blob: b4af4a08819138bf696926133f04e1edee3a6020 [file] [log] [blame]
/*
* Copyright (c) 2018 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 "api/audio_codecs/audio_decoder_factory_template.h"
#include "api/audio_codecs/audio_encoder_factory_template.h"
#include "api/audio_codecs/opus/audio_decoder_opus.h"
#include "api/audio_codecs/opus/audio_encoder_opus.h"
#include "api/test/loopback_media_transport.h"
#include "api/test/mock_audio_mixer.h"
#include "audio/audio_receive_stream.h"
#include "audio/audio_send_stream.h"
#include "call/test/mock_bitrate_allocator.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "modules/audio_device/include/test_audio_device.h"
#include "modules/audio_mixer/audio_mixer_impl.h"
#include "modules/audio_processing/include/mock_audio_processing.h"
#include "modules/utility/include/process_thread.h"
#include "rtc_base/task_queue.h"
#include "rtc_base/timeutils.h"
#include "test/gtest.h"
#include "test/mock_transport.h"
namespace webrtc {
namespace test {
namespace {
constexpr int kPayloadTypeOpus = 17;
constexpr int kSamplingFrequency = 48000;
constexpr int kNumChannels = 2;
constexpr int kWantedSamples = 3000;
constexpr int kTestTimeoutMs = 2 * rtc::kNumMillisecsPerSec;
class TestRenderer : public TestAudioDeviceModule::Renderer {
public:
TestRenderer(int sampling_frequency, int num_channels, size_t wanted_samples)
: sampling_frequency_(sampling_frequency),
num_channels_(num_channels),
wanted_samples_(wanted_samples) {}
~TestRenderer() override = default;
int SamplingFrequency() const override { return sampling_frequency_; }
int NumChannels() const override { return num_channels_; }
bool Render(rtc::ArrayView<const int16_t> data) override {
if (data.size() >= wanted_samples_) {
return false;
}
wanted_samples_ -= data.size();
return true;
}
private:
const int sampling_frequency_;
const int num_channels_;
size_t wanted_samples_;
};
} // namespace
TEST(AudioWithMediaTransport, DeliversAudio) {
std::unique_ptr<rtc::Thread> transport_thread = rtc::Thread::Create();
transport_thread->Start();
MediaTransportPair transport_pair(transport_thread.get());
MockTransport rtcp_send_transport;
MockTransport send_transport;
std::unique_ptr<RtcEventLog> null_event_log = RtcEventLog::CreateNull();
MockBitrateAllocator bitrate_allocator;
rtc::scoped_refptr<TestAudioDeviceModule> audio_device =
TestAudioDeviceModule::CreateTestAudioDeviceModule(
TestAudioDeviceModule::CreatePulsedNoiseCapturer(
/* max_amplitude= */ 10000, kSamplingFrequency, kNumChannels),
absl::make_unique<TestRenderer>(kSamplingFrequency, kNumChannels,
kWantedSamples));
AudioState::Config audio_config;
audio_config.audio_mixer = AudioMixerImpl::Create();
// TODO(nisse): Is a mock AudioProcessing enough?
audio_config.audio_processing =
new rtc::RefCountedObject<MockAudioProcessing>();
audio_config.audio_device_module = audio_device;
rtc::scoped_refptr<AudioState> audio_state = AudioState::Create(audio_config);
// TODO(nisse): Use some lossless codec?
const SdpAudioFormat audio_format("opus", kSamplingFrequency, kNumChannels);
// Setup receive stream;
webrtc::AudioReceiveStream::Config receive_config;
// TODO(nisse): Update AudioReceiveStream to not require rtcp_send_transport
// when a MediaTransport is provided.
receive_config.rtcp_send_transport = &rtcp_send_transport;
receive_config.media_transport = transport_pair.first();
receive_config.decoder_map.emplace(kPayloadTypeOpus, audio_format);
receive_config.decoder_factory =
CreateAudioDecoderFactory<AudioDecoderOpus>();
std::unique_ptr<ProcessThread> receive_process_thread =
ProcessThread::Create("audio recv thread");
webrtc::internal::AudioReceiveStream receive_stream(
/*rtp_stream_receiver_controller=*/nullptr,
/*packet_router=*/nullptr, receive_process_thread.get(), receive_config,
audio_state, null_event_log.get());
// TODO(nisse): Update AudioSendStream to not require send_transport when a
// MediaTransport is provided.
AudioSendStream::Config send_config(&send_transport, transport_pair.second());
send_config.send_codec_spec =
AudioSendStream::Config::SendCodecSpec(kPayloadTypeOpus, audio_format);
send_config.encoder_factory = CreateAudioEncoderFactory<AudioEncoderOpus>();
rtc::TaskQueue send_tq("audio send queue");
std::unique_ptr<ProcessThread> send_process_thread =
ProcessThread::Create("audio send thread");
webrtc::internal::AudioSendStream send_stream(
send_config, audio_state, &send_tq, send_process_thread.get(),
/*transport=*/nullptr, &bitrate_allocator, null_event_log.get(),
/*rtcp_rtt_stats=*/nullptr, absl::optional<RtpState>());
audio_device->Init(); // Starts thread.
audio_device->RegisterAudioCallback(audio_state->audio_transport());
receive_stream.Start();
send_stream.Start();
audio_device->StartPlayout();
audio_device->StartRecording();
EXPECT_TRUE(audio_device->WaitForPlayoutEnd(kTestTimeoutMs));
audio_device->StopRecording();
audio_device->StopPlayout();
receive_stream.Stop();
send_stream.Stop();
}
} // namespace test
} // namespace webrtc