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Reland "Remove unused APM voice activity detection sub-module"
This reverts commita751f167c6. Reason for revert: dependency in a downstream project removed Original change's description: > Revert "Remove unused APM voice activity detection sub-module" > > This reverts commitb4e06d032e. > > 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}
This commit is contained in:
Alessio Bazzica
WebRTC LUCI CQ
parent
f342d6054a
commit
54d1344d98
23 changed files with 18 additions and 483 deletions
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@ -60,6 +60,8 @@ struct RTC_EXPORT AudioOptions {
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absl::optional<int> audio_jitter_buffer_min_delay_ms;
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// Audio receiver jitter buffer (NetEq) should handle retransmitted packets.
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absl::optional<bool> audio_jitter_buffer_enable_rtx_handling;
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// Deprecated.
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// TODO(bugs.webrtc.org/11226): Remove.
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// Audio processing to detect typing.
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absl::optional<bool> typing_detection;
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absl::optional<bool> experimental_agc;
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@ -648,6 +648,7 @@ const char* StatsReport::Value::display_name() const {
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return "googTrackId";
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case kStatsValueNameTimingFrameInfo:
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return "googTimingFrameInfo";
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// TODO(bugs.webrtc.org/11226): Remove.
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case kStatsValueNameTypingNoiseState:
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return "googTypingNoiseState";
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case kStatsValueNameWritable:
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@ -235,6 +235,7 @@ class RTC_EXPORT StatsReport {
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kStatsValueNameTrackId,
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kStatsValueNameTransmitBitrate,
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kStatsValueNameTransportType,
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// TODO(bugs.webrtc.org/11226): Remove.
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kStatsValueNameTypingNoiseState,
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kStatsValueNameWritable,
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kStatsValueNameAudioDeviceUnderrunCounter,
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@ -165,24 +165,6 @@ int32_t AudioTransportImpl::RecordedDataIsAvailable(
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audio_frame.get());
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audio_frame->set_absolute_capture_timestamp_ms(estimated_capture_time_ns /
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1000000);
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// Typing detection (utilizes the APM/VAD decision). We let the VAD determine
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// if we're using this feature or not.
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// TODO(solenberg): GetConfig() takes a lock. Work around that.
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bool typing_detected = false;
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if (audio_processing_ &&
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audio_processing_->GetConfig().voice_detection.enabled) {
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if (audio_frame->vad_activity_ != AudioFrame::kVadUnknown) {
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bool vad_active = audio_frame->vad_activity_ == AudioFrame::kVadActive;
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typing_detected = typing_detection_.Process(key_pressed, vad_active);
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}
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}
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// Copy frame and push to each sending stream. The copy is required since an
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// encoding task will be posted internally to each stream.
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{
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MutexLock lock(&capture_lock_);
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typing_noise_detected_ = typing_detected;
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}
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RTC_DCHECK_GT(audio_frame->samples_per_channel_, 0);
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if (async_audio_processing_)
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@ -290,8 +272,4 @@ void AudioTransportImpl::SetStereoChannelSwapping(bool enable) {
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swap_stereo_channels_ = enable;
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}
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bool AudioTransportImpl::typing_noise_detected() const {
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MutexLock lock(&capture_lock_);
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return typing_noise_detected_;
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}
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} // namespace webrtc
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@ -86,7 +86,9 @@ class AudioTransportImpl : public AudioTransport {
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int send_sample_rate_hz,
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size_t send_num_channels);
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void SetStereoChannelSwapping(bool enable);
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bool typing_noise_detected() const;
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// Deprecated.
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// TODO(bugs.webrtc.org/11226): Remove.
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bool typing_noise_detected() const { return false; }
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private:
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void SendProcessedData(std::unique_ptr<AudioFrame> audio_frame);
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@ -103,7 +105,6 @@ class AudioTransportImpl : public AudioTransport {
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std::vector<AudioSender*> audio_senders_ RTC_GUARDED_BY(capture_lock_);
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int send_sample_rate_hz_ RTC_GUARDED_BY(capture_lock_) = 8000;
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size_t send_num_channels_ RTC_GUARDED_BY(capture_lock_) = 1;
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bool typing_noise_detected_ RTC_GUARDED_BY(capture_lock_) = false;
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bool swap_stereo_channels_ RTC_GUARDED_BY(capture_lock_) = false;
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PushResampler<int16_t> capture_resampler_;
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TypingDetection typing_detection_;
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@ -634,9 +634,7 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
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}
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if (options.typing_detection) {
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RTC_LOG(LS_INFO) << "Typing detection is enabled? "
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<< *options.typing_detection;
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apm_config.voice_detection.enabled = *options.typing_detection;
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RTC_LOG(LS_WARNING) << "Typing detection is requested, but unsupported.";
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}
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ap->ApplyConfig(apm_config);
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@ -221,11 +221,6 @@ class WebRtcVoiceEngineTestFake : public ::testing::TestWithParam<bool> {
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// Default Options.
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VerifyEchoCancellationSettings(/*enabled=*/true);
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EXPECT_TRUE(IsHighPassFilterEnabled());
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#if defined(WEBRTC_ANDROID)
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EXPECT_FALSE(IsTypingDetectionEnabled());
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#else
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EXPECT_TRUE(IsTypingDetectionEnabled());
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#endif
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EXPECT_TRUE(apm_config_.noise_suppression.enabled);
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EXPECT_EQ(apm_config_.noise_suppression.level, kDefaultNsLevel);
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VerifyGainControlEnabledCorrectly();
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@ -793,10 +788,6 @@ class WebRtcVoiceEngineTestFake : public ::testing::TestWithParam<bool> {
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return apm_config_.high_pass_filter.enabled;
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}
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bool IsTypingDetectionEnabled() {
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return apm_config_.voice_detection.enabled;
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}
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protected:
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const bool use_null_apm_;
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std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory_;
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@ -3041,40 +3032,10 @@ TEST_P(WebRtcVoiceEngineTestFake, SetAudioOptions) {
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if (!use_null_apm_) {
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VerifyEchoCancellationSettings(/*enabled=*/true);
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EXPECT_TRUE(IsHighPassFilterEnabled());
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#if defined(WEBRTC_ANDROID)
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EXPECT_FALSE(IsTypingDetectionEnabled());
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#else
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EXPECT_TRUE(IsTypingDetectionEnabled());
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#endif
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}
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EXPECT_EQ(200u, GetRecvStreamConfig(kSsrcY).jitter_buffer_max_packets);
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EXPECT_FALSE(GetRecvStreamConfig(kSsrcY).jitter_buffer_fast_accelerate);
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// Turn typing detection off.
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send_parameters_.options.typing_detection = false;
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SetSendParameters(send_parameters_);
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if (!use_null_apm_) {
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EXPECT_FALSE(IsTypingDetectionEnabled());
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}
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// Leave typing detection unchanged, but non-default.
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send_parameters_.options.typing_detection = absl::nullopt;
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SetSendParameters(send_parameters_);
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if (!use_null_apm_) {
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EXPECT_FALSE(IsTypingDetectionEnabled());
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}
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// Turn typing detection on.
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send_parameters_.options.typing_detection = true;
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SetSendParameters(send_parameters_);
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if (!use_null_apm_) {
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#if defined(WEBRTC_ANDROID)
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EXPECT_FALSE(IsTypingDetectionEnabled());
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#else
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EXPECT_TRUE(IsTypingDetectionEnabled());
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#endif
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}
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// Turn echo cancellation off
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send_parameters_.options.echo_cancellation = false;
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SetSendParameters(send_parameters_);
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@ -168,7 +168,6 @@ rtc_library("audio_processing") {
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":high_pass_filter",
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":optionally_built_submodule_creators",
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":rms_level",
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":voice_detection",
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"../../api:array_view",
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"../../api:function_view",
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"../../api/audio:aec3_config",
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@ -218,20 +217,6 @@ rtc_library("audio_processing") {
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}
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}
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rtc_library("voice_detection") {
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sources = [
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"voice_detection.cc",
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"voice_detection.h",
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]
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deps = [
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":api",
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":audio_buffer",
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"../../api/audio:audio_frame_api",
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"../../common_audio:common_audio_c",
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"../../rtc_base:checks",
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]
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}
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rtc_library("residual_echo_detector") {
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poisonous = [ "default_echo_detector" ]
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configs += [ ":apm_debug_dump" ]
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@ -379,7 +364,6 @@ if (rtc_include_tests) {
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":gain_controller2",
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":high_pass_filter",
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":mocks",
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":voice_detection",
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"../../api:array_view",
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"../../api:scoped_refptr",
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"../../api/audio:aec3_config",
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@ -474,7 +458,6 @@ if (rtc_include_tests) {
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"test/echo_canceller_test_tools_unittest.cc",
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"test/echo_control_mock.h",
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"test/test_utils.h",
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"voice_detection_unittest.cc",
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]
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}
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}
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@ -141,7 +141,6 @@ bool AudioProcessingImpl::SubmoduleStates::Update(
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bool gain_controller2_enabled,
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bool gain_adjustment_enabled,
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bool echo_controller_enabled,
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bool voice_detector_enabled,
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bool transient_suppressor_enabled) {
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bool changed = false;
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changed |= (high_pass_filter_enabled != high_pass_filter_enabled_);
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@ -153,7 +152,6 @@ bool AudioProcessingImpl::SubmoduleStates::Update(
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changed |= (gain_controller2_enabled != gain_controller2_enabled_);
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changed |= (gain_adjustment_enabled != gain_adjustment_enabled_);
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changed |= (echo_controller_enabled != echo_controller_enabled_);
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changed |= (voice_detector_enabled != voice_detector_enabled_);
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changed |= (transient_suppressor_enabled != transient_suppressor_enabled_);
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if (changed) {
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high_pass_filter_enabled_ = high_pass_filter_enabled;
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@ -163,7 +161,6 @@ bool AudioProcessingImpl::SubmoduleStates::Update(
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gain_controller2_enabled_ = gain_controller2_enabled;
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gain_adjustment_enabled_ = gain_adjustment_enabled;
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echo_controller_enabled_ = echo_controller_enabled;
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voice_detector_enabled_ = voice_detector_enabled;
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transient_suppressor_enabled_ = transient_suppressor_enabled;
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}
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@ -174,7 +171,7 @@ bool AudioProcessingImpl::SubmoduleStates::Update(
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bool AudioProcessingImpl::SubmoduleStates::CaptureMultiBandSubModulesActive()
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const {
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return CaptureMultiBandProcessingPresent() || voice_detector_enabled_;
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return CaptureMultiBandProcessingPresent();
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}
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bool AudioProcessingImpl::SubmoduleStates::CaptureMultiBandProcessingPresent()
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@ -371,7 +368,6 @@ void AudioProcessingImpl::InitializeLocked() {
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InitializeGainController1();
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InitializeTransientSuppressor();
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InitializeHighPassFilter(true);
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InitializeVoiceDetector();
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InitializeResidualEchoDetector();
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InitializeEchoController();
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InitializeGainController2(/*config_has_changed=*/true);
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@ -506,9 +502,6 @@ void AudioProcessingImpl::ApplyConfig(const AudioProcessing::Config& config) {
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const bool agc2_config_changed =
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config_.gain_controller2 != config.gain_controller2;
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const bool voice_detection_config_changed =
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config_.voice_detection.enabled != config.voice_detection.enabled;
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const bool ns_config_changed =
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config_.noise_suppression.enabled != config.noise_suppression.enabled ||
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config_.noise_suppression.level != config.noise_suppression.level;
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@ -557,10 +550,6 @@ void AudioProcessingImpl::ApplyConfig(const AudioProcessing::Config& config) {
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InitializeCaptureLevelsAdjuster();
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}
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if (voice_detection_config_changed) {
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InitializeVoiceDetector();
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}
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// Reinitialization must happen after all submodule configuration to avoid
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// additional reinitializations on the next capture / render processing call.
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if (pipeline_config_changed) {
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@ -1215,13 +1204,6 @@ int AudioProcessingImpl::ProcessCaptureStreamLocked() {
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}
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}
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if (config_.voice_detection.enabled) {
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capture_.stats.voice_detected =
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submodules_.voice_detector->ProcessCaptureAudio(capture_buffer);
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} else {
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capture_.stats.voice_detected = absl::nullopt;
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}
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if (submodules_.agc_manager) {
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submodules_.agc_manager->Process(capture_buffer);
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@ -1682,7 +1664,7 @@ bool AudioProcessingImpl::UpdateActiveSubmoduleStates() {
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!!submodules_.gain_controller2,
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config_.pre_amplifier.enabled || config_.capture_level_adjustment.enabled,
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capture_nonlocked_.echo_controller_enabled,
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config_.voice_detection.enabled, !!submodules_.transient_suppressor);
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!!submodules_.transient_suppressor);
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}
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void AudioProcessingImpl::InitializeTransientSuppressor() {
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@ -1732,14 +1714,6 @@ void AudioProcessingImpl::InitializeHighPassFilter(bool forced_reset) {
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}
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}
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void AudioProcessingImpl::InitializeVoiceDetector() {
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if (config_.voice_detection.enabled) {
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submodules_.voice_detector = std::make_unique<VoiceDetection>(
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proc_split_sample_rate_hz(), VoiceDetection::kVeryLowLikelihood);
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} else {
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submodules_.voice_detector.reset();
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}
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}
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void AudioProcessingImpl::InitializeEchoController() {
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bool use_echo_controller =
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echo_control_factory_ ||
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@ -39,7 +39,6 @@
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#include "modules/audio_processing/render_queue_item_verifier.h"
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#include "modules/audio_processing/rms_level.h"
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#include "modules/audio_processing/transient/transient_suppressor.h"
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#include "modules/audio_processing/voice_detection.h"
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#include "rtc_base/gtest_prod_util.h"
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#include "rtc_base/ignore_wundef.h"
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#include "rtc_base/swap_queue.h"
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@ -208,7 +207,6 @@ class AudioProcessingImpl : public AudioProcessing {
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bool gain_controller2_enabled,
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bool gain_adjustment_enabled,
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bool echo_controller_enabled,
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bool voice_detector_enabled,
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bool transient_suppressor_enabled);
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bool CaptureMultiBandSubModulesActive() const;
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bool CaptureMultiBandProcessingPresent() const;
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@ -231,7 +229,6 @@ class AudioProcessingImpl : public AudioProcessing {
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bool gain_controller2_enabled_ = false;
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bool gain_adjustment_enabled_ = false;
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bool echo_controller_enabled_ = false;
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bool voice_detector_enabled_ = false;
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bool transient_suppressor_enabled_ = false;
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bool first_update_ = true;
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};
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@ -267,7 +264,6 @@ class AudioProcessingImpl : public AudioProcessing {
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// already acquired.
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void InitializeHighPassFilter(bool forced_reset)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
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void InitializeVoiceDetector() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
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void InitializeGainController1() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
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void InitializeTransientSuppressor()
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RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
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@ -400,7 +396,6 @@ class AudioProcessingImpl : public AudioProcessing {
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std::unique_ptr<EchoControlMobileImpl> echo_control_mobile;
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std::unique_ptr<NoiseSuppressor> noise_suppressor;
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std::unique_ptr<TransientSuppressor> transient_suppressor;
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std::unique_ptr<VoiceDetection> voice_detector;
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std::unique_ptr<CaptureLevelsAdjuster> capture_levels_adjuster;
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} submodules_;
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@ -483,7 +483,6 @@ AudioProcessing::Config GetApmTestConfig(AecType aec_type) {
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apm_config.gain_controller1.mode =
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AudioProcessing::Config::GainController1::kAdaptiveDigital;
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apm_config.noise_suppression.enabled = true;
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apm_config.voice_detection.enabled = true;
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return apm_config;
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}
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@ -441,7 +441,6 @@ class CallSimulator : public ::testing::TestWithParam<SimulationConfig> {
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apm_config.gain_controller1.enabled = true;
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apm_config.gain_controller1.mode =
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AudioProcessing::Config::GainController1::kAdaptiveDigital;
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apm_config.voice_detection.enabled = true;
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apm->ApplyConfig(apm_config);
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};
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|
|
@ -453,7 +452,6 @@ class CallSimulator : public ::testing::TestWithParam<SimulationConfig> {
|
|||
apm_config.noise_suppression.enabled = true;
|
||||
apm_config.gain_controller1.mode =
|
||||
AudioProcessing::Config::GainController1::kAdaptiveDigital;
|
||||
apm_config.voice_detection.enabled = true;
|
||||
apm->ApplyConfig(apm_config);
|
||||
};
|
||||
|
||||
|
|
@ -464,7 +462,6 @@ class CallSimulator : public ::testing::TestWithParam<SimulationConfig> {
|
|||
apm_config.echo_canceller.enabled = false;
|
||||
apm_config.gain_controller1.enabled = false;
|
||||
apm_config.noise_suppression.enabled = false;
|
||||
apm_config.voice_detection.enabled = false;
|
||||
apm->ApplyConfig(apm_config);
|
||||
};
|
||||
|
||||
|
|
|
|||
|
|
@ -190,7 +190,6 @@ 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);
|
||||
}
|
||||
|
|
@ -1226,7 +1225,6 @@ 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) {
|
||||
|
|
@ -1367,48 +1365,6 @@ 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;
|
||||
|
|
@ -1736,7 +1692,6 @@ 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;
|
||||
|
|
@ -1772,8 +1727,6 @@ 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 =
|
||||
|
|
@ -1829,33 +1782,23 @@ TEST_F(ApmTest, Process) {
|
|||
|
||||
if (!absl::GetFlag(FLAGS_write_apm_ref_data)) {
|
||||
const int kIntNear = 1;
|
||||
// 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.
|
||||
// All numbers being consistently higher on N7 compare to the reference
|
||||
// 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);
|
||||
}
|
||||
|
|
@ -2685,7 +2628,6 @@ 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;
|
||||
}
|
||||
|
|
@ -2794,10 +2736,9 @@ TEST(MAYBE_ApmStatistics, AECMEnabledTest) {
|
|||
EXPECT_FALSE(stats.echo_return_loss_enhancement.has_value());
|
||||
}
|
||||
|
||||
TEST(ApmStatistics, ReportHasVoice) {
|
||||
TEST(ApmStatistics, DoNotReportVoiceDetectedStat) {
|
||||
ProcessingConfig processing_config = {
|
||||
{{32000, 1}, {32000, 1}, {32000, 1}, {32000, 1}}};
|
||||
AudioProcessing::Config config;
|
||||
|
||||
// Set up an audioframe.
|
||||
Int16FrameData frame;
|
||||
|
|
@ -2814,37 +2755,14 @@ TEST(ApmStatistics, ReportHasVoice) {
|
|||
AudioProcessingBuilderForTesting().Create();
|
||||
apm->Initialize(processing_config);
|
||||
|
||||
// If not enabled, no metric should be reported.
|
||||
// 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);
|
||||
|
||||
// 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);
|
||||
EXPECT_FALSE(apm->GetStatistics().voice_detected.has_value());
|
||||
}
|
||||
|
||||
TEST(ApmStatistics, GetStatisticsReportsNoEchoDetectorStatsWhenDisabled) {
|
||||
|
|
|
|||
|
|
@ -145,7 +145,6 @@ std::string AudioProcessing::Config::ToString() const {
|
|||
<< NoiseSuppressionLevelToString(noise_suppression.level)
|
||||
<< " }, transient_suppression: { enabled: "
|
||||
<< transient_suppression.enabled
|
||||
<< " }, voice_detection: { enabled: " << voice_detection.enabled
|
||||
<< " }, gain_controller1: { enabled: " << gain_controller1.enabled
|
||||
<< ", mode: " << GainController1ModeToString(gain_controller1.mode)
|
||||
<< ", target_level_dbfs: " << gain_controller1.target_level_dbfs
|
||||
|
|
|
|||
|
|
@ -113,8 +113,6 @@ static constexpr int kClippedLevelMin = 70;
|
|||
//
|
||||
// config.high_pass_filter.enabled = true;
|
||||
//
|
||||
// config.voice_detection.enabled = true;
|
||||
//
|
||||
// apm->ApplyConfig(config)
|
||||
//
|
||||
// apm->noise_reduction()->set_level(kHighSuppression);
|
||||
|
|
@ -232,11 +230,6 @@ class RTC_EXPORT AudioProcessing : public rtc::RefCountInterface {
|
|||
bool enabled = false;
|
||||
} transient_suppression;
|
||||
|
||||
// Enables reporting of `voice_detected` in webrtc::AudioProcessingStats.
|
||||
struct VoiceDetection {
|
||||
bool enabled = false;
|
||||
} voice_detection;
|
||||
|
||||
// Enables automatic gain control (AGC) functionality.
|
||||
// The automatic gain control (AGC) component brings the signal to an
|
||||
// appropriate range. This is done by applying a digital gain directly and,
|
||||
|
|
|
|||
|
|
@ -24,6 +24,8 @@ struct RTC_EXPORT AudioProcessingStats {
|
|||
AudioProcessingStats(const AudioProcessingStats& other);
|
||||
~AudioProcessingStats();
|
||||
|
||||
// Deprecated.
|
||||
// TODO(bugs.webrtc.org/11226): Remove.
|
||||
// True if voice is detected in the last capture frame, after processing.
|
||||
// It is conservative in flagging audio as speech, with low likelihood of
|
||||
// incorrectly flagging a frame as voice.
|
||||
|
|
|
|||
|
|
@ -543,10 +543,6 @@ void AudioProcessingSimulator::ConfigureAudioProcessor() {
|
|||
apm_config.high_pass_filter.enabled = *settings_.use_hpf;
|
||||
}
|
||||
|
||||
if (settings_.use_vad) {
|
||||
apm_config.voice_detection.enabled = *settings_.use_vad;
|
||||
}
|
||||
|
||||
if (settings_.use_agc) {
|
||||
apm_config.gain_controller1.enabled = *settings_.use_agc;
|
||||
}
|
||||
|
|
|
|||
|
|
@ -105,7 +105,6 @@ struct SimulationSettings {
|
|||
absl::optional<bool> use_ns;
|
||||
absl::optional<int> use_ts;
|
||||
absl::optional<bool> use_analog_agc;
|
||||
absl::optional<bool> use_vad;
|
||||
absl::optional<bool> use_all;
|
||||
absl::optional<bool> analog_agc_disable_digital_adaptive;
|
||||
absl::optional<int> agc_mode;
|
||||
|
|
|
|||
|
|
@ -117,10 +117,6 @@ ABSL_FLAG(int,
|
|||
analog_agc,
|
||||
kParameterNotSpecifiedValue,
|
||||
"Activate (1) or deactivate (0) the analog AGC");
|
||||
ABSL_FLAG(int,
|
||||
vad,
|
||||
kParameterNotSpecifiedValue,
|
||||
"Activate (1) or deactivate (0) the voice activity detector");
|
||||
ABSL_FLAG(bool,
|
||||
all_default,
|
||||
false,
|
||||
|
|
@ -365,7 +361,6 @@ void SetSettingIfFlagSet(int32_t flag, absl::optional<bool>* parameter) {
|
|||
SimulationSettings CreateSettings() {
|
||||
SimulationSettings settings;
|
||||
if (absl::GetFlag(FLAGS_all_default)) {
|
||||
settings.use_vad = true;
|
||||
settings.use_ts = true;
|
||||
settings.use_analog_agc = true;
|
||||
settings.use_ns = true;
|
||||
|
|
@ -417,7 +412,6 @@ SimulationSettings CreateSettings() {
|
|||
SetSettingIfSpecified(absl::GetFlag(FLAGS_ts), &settings.use_ts);
|
||||
SetSettingIfFlagSet(absl::GetFlag(FLAGS_analog_agc),
|
||||
&settings.use_analog_agc);
|
||||
SetSettingIfFlagSet(absl::GetFlag(FLAGS_vad), &settings.use_vad);
|
||||
SetSettingIfFlagSet(absl::GetFlag(FLAGS_analog_agc_disable_digital_adaptive),
|
||||
&settings.analog_agc_disable_digital_adaptive);
|
||||
SetSettingIfSpecified(absl::GetFlag(FLAGS_agc_mode), &settings.agc_mode);
|
||||
|
|
|
|||
|
|
@ -1,92 +0,0 @@
|
|||
/*
|
||||
* Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
|
||||
*
|
||||
* Use of this source code is governed by a BSD-style license
|
||||
* that can be found in the LICENSE file in the root of the source
|
||||
* tree. An additional intellectual property rights grant can be found
|
||||
* in the file PATENTS. All contributing project authors may
|
||||
* be found in the AUTHORS file in the root of the source tree.
|
||||
*/
|
||||
|
||||
#include "modules/audio_processing/voice_detection.h"
|
||||
|
||||
#include "common_audio/vad/include/webrtc_vad.h"
|
||||
#include "modules/audio_processing/audio_buffer.h"
|
||||
#include "rtc_base/checks.h"
|
||||
|
||||
namespace webrtc {
|
||||
class VoiceDetection::Vad {
|
||||
public:
|
||||
Vad() {
|
||||
state_ = WebRtcVad_Create();
|
||||
RTC_CHECK(state_);
|
||||
int error = WebRtcVad_Init(state_);
|
||||
RTC_DCHECK_EQ(0, error);
|
||||
}
|
||||
~Vad() { WebRtcVad_Free(state_); }
|
||||
|
||||
Vad(Vad&) = delete;
|
||||
Vad& operator=(Vad&) = delete;
|
||||
|
||||
VadInst* state() { return state_; }
|
||||
|
||||
private:
|
||||
VadInst* state_ = nullptr;
|
||||
};
|
||||
|
||||
VoiceDetection::VoiceDetection(int sample_rate_hz, Likelihood likelihood)
|
||||
: sample_rate_hz_(sample_rate_hz),
|
||||
frame_size_samples_(static_cast<size_t>(sample_rate_hz_ / 100)),
|
||||
likelihood_(likelihood),
|
||||
vad_(new Vad()) {
|
||||
int mode = 2;
|
||||
switch (likelihood) {
|
||||
case VoiceDetection::kVeryLowLikelihood:
|
||||
mode = 3;
|
||||
break;
|
||||
case VoiceDetection::kLowLikelihood:
|
||||
mode = 2;
|
||||
break;
|
||||
case VoiceDetection::kModerateLikelihood:
|
||||
mode = 1;
|
||||
break;
|
||||
case VoiceDetection::kHighLikelihood:
|
||||
mode = 0;
|
||||
break;
|
||||
default:
|
||||
RTC_DCHECK_NOTREACHED();
|
||||
break;
|
||||
}
|
||||
int error = WebRtcVad_set_mode(vad_->state(), mode);
|
||||
RTC_DCHECK_EQ(0, error);
|
||||
}
|
||||
|
||||
VoiceDetection::~VoiceDetection() {}
|
||||
|
||||
bool VoiceDetection::ProcessCaptureAudio(AudioBuffer* audio) {
|
||||
RTC_DCHECK_GE(AudioBuffer::kMaxSplitFrameLength,
|
||||
audio->num_frames_per_band());
|
||||
std::array<int16_t, AudioBuffer::kMaxSplitFrameLength> mixed_low_pass_data;
|
||||
rtc::ArrayView<const int16_t> mixed_low_pass(mixed_low_pass_data.data(),
|
||||
audio->num_frames_per_band());
|
||||
if (audio->num_channels() == 1) {
|
||||
FloatS16ToS16(audio->split_bands_const(0)[kBand0To8kHz],
|
||||
audio->num_frames_per_band(), mixed_low_pass_data.data());
|
||||
} else {
|
||||
const int num_channels = static_cast<int>(audio->num_channels());
|
||||
for (size_t i = 0; i < audio->num_frames_per_band(); ++i) {
|
||||
int32_t value =
|
||||
FloatS16ToS16(audio->split_channels_const(kBand0To8kHz)[0][i]);
|
||||
for (int j = 1; j < num_channels; ++j) {
|
||||
value += FloatS16ToS16(audio->split_channels_const(kBand0To8kHz)[j][i]);
|
||||
}
|
||||
mixed_low_pass_data[i] = value / num_channels;
|
||||
}
|
||||
}
|
||||
|
||||
int vad_ret = WebRtcVad_Process(vad_->state(), sample_rate_hz_,
|
||||
mixed_low_pass.data(), frame_size_samples_);
|
||||
RTC_DCHECK(vad_ret == 0 || vad_ret == 1);
|
||||
return vad_ret == 0 ? false : true;
|
||||
}
|
||||
} // namespace webrtc
|
||||
|
|
@ -1,59 +0,0 @@
|
|||
/*
|
||||
* Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
|
||||
*
|
||||
* Use of this source code is governed by a BSD-style license
|
||||
* that can be found in the LICENSE file in the root of the source
|
||||
* tree. An additional intellectual property rights grant can be found
|
||||
* in the file PATENTS. All contributing project authors may
|
||||
* be found in the AUTHORS file in the root of the source tree.
|
||||
*/
|
||||
|
||||
#ifndef MODULES_AUDIO_PROCESSING_VOICE_DETECTION_H_
|
||||
#define MODULES_AUDIO_PROCESSING_VOICE_DETECTION_H_
|
||||
|
||||
#include <stddef.h>
|
||||
|
||||
#include <memory>
|
||||
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
||||
class AudioBuffer;
|
||||
|
||||
// The voice activity detection (VAD) component analyzes the stream to
|
||||
// determine if voice is present.
|
||||
class VoiceDetection {
|
||||
public:
|
||||
// Specifies the likelihood that a frame will be declared to contain voice.
|
||||
// A higher value makes it more likely that speech will not be clipped, at
|
||||
// the expense of more noise being detected as voice.
|
||||
enum Likelihood {
|
||||
kVeryLowLikelihood,
|
||||
kLowLikelihood,
|
||||
kModerateLikelihood,
|
||||
kHighLikelihood
|
||||
};
|
||||
|
||||
VoiceDetection(int sample_rate_hz, Likelihood likelihood);
|
||||
~VoiceDetection();
|
||||
|
||||
VoiceDetection(VoiceDetection&) = delete;
|
||||
VoiceDetection& operator=(VoiceDetection&) = delete;
|
||||
|
||||
// Returns true if voice is detected in the current frame.
|
||||
bool ProcessCaptureAudio(AudioBuffer* audio);
|
||||
|
||||
Likelihood likelihood() const { return likelihood_; }
|
||||
|
||||
private:
|
||||
class Vad;
|
||||
|
||||
int sample_rate_hz_;
|
||||
size_t frame_size_samples_;
|
||||
Likelihood likelihood_;
|
||||
std::unique_ptr<Vad> vad_;
|
||||
};
|
||||
} // namespace webrtc
|
||||
|
||||
#endif // MODULES_AUDIO_PROCESSING_VOICE_DETECTION_H_
|
||||
|
|
@ -1,104 +0,0 @@
|
|||
/*
|
||||
* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
|
||||
*
|
||||
* Use of this source code is governed by a BSD-style license
|
||||
* that can be found in the LICENSE file in the root of the source
|
||||
* tree. An additional intellectual property rights grant can be found
|
||||
* in the file PATENTS. All contributing project authors may
|
||||
* be found in the AUTHORS file in the root of the source tree.
|
||||
*/
|
||||
#include <vector>
|
||||
|
||||
#include "api/array_view.h"
|
||||
#include "modules/audio_processing/audio_buffer.h"
|
||||
#include "modules/audio_processing/test/audio_buffer_tools.h"
|
||||
#include "modules/audio_processing/test/bitexactness_tools.h"
|
||||
#include "modules/audio_processing/voice_detection.h"
|
||||
#include "test/gtest.h"
|
||||
|
||||
namespace webrtc {
|
||||
namespace {
|
||||
|
||||
const int kNumFramesToProcess = 1000;
|
||||
|
||||
// Process one frame of data and produce the output.
|
||||
bool ProcessOneFrame(int sample_rate_hz,
|
||||
AudioBuffer* audio_buffer,
|
||||
VoiceDetection* voice_detection) {
|
||||
if (sample_rate_hz > AudioProcessing::kSampleRate16kHz) {
|
||||
audio_buffer->SplitIntoFrequencyBands();
|
||||
}
|
||||
|
||||
return voice_detection->ProcessCaptureAudio(audio_buffer);
|
||||
}
|
||||
|
||||
// Processes a specified amount of frames, verifies the results and reports
|
||||
// any errors.
|
||||
void RunBitexactnessTest(int sample_rate_hz,
|
||||
size_t num_channels,
|
||||
bool stream_has_voice_reference) {
|
||||
int sample_rate_to_use = std::min(sample_rate_hz, 16000);
|
||||
VoiceDetection voice_detection(sample_rate_to_use,
|
||||
VoiceDetection::kLowLikelihood);
|
||||
|
||||
int samples_per_channel = rtc::CheckedDivExact(sample_rate_hz, 100);
|
||||
const StreamConfig capture_config(sample_rate_hz, num_channels);
|
||||
AudioBuffer capture_buffer(
|
||||
capture_config.sample_rate_hz(), capture_config.num_channels(),
|
||||
capture_config.sample_rate_hz(), capture_config.num_channels(),
|
||||
capture_config.sample_rate_hz(), capture_config.num_channels());
|
||||
test::InputAudioFile capture_file(
|
||||
test::GetApmCaptureTestVectorFileName(sample_rate_hz));
|
||||
std::vector<float> capture_input(samples_per_channel * num_channels);
|
||||
bool stream_has_voice = false;
|
||||
for (int frame_no = 0; frame_no < kNumFramesToProcess; ++frame_no) {
|
||||
ReadFloatSamplesFromStereoFile(samples_per_channel, num_channels,
|
||||
&capture_file, capture_input);
|
||||
|
||||
test::CopyVectorToAudioBuffer(capture_config, capture_input,
|
||||
&capture_buffer);
|
||||
|
||||
stream_has_voice =
|
||||
ProcessOneFrame(sample_rate_hz, &capture_buffer, &voice_detection);
|
||||
}
|
||||
|
||||
EXPECT_EQ(stream_has_voice_reference, stream_has_voice);
|
||||
}
|
||||
|
||||
const bool kStreamHasVoiceReference = true;
|
||||
|
||||
} // namespace
|
||||
|
||||
TEST(VoiceDetectionBitExactnessTest, Mono8kHz) {
|
||||
RunBitexactnessTest(8000, 1, kStreamHasVoiceReference);
|
||||
}
|
||||
|
||||
TEST(VoiceDetectionBitExactnessTest, Mono16kHz) {
|
||||
RunBitexactnessTest(16000, 1, kStreamHasVoiceReference);
|
||||
}
|
||||
|
||||
TEST(VoiceDetectionBitExactnessTest, Mono32kHz) {
|
||||
RunBitexactnessTest(32000, 1, kStreamHasVoiceReference);
|
||||
}
|
||||
|
||||
TEST(VoiceDetectionBitExactnessTest, Mono48kHz) {
|
||||
RunBitexactnessTest(48000, 1, kStreamHasVoiceReference);
|
||||
}
|
||||
|
||||
TEST(VoiceDetectionBitExactnessTest, Stereo8kHz) {
|
||||
RunBitexactnessTest(8000, 2, kStreamHasVoiceReference);
|
||||
}
|
||||
|
||||
TEST(VoiceDetectionBitExactnessTest, Stereo16kHz) {
|
||||
RunBitexactnessTest(16000, 2, kStreamHasVoiceReference);
|
||||
}
|
||||
|
||||
TEST(VoiceDetectionBitExactnessTest, Stereo32kHz) {
|
||||
RunBitexactnessTest(32000, 2, kStreamHasVoiceReference);
|
||||
}
|
||||
|
||||
TEST(VoiceDetectionBitExactnessTest, Stereo48kHz) {
|
||||
RunBitexactnessTest(48000, 2, kStreamHasVoiceReference);
|
||||
}
|
||||
|
||||
} // namespace webrtc
|
||||
|
|
@ -54,7 +54,7 @@ rtc::scoped_refptr<AudioProcessing> CreateApm(test::FuzzDataHelper* fuzz_data,
|
|||
bool use_agc = fuzz_data->ReadOrDefaultValue(true);
|
||||
bool use_ns = fuzz_data->ReadOrDefaultValue(true);
|
||||
static_cast<void>(fuzz_data->ReadOrDefaultValue(true));
|
||||
bool use_vad = fuzz_data->ReadOrDefaultValue(true);
|
||||
static_cast<void>(fuzz_data->ReadOrDefaultValue(true));
|
||||
bool use_agc_limiter = fuzz_data->ReadOrDefaultValue(true);
|
||||
bool use_agc2 = fuzz_data->ReadOrDefaultValue(true);
|
||||
|
||||
|
|
@ -114,7 +114,6 @@ rtc::scoped_refptr<AudioProcessing> CreateApm(test::FuzzDataHelper* fuzz_data,
|
|||
use_agc2_adaptive_digital;
|
||||
apm_config.noise_suppression.enabled = use_ns;
|
||||
apm_config.transient_suppression.enabled = use_ts;
|
||||
apm_config.voice_detection.enabled = use_vad;
|
||||
|
||||
rtc::scoped_refptr<AudioProcessing> apm =
|
||||
AudioProcessingBuilderForTesting()
|
||||
|
|
|
|||
Loading…
Reference in a new issue