);
    streams.push_back(stream);

    config.remote_ssrc = 99383;
    stream = call->CreateFlexfecReceiveStream(config);
    EXPECT_NE(stream, nullptr);
    streams.push_back(stream);

    config.remote_ssrc = 5548;
    stream = call->CreateFlexfecReceiveStream(config);
    EXPECT_NE(stream, nullptr);
    streams.push_back(stream);

    for (auto s : streams) {
      call->DestroyFlexfecReceiveStream(s);
    }
  }
}

TEST(CallTest, RecreatingAudioStreamWithSameSsrcReusesRtpState) {
  constexpr uint32_t kSSRC = 12345;
  for (bool use_null_audio_processing : {false, true}) {
    CallHelper call(use_null_audio_processing);

    auto create_stream_and_get_rtp_state = [&](uint32_t ssrc) {
      MockTransport send_transport;
      AudioSendStream::Config config(&send_transport);
      config.rtp.ssrc = ssrc;
      AudioSendStream* stream = call->CreateAudioSendStream(config);
      const RtpState rtp_state =
          static_cast<internal::AudioSendStream*>(stream)->GetRtpState();
      call->DestroyAudioSendStream(stream);
      return rtp_state;
    };

    const RtpState rtp_state1 = create_stream_and_get_rtp_state(kSSRC);
    const RtpState rtp_state2 = create_stream_and_get_rtp_state(kSSRC);

    EXPECT_EQ(rtp_state1.sequence_number, rtp_state2.sequence_number);
    EXPECT_EQ(rtp_state1.start_timestamp, rtp_state2.start_timestamp);
    EXPECT_EQ(rtp_state1.timestamp, rtp_state2.timestamp);
    EXPECT_EQ(rtp_state1.capture_time_ms, rtp_state2.capture_time_ms);
    EXPECT_EQ(rtp_state1.last_timestamp_time_ms,
              rtp_state2.last_timestamp_time_ms);
    EXPECT_EQ(rtp_state1.media_has_been_sent, rtp_state2.media_has_been_sent);
  }
}

}  // namespace webrtc