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f254e9e9e5
This reverts commit 81c0cf287c.
Reason for revert: internal test failures
Original change's description:
> Simplification and refactoring of the AudioBuffer code
>
> This CL performs a major refactoring and simplification
> of the AudioBuffer code that.
> -Removes 7 of the 9 internal buffers of the AudioBuffer.
> -Avoids the implicit copying required to keep the
> internal buffers in sync.
> -Removes all code relating to handling of fixed-point
> sample data in the AudioBuffer.
> -Changes the naming of the class methods to reflect
> that only floating point is handled.
> -Corrects some bugs in the code.
> -Extends the handling of internal downmixing to be
> more generic.
>
> Bug: webrtc:10882
> Change-Id: I12c8af156fbe366b154744a0a1b3d926bf7be572
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149828
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28928}
TBR=gustaf@webrtc.org,peah@webrtc.org
Change-Id: I2729e3ad24b3a9b40b368b84cb565c859e79b51e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10882
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150084
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28931}
123 lines
4.6 KiB
C++
123 lines
4.6 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_impl.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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void ProcessOneFrame(int sample_rate_hz,
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AudioBuffer* audio_buffer,
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VoiceDetectionImpl* 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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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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int frame_size_ms_reference,
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bool stream_has_voice_reference,
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VoiceDetection::Likelihood likelihood_reference) {
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rtc::CriticalSection crit_capture;
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VoiceDetectionImpl voice_detection(&crit_capture);
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voice_detection.Initialize(sample_rate_hz > 16000 ? 16000 : sample_rate_hz);
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voice_detection.Enable(true);
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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, false);
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AudioBuffer capture_buffer(
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capture_config.num_frames(), capture_config.num_channels(),
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capture_config.num_frames(), capture_config.num_channels(),
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capture_config.num_frames());
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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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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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ProcessOneFrame(sample_rate_hz, &capture_buffer, &voice_detection);
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}
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int frame_size_ms = voice_detection.frame_size_ms();
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bool stream_has_voice = voice_detection.stream_has_voice();
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VoiceDetection::Likelihood likelihood = voice_detection.likelihood();
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// Compare the outputs to the references.
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EXPECT_EQ(frame_size_ms_reference, frame_size_ms);
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EXPECT_EQ(stream_has_voice_reference, stream_has_voice);
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EXPECT_EQ(likelihood_reference, likelihood);
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}
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const int kFrameSizeMsReference = 10;
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const bool kStreamHasVoiceReference = true;
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const VoiceDetection::Likelihood kLikelihoodReference =
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VoiceDetection::kLowLikelihood;
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} // namespace
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TEST(VoiceDetectionBitExactnessTest, Mono8kHz) {
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RunBitexactnessTest(8000, 1, kFrameSizeMsReference, kStreamHasVoiceReference,
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kLikelihoodReference);
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}
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TEST(VoiceDetectionBitExactnessTest, Mono16kHz) {
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RunBitexactnessTest(16000, 1, kFrameSizeMsReference, kStreamHasVoiceReference,
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kLikelihoodReference);
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}
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TEST(VoiceDetectionBitExactnessTest, Mono32kHz) {
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RunBitexactnessTest(32000, 1, kFrameSizeMsReference, kStreamHasVoiceReference,
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kLikelihoodReference);
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}
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TEST(VoiceDetectionBitExactnessTest, Mono48kHz) {
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RunBitexactnessTest(48000, 1, kFrameSizeMsReference, kStreamHasVoiceReference,
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kLikelihoodReference);
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}
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TEST(VoiceDetectionBitExactnessTest, Stereo8kHz) {
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RunBitexactnessTest(8000, 2, kFrameSizeMsReference, kStreamHasVoiceReference,
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kLikelihoodReference);
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}
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TEST(VoiceDetectionBitExactnessTest, Stereo16kHz) {
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RunBitexactnessTest(16000, 2, kFrameSizeMsReference, kStreamHasVoiceReference,
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kLikelihoodReference);
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}
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TEST(VoiceDetectionBitExactnessTest, Stereo32kHz) {
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RunBitexactnessTest(32000, 2, kFrameSizeMsReference, kStreamHasVoiceReference,
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kLikelihoodReference);
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}
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TEST(VoiceDetectionBitExactnessTest, Stereo48kHz) {
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RunBitexactnessTest(48000, 2, kFrameSizeMsReference, kStreamHasVoiceReference,
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kLikelihoodReference);
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}
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} // namespace webrtc
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