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Adds delta-time logging for audio playout

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BUG=NONE
R=magjed@webrtc.org

Review URL: https://codereview.webrtc.org/2190343002 .

Cr-Commit-Position: refs/heads/master@{#13576}
This commit is contained in:
henrika
2016-07-29 16:20:47 +02:00
parent b77bd81a4a
commit 3d7346fd46
2 changed files with 47 additions and 0 deletions
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34
webrtc/modules/audio_device/audio_device_buffer.cc
View File

@ -8,8 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include <algorithm>
#include "webrtc/modules/audio_device/audio_device_buffer.h"
#include "webrtc/base/arraysize.h"
#include "webrtc/base/bind.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
@ -72,6 +75,25 @@ AudioDeviceBuffer::AudioDeviceBuffer()
AudioDeviceBuffer::~AudioDeviceBuffer() {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
LOG(INFO) << "AudioDeviceBuffer::~dtor";
size_t total_diff_time = 0;
int num_measurements = 0;
LOG(INFO) << "[playout diff time => #measurements]";
for (size_t diff = 0; diff < arraysize(playout_diff_times_); ++diff) {
uint32_t num_elements = playout_diff_times_[diff];
if (num_elements > 0) {
total_diff_time += num_elements * diff;
num_measurements += num_elements;
LOG(INFO) << "[" << diff << " => " << num_elements << "]";
}
}
if (num_measurements > 0) {
LOG(INFO) << "total_diff_time: " << total_diff_time;
LOG(INFO) << "num_measurements: " << num_measurements;
LOG(INFO) << "average: "
<< static_cast<float>(total_diff_time) / num_measurements;
}
_recFile.Flush();
_recFile.CloseFile();
delete &_recFile;
@ -92,6 +114,7 @@ int32_t AudioDeviceBuffer::RegisterAudioCallback(
int32_t AudioDeviceBuffer::InitPlayout() {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
LOG(INFO) << __FUNCTION__;
last_playout_time_ = rtc::TimeMillis();
if (!timer_has_started_) {
StartTimer();
timer_has_started_ = true;
@ -333,6 +356,17 @@ int32_t AudioDeviceBuffer::RequestPlayoutData(size_t nSamples) {
size_t playBytesPerSample = 0;
size_t playChannels = 0;
// Measure time since last function call and update an array where the
// position/index corresponds to time differences (in milliseconds) between
// two successive playout callbacks, and the stored value is the number of
// times a given time difference was found.
int64_t now_time = rtc::TimeMillis();
size_t diff_time = rtc::TimeDiff(now_time, last_playout_time_);
// Truncate at 500ms to limit the size of the array.
diff_time = std::min(kMaxDeltaTimeInMs, diff_time);
last_playout_time_ = now_time;
playout_diff_times_[diff_time]++;
// TOOD(henrika): improve bad locking model and make it more clear that only
// 10ms buffer sizes is supported in WebRTC.
{

13
webrtc/modules/audio_device/audio_device_buffer.h
View File

@ -23,6 +23,9 @@ class CriticalSectionWrapper;
const uint32_t kPulsePeriodMs = 1000;
const size_t kMaxBufferSizeBytes = 3840; // 10ms in stereo @ 96kHz
// Delta times between two successive playout callbacks are limited to this
// value before added to an internal array.
const size_t kMaxDeltaTimeInMs = 500;
class AudioDeviceObserver;
@ -167,6 +170,16 @@ class AudioDeviceBuffer {
// Time stamp of last stat report.
uint64_t last_log_stat_time_;
// Time stamp of last playout callback.
uint64_t last_playout_time_;
// An array where the position corresponds to time differences (in
// milliseconds) between two successive playout callbacks, and the stored
// value is the number of times a given time difference was found.
// Writing to the array is done without a lock since it is only read once at
// destruction when no audio is running.
uint32_t playout_diff_times_[kMaxDeltaTimeInMs + 1] = {0};
};
} // namespace webrtc