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d319534143
Bug: webrtc:4690 Change-Id: I3b8950fdb13835964c5bf41162731eff5048bf1a Reviewed-on: https://webrtc-review.googlesource.com/23820 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20823}
134 lines
4.6 KiB
C++
134 lines
4.6 KiB
C++
/*
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* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include <memory>
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#include "audio/audio_state.h"
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#include "modules/audio_mixer/audio_mixer_impl.h"
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#include "modules/audio_processing/include/mock_audio_processing.h"
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#include "test/gtest.h"
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#include "test/mock_voice_engine.h"
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namespace webrtc {
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namespace test {
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namespace {
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const int kSampleRate = 8000;
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const int kNumberOfChannels = 1;
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const int kBytesPerSample = 2;
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struct ConfigHelper {
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ConfigHelper() : audio_mixer(AudioMixerImpl::Create()) {
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EXPECT_CALL(mock_voice_engine, audio_transport())
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.WillRepeatedly(testing::Return(&audio_transport));
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audio_state_config.voice_engine = &mock_voice_engine;
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audio_state_config.audio_mixer = audio_mixer;
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audio_state_config.audio_processing =
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new rtc::RefCountedObject<MockAudioProcessing>();
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}
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AudioState::Config& config() { return audio_state_config; }
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MockVoiceEngine& voice_engine() { return mock_voice_engine; }
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rtc::scoped_refptr<AudioMixer> mixer() { return audio_mixer; }
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MockAudioTransport& original_audio_transport() { return audio_transport; }
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private:
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testing::StrictMock<MockVoiceEngine> mock_voice_engine;
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AudioState::Config audio_state_config;
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rtc::scoped_refptr<AudioMixer> audio_mixer;
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MockAudioTransport audio_transport;
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};
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class FakeAudioSource : public AudioMixer::Source {
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public:
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// TODO(aleloi): Valid overrides commented out, because the gmock
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// methods don't use any override declarations, and we want to avoid
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// warnings from -Winconsistent-missing-override. See
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// http://crbug.com/428099.
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int Ssrc() const /*override*/ { return 0; }
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int PreferredSampleRate() const /*override*/ { return kSampleRate; }
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MOCK_METHOD2(GetAudioFrameWithInfo,
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AudioFrameInfo(int sample_rate_hz, AudioFrame* audio_frame));
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};
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} // namespace
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TEST(AudioStateTest, Create) {
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ConfigHelper helper;
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rtc::scoped_refptr<AudioState> audio_state =
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AudioState::Create(helper.config());
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EXPECT_TRUE(audio_state.get());
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}
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TEST(AudioStateTest, ConstructDestruct) {
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ConfigHelper helper;
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std::unique_ptr<internal::AudioState> audio_state(
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new internal::AudioState(helper.config()));
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}
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TEST(AudioStateTest, GetVoiceEngine) {
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ConfigHelper helper;
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std::unique_ptr<internal::AudioState> audio_state(
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new internal::AudioState(helper.config()));
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EXPECT_EQ(audio_state->voice_engine(), &helper.voice_engine());
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}
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// Test that RecordedDataIsAvailable calls get to the original transport.
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TEST(AudioStateAudioPathTest, RecordedAudioArrivesAtOriginalTransport) {
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ConfigHelper helper;
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rtc::scoped_refptr<AudioState> audio_state =
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AudioState::Create(helper.config());
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// Setup completed. Ensure call of original transport is forwarded to new.
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uint32_t new_mic_level;
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EXPECT_CALL(
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helper.original_audio_transport(),
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RecordedDataIsAvailable(nullptr, kSampleRate / 100, kBytesPerSample,
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kNumberOfChannels, kSampleRate, 0, 0, 0, false,
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testing::Ref(new_mic_level)));
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audio_state->audio_transport()->RecordedDataIsAvailable(
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nullptr, kSampleRate / 100, kBytesPerSample, kNumberOfChannels,
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kSampleRate, 0, 0, 0, false, new_mic_level);
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}
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TEST(AudioStateAudioPathTest,
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QueryingProxyForAudioShouldResultInGetAudioCallOnMixerSource) {
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ConfigHelper helper;
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rtc::scoped_refptr<AudioState> audio_state =
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AudioState::Create(helper.config());
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FakeAudioSource fake_source;
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helper.mixer()->AddSource(&fake_source);
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EXPECT_CALL(fake_source, GetAudioFrameWithInfo(testing::_, testing::_))
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.WillOnce(
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testing::Invoke([](int sample_rate_hz, AudioFrame* audio_frame) {
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audio_frame->sample_rate_hz_ = sample_rate_hz;
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audio_frame->samples_per_channel_ = sample_rate_hz / 100;
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audio_frame->num_channels_ = kNumberOfChannels;
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return AudioMixer::Source::AudioFrameInfo::kNormal;
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}));
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int16_t audio_buffer[kSampleRate / 100 * kNumberOfChannels];
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size_t n_samples_out;
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int64_t elapsed_time_ms;
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int64_t ntp_time_ms;
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audio_state->audio_transport()->NeedMorePlayData(
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kSampleRate / 100, kBytesPerSample, kNumberOfChannels, kSampleRate,
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audio_buffer, n_samples_out, &elapsed_time_ms, &ntp_time_ms);
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}
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} // namespace test
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} // namespace webrtc
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