a751f167c6
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}
105 lines
3.6 KiB
C++
105 lines
3.6 KiB
C++
/*
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* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include <vector>
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#include "api/array_view.h"
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#include "modules/audio_processing/audio_buffer.h"
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#include "modules/audio_processing/test/audio_buffer_tools.h"
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#include "modules/audio_processing/test/bitexactness_tools.h"
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#include "modules/audio_processing/voice_detection.h"
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#include "test/gtest.h"
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namespace webrtc {
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namespace {
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const int kNumFramesToProcess = 1000;
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// Process one frame of data and produce the output.
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bool ProcessOneFrame(int sample_rate_hz,
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AudioBuffer* audio_buffer,
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VoiceDetection* voice_detection) {
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if (sample_rate_hz > AudioProcessing::kSampleRate16kHz) {
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audio_buffer->SplitIntoFrequencyBands();
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}
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return voice_detection->ProcessCaptureAudio(audio_buffer);
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}
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// Processes a specified amount of frames, verifies the results and reports
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// any errors.
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void RunBitexactnessTest(int sample_rate_hz,
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size_t num_channels,
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bool stream_has_voice_reference) {
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int sample_rate_to_use = std::min(sample_rate_hz, 16000);
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VoiceDetection voice_detection(sample_rate_to_use,
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VoiceDetection::kLowLikelihood);
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int samples_per_channel = rtc::CheckedDivExact(sample_rate_hz, 100);
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const StreamConfig capture_config(sample_rate_hz, num_channels);
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AudioBuffer capture_buffer(
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capture_config.sample_rate_hz(), capture_config.num_channels(),
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capture_config.sample_rate_hz(), capture_config.num_channels(),
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capture_config.sample_rate_hz(), capture_config.num_channels());
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test::InputAudioFile capture_file(
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test::GetApmCaptureTestVectorFileName(sample_rate_hz));
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std::vector<float> capture_input(samples_per_channel * num_channels);
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bool stream_has_voice = false;
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for (int frame_no = 0; frame_no < kNumFramesToProcess; ++frame_no) {
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ReadFloatSamplesFromStereoFile(samples_per_channel, num_channels,
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&capture_file, capture_input);
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test::CopyVectorToAudioBuffer(capture_config, capture_input,
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&capture_buffer);
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stream_has_voice =
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ProcessOneFrame(sample_rate_hz, &capture_buffer, &voice_detection);
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}
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EXPECT_EQ(stream_has_voice_reference, stream_has_voice);
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}
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const bool kStreamHasVoiceReference = true;
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} // namespace
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TEST(VoiceDetectionBitExactnessTest, Mono8kHz) {
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RunBitexactnessTest(8000, 1, kStreamHasVoiceReference);
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}
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TEST(VoiceDetectionBitExactnessTest, Mono16kHz) {
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RunBitexactnessTest(16000, 1, kStreamHasVoiceReference);
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}
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TEST(VoiceDetectionBitExactnessTest, Mono32kHz) {
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RunBitexactnessTest(32000, 1, kStreamHasVoiceReference);
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}
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TEST(VoiceDetectionBitExactnessTest, Mono48kHz) {
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RunBitexactnessTest(48000, 1, kStreamHasVoiceReference);
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}
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TEST(VoiceDetectionBitExactnessTest, Stereo8kHz) {
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RunBitexactnessTest(8000, 2, kStreamHasVoiceReference);
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}
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TEST(VoiceDetectionBitExactnessTest, Stereo16kHz) {
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RunBitexactnessTest(16000, 2, kStreamHasVoiceReference);
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}
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TEST(VoiceDetectionBitExactnessTest, Stereo32kHz) {
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RunBitexactnessTest(32000, 2, kStreamHasVoiceReference);
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}
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TEST(VoiceDetectionBitExactnessTest, Stereo48kHz) {
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RunBitexactnessTest(48000, 2, kStreamHasVoiceReference);
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}
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} // namespace webrtc
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