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Remove the reporting of histogram data for AEC2

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This CL removes the legacy reporting of histogram data for AEC2.

Bug: webrtc:5298
Change-Id: I838e729e0fb78d28e16de0fa79ddf5c857682d65
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135101
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27834}
This commit is contained in:
Per Åhgren
2019-05-03 09:00:08 +02:00
committed by Commit Bot
parent 4731f0062e
commit ea4c5df366
3 changed files with 4 additions and 85 deletions
Download Patch File Download Diff File Expand all files Collapse all files

82
modules/audio_processing/audio_processing_impl.cc
View File

@ -1253,8 +1253,6 @@ int AudioProcessingImpl::ProcessCaptureStreamLocked() {
!!private_submodules_->echo_control_mobile,
1);
MaybeUpdateHistograms();
AudioBuffer* capture_buffer = capture_.capture_audio.get(); // For brevity.
if (private_submodules_->pre_amplifier) {
@ -1951,79 +1949,7 @@ void AudioProcessingImpl::InitializePreProcessor() {
}
}
void AudioProcessingImpl::MaybeUpdateHistograms() {
static const int kMinDiffDelayMs = 60;
if (private_submodules_->echo_cancellation &&
private_submodules_->echo_cancellation->is_enabled()) {
// Activate delay_jumps_ counters if we know echo_cancellation is running.
// If a stream has echo we know that the echo_cancellation is in process.
if (capture_.stream_delay_jumps == -1 &&
private_submodules_->echo_cancellation->stream_has_echo()) {
capture_.stream_delay_jumps = 0;
}
if (capture_.aec_system_delay_jumps == -1 &&
private_submodules_->echo_cancellation->stream_has_echo()) {
capture_.aec_system_delay_jumps = 0;
}
// Detect a jump in platform reported system delay and log the difference.
const int diff_stream_delay_ms =
capture_nonlocked_.stream_delay_ms - capture_.last_stream_delay_ms;
if (diff_stream_delay_ms > kMinDiffDelayMs &&
capture_.last_stream_delay_ms != 0) {
RTC_HISTOGRAM_COUNTS("WebRTC.Audio.PlatformReportedStreamDelayJump",
diff_stream_delay_ms, kMinDiffDelayMs, 1000, 100);
if (capture_.stream_delay_jumps == -1) {
capture_.stream_delay_jumps = 0; // Activate counter if needed.
}
capture_.stream_delay_jumps++;
}
capture_.last_stream_delay_ms = capture_nonlocked_.stream_delay_ms;
// Detect a jump in AEC system delay and log the difference.
const int samples_per_ms =
rtc::CheckedDivExact(capture_nonlocked_.split_rate, 1000);
RTC_DCHECK_LT(0, samples_per_ms);
const int aec_system_delay_ms =
private_submodules_->echo_cancellation->GetSystemDelayInSamples() /
samples_per_ms;
const int diff_aec_system_delay_ms =
aec_system_delay_ms - capture_.last_aec_system_delay_ms;
if (diff_aec_system_delay_ms > kMinDiffDelayMs &&
capture_.last_aec_system_delay_ms != 0) {
RTC_HISTOGRAM_COUNTS("WebRTC.Audio.AecSystemDelayJump",
diff_aec_system_delay_ms, kMinDiffDelayMs, 1000,
100);
if (capture_.aec_system_delay_jumps == -1) {
capture_.aec_system_delay_jumps = 0; // Activate counter if needed.
}
capture_.aec_system_delay_jumps++;
}
capture_.last_aec_system_delay_ms = aec_system_delay_ms;
}
}
void AudioProcessingImpl::UpdateHistogramsOnCallEnd() {
// Run in a single-threaded manner.
rtc::CritScope cs_render(&crit_render_);
rtc::CritScope cs_capture(&crit_capture_);
if (capture_.stream_delay_jumps > -1) {
RTC_HISTOGRAM_ENUMERATION(
"WebRTC.Audio.NumOfPlatformReportedStreamDelayJumps",
capture_.stream_delay_jumps, 51);
}
capture_.stream_delay_jumps = -1;
capture_.last_stream_delay_ms = 0;
if (capture_.aec_system_delay_jumps > -1) {
RTC_HISTOGRAM_ENUMERATION("WebRTC.Audio.NumOfAecSystemDelayJumps",
capture_.aec_system_delay_jumps, 51);
}
capture_.aec_system_delay_jumps = -1;
capture_.last_aec_system_delay_ms = 0;
}
void AudioProcessingImpl::UpdateHistogramsOnCallEnd() {}
void AudioProcessingImpl::WriteAecDumpConfigMessage(bool forced) {
if (!aec_dump_) {
@ -2160,12 +2086,8 @@ void AudioProcessingImpl::RecordAudioProcessingState() {
AudioProcessingImpl::ApmCaptureState::ApmCaptureState(
bool transient_suppressor_enabled)
: aec_system_delay_jumps(-1),
delay_offset_ms(0),
: delay_offset_ms(0),
was_stream_delay_set(false),
last_stream_delay_ms(0),
last_aec_system_delay_ms(0),
stream_delay_jumps(-1),
output_will_be_muted(false),
key_pressed(false),
transient_suppressor_enabled(transient_suppressor_enabled),

5
modules/audio_processing/audio_processing_impl.h
View File

@ -277,7 +277,6 @@ class AudioProcessingImpl : public AudioProcessing {
// Capture-side exclusive methods possibly running APM in a multi-threaded
// manner that are called with the render lock already acquired.
int ProcessCaptureStreamLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
void MaybeUpdateHistograms() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
// Render-side exclusive methods possibly running APM in a multi-threaded
// manner that are called with the render lock already acquired.
@ -385,12 +384,8 @@ class AudioProcessingImpl : public AudioProcessing {
struct ApmCaptureState {
ApmCaptureState(bool transient_suppressor_enabled);
~ApmCaptureState();
int aec_system_delay_jumps;
int delay_offset_ms;
bool was_stream_delay_set;
int last_stream_delay_ms;
int last_aec_system_delay_ms;
int stream_delay_jumps;
bool output_will_be_muted;
bool key_pressed;
bool transient_suppressor_enabled;

2
modules/audio_processing/include/audio_processing.h
View File

@ -630,6 +630,8 @@ class AudioProcessing : public rtc::RefCountInterface {
// Use to send UMA histograms at end of a call. Note that all histogram
// specific member variables are reset.
// Deprecated. This method is deprecated and will be removed.
// TODO(peah): Remove this method.
virtual void UpdateHistogramsOnCallEnd() = 0;
// Get audio processing statistics. The |has_remote_tracks| argument should be