mirror of
https://github.com/mollyim/webrtc.git
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d47941e018
This is a reland of 81c0cf287c
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}
Bug: webrtc:10882
Change-Id: I2ddf327e80a03468c41662ae63c619ff34f2363a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150101
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28938}
353 lines
14 KiB
C++
353 lines
14 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/echo_cancellation_impl.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 "test/gtest.h"
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namespace webrtc {
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namespace {
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const int kNumFramesToProcess = 100;
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void SetupComponent(int sample_rate_hz,
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EchoCancellationImpl::SuppressionLevel suppression_level,
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bool drift_compensation_enabled,
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EchoCancellationImpl* echo_canceller) {
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echo_canceller->Initialize(sample_rate_hz, 1, 1, 1);
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echo_canceller->set_suppression_level(suppression_level);
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echo_canceller->enable_drift_compensation(drift_compensation_enabled);
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Config config;
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config.Set<DelayAgnostic>(new DelayAgnostic(true));
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config.Set<ExtendedFilter>(new ExtendedFilter(true));
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echo_canceller->SetExtraOptions(true, true, false);
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}
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void ProcessOneFrame(int sample_rate_hz,
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int stream_delay_ms,
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bool drift_compensation_enabled,
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int stream_drift_samples,
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AudioBuffer* render_audio_buffer,
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AudioBuffer* capture_audio_buffer,
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EchoCancellationImpl* echo_canceller) {
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if (sample_rate_hz > AudioProcessing::kSampleRate16kHz) {
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render_audio_buffer->SplitIntoFrequencyBands();
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capture_audio_buffer->SplitIntoFrequencyBands();
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}
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std::vector<float> render_audio;
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EchoCancellationImpl::PackRenderAudioBuffer(
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render_audio_buffer, 1, render_audio_buffer->num_channels(),
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&render_audio);
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echo_canceller->ProcessRenderAudio(render_audio);
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if (drift_compensation_enabled) {
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echo_canceller->set_stream_drift_samples(stream_drift_samples);
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}
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echo_canceller->ProcessCaptureAudio(capture_audio_buffer, stream_delay_ms);
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if (sample_rate_hz > AudioProcessing::kSampleRate16kHz) {
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capture_audio_buffer->MergeFrequencyBands();
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}
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}
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void RunBitexactnessTest(
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int sample_rate_hz,
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size_t num_channels,
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int stream_delay_ms,
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bool drift_compensation_enabled,
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int stream_drift_samples,
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EchoCancellationImpl::SuppressionLevel suppression_level,
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bool stream_has_echo_reference,
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const rtc::ArrayView<const float>& output_reference) {
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EchoCancellationImpl echo_canceller;
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SetupComponent(sample_rate_hz, suppression_level, drift_compensation_enabled,
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&echo_canceller);
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const int samples_per_channel = rtc::CheckedDivExact(sample_rate_hz, 100);
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const StreamConfig render_config(sample_rate_hz, num_channels, false);
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AudioBuffer render_buffer(
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render_config.sample_rate_hz(), render_config.num_channels(),
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render_config.sample_rate_hz(), 1, render_config.sample_rate_hz(), 1);
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test::InputAudioFile render_file(
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test::GetApmRenderTestVectorFileName(sample_rate_hz));
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std::vector<float> render_input(samples_per_channel * num_channels);
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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.sample_rate_hz(), capture_config.num_channels(),
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capture_config.sample_rate_hz(), 1, capture_config.sample_rate_hz(), 1);
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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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&render_file, render_input);
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ReadFloatSamplesFromStereoFile(samples_per_channel, num_channels,
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&capture_file, capture_input);
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test::CopyVectorToAudioBuffer(render_config, render_input, &render_buffer);
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test::CopyVectorToAudioBuffer(capture_config, capture_input,
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&capture_buffer);
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ProcessOneFrame(sample_rate_hz, stream_delay_ms, drift_compensation_enabled,
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stream_drift_samples, &render_buffer, &capture_buffer,
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&echo_canceller);
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}
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// Extract and verify the test results.
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std::vector<float> capture_output;
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test::ExtractVectorFromAudioBuffer(capture_config, &capture_buffer,
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&capture_output);
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EXPECT_EQ(stream_has_echo_reference, echo_canceller.stream_has_echo());
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// Compare the output with the reference. Only the first values of the output
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// from last frame processed are compared in order not having to specify all
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// preceeding frames as testvectors. As the algorithm being tested has a
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// memory, testing only the last frame implicitly also tests the preceeding
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// frames.
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const float kElementErrorBound = 1.0f / 32768.0f;
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EXPECT_TRUE(test::VerifyDeinterleavedArray(
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capture_config.num_frames(), capture_config.num_channels(),
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output_reference, capture_output, kElementErrorBound));
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}
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const bool kStreamHasEchoReference = true;
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} // namespace
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// TODO(peah): Activate all these tests for ARM and ARM64 once the issue on the
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// Chromium ARM and ARM64 boths have been identified. This is tracked in the
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// issue https://bugs.chromium.org/p/webrtc/issues/detail?id=5711.
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#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
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defined(WEBRTC_ANDROID))
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TEST(EchoCancellationBitExactnessTest,
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Mono8kHz_HighLevel_NoDrift_StreamDelay0) {
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#else
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TEST(EchoCancellationBitExactnessTest,
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DISABLED_Mono8kHz_HighLevel_NoDrift_StreamDelay0) {
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#endif
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const float kOutputReference[] = {-0.000646f, -0.001525f, 0.002688f};
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RunBitexactnessTest(8000, 1, 0, false, 0,
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EchoCancellationImpl::SuppressionLevel::kHighSuppression,
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kStreamHasEchoReference, kOutputReference);
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}
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#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
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defined(WEBRTC_ANDROID))
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TEST(EchoCancellationBitExactnessTest,
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Mono16kHz_HighLevel_NoDrift_StreamDelay0) {
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#else
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TEST(EchoCancellationBitExactnessTest,
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DISABLED_Mono16kHz_HighLevel_NoDrift_StreamDelay0) {
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#endif
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const float kOutputReference[] = {0.000055f, 0.000421f, 0.001149f};
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RunBitexactnessTest(16000, 1, 0, false, 0,
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EchoCancellationImpl::SuppressionLevel::kHighSuppression,
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kStreamHasEchoReference, kOutputReference);
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}
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#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
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defined(WEBRTC_ANDROID))
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TEST(EchoCancellationBitExactnessTest,
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Mono32kHz_HighLevel_NoDrift_StreamDelay0) {
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#else
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TEST(EchoCancellationBitExactnessTest,
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DISABLED_Mono32kHz_HighLevel_NoDrift_StreamDelay0) {
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#endif
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const float kOutputReference[] = {-0.000671f, 0.000061f, -0.000031f};
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RunBitexactnessTest(32000, 1, 0, false, 0,
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EchoCancellationImpl::SuppressionLevel::kHighSuppression,
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kStreamHasEchoReference, kOutputReference);
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}
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#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
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defined(WEBRTC_ANDROID))
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TEST(EchoCancellationBitExactnessTest,
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Mono48kHz_HighLevel_NoDrift_StreamDelay0) {
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#else
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TEST(EchoCancellationBitExactnessTest,
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DISABLED_Mono48kHz_HighLevel_NoDrift_StreamDelay0) {
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#endif
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const float kOutputReference[] = {-0.001403f, -0.001411f, -0.000755f};
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RunBitexactnessTest(48000, 1, 0, false, 0,
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EchoCancellationImpl::SuppressionLevel::kHighSuppression,
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kStreamHasEchoReference, kOutputReference);
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}
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#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
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defined(WEBRTC_ANDROID))
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TEST(EchoCancellationBitExactnessTest,
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Mono16kHz_LowLevel_NoDrift_StreamDelay0) {
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#else
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TEST(EchoCancellationBitExactnessTest,
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DISABLED_Mono16kHz_LowLevel_NoDrift_StreamDelay0) {
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#endif
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#if defined(WEBRTC_MAC)
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const float kOutputReference[] = {-0.000145f, 0.000179f, 0.000917f};
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#else
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const float kOutputReference[] = {-0.000009f, 0.000363f, 0.001094f};
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#endif
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RunBitexactnessTest(16000, 1, 0, false, 0,
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EchoCancellationImpl::SuppressionLevel::kLowSuppression,
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kStreamHasEchoReference, kOutputReference);
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}
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#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
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defined(WEBRTC_ANDROID))
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TEST(EchoCancellationBitExactnessTest,
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Mono16kHz_ModerateLevel_NoDrift_StreamDelay0) {
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#else
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TEST(EchoCancellationBitExactnessTest,
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DISABLED_Mono16kHz_ModerateLevel_NoDrift_StreamDelay0) {
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#endif
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const float kOutputReference[] = {0.000055f, 0.000421f, 0.001149f};
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RunBitexactnessTest(
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16000, 1, 0, false, 0,
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EchoCancellationImpl::SuppressionLevel::kModerateSuppression,
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kStreamHasEchoReference, kOutputReference);
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}
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#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
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defined(WEBRTC_ANDROID))
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TEST(EchoCancellationBitExactnessTest,
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Mono16kHz_HighLevel_NoDrift_StreamDelay10) {
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#else
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TEST(EchoCancellationBitExactnessTest,
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DISABLED_Mono16kHz_HighLevel_NoDrift_StreamDelay10) {
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#endif
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const float kOutputReference[] = {0.000055f, 0.000421f, 0.001149f};
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RunBitexactnessTest(16000, 1, 10, false, 0,
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EchoCancellationImpl::SuppressionLevel::kHighSuppression,
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kStreamHasEchoReference, kOutputReference);
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}
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#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
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defined(WEBRTC_ANDROID))
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TEST(EchoCancellationBitExactnessTest,
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Mono16kHz_HighLevel_NoDrift_StreamDelay20) {
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#else
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TEST(EchoCancellationBitExactnessTest,
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DISABLED_Mono16kHz_HighLevel_NoDrift_StreamDelay20) {
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#endif
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const float kOutputReference[] = {0.000055f, 0.000421f, 0.001149f};
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RunBitexactnessTest(16000, 1, 20, false, 0,
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EchoCancellationImpl::SuppressionLevel::kHighSuppression,
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kStreamHasEchoReference, kOutputReference);
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}
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#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
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defined(WEBRTC_ANDROID))
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TEST(EchoCancellationBitExactnessTest,
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Mono16kHz_HighLevel_Drift0_StreamDelay0) {
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#else
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TEST(EchoCancellationBitExactnessTest,
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DISABLED_Mono16kHz_HighLevel_Drift0_StreamDelay0) {
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#endif
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const float kOutputReference[] = {0.000055f, 0.000421f, 0.001149f};
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RunBitexactnessTest(16000, 1, 0, true, 0,
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EchoCancellationImpl::SuppressionLevel::kHighSuppression,
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kStreamHasEchoReference, kOutputReference);
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}
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#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
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defined(WEBRTC_ANDROID))
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TEST(EchoCancellationBitExactnessTest,
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Mono16kHz_HighLevel_Drift5_StreamDelay0) {
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#else
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TEST(EchoCancellationBitExactnessTest,
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DISABLED_Mono16kHz_HighLevel_Drift5_StreamDelay0) {
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#endif
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const float kOutputReference[] = {0.000055f, 0.000421f, 0.001149f};
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RunBitexactnessTest(16000, 1, 0, true, 5,
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EchoCancellationImpl::SuppressionLevel::kHighSuppression,
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kStreamHasEchoReference, kOutputReference);
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}
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#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
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defined(WEBRTC_ANDROID))
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TEST(EchoCancellationBitExactnessTest,
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Stereo8kHz_HighLevel_NoDrift_StreamDelay0) {
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#else
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TEST(EchoCancellationBitExactnessTest,
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DISABLED_Stereo8kHz_HighLevel_NoDrift_StreamDelay0) {
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#endif
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#if defined(WEBRTC_MAC)
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const float kOutputReference[] = {-0.000392f, -0.001449f, 0.003004f,
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-0.000392f, -0.001449f, 0.003004f};
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#else
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const float kOutputReference[] = {-0.000464f, -0.001525f, 0.002933f,
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-0.000464f, -0.001525f, 0.002933f};
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#endif
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RunBitexactnessTest(8000, 2, 0, false, 0,
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EchoCancellationImpl::SuppressionLevel::kHighSuppression,
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kStreamHasEchoReference, kOutputReference);
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}
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#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
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defined(WEBRTC_ANDROID))
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TEST(EchoCancellationBitExactnessTest,
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Stereo16kHz_HighLevel_NoDrift_StreamDelay0) {
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#else
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TEST(EchoCancellationBitExactnessTest,
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DISABLED_Stereo16kHz_HighLevel_NoDrift_StreamDelay0) {
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#endif
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const float kOutputReference[] = {0.000166f, 0.000735f, 0.000841f,
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0.000166f, 0.000735f, 0.000841f};
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RunBitexactnessTest(16000, 2, 0, false, 0,
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EchoCancellationImpl::SuppressionLevel::kHighSuppression,
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kStreamHasEchoReference, kOutputReference);
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}
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#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
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defined(WEBRTC_ANDROID))
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TEST(EchoCancellationBitExactnessTest,
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Stereo32kHz_HighLevel_NoDrift_StreamDelay0) {
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#else
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TEST(EchoCancellationBitExactnessTest,
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DISABLED_Stereo32kHz_HighLevel_NoDrift_StreamDelay0) {
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#endif
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#if defined(WEBRTC_MAC)
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const float kOutputReference[] = {-0.000458f, 0.000214f, 0.000122f,
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-0.000458f, 0.000214f, 0.000122f};
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#else
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const float kOutputReference[] = {-0.000427f, 0.000183f, 0.000183f,
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-0.000427f, 0.000183f, 0.000183f};
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#endif
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RunBitexactnessTest(32000, 2, 0, false, 0,
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EchoCancellationImpl::SuppressionLevel::kHighSuppression,
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kStreamHasEchoReference, kOutputReference);
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}
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#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
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defined(WEBRTC_ANDROID))
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TEST(EchoCancellationBitExactnessTest,
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Stereo48kHz_HighLevel_NoDrift_StreamDelay0) {
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#else
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TEST(EchoCancellationBitExactnessTest,
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DISABLED_Stereo48kHz_HighLevel_NoDrift_StreamDelay0) {
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#endif
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const float kOutputReference[] = {-0.001101f, -0.001101f, -0.000449f,
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-0.001101f, -0.001101f, -0.000449f};
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RunBitexactnessTest(48000, 2, 0, false, 0,
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EchoCancellationImpl::SuppressionLevel::kHighSuppression,
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kStreamHasEchoReference, kOutputReference);
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}
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} // namespace webrtc
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