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Revert "Reland "Remove unused APM voice activity detection sub-module""

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This reverts commit 54d1344d985b00d4d1580dd18057d4618c11ad1f.

Reason for revert: Breaks chromium roll, see 
https://ci.chromium.org/ui/p/chromium/builders/try/linux_chromium_tsan_rel_ng/1080583/overview

https://chromium-review.googlesource.com/c/chromium/src/+/3461512

Original change's description:
> Reland "Remove unused APM voice activity detection sub-module"
>
> This reverts commit a751f167c68343f76528436defdbc61600a8d7b3.
>
> Reason for revert: dependency in a downstream project removed
>
> Original change's description:
> > Revert "Remove unused APM voice activity detection sub-module"
> >
> > This reverts commit b4e06d032e6f82a65c52ed0c5364ae9e7c0a0215.
> >
> > Reason for revert: breaking downstream projects
> >
> > Original change's description:
> > > Remove unused APM voice activity detection sub-module
> > >
> > > API changes:
> > > - webrtc::AudioProcessing::Config::VoiceDetection removed
> > > - webrtc::AudioProcessingStats::voice_detected deprecated
> > > - cricket::AudioOptions::typing_detection deprecated
> > > - webrtc::StatsReport::StatsValueName::
> > >   kStatsValueNameTypingNoiseState deprecated
> > >
> > > PSA: https://groups.google.com/g/discuss-webrtc/c/7X6uwmJarE0
> > >
> > > Bug: webrtc:11226,webrtc:11292
> > > Change-Id: I8d008b56708cf62961b9857ec052b59fda3b41bf
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250666
> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > > Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> > > Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> > > Reviewed-by: Björn Terelius <terelius@webrtc.org>
> > > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> > > Cr-Commit-Position: refs/heads/main@{#35975}
> >
> > TBR=gustaf@webrtc.org,saza@webrtc.org,alessiob@webrtc.org,terelius@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
> >
> > Change-Id: Iee01fdb874b4e0331277f3ffe60dacaabc3859a2
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: webrtc:11226,webrtc:11292
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251600
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#35977}
>
> # Not skipping CQ checks because this is a reland.
>
> Bug: webrtc:11226,webrtc:11292
> Change-Id: I2fcbc5fdade16bfe6a0f0a02841a33a598d4f2ad
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251660
> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35984}

TBR=mbonadei@webrtc.org,gustaf@webrtc.org,saza@webrtc.org,alessiob@webrtc.org,terelius@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com

Change-Id: Ib308a3af2dcce85a0074ef5a4680ccec3f82712f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11226,webrtc:11292
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251688
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35990}
This commit is contained in:
Henrik Boström
2022-02-14 12:02:45 +00:00
committed by WebRTC LUCI CQ
parent eb6c6fcf27
commit 09aaf6f7bc
23 changed files with 483 additions and 18 deletions
Download Patch File Download Diff File Expand all files Collapse all files

92
modules/audio_processing/audio_processing_unittest.cc
View File

@ -190,6 +190,7 @@ void EnableAllAPComponents(AudioProcessing* ap) {
apm_config.noise_suppression.enabled = true;
apm_config.high_pass_filter.enabled = true;
apm_config.voice_detection.enabled = true;
apm_config.pipeline.maximum_internal_processing_rate = 48000;
ap->ApplyConfig(apm_config);
}
@ -1225,6 +1226,7 @@ TEST_F(ApmTest, AllProcessingDisabledByDefault) {
EXPECT_FALSE(config.high_pass_filter.enabled);
EXPECT_FALSE(config.gain_controller1.enabled);
EXPECT_FALSE(config.noise_suppression.enabled);
EXPECT_FALSE(config.voice_detection.enabled);
}
TEST_F(ApmTest, NoProcessingWhenAllComponentsDisabled) {
@ -1365,6 +1367,48 @@ TEST_F(ApmTest, SplittingFilter) {
EXPECT_TRUE(FrameDataAreEqual(frame_, frame_copy));
apm_->ApplyConfig(apm_config);
// 3. Only GetStatistics-reporting VAD is enabled...
SetFrameTo(&frame_, 1000);
frame_copy.CopyFrom(frame_);
apm_config.voice_detection.enabled = true;
apm_->ApplyConfig(apm_config);
EXPECT_EQ(apm_->kNoError,
apm_->ProcessStream(
frame_.data.data(),
StreamConfig(frame_.sample_rate_hz, frame_.num_channels),
StreamConfig(frame_.sample_rate_hz, frame_.num_channels),
frame_.data.data()));
EXPECT_EQ(apm_->kNoError,
apm_->ProcessStream(
frame_.data.data(),
StreamConfig(frame_.sample_rate_hz, frame_.num_channels),
StreamConfig(frame_.sample_rate_hz, frame_.num_channels),
frame_.data.data()));
EXPECT_TRUE(FrameDataAreEqual(frame_, frame_copy));
apm_config.voice_detection.enabled = false;
apm_->ApplyConfig(apm_config);
// 4. The VAD is enabled...
SetFrameTo(&frame_, 1000);
frame_copy.CopyFrom(frame_);
apm_config.voice_detection.enabled = true;
apm_->ApplyConfig(apm_config);
EXPECT_EQ(apm_->kNoError,
apm_->ProcessStream(
frame_.data.data(),
StreamConfig(frame_.sample_rate_hz, frame_.num_channels),
StreamConfig(frame_.sample_rate_hz, frame_.num_channels),
frame_.data.data()));
EXPECT_EQ(apm_->kNoError,
apm_->ProcessStream(
frame_.data.data(),
StreamConfig(frame_.sample_rate_hz, frame_.num_channels),
StreamConfig(frame_.sample_rate_hz, frame_.num_channels),
frame_.data.data()));
EXPECT_TRUE(FrameDataAreEqual(frame_, frame_copy));
apm_config.voice_detection.enabled = false;
apm_->ApplyConfig(apm_config);
// Check the test is valid. We should have distortion from the filter
// when AEC is enabled (which won't affect the audio).
apm_config.echo_canceller.enabled = true;
@ -1692,6 +1736,7 @@ TEST_F(ApmTest, Process) {
static_cast<size_t>(test->num_reverse_channels()), true);
int frame_count = 0;
int has_voice_count = 0;
int analog_level = 127;
int analog_level_average = 0;
int max_output_average = 0;
@ -1727,6 +1772,8 @@ TEST_F(ApmTest, Process) {
analog_level = apm_->recommended_stream_analog_level();
analog_level_average += analog_level;
AudioProcessingStats stats = apm_->GetStatistics();
EXPECT_TRUE(stats.voice_detected);
has_voice_count += *stats.voice_detected ? 1 : 0;
size_t frame_size = frame_.samples_per_channel * frame_.num_channels;
size_t write_count =
@ -1782,23 +1829,33 @@ TEST_F(ApmTest, Process) {
if (!absl::GetFlag(FLAGS_write_apm_ref_data)) {
const int kIntNear = 1;
// All numbers being consistently higher on N7 compare to the reference
// data.
// When running the test on a N7 we get a {2, 6} difference of
// `has_voice_count` and `max_output_average` is up to 18 higher.
// All numbers being consistently higher on N7 compare to ref_data.
// TODO(bjornv): If we start getting more of these offsets on Android we
// should consider a different approach. Either using one slack for all,
// or generate a separate android reference.
#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
const int kHasVoiceCountOffset = 3;
const int kHasVoiceCountNear = 8;
const int kMaxOutputAverageOffset = 9;
const int kMaxOutputAverageNear = 26;
#else
const int kHasVoiceCountOffset = 0;
const int kHasVoiceCountNear = kIntNear;
const int kMaxOutputAverageOffset = 0;
const int kMaxOutputAverageNear = kIntNear;
#endif
EXPECT_NEAR(test->has_voice_count(),
has_voice_count - kHasVoiceCountOffset, kHasVoiceCountNear);
EXPECT_NEAR(test->analog_level_average(), analog_level_average, kIntNear);
EXPECT_NEAR(test->max_output_average(),
max_output_average - kMaxOutputAverageOffset,
kMaxOutputAverageNear);
} else {
test->set_has_voice_count(has_voice_count);
test->set_analog_level_average(analog_level_average);
test->set_max_output_average(max_output_average);
}
@ -2628,6 +2685,7 @@ rtc::scoped_refptr<AudioProcessing> CreateApm(bool mobile_aec) {
apm_config.echo_canceller.enabled = true;
apm_config.echo_canceller.mobile_mode = mobile_aec;
apm_config.noise_suppression.enabled = false;
apm_config.voice_detection.enabled = false;
apm->ApplyConfig(apm_config);
return apm;
}
@ -2736,9 +2794,10 @@ TEST(MAYBE_ApmStatistics, AECMEnabledTest) {
EXPECT_FALSE(stats.echo_return_loss_enhancement.has_value());
}
TEST(ApmStatistics, DoNotReportVoiceDetectedStat) {
TEST(ApmStatistics, ReportHasVoice) {
ProcessingConfig processing_config = {
{{32000, 1}, {32000, 1}, {32000, 1}, {32000, 1}}};
AudioProcessing::Config config;
// Set up an audioframe.
Int16FrameData frame;
@ -2755,14 +2814,37 @@ TEST(ApmStatistics, DoNotReportVoiceDetectedStat) {
AudioProcessingBuilderForTesting().Create();
apm->Initialize(processing_config);
// No metric should be reported.
// If not enabled, no metric should be reported.
EXPECT_EQ(
apm->ProcessStream(frame.data.data(),
StreamConfig(frame.sample_rate_hz, frame.num_channels),
StreamConfig(frame.sample_rate_hz, frame.num_channels),
frame.data.data()),
0);
EXPECT_FALSE(apm->GetStatistics().voice_detected.has_value());
EXPECT_FALSE(apm->GetStatistics().voice_detected);
// If enabled, metrics should be reported.
config.voice_detection.enabled = true;
apm->ApplyConfig(config);
EXPECT_EQ(
apm->ProcessStream(frame.data.data(),
StreamConfig(frame.sample_rate_hz, frame.num_channels),
StreamConfig(frame.sample_rate_hz, frame.num_channels),
frame.data.data()),
0);
auto stats = apm->GetStatistics();
EXPECT_TRUE(stats.voice_detected);
// If re-disabled, the value is again not reported.
config.voice_detection.enabled = false;
apm->ApplyConfig(config);
EXPECT_EQ(
apm->ProcessStream(frame.data.data(),
StreamConfig(frame.sample_rate_hz, frame.num_channels),
StreamConfig(frame.sample_rate_hz, frame.num_channels),
frame.data.data()),
0);
EXPECT_FALSE(apm->GetStatistics().voice_detected);
}
TEST(ApmStatistics, GetStatisticsReportsNoEchoDetectorStatsWhenDisabled) {