webrtc_m130/webrtc/api/test/fakeaudiocapturemodule_unittest.cc

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/*
* 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 "webrtc/api/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,
Update a ton of audio code to use size_t more correctly and in general reduce use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
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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;
Match existing type usage better. This makes a variety of small changes to synchronize bits of code using different types, remove useless code or casts, and add explicit casts in some places previously doing implicit ones. For example: * Change a few type declarations to better match how the majority of code uses those objects. * Eliminate "< 0" check for unsigned values. * Replace "(float)sin(x)", where |x| is also a float, with "sinf(x)", and similar. * Add casts to uint32_t in many places timestamps were used and the existing code stored signed values into the unsigned objects. * Remove downcasts when the results would be passed to a larger type, e.g. calling "foo((int16_t)x)" with an int |x| when foo() takes an int instead of an int16_t. * Similarly, add casts when passing a larger type to a function taking a smaller one. * Add casts to int16_t when doing something like "int16_t = int16_t + int16_t" as the "+" operation would implicitly upconvert to int, and similar. * Use "false" instead of "0" for setting a bool. * Shift a few temp types when doing a multi-stage calculation involving typecasts, so as to put the most logical/semantically correct type possible into the temps. For example, when doing "int foo = int + int; size_t bar = (size_t)foo + size_t;", we might change |foo| to a size_t and move the cast if it makes more sense for |foo| to be represented as a size_t. BUG=none R=andrew@webrtc.org, asapersson@webrtc.org, henrika@webrtc.org, juberti@webrtc.org, kwiberg@webrtc.org TBR=andrew, asapersson, henrika Review URL: https://codereview.webrtc.org/1168753002 Cr-Commit-Position: refs/heads/master@{#9419}
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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.
Update a ton of audio code to use size_t more correctly and in general reduce use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
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int32_t NeedMorePlayData(const size_t nSamples,
const size_t nBytesPerSample,
const size_t nChannels,
const uint32_t samplesPerSec,
void* audioSamples,
Update a ton of audio code to use size_t more correctly and in general reduce use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
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size_t& nSamplesOut,
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms) override {
rtc::CritScope cs(&crit_);
++pull_iterations_;
Update a ton of audio code to use size_t more correctly and in general reduce use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
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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;
}
Update a ton of audio code to use size_t more correctly and in general reduce use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
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size_t GenerateZeroBuffer(void* audio_buffer, size_t audio_buffer_size) {
memset(audio_buffer, 0, audio_buffer_size);
return audio_buffer_size;
}
Update a ton of audio code to use size_t more correctly and in general reduce use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
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size_t CopyFromRecBuffer(void* audio_buffer, size_t audio_buffer_size) {
EXPECT_EQ(audio_buffer_size, rec_buffer_bytes_);
Update a ton of audio code to use size_t more correctly and in general reduce use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
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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;
}
rtc::CriticalSection crit_;
int push_iterations_;
int pull_iterations_;
char rec_buffer_[FakeAudioCaptureModule::kNumberSamples *
FakeAudioCaptureModule::kNumberBytesPerSample];
Update a ton of audio code to use size_t more correctly and in general reduce use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
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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());
}