talk/ and webrtc/ are hosted in the same repository and it no longer makes sense to support building talk/ without the corresponding webrtc/ catalog. R=bjornv@webrtc.org, juberti@webrtc.org BUG= Review URL: https://webrtc-codereview.appspot.com/39849004 Cr-Commit-Position: refs/heads/master@{#8291} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8291 4adac7df-926f-26a2-2b94-8c16560cd09d
207 lines
7.9 KiB
C++
207 lines
7.9 KiB
C++
/*
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* libjingle
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* Copyright 2012 Google Inc.
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions are met:
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*
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* 1. Redistributions of source code must retain the above copyright notice,
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* this list of conditions and the following disclaimer.
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* 2. Redistributions in binary form must reproduce the above copyright notice,
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* this list of conditions and the following disclaimer in the documentation
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* and/or other materials provided with the distribution.
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* 3. The name of the author may not be used to endorse or promote products
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* derived from this software without specific prior written permission.
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*
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* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
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* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
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* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
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* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
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* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
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* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
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* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
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* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
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* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
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* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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*/
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#include "talk/app/webrtc/test/fakeaudiocapturemodule.h"
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#include <algorithm>
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#include "webrtc/base/gunit.h"
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#include "webrtc/base/scoped_ref_ptr.h"
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#include "webrtc/base/thread.h"
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using std::min;
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class FakeAdmTest : public testing::Test,
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public webrtc::AudioTransport {
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protected:
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static const int kMsInSecond = 1000;
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FakeAdmTest()
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: push_iterations_(0),
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pull_iterations_(0),
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rec_buffer_bytes_(0) {
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memset(rec_buffer_, 0, sizeof(rec_buffer_));
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}
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virtual void SetUp() {
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fake_audio_capture_module_ = FakeAudioCaptureModule::Create(
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rtc::Thread::Current());
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EXPECT_TRUE(fake_audio_capture_module_.get() != NULL);
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}
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// Callbacks inherited from webrtc::AudioTransport.
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// ADM is pushing data.
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virtual int32_t RecordedDataIsAvailable(const void* audioSamples,
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const uint32_t nSamples,
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const uint8_t nBytesPerSample,
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const uint8_t nChannels,
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const uint32_t samplesPerSec,
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const uint32_t totalDelayMS,
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const int32_t clockDrift,
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const uint32_t currentMicLevel,
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const bool keyPressed,
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uint32_t& newMicLevel) {
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rec_buffer_bytes_ = nSamples * nBytesPerSample;
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if ((rec_buffer_bytes_ <= 0) ||
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(rec_buffer_bytes_ > FakeAudioCaptureModule::kNumberSamples *
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FakeAudioCaptureModule::kNumberBytesPerSample)) {
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ADD_FAILURE();
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return -1;
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}
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memcpy(rec_buffer_, audioSamples, rec_buffer_bytes_);
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++push_iterations_;
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newMicLevel = currentMicLevel;
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return 0;
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}
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// ADM is pulling data.
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virtual int32_t NeedMorePlayData(const uint32_t nSamples,
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const uint8_t nBytesPerSample,
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const uint8_t nChannels,
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const uint32_t samplesPerSec,
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void* audioSamples,
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uint32_t& nSamplesOut,
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int64_t* elapsed_time_ms,
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int64_t* ntp_time_ms) {
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++pull_iterations_;
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const uint32_t audio_buffer_size = nSamples * nBytesPerSample;
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const uint32_t bytes_out = RecordedDataReceived() ?
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CopyFromRecBuffer(audioSamples, audio_buffer_size):
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GenerateZeroBuffer(audioSamples, audio_buffer_size);
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nSamplesOut = bytes_out / nBytesPerSample;
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*elapsed_time_ms = 0;
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*ntp_time_ms = 0;
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return 0;
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}
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int push_iterations() const { return push_iterations_; }
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int pull_iterations() const { return pull_iterations_; }
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rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
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private:
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bool RecordedDataReceived() const {
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return rec_buffer_bytes_ != 0;
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}
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int32_t GenerateZeroBuffer(void* audio_buffer, uint32_t audio_buffer_size) {
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memset(audio_buffer, 0, audio_buffer_size);
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return audio_buffer_size;
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}
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int32_t CopyFromRecBuffer(void* audio_buffer, uint32_t audio_buffer_size) {
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EXPECT_EQ(audio_buffer_size, rec_buffer_bytes_);
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const uint32_t min_buffer_size = min(audio_buffer_size, rec_buffer_bytes_);
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memcpy(audio_buffer, rec_buffer_, min_buffer_size);
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return min_buffer_size;
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}
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int push_iterations_;
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int pull_iterations_;
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char rec_buffer_[FakeAudioCaptureModule::kNumberSamples *
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FakeAudioCaptureModule::kNumberBytesPerSample];
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uint32_t rec_buffer_bytes_;
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};
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TEST_F(FakeAdmTest, TestProccess) {
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// Next process call must be some time in the future (or now).
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EXPECT_LE(0, fake_audio_capture_module_->TimeUntilNextProcess());
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// Process call updates TimeUntilNextProcess() but there are no guarantees on
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// timing so just check that Process can ba called successfully.
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EXPECT_LE(0, fake_audio_capture_module_->Process());
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}
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TEST_F(FakeAdmTest, PlayoutTest) {
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EXPECT_EQ(0, fake_audio_capture_module_->RegisterAudioCallback(this));
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bool stereo_available = false;
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EXPECT_EQ(0,
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fake_audio_capture_module_->StereoPlayoutIsAvailable(
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&stereo_available));
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EXPECT_TRUE(stereo_available);
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EXPECT_NE(0, fake_audio_capture_module_->StartPlayout());
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EXPECT_FALSE(fake_audio_capture_module_->PlayoutIsInitialized());
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EXPECT_FALSE(fake_audio_capture_module_->Playing());
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EXPECT_EQ(0, fake_audio_capture_module_->StopPlayout());
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EXPECT_EQ(0, fake_audio_capture_module_->InitPlayout());
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EXPECT_TRUE(fake_audio_capture_module_->PlayoutIsInitialized());
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EXPECT_FALSE(fake_audio_capture_module_->Playing());
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EXPECT_EQ(0, fake_audio_capture_module_->StartPlayout());
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EXPECT_TRUE(fake_audio_capture_module_->Playing());
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uint16_t delay_ms = 10;
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EXPECT_EQ(0, fake_audio_capture_module_->PlayoutDelay(&delay_ms));
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EXPECT_EQ(0, delay_ms);
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EXPECT_TRUE_WAIT(pull_iterations() > 0, kMsInSecond);
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EXPECT_GE(0, push_iterations());
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EXPECT_EQ(0, fake_audio_capture_module_->StopPlayout());
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EXPECT_FALSE(fake_audio_capture_module_->Playing());
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}
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TEST_F(FakeAdmTest, RecordTest) {
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EXPECT_EQ(0, fake_audio_capture_module_->RegisterAudioCallback(this));
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bool stereo_available = false;
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EXPECT_EQ(0, fake_audio_capture_module_->StereoRecordingIsAvailable(
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&stereo_available));
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EXPECT_FALSE(stereo_available);
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EXPECT_NE(0, fake_audio_capture_module_->StartRecording());
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EXPECT_FALSE(fake_audio_capture_module_->Recording());
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EXPECT_EQ(0, fake_audio_capture_module_->StopRecording());
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EXPECT_EQ(0, fake_audio_capture_module_->InitRecording());
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EXPECT_EQ(0, fake_audio_capture_module_->StartRecording());
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EXPECT_TRUE(fake_audio_capture_module_->Recording());
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EXPECT_TRUE_WAIT(push_iterations() > 0, kMsInSecond);
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EXPECT_GE(0, pull_iterations());
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EXPECT_EQ(0, fake_audio_capture_module_->StopRecording());
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EXPECT_FALSE(fake_audio_capture_module_->Recording());
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}
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TEST_F(FakeAdmTest, DuplexTest) {
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EXPECT_EQ(0, fake_audio_capture_module_->RegisterAudioCallback(this));
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EXPECT_EQ(0, fake_audio_capture_module_->InitPlayout());
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EXPECT_EQ(0, fake_audio_capture_module_->StartPlayout());
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EXPECT_EQ(0, fake_audio_capture_module_->InitRecording());
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EXPECT_EQ(0, fake_audio_capture_module_->StartRecording());
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EXPECT_TRUE_WAIT(push_iterations() > 0, kMsInSecond);
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EXPECT_TRUE_WAIT(pull_iterations() > 0, kMsInSecond);
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EXPECT_EQ(0, fake_audio_capture_module_->StopPlayout());
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EXPECT_EQ(0, fake_audio_capture_module_->StopRecording());
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}
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