Revert of Adds data logging in native AudioDeviceBuffer class (patchset #10 id:180001 of https://codereview.webrtc.org/2132613002/ )
Reason for revert: Seems to break things upstream. Original issue's description: > Adds data logging in native AudioDeviceBuffer class. > > Goal is to provide periodic logging of most essential audio parameters > for playout and recording sides. It will allow us to track if the native audio layer is working as intended. > > BUG=NONE > > Committed: https://crrev.com/348e411dd27e6dbe9b84b27ce46e9b7c657c1eae > Cr-Commit-Position: refs/heads/master@{#13440} TBR=stefan@webrtc.org,henrika@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=NONE Review-Url: https://codereview.webrtc.org/2139233002 Cr-Commit-Position: refs/heads/master@{#13441}
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348e411dd2
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025aa94ccb
@ -10,29 +10,21 @@
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#include "webrtc/modules/audio_device/audio_device_buffer.h"
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#include "webrtc/base/bind.h"
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#include "webrtc/base/checks.h"
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#include "webrtc/base/logging.h"
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#include "webrtc/base/format_macros.h"
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#include "webrtc/base/timeutils.h"
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#include "webrtc/modules/audio_device/audio_device_config.h"
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#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
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namespace webrtc {
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static const int kHighDelayThresholdMs = 300;
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static const int kLogHighDelayIntervalFrames = 500; // 5 seconds.
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static const char kTimerQueueName[] = "AudioDeviceBufferTimer";
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// Time between two sucessive calls to LogStats().
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static const size_t kTimerIntervalInSeconds = 10;
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static const size_t kTimerIntervalInMilliseconds =
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kTimerIntervalInSeconds * rtc::kNumMillisecsPerSec;
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AudioDeviceBuffer::AudioDeviceBuffer()
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: _ptrCbAudioTransport(nullptr),
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task_queue_(kTimerQueueName),
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timer_has_started_(false),
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: _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
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_critSectCb(*CriticalSectionWrapper::CreateCriticalSection()),
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_ptrCbAudioTransport(nullptr),
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_recSampleRate(0),
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_playSampleRate(0),
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_recChannels(0),
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@ -53,72 +45,58 @@ AudioDeviceBuffer::AudioDeviceBuffer()
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_recDelayMS(0),
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_clockDrift(0),
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// Set to the interval in order to log on the first occurrence.
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high_delay_counter_(kLogHighDelayIntervalFrames),
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num_stat_reports_(0),
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rec_callbacks_(0),
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last_rec_callbacks_(0),
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play_callbacks_(0),
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last_play_callbacks_(0),
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rec_samples_(0),
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last_rec_samples_(0),
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play_samples_(0),
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last_play_samples_(0),
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last_log_stat_time_(0) {
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high_delay_counter_(kLogHighDelayIntervalFrames) {
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LOG(INFO) << "AudioDeviceBuffer::ctor";
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memset(_recBuffer, 0, kMaxBufferSizeBytes);
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memset(_playBuffer, 0, kMaxBufferSizeBytes);
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}
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AudioDeviceBuffer::~AudioDeviceBuffer() {
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RTC_DCHECK(thread_checker_.CalledOnValidThread());
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LOG(INFO) << "AudioDeviceBuffer::~dtor";
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_recFile.Flush();
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_recFile.CloseFile();
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delete &_recFile;
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{
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CriticalSectionScoped lock(&_critSect);
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_playFile.Flush();
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_playFile.CloseFile();
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delete &_playFile;
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_recFile.Flush();
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_recFile.CloseFile();
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delete &_recFile;
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_playFile.Flush();
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_playFile.CloseFile();
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delete &_playFile;
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}
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delete &_critSect;
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delete &_critSectCb;
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}
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int32_t AudioDeviceBuffer::RegisterAudioCallback(
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AudioTransport* audioCallback) {
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LOG(INFO) << __FUNCTION__;
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rtc::CritScope lock(&_critSectCb);
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CriticalSectionScoped lock(&_critSectCb);
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_ptrCbAudioTransport = audioCallback;
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return 0;
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}
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int32_t AudioDeviceBuffer::InitPlayout() {
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RTC_DCHECK(thread_checker_.CalledOnValidThread());
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LOG(INFO) << __FUNCTION__;
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if (!timer_has_started_) {
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StartTimer();
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timer_has_started_ = true;
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}
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return 0;
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}
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int32_t AudioDeviceBuffer::InitRecording() {
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RTC_DCHECK(thread_checker_.CalledOnValidThread());
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LOG(INFO) << __FUNCTION__;
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if (!timer_has_started_) {
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StartTimer();
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timer_has_started_ = true;
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}
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return 0;
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}
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int32_t AudioDeviceBuffer::SetRecordingSampleRate(uint32_t fsHz) {
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LOG(INFO) << "SetRecordingSampleRate(" << fsHz << ")";
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rtc::CritScope lock(&_critSect);
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CriticalSectionScoped lock(&_critSect);
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_recSampleRate = fsHz;
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return 0;
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}
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int32_t AudioDeviceBuffer::SetPlayoutSampleRate(uint32_t fsHz) {
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LOG(INFO) << "SetPlayoutSampleRate(" << fsHz << ")";
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rtc::CritScope lock(&_critSect);
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CriticalSectionScoped lock(&_critSect);
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_playSampleRate = fsHz;
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return 0;
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}
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@ -132,7 +110,7 @@ int32_t AudioDeviceBuffer::PlayoutSampleRate() const {
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}
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int32_t AudioDeviceBuffer::SetRecordingChannels(size_t channels) {
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rtc::CritScope lock(&_critSect);
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CriticalSectionScoped lock(&_critSect);
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_recChannels = channels;
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_recBytesPerSample =
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2 * channels; // 16 bits per sample in mono, 32 bits in stereo
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@ -140,7 +118,7 @@ int32_t AudioDeviceBuffer::SetRecordingChannels(size_t channels) {
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}
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int32_t AudioDeviceBuffer::SetPlayoutChannels(size_t channels) {
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rtc::CritScope lock(&_critSect);
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CriticalSectionScoped lock(&_critSect);
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_playChannels = channels;
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// 16 bits per sample in mono, 32 bits in stereo
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_playBytesPerSample = 2 * channels;
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@ -149,7 +127,7 @@ int32_t AudioDeviceBuffer::SetPlayoutChannels(size_t channels) {
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int32_t AudioDeviceBuffer::SetRecordingChannel(
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const AudioDeviceModule::ChannelType channel) {
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rtc::CritScope lock(&_critSect);
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CriticalSectionScoped lock(&_critSect);
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if (_recChannels == 1) {
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return -1;
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@ -215,7 +193,7 @@ void AudioDeviceBuffer::SetVQEData(int playDelayMs,
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int32_t AudioDeviceBuffer::StartInputFileRecording(
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const char fileName[kAdmMaxFileNameSize]) {
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rtc::CritScope lock(&_critSect);
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CriticalSectionScoped lock(&_critSect);
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_recFile.Flush();
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_recFile.CloseFile();
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@ -224,7 +202,7 @@ int32_t AudioDeviceBuffer::StartInputFileRecording(
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}
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int32_t AudioDeviceBuffer::StopInputFileRecording() {
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rtc::CritScope lock(&_critSect);
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CriticalSectionScoped lock(&_critSect);
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_recFile.Flush();
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_recFile.CloseFile();
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@ -234,7 +212,7 @@ int32_t AudioDeviceBuffer::StopInputFileRecording() {
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int32_t AudioDeviceBuffer::StartOutputFileRecording(
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const char fileName[kAdmMaxFileNameSize]) {
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rtc::CritScope lock(&_critSect);
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CriticalSectionScoped lock(&_critSect);
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_playFile.Flush();
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_playFile.CloseFile();
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@ -243,7 +221,7 @@ int32_t AudioDeviceBuffer::StartOutputFileRecording(
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}
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int32_t AudioDeviceBuffer::StopOutputFileRecording() {
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rtc::CritScope lock(&_critSect);
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CriticalSectionScoped lock(&_critSect);
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_playFile.Flush();
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_playFile.CloseFile();
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@ -253,7 +231,7 @@ int32_t AudioDeviceBuffer::StopOutputFileRecording() {
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int32_t AudioDeviceBuffer::SetRecordedBuffer(const void* audioBuffer,
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size_t nSamples) {
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rtc::CritScope lock(&_critSect);
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CriticalSectionScoped lock(&_critSect);
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if (_recBytesPerSample == 0) {
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assert(false);
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@ -292,16 +270,11 @@ int32_t AudioDeviceBuffer::SetRecordedBuffer(const void* audioBuffer,
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_recFile.Write(&_recBuffer[0], _recSize);
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}
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// Update some stats but do it on the task queue to ensure that the members
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// are modified and read on the same thread.
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task_queue_.PostTask(
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rtc::Bind(&AudioDeviceBuffer::UpdateRecStats, this, nSamples));
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return 0;
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}
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int32_t AudioDeviceBuffer::DeliverRecordedData() {
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rtc::CritScope lock(&_critSectCb);
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CriticalSectionScoped lock(&_critSectCb);
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// Ensure that user has initialized all essential members
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if ((_recSampleRate == 0) || (_recSamples == 0) ||
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(_recBytesPerSample == 0) || (_recChannels == 0)) {
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@ -336,7 +309,7 @@ int32_t AudioDeviceBuffer::RequestPlayoutData(size_t nSamples) {
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// TOOD(henrika): improve bad locking model and make it more clear that only
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// 10ms buffer sizes is supported in WebRTC.
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{
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rtc::CritScope lock(&_critSect);
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CriticalSectionScoped lock(&_critSect);
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// Store copies under lock and use copies hereafter to avoid race with
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// setter methods.
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@ -359,7 +332,7 @@ int32_t AudioDeviceBuffer::RequestPlayoutData(size_t nSamples) {
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size_t nSamplesOut(0);
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rtc::CritScope lock(&_critSectCb);
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CriticalSectionScoped lock(&_critSectCb);
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// It is currently supported to start playout without a valid audio
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// transport object. Leads to warning and silence.
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@ -378,16 +351,11 @@ int32_t AudioDeviceBuffer::RequestPlayoutData(size_t nSamples) {
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LOG(LS_ERROR) << "NeedMorePlayData() failed";
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}
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// Update some stats but do it on the task queue to ensure that access of
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// members is serialized hence avoiding usage of locks.
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task_queue_.PostTask(
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rtc::Bind(&AudioDeviceBuffer::UpdatePlayStats, this, nSamplesOut));
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return static_cast<int32_t>(nSamplesOut);
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}
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int32_t AudioDeviceBuffer::GetPlayoutData(void* audioBuffer) {
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rtc::CritScope lock(&_critSect);
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CriticalSectionScoped lock(&_critSect);
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RTC_CHECK_LE(_playSize, kMaxBufferSizeBytes);
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memcpy(audioBuffer, &_playBuffer[0], _playSize);
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@ -400,67 +368,4 @@ int32_t AudioDeviceBuffer::GetPlayoutData(void* audioBuffer) {
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return static_cast<int32_t>(_playSamples);
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}
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void AudioDeviceBuffer::StartTimer() {
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last_log_stat_time_ = rtc::TimeMillis();
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task_queue_.PostDelayedTask(rtc::Bind(&AudioDeviceBuffer::LogStats, this),
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kTimerIntervalInMilliseconds);
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}
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void AudioDeviceBuffer::LogStats() {
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RTC_DCHECK(task_queue_.IsCurrent());
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int64_t now_time = rtc::TimeMillis();
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int64_t next_callback_time = now_time + kTimerIntervalInMilliseconds;
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int64_t time_since_last = rtc::TimeDiff(now_time, last_log_stat_time_);
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last_log_stat_time_ = now_time;
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// Log the latest statistics but skip the first 10 seconds since we are not
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// sure of the exact starting point. I.e., the first log printout will be
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// after ~20 seconds.
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if (++num_stat_reports_ > 1) {
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uint32_t diff_samples = rec_samples_ - last_rec_samples_;
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uint32_t rate = diff_samples / kTimerIntervalInSeconds;
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LOG(INFO) << "[REC : " << time_since_last << "msec, "
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<< _recSampleRate / 1000
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<< "kHz] callbacks: " << rec_callbacks_ - last_rec_callbacks_
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<< ", "
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<< "samples: " << diff_samples << ", "
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<< "rate: " << rate;
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diff_samples = play_samples_ - last_play_samples_;
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rate = diff_samples / kTimerIntervalInSeconds;
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LOG(INFO) << "[PLAY: " << time_since_last << "msec, "
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<< _playSampleRate / 1000
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<< "kHz] callbacks: " << play_callbacks_ - last_play_callbacks_
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<< ", "
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<< "samples: " << diff_samples << ", "
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<< "rate: " << rate;
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}
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last_rec_callbacks_ = rec_callbacks_;
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last_play_callbacks_ = play_callbacks_;
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last_rec_samples_ = rec_samples_;
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last_play_samples_ = play_samples_;
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int64_t time_to_wait_ms = next_callback_time - rtc::TimeMillis();
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RTC_DCHECK_GT(time_to_wait_ms, 0) << "Invalid timer interval";
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// Update some stats but do it on the task queue to ensure that access of
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// members is serialized hence avoiding usage of locks.
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task_queue_.PostDelayedTask(rtc::Bind(&AudioDeviceBuffer::LogStats, this),
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time_to_wait_ms);
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}
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void AudioDeviceBuffer::UpdateRecStats(size_t num_samples) {
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RTC_DCHECK(task_queue_.IsCurrent());
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++rec_callbacks_;
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rec_samples_ += num_samples;
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}
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void AudioDeviceBuffer::UpdatePlayStats(size_t num_samples) {
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RTC_DCHECK(task_queue_.IsCurrent());
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++play_callbacks_;
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play_samples_ += num_samples;
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}
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} // namespace webrtc
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@ -8,12 +8,9 @@
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_
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#define WEBRTC_MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_
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#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H
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#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H
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#include "webrtc/base/criticalsection.h"
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#include "webrtc/base/task_queue.h"
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#include "webrtc/base/thread_checker.h"
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#include "webrtc/modules/audio_device/include/audio_device.h"
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#include "webrtc/system_wrappers/include/file_wrapper.h"
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#include "webrtc/typedefs.h"
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@ -66,36 +63,11 @@ class AudioDeviceBuffer {
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int32_t SetTypingStatus(bool typingStatus);
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private:
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// Posts the first delayed task in the task queue and starts the periodic
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// timer.
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void StartTimer();
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// Called periodically on the internal thread created by the TaskQueue.
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void LogStats();
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// Updates counters in each play/record callback but does it on the task
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// queue to ensure that they can be read by LogStats() without any locks since
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// each task is serialized by the task queue.
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void UpdateRecStats(size_t num_samples);
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void UpdatePlayStats(size_t num_samples);
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// Ensures that methods are called on the same thread as the thread that
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// creates this object.
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rtc::ThreadChecker thread_checker_;
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rtc::CriticalSection _critSect;
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rtc::CriticalSection _critSectCb;
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CriticalSectionWrapper& _critSect;
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CriticalSectionWrapper& _critSectCb;
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AudioTransport* _ptrCbAudioTransport;
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// Task queue used to invoke LogStats() periodically. Tasks are executed on a
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// worker thread but it does not necessarily have to be the same thread for
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// each task.
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rtc::TaskQueue task_queue_;
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// Ensures that the timer is only started once.
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bool timer_has_started_;
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uint32_t _recSampleRate;
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uint32_t _playSampleRate;
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@ -135,40 +107,8 @@ class AudioDeviceBuffer {
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int _recDelayMS;
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int _clockDrift;
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int high_delay_counter_;
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// Counts number of times LogStats() has been called.
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size_t num_stat_reports_;
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// Total number of recording callbacks where the source provides 10ms audio
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// data each time.
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uint64_t rec_callbacks_;
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// Total number of recording callbacks stored at the last timer task.
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uint64_t last_rec_callbacks_;
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// Total number of playback callbacks where the sink asks for 10ms audio
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// data each time.
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uint64_t play_callbacks_;
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// Total number of playout callbacks stored at the last timer task.
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uint64_t last_play_callbacks_;
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// Total number of recorded audio samples.
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uint64_t rec_samples_;
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// Total number of recorded samples stored at the previous timer task.
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uint64_t last_rec_samples_;
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// Total number of played audio samples.
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uint64_t play_samples_;
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// Total number of played samples stored at the previous timer task.
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uint64_t last_play_samples_;
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// Time stamp of last stat report.
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uint64_t last_log_stat_time_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_
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#endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H
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