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Revert "Adds WebRTC.Audio.Record/PlayoutSampleRateOffsetInPercent UMA stats to native WebRTC."

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This reverts commit f217903a67995496a1d67674d77d5f237772b01b.

Reason for revert: Breaks downstream tests

Original change's description:
> Adds WebRTC.Audio.Record/PlayoutSampleRateOffsetInPercent UMA stats to native WebRTC.
> 
> Also ensures that audio parameters are accessed atomically.
> 
> Bug: b/113648245
> Change-Id: Ic812bfe2b2c4cfb3b00d9d411bb4986dfeda1028
> Reviewed-on: https://webrtc-review.googlesource.com/97331
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24550}

TBR=henrika@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org

Change-Id: I620406f25762cf76db0470b3b29b50bc146935c7
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: b/113648245
Reviewed-on: https://webrtc-review.googlesource.com/97941
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24569}
This commit is contained in:
Patrik Höglund
2018-09-05 08:52:40 +00:00
committed by Commit Bot
parent 2ac2739d06
commit e2924d555d
3 changed files with 33 additions and 48 deletions
Download Patch File Download Diff File Expand all files Collapse all files

4
modules/audio_device/android/audio_device_unittest.cc
View File

@ -755,9 +755,9 @@ TEST_F(AudioDeviceTest, UsesCorrectDelayEstimateForLowLatencyOutputPath) {
// correct set of parameters. // correct set of parameters.
TEST_F(AudioDeviceTest, VerifyAudioDeviceBufferParameters) { TEST_F(AudioDeviceTest, VerifyAudioDeviceBufferParameters) {
EXPECT_EQ(playout_parameters_.sample_rate(), EXPECT_EQ(playout_parameters_.sample_rate(),
static_cast<int>(audio_device_buffer()->PlayoutSampleRate())); audio_device_buffer()->PlayoutSampleRate());
EXPECT_EQ(record_parameters_.sample_rate(), EXPECT_EQ(record_parameters_.sample_rate(),
static_cast<int>(audio_device_buffer()->RecordingSampleRate())); audio_device_buffer()->RecordingSampleRate());
EXPECT_EQ(playout_parameters_.channels(), EXPECT_EQ(playout_parameters_.channels(),
audio_device_buffer()->PlayoutChannels()); audio_device_buffer()->PlayoutChannels());
EXPECT_EQ(record_parameters_.channels(), EXPECT_EQ(record_parameters_.channels(),

49
modules/audio_device/audio_device_buffer.cc
View File

@ -188,36 +188,43 @@ int32_t AudioDeviceBuffer::SetRecordingSampleRate(uint32_t fsHz) {
} }
int32_t AudioDeviceBuffer::SetPlayoutSampleRate(uint32_t fsHz) { int32_t AudioDeviceBuffer::SetPlayoutSampleRate(uint32_t fsHz) {
RTC_DCHECK(main_thread_checker_.CalledOnValidThread());
RTC_LOG(INFO) << "SetPlayoutSampleRate(" << fsHz << ")"; RTC_LOG(INFO) << "SetPlayoutSampleRate(" << fsHz << ")";
play_sample_rate_ = fsHz; play_sample_rate_ = fsHz;
return 0; return 0;
} }
uint32_t AudioDeviceBuffer::RecordingSampleRate() const { int32_t AudioDeviceBuffer::RecordingSampleRate() const {
RTC_DCHECK(main_thread_checker_.CalledOnValidThread());
return rec_sample_rate_; return rec_sample_rate_;
} }
uint32_t AudioDeviceBuffer::PlayoutSampleRate() const { int32_t AudioDeviceBuffer::PlayoutSampleRate() const {
RTC_DCHECK(main_thread_checker_.CalledOnValidThread());
return play_sample_rate_; return play_sample_rate_;
} }
int32_t AudioDeviceBuffer::SetRecordingChannels(size_t channels) { int32_t AudioDeviceBuffer::SetRecordingChannels(size_t channels) {
RTC_DCHECK(main_thread_checker_.CalledOnValidThread());
RTC_LOG(INFO) << "SetRecordingChannels(" << channels << ")"; RTC_LOG(INFO) << "SetRecordingChannels(" << channels << ")";
rec_channels_ = channels; rec_channels_ = channels;
return 0; return 0;
} }
int32_t AudioDeviceBuffer::SetPlayoutChannels(size_t channels) { int32_t AudioDeviceBuffer::SetPlayoutChannels(size_t channels) {
RTC_DCHECK(main_thread_checker_.CalledOnValidThread());
RTC_LOG(INFO) << "SetPlayoutChannels(" << channels << ")"; RTC_LOG(INFO) << "SetPlayoutChannels(" << channels << ")";
play_channels_ = channels; play_channels_ = channels;
return 0; return 0;
} }
size_t AudioDeviceBuffer::RecordingChannels() const { size_t AudioDeviceBuffer::RecordingChannels() const {
RTC_DCHECK(main_thread_checker_.CalledOnValidThread());
return rec_channels_; return rec_channels_;
} }
size_t AudioDeviceBuffer::PlayoutChannels() const { size_t AudioDeviceBuffer::PlayoutChannels() const {
RTC_DCHECK(main_thread_checker_.CalledOnValidThread());
return play_channels_; return play_channels_;
} }
@ -412,54 +419,28 @@ void AudioDeviceBuffer::LogStats(LogState state) {
stats_.max_play_level = 0; stats_.max_play_level = 0;
} }
// Cache current sample rate from atomic members. // Log the latest statistics but skip the first round just after state was
const uint32_t rec_sample_rate = rec_sample_rate_; // set to LOG_START. Hence, first printed log will be after ~10 seconds.
const uint32_t play_sample_rate = play_sample_rate_; if (++num_stat_reports_ > 1 && time_since_last > 0) {
RTC_DCHECK_GT(rec_sample_rate, 0u);
RTC_DCHECK_GT(play_sample_rate, 0u);
// Log the latest statistics but skip the first two rounds just after state
// was set to LOG_START to ensure that we have at least one full stable
// 10-second interval for sample-rate estimation. Hence, first printed log
// will be after ~20 seconds.
if (++num_stat_reports_ > 2 && time_since_last > 0) {
uint32_t diff_samples = stats.rec_samples - last_stats_.rec_samples; uint32_t diff_samples = stats.rec_samples - last_stats_.rec_samples;
float rate = diff_samples / (static_cast<float>(time_since_last) / 1000.0); float rate = diff_samples / (static_cast<float>(time_since_last) / 1000.0);
uint32_t abs_diff_rate_in_percent = 0;
if (rec_sample_rate > 0) {
abs_diff_rate_in_percent = static_cast<uint32_t>(
0.5f +
((100.0f * std::abs(rate - rec_sample_rate)) / rec_sample_rate));
RTC_HISTOGRAM_PERCENTAGE("WebRTC.Audio.RecordSampleRateOffsetInPercent",
abs_diff_rate_in_percent);
}
RTC_LOG(INFO) << "[REC : " << time_since_last << "msec, " RTC_LOG(INFO) << "[REC : " << time_since_last << "msec, "
<< rec_sample_rate / 1000 << "kHz] callbacks: " << rec_sample_rate_ / 1000 << "kHz] callbacks: "
<< stats.rec_callbacks - last_stats_.rec_callbacks << ", " << stats.rec_callbacks - last_stats_.rec_callbacks << ", "
<< "samples: " << diff_samples << ", " << "samples: " << diff_samples << ", "
<< "rate: " << static_cast<int>(rate + 0.5) << ", " << "rate: " << static_cast<int>(rate + 0.5) << ", "
<< "rate diff: " << abs_diff_rate_in_percent << "%, "
<< "level: " << stats.max_rec_level; << "level: " << stats.max_rec_level;
diff_samples = stats.play_samples - last_stats_.play_samples; diff_samples = stats.play_samples - last_stats_.play_samples;
rate = diff_samples / (static_cast<float>(time_since_last) / 1000.0); rate = diff_samples / (static_cast<float>(time_since_last) / 1000.0);
abs_diff_rate_in_percent = 0;
if (play_sample_rate > 0) {
abs_diff_rate_in_percent = static_cast<uint32_t>(
0.5f +
((100.0f * std::abs(rate - play_sample_rate)) / play_sample_rate));
RTC_HISTOGRAM_PERCENTAGE("WebRTC.Audio.PlayoutSampleRateOffsetInPercent",
abs_diff_rate_in_percent);
}
RTC_LOG(INFO) << "[PLAY: " << time_since_last << "msec, " RTC_LOG(INFO) << "[PLAY: " << time_since_last << "msec, "
<< play_sample_rate / 1000 << "kHz] callbacks: " << play_sample_rate_ / 1000 << "kHz] callbacks: "
<< stats.play_callbacks - last_stats_.play_callbacks << ", " << stats.play_callbacks - last_stats_.play_callbacks << ", "
<< "samples: " << diff_samples << ", " << "samples: " << diff_samples << ", "
<< "rate: " << static_cast<int>(rate + 0.5) << ", " << "rate: " << static_cast<int>(rate + 0.5) << ", "
<< "rate diff: " << abs_diff_rate_in_percent << "%, "
<< "level: " << stats.max_play_level; << "level: " << stats.max_play_level;
}
last_stats_ = stats; last_stats_ = stats;
}
int64_t time_to_wait_ms = next_callback_time - rtc::TimeMillis(); int64_t time_to_wait_ms = next_callback_time - rtc::TimeMillis();
RTC_DCHECK_GT(time_to_wait_ms, 0) << "Invalid timer interval"; RTC_DCHECK_GT(time_to_wait_ms, 0) << "Invalid timer interval";

28
modules/audio_device/audio_device_buffer.h
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@ -11,8 +11,6 @@
#ifndef MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_ #ifndef MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_
#define MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_ #define MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_
#include <atomic>
#include "modules/audio_device/include/audio_device.h" #include "modules/audio_device/include/audio_device.h"
#include "rtc_base/buffer.h" #include "rtc_base/buffer.h"
#include "rtc_base/criticalsection.h" #include "rtc_base/criticalsection.h"
@ -85,8 +83,8 @@ class AudioDeviceBuffer {
int32_t SetRecordingSampleRate(uint32_t fsHz); int32_t SetRecordingSampleRate(uint32_t fsHz);
int32_t SetPlayoutSampleRate(uint32_t fsHz); int32_t SetPlayoutSampleRate(uint32_t fsHz);
uint32_t RecordingSampleRate() const; int32_t RecordingSampleRate() const;
uint32_t PlayoutSampleRate() const; int32_t PlayoutSampleRate() const;
int32_t SetRecordingChannels(size_t channels); int32_t SetRecordingChannels(size_t channels);
int32_t SetPlayoutChannels(size_t channels); int32_t SetPlayoutChannels(size_t channels);
@ -138,7 +136,7 @@ class AudioDeviceBuffer {
// called on that same thread. When audio has started some methods will be // called on that same thread. When audio has started some methods will be
// called on either a native audio thread for playout or a native thread for // called on either a native audio thread for playout or a native thread for
// recording. Some members are not annotated since they are "protected by // recording. Some members are not annotated since they are "protected by
// design" and adding e.g. a race checker can cause failures for very few // design" and adding e.g. a race checker can cause failuries for very few
// edge cases and it is IMHO not worth the risk to use them in this class. // edge cases and it is IMHO not worth the risk to use them in this class.
// TODO(henrika): see if it is possible to refactor and annotate all members. // TODO(henrika): see if it is possible to refactor and annotate all members.
@ -162,17 +160,23 @@ class AudioDeviceBuffer {
// and it must outlive this object. It is not possible to change this member // and it must outlive this object. It is not possible to change this member
// while any media is active. It is possible to start media without calling // while any media is active. It is possible to start media without calling
// RegisterAudioCallback() but that will lead to ignored audio callbacks in // RegisterAudioCallback() but that will lead to ignored audio callbacks in
// both directions where native audio will be active but no audio samples will // both directions where native audio will be acive but no audio samples will
// be transported. // be transported.
AudioTransport* audio_transport_cb_; AudioTransport* audio_transport_cb_;
// Sample rate in Hertz. Accessed atomically. // The members below that are not annotated are protected by design. They are
std::atomic<uint32_t> rec_sample_rate_; // all set on the main thread (verified by |main_thread_checker_|) and then
std::atomic<uint32_t> play_sample_rate_; // read on either the playout or recording audio thread. But, media will never
// be active when the member is set; hence no conflict exists. It is too
// complex to ensure and verify that this is actually the case.
// Number of audio channels. Accessed atomically. // Sample rate in Hertz.
std::atomic<size_t> rec_channels_; uint32_t rec_sample_rate_;
std::atomic<size_t> play_channels_; uint32_t play_sample_rate_;
// Number of audio channels.
size_t rec_channels_;
size_t play_channels_;
// Keeps track of if playout/recording are active or not. A combination // Keeps track of if playout/recording are active or not. A combination
// of these states are used to determine when to start and stop the timer. // of these states are used to determine when to start and stop the timer.