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/*
* Copyright 2012 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 "pc/test/fake_audio_capture_module.h"
#include <string.h>
#include <algorithm>
#include "api/scoped_refptr.h"
#include "rtc_base/gunit.h"
#include "rtc_base/synchronization/mutex.h"
#include "test/gtest.h"
class FakeAdmTest : public ::testing::Test, public webrtc::AudioTransport {
protected:
static const int kMsInSecond = 1000;
FakeAdmTest()
: push_iterations_(0), pull_iterations_(0), rec_buffer_bytes_(0) {
memset(rec_buffer_, 0, sizeof(rec_buffer_));
}
void SetUp() override {
fake_audio_capture_module_ = FakeAudioCaptureModule::Create();
EXPECT_TRUE(fake_audio_capture_module_.get() != NULL);
}
// Callbacks inherited from webrtc::AudioTransport.
// ADM is pushing data.
int32_t RecordedDataIsAvailable(const void* audioSamples,
const size_t nSamples,
const size_t nBytesPerSample,
const size_t nChannels,
const uint32_t samplesPerSec,
const uint32_t totalDelayMS,
const int32_t clockDrift,
const uint32_t currentMicLevel,
const bool keyPressed,
uint32_t& newMicLevel) override {
webrtc::MutexLock lock(&mutex_);
rec_buffer_bytes_ = nSamples * nBytesPerSample;
if ((rec_buffer_bytes_ == 0) ||
(rec_buffer_bytes_ >
FakeAudioCaptureModule::kNumberSamples *
FakeAudioCaptureModule::kNumberBytesPerSample)) {
ADD_FAILURE();
return -1;
}
memcpy(rec_buffer_, audioSamples, rec_buffer_bytes_);
++push_iterations_;
newMicLevel = currentMicLevel;
return 0;
}
void PullRenderData(int bits_per_sample,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames,
void* audio_data,
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms) override {}
// ADM is pulling data.
int32_t NeedMorePlayData(const size_t nSamples,
const size_t nBytesPerSample,
const size_t nChannels,
const uint32_t samplesPerSec,
void* audioSamples,
size_t& nSamplesOut,
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms) override {
webrtc::MutexLock lock(&mutex_);
++pull_iterations_;
const size_t audio_buffer_size = nSamples * nBytesPerSample;
const size_t bytes_out =
RecordedDataReceived()
? CopyFromRecBuffer(audioSamples, audio_buffer_size)
: GenerateZeroBuffer(audioSamples, audio_buffer_size);
nSamplesOut = bytes_out / nBytesPerSample;
*elapsed_time_ms = 0;
*ntp_time_ms = 0;
return 0;
}
int push_iterations() const {
webrtc::MutexLock lock(&mutex_);
return push_iterations_;
}
int pull_iterations() const {
webrtc::MutexLock lock(&mutex_);
return pull_iterations_;
}
rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
private:
bool RecordedDataReceived() const { return rec_buffer_bytes_ != 0; }
size_t GenerateZeroBuffer(void* audio_buffer, size_t audio_buffer_size) {
memset(audio_buffer, 0, audio_buffer_size);
return audio_buffer_size;
}
size_t CopyFromRecBuffer(void* audio_buffer, size_t audio_buffer_size) {
EXPECT_EQ(audio_buffer_size, rec_buffer_bytes_);
const size_t min_buffer_size =
std::min(audio_buffer_size, rec_buffer_bytes_);
memcpy(audio_buffer, rec_buffer_, min_buffer_size);
return min_buffer_size;
}
rtc::AutoThread main_thread_;
mutable webrtc::Mutex mutex_;
int push_iterations_;
int pull_iterations_;
char rec_buffer_[FakeAudioCaptureModule::kNumberSamples *
FakeAudioCaptureModule::kNumberBytesPerSample];
size_t rec_buffer_bytes_;
};
TEST_F(FakeAdmTest, PlayoutTest) {
EXPECT_EQ(0, fake_audio_capture_module_->RegisterAudioCallback(this));
bool stereo_available = false;
EXPECT_EQ(0, fake_audio_capture_module_->StereoPlayoutIsAvailable(
&stereo_available));
EXPECT_TRUE(stereo_available);
EXPECT_NE(0, fake_audio_capture_module_->StartPlayout());
EXPECT_FALSE(fake_audio_capture_module_->PlayoutIsInitialized());
EXPECT_FALSE(fake_audio_capture_module_->Playing());
EXPECT_EQ(0, fake_audio_capture_module_->StopPlayout());
EXPECT_EQ(0, fake_audio_capture_module_->InitPlayout());
EXPECT_TRUE(fake_audio_capture_module_->PlayoutIsInitialized());
EXPECT_FALSE(fake_audio_capture_module_->Playing());
EXPECT_EQ(0, fake_audio_capture_module_->StartPlayout());
EXPECT_TRUE(fake_audio_capture_module_->Playing());
uint16_t delay_ms = 10;
EXPECT_EQ(0, fake_audio_capture_module_->PlayoutDelay(&delay_ms));
EXPECT_EQ(0, delay_ms);
EXPECT_TRUE_WAIT(pull_iterations() > 0, kMsInSecond);
EXPECT_GE(0, push_iterations());
EXPECT_EQ(0, fake_audio_capture_module_->StopPlayout());
EXPECT_FALSE(fake_audio_capture_module_->Playing());
}
TEST_F(FakeAdmTest, RecordTest) {
EXPECT_EQ(0, fake_audio_capture_module_->RegisterAudioCallback(this));
bool stereo_available = false;
EXPECT_EQ(0, fake_audio_capture_module_->StereoRecordingIsAvailable(
&stereo_available));
EXPECT_FALSE(stereo_available);
EXPECT_NE(0, fake_audio_capture_module_->StartRecording());
EXPECT_FALSE(fake_audio_capture_module_->Recording());
EXPECT_EQ(0, fake_audio_capture_module_->StopRecording());
EXPECT_EQ(0, fake_audio_capture_module_->InitRecording());
EXPECT_EQ(0, fake_audio_capture_module_->StartRecording());
EXPECT_TRUE(fake_audio_capture_module_->Recording());
EXPECT_TRUE_WAIT(push_iterations() > 0, kMsInSecond);
EXPECT_GE(0, pull_iterations());
EXPECT_EQ(0, fake_audio_capture_module_->StopRecording());
EXPECT_FALSE(fake_audio_capture_module_->Recording());
}
TEST_F(FakeAdmTest, DuplexTest) {
EXPECT_EQ(0, fake_audio_capture_module_->RegisterAudioCallback(this));
EXPECT_EQ(0, fake_audio_capture_module_->InitPlayout());
EXPECT_EQ(0, fake_audio_capture_module_->StartPlayout());
EXPECT_EQ(0, fake_audio_capture_module_->InitRecording());
EXPECT_EQ(0, fake_audio_capture_module_->StartRecording());
EXPECT_TRUE_WAIT(push_iterations() > 0, kMsInSecond);
EXPECT_TRUE_WAIT(pull_iterations() > 0, kMsInSecond);
EXPECT_EQ(0, fake_audio_capture_module_->StopPlayout());
EXPECT_EQ(0, fake_audio_capture_module_->StopRecording());
}