/*
|
* libjingle
|
* Copyright 2012 Google Inc.
|
*
|
* Redistribution and use in source and binary forms, with or without
|
* modification, are permitted provided that the following conditions are met:
|
*
|
* 1. Redistributions of source code must retain the above copyright notice,
|
* this list of conditions and the following disclaimer.
|
* 2. Redistributions in binary form must reproduce the above copyright notice,
|
* this list of conditions and the following disclaimer in the documentation
|
* and/or other materials provided with the distribution.
|
* 3. The name of the author may not be used to endorse or promote products
|
* derived from this software without specific prior written permission.
|
*
|
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
|
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
|
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
|
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
|
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
|
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
|
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
|
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
|
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
|
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
|
*/
|
|
#include "talk/app/webrtc/test/fakeaudiocapturemodule.h"
|
|
#include <algorithm>
|
|
#include "webrtc/base/criticalsection.h"
|
#include "webrtc/base/gunit.h"
|
#include "webrtc/base/scoped_ref_ptr.h"
|
#include "webrtc/base/thread.h"
|
|
using std::min;
|
|
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_));
|
}
|
|
virtual void SetUp() {
|
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 {
|
rtc::CritScope cs(&crit_);
|
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;
|
}
|
|
// 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 {
|
rtc::CritScope cs(&crit_);
|
++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 {
|
rtc::CritScope cs(&crit_);
|
return push_iterations_;
|
}
|
int pull_iterations() const {
|
rtc::CritScope cs(&crit_);
|
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 = min(audio_buffer_size, rec_buffer_bytes_);
|
memcpy(audio_buffer, rec_buffer_, min_buffer_size);
|
return min_buffer_size;
|
}
|
|
mutable rtc::CriticalSection crit_;
|
|
int push_iterations_;
|
int pull_iterations_;
|
|
char rec_buffer_[FakeAudioCaptureModule::kNumberSamples *
|
FakeAudioCaptureModule::kNumberBytesPerSample];
|
size_t rec_buffer_bytes_;
|
};
|
|
TEST_F(FakeAdmTest, TestProccess) {
|
// Next process call must be some time in the future (or now).
|
EXPECT_LE(0, fake_audio_capture_module_->TimeUntilNextProcess());
|
// Process call updates TimeUntilNextProcess() but there are no guarantees on
|
// timing so just check that Process can ba called successfully.
|
EXPECT_LE(0, fake_audio_capture_module_->Process());
|
}
|
|
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());
|
}
|
