Adds logging of native audio levels and UMA stats to track issues.
This changes added a simple measurement of levels "close to the audio hardware"
both for playout and for recording. These levels are logged once each 10 seconds.
It also adds WebRTC.Audio.RecordedOnlyZeros UMA stat and it is updated at
destuction. It will report true iff all reported recording leves are zero.
BUG=NONE
R=peah@webrtc.org
Review URL: https://codereview.webrtc.org/2328433003 .
Cr-Commit-Position: refs/heads/master@{#14160}
diff --git a/webrtc/modules/audio_device/audio_device_buffer.cc b/webrtc/modules/audio_device/audio_device_buffer.cc
index 5a65288..f87e068 100644
--- a/webrtc/modules/audio_device/audio_device_buffer.cc
+++ b/webrtc/modules/audio_device/audio_device_buffer.cc
@@ -18,7 +18,9 @@
#include "webrtc/base/logging.h"
#include "webrtc/base/format_macros.h"
#include "webrtc/base/timeutils.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/modules/audio_device/audio_device_config.h"
+#include "webrtc/system_wrappers/include/metrics.h"
namespace webrtc {
@@ -59,7 +61,10 @@
last_rec_samples_(0),
play_samples_(0),
last_play_samples_(0),
- last_log_stat_time_(0) {
+ last_log_stat_time_(0),
+ max_rec_level_(0),
+ max_play_level_(0),
+ num_rec_level_is_zero_(0) {
LOG(INFO) << "AudioDeviceBuffer::ctor";
// TODO(henrika): improve buffer handling and ensure that we don't allocate
// more than what is required.
@@ -88,6 +93,16 @@
LOG(INFO) << "average: "
<< static_cast<float>(total_diff_time) / num_measurements;
}
+
+ // Add UMA histogram to keep track of the case when only zeros have been
+ // recorded. Ensure that recording callbacks have started and that at least
+ // one timer event has been able to update |num_rec_level_is_zero_|.
+ // I am avoiding use of the task queue here since we are under destruction
+ // and reading these members on the creating thread feels safe.
+ if (rec_callbacks_ > 0 && num_stat_reports_ > 0) {
+ RTC_LOGGED_HISTOGRAM_BOOLEAN("WebRTC.Audio.RecordedOnlyZeros",
+ static_cast<int>(num_stat_reports_ == num_rec_level_is_zero_));
+ }
}
int32_t AudioDeviceBuffer::RegisterAudioCallback(
@@ -101,7 +116,7 @@
int32_t AudioDeviceBuffer::InitPlayout() {
LOG(INFO) << __FUNCTION__;
RTC_DCHECK(thread_checker_.CalledOnValidThread());
- last_playout_time_ = rtc::TimeMillis();
+ ResetPlayStats();
if (!timer_has_started_) {
StartTimer();
timer_has_started_ = true;
@@ -112,6 +127,7 @@
int32_t AudioDeviceBuffer::InitRecording() {
LOG(INFO) << __FUNCTION__;
RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ ResetRecStats();
if (!timer_has_started_) {
StartTimer();
timer_has_started_ = true;
@@ -266,8 +282,8 @@
// Update some stats but do it on the task queue to ensure that the members
// are modified and read on the same thread.
- task_queue_.PostTask(
- rtc::Bind(&AudioDeviceBuffer::UpdateRecStats, this, num_samples));
+ task_queue_.PostTask(rtc::Bind(&AudioDeviceBuffer::UpdateRecStats, this,
+ audio_buffer, num_samples));
return 0;
}
@@ -336,8 +352,8 @@
// Update some stats but do it on the task queue to ensure that access of
// members is serialized hence avoiding usage of locks.
- task_queue_.PostTask(
- rtc::Bind(&AudioDeviceBuffer::UpdatePlayStats, this, num_samples_out));
+ task_queue_.PostTask(rtc::Bind(&AudioDeviceBuffer::UpdatePlayStats, this,
+ &play_buffer_[0], num_samples_out));
return static_cast<int32_t>(num_samples_out);
}
@@ -366,6 +382,7 @@
}
void AudioDeviceBuffer::StartTimer() {
+ num_stat_reports_ = 0;
last_log_stat_time_ = rtc::TimeMillis();
task_queue_.PostDelayedTask(rtc::Bind(&AudioDeviceBuffer::LogStats, this),
kTimerIntervalInMilliseconds);
@@ -390,7 +407,8 @@
<< "kHz] callbacks: " << rec_callbacks_ - last_rec_callbacks_
<< ", "
<< "samples: " << diff_samples << ", "
- << "rate: " << rate;
+ << "rate: " << rate << ", "
+ << "level: " << max_rec_level_;
diff_samples = play_samples_ - last_play_samples_;
rate = diff_samples / kTimerIntervalInSeconds;
@@ -399,13 +417,22 @@
<< "kHz] callbacks: " << play_callbacks_ - last_play_callbacks_
<< ", "
<< "samples: " << diff_samples << ", "
- << "rate: " << rate;
+ << "rate: " << rate << ", "
+ << "level: " << max_play_level_;
+ }
+
+ // Count number of times we detect "no audio" corresponding to a case where
+ // all level measurements have been zero.
+ if (max_rec_level_ == 0) {
+ ++num_rec_level_is_zero_;
}
last_rec_callbacks_ = rec_callbacks_;
last_play_callbacks_ = play_callbacks_;
last_rec_samples_ = rec_samples_;
last_play_samples_ = play_samples_;
+ max_rec_level_ = 0;
+ max_play_level_ = 0;
int64_t time_to_wait_ms = next_callback_time - rtc::TimeMillis();
RTC_DCHECK_GT(time_to_wait_ms, 0) << "Invalid timer interval";
@@ -416,16 +443,58 @@
time_to_wait_ms);
}
-void AudioDeviceBuffer::UpdateRecStats(size_t num_samples) {
+void AudioDeviceBuffer::ResetRecStats() {
+ rec_callbacks_ = 0;
+ last_rec_callbacks_ = 0;
+ rec_samples_ = 0;
+ last_rec_samples_ = 0;
+ max_rec_level_ = 0;
+ num_rec_level_is_zero_ = 0;
+}
+
+void AudioDeviceBuffer::ResetPlayStats() {
+ last_playout_time_ = rtc::TimeMillis();
+ play_callbacks_ = 0;
+ last_play_callbacks_ = 0;
+ play_samples_ = 0;
+ last_play_samples_ = 0;
+ max_play_level_ = 0;
+}
+
+void AudioDeviceBuffer::UpdateRecStats(const void* audio_buffer,
+ size_t num_samples) {
RTC_DCHECK(task_queue_.IsCurrent());
++rec_callbacks_;
rec_samples_ += num_samples;
+
+ // Find the max absolute value in an audio packet twice per second and update
+ // |max_rec_level_| to track the largest value.
+ if (rec_callbacks_ % 50 == 0) {
+ int16_t max_abs = WebRtcSpl_MaxAbsValueW16(
+ static_cast<int16_t*>(const_cast<void*>(audio_buffer)),
+ num_samples * rec_channels_);
+ if (max_abs > max_rec_level_) {
+ max_rec_level_ = max_abs;
+ }
+ }
}
-void AudioDeviceBuffer::UpdatePlayStats(size_t num_samples) {
+void AudioDeviceBuffer::UpdatePlayStats(const void* audio_buffer,
+ size_t num_samples) {
RTC_DCHECK(task_queue_.IsCurrent());
++play_callbacks_;
play_samples_ += num_samples;
+
+ // Find the max absolute value in an audio packet twice per second and update
+ // |max_play_level_| to track the largest value.
+ if (play_callbacks_ % 50 == 0) {
+ int16_t max_abs = WebRtcSpl_MaxAbsValueW16(
+ static_cast<int16_t*>(const_cast<void*>(audio_buffer)),
+ num_samples * play_channels_);
+ if (max_abs > max_play_level_) {
+ max_play_level_ = max_abs;
+ }
+ }
}
} // namespace webrtc