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Revert "Reland "Refactor TestAudioDeviceModule to not depend on EventTimerWrapper.""

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This reverts commit cd87e014f34069fd5a73c1ed5b74ddf251a95c2d.

Reason for revert: Somehow introduces a race where rtc::Thread auto-wrapping may be applied to the TestAudioDevice thread rather than the main thread. This causes failures when running video_engine_tests without any test filter.

Original change's description:
> Reland "Refactor TestAudioDeviceModule to not depend on EventTimerWrapper."
>
> This is a reland of 9ea5765f78ed3d0d7b0d483e81f08fb8a2e1110a
>
> Original change's description:
> > Refactor TestAudioDeviceModule to not depend on EventTimerWrapper.
> >
> > In addition, let the processing thread loop explicitly, and not use
> > the deprecated builtin looping in PlatformThread.
> >
> > Bug: webrtc:3380
> > Change-Id: I5171ce3457b80f922c8284259882da63c8f146f1
> > Reviewed-on: https://webrtc-review.googlesource.com/96544
> > Commit-Queue: Niels Moller <nisse@webrtc.org>
> > Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> > Reviewed-by: Artem Titov <titovartem@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#24492}
>
> Bug: webrtc:3380
> Change-Id: I671e3a60ace6ade765a8537b7e20e36f1782a60d
> Reviewed-on: https://webrtc-review.googlesource.com/97320
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24528}

TBR=henrika@webrtc.org,nisse@webrtc.org,titovartem@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.
# Skip anyway, needed for a two-step revert.

No-try: True
Bug: webrtc:3380
Change-Id: Ia7c8cfab36b8259f150b5ccd0c28defd0e7237f6
Reviewed-on: https://webrtc-review.googlesource.com/97682
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24552}
This commit is contained in:
Niels Moller 2018-09-04 11:52:23 +00:00 committed by Commit Bot
parent 85e6e826a2
commit 4c6747c9db
1 changed files with 43 additions and 67 deletions
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110
modules/audio_device/include/test_audio_device.cc
View File

@ -22,20 +22,20 @@
#include "rtc_base/checks.h"
#include "rtc_base/criticalsection.h"
#include "rtc_base/event.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/platform_thread.h"
#include "rtc_base/random.h"
#include "rtc_base/refcountedobject.h"
#include "rtc_base/thread.h"
#include "rtc_base/timeutils.h"
#include "system_wrappers/include/event_wrapper.h"
namespace webrtc {
class EventTimerWrapper;
namespace {
constexpr int kFrameLengthUs = 10000;
constexpr int kFramesPerSecond = rtc::kNumMicrosecsPerSec / kFrameLengthUs;
constexpr int kFrameLengthMs = 10;
constexpr int kFramesPerSecond = 1000 / kFrameLengthMs;
// TestAudioDeviceModule implements an AudioDevice module that can act both as a
// capturer and a renderer. It will use 10ms audio frames.
@ -54,13 +54,13 @@ class TestAudioDeviceModuleImpl
float speed = 1)
: capturer_(std::move(capturer)),
renderer_(std::move(renderer)),
process_interval_us_(kFrameLengthUs / speed),
speed_(speed),
audio_callback_(nullptr),
rendering_(false),
capturing_(false),
done_rendering_(true, true),
done_capturing_(true, true),
stop_thread_(false) {
tick_(EventTimerWrapper::Create()) {
auto good_sample_rate = [](int sr) {
return sr == 8000 || sr == 16000 || sr == 32000 || sr == 44100 ||
sr == 48000;
@ -81,19 +81,16 @@ class TestAudioDeviceModuleImpl
StopPlayout();
StopRecording();
if (thread_) {
{
rtc::CritScope cs(&lock_);
stop_thread_ = true;
}
thread_->Stop();
}
}
int32_t Init() {
RTC_CHECK(tick_->StartTimer(true, kFrameLengthMs / speed_));
thread_ = absl::make_unique<rtc::PlatformThread>(
TestAudioDeviceModuleImpl::Run, this, "TestAudioDeviceModuleImpl",
rtc::kHighPriority);
TestAudioDeviceModuleImpl::Run, this, "TestAudioDeviceModuleImpl");
thread_->Start();
thread_->SetPriority(rtc::kHighPriority);
return 0;
}
@ -158,72 +155,51 @@ class TestAudioDeviceModuleImpl
private:
void ProcessAudio() {
int64_t time_us = rtc::TimeMicros();
bool logged_once = false;
for (;;) {
{
rtc::CritScope cs(&lock_);
if (stop_thread_) {
return;
{
rtc::CritScope cs(&lock_);
if (capturing_) {
// Capture 10ms of audio. 2 bytes per sample.
const bool keep_capturing = capturer_->Capture(&recording_buffer_);
uint32_t new_mic_level = 0;
if (recording_buffer_.size() > 0) {
audio_callback_->RecordedDataIsAvailable(
recording_buffer_.data(), recording_buffer_.size(), 2,
capturer_->NumChannels(), capturer_->SamplingFrequency(), 0, 0, 0,
false, new_mic_level);
}
if (capturing_) {
// Capture 10ms of audio. 2 bytes per sample.
const bool keep_capturing = capturer_->Capture(&recording_buffer_);
uint32_t new_mic_level = 0;
if (recording_buffer_.size() > 0) {
audio_callback_->RecordedDataIsAvailable(
recording_buffer_.data(), recording_buffer_.size(), 2,
capturer_->NumChannels(), capturer_->SamplingFrequency(), 0, 0,
0, false, new_mic_level);
}
if (!keep_capturing) {
capturing_ = false;
done_capturing_.Set();
}
}
if (rendering_) {
size_t samples_out = 0;
int64_t elapsed_time_ms = -1;
int64_t ntp_time_ms = -1;
const int sampling_frequency = renderer_->SamplingFrequency();
audio_callback_->NeedMorePlayData(
SamplesPerFrame(sampling_frequency), 2, renderer_->NumChannels(),
sampling_frequency, playout_buffer_.data(), samples_out,
&elapsed_time_ms, &ntp_time_ms);
const bool keep_rendering =
renderer_->Render(rtc::ArrayView<const int16_t>(
playout_buffer_.data(), samples_out));
if (!keep_rendering) {
rendering_ = false;
done_rendering_.Set();
}
if (!keep_capturing) {
capturing_ = false;
done_capturing_.Set();
}
}
time_us += process_interval_us_;
int64_t time_left_us = time_us - rtc::TimeMicros();
if (time_left_us < 0) {
if (!logged_once) {
RTC_LOG(LS_ERROR) << "ProcessAudio is too slow";
logged_once = true;
}
} else {
while (time_left_us > 1000) {
if (rtc::Thread::SleepMs(time_left_us / 1000))
break;
time_left_us = time_us - rtc::TimeMicros();
if (rendering_) {
size_t samples_out = 0;
int64_t elapsed_time_ms = -1;
int64_t ntp_time_ms = -1;
const int sampling_frequency = renderer_->SamplingFrequency();
audio_callback_->NeedMorePlayData(
SamplesPerFrame(sampling_frequency), 2, renderer_->NumChannels(),
sampling_frequency, playout_buffer_.data(), samples_out,
&elapsed_time_ms, &ntp_time_ms);
const bool keep_rendering = renderer_->Render(
rtc::ArrayView<const int16_t>(playout_buffer_.data(), samples_out));
if (!keep_rendering) {
rendering_ = false;
done_rendering_.Set();
}
}
}
tick_->Wait(WEBRTC_EVENT_INFINITE);
}
static void Run(void* obj) {
static bool Run(void* obj) {
static_cast<TestAudioDeviceModuleImpl*>(obj)->ProcessAudio();
return true;
}
const std::unique_ptr<Capturer> capturer_ RTC_GUARDED_BY(lock_);
const std::unique_ptr<Renderer> renderer_ RTC_GUARDED_BY(lock_);
const int64_t process_interval_us_;
const float speed_;
rtc::CriticalSection lock_;
AudioTransport* audio_callback_ RTC_GUARDED_BY(lock_);
@ -235,8 +211,8 @@ class TestAudioDeviceModuleImpl
std::vector<int16_t> playout_buffer_ RTC_GUARDED_BY(lock_);
rtc::BufferT<int16_t> recording_buffer_ RTC_GUARDED_BY(lock_);
std::unique_ptr<EventTimerWrapper> tick_;
std::unique_ptr<rtc::PlatformThread> thread_;
bool stop_thread_ RTC_GUARDED_BY(lock_);
};
// A fake capturer that generates pulses with random samples between