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ff571c60a9
Internal counters in the RenderDelayBuffer can slip out of sync with external counters, leading to buffer misalignment. This CL gives the RenderDelayBuffer an opportunity to update its counters. Tested: Passes: modules_unittests --gtest_filter=BlockProcessor.* Fails as expected due to new unit test: modules_unittests --gtest_filter=BlockProcessor.* --force_fieldtrials="WebRTC-Aec3RenderBufferCallCounterUpdateKillSwitch/Enabled/" audioproc_f with default AEC settings has been verified to be bit-exact on a large number of aecdumps. Bug: webrtc:11803 Change-Id: I9363b834c8c8c934add0335013df60bf131da4bc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180126 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31795}
67 lines
2.5 KiB
C++
67 lines
2.5 KiB
C++
/*
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* Copyright (c) 2017 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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#ifndef MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_RENDER_DELAY_BUFFER_H_
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#define MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_RENDER_DELAY_BUFFER_H_
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#include <vector>
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#include "modules/audio_processing/aec3/aec3_common.h"
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#include "modules/audio_processing/aec3/downsampled_render_buffer.h"
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#include "modules/audio_processing/aec3/render_buffer.h"
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#include "modules/audio_processing/aec3/render_delay_buffer.h"
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#include "test/gmock.h"
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namespace webrtc {
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namespace test {
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class MockRenderDelayBuffer : public RenderDelayBuffer {
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public:
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MockRenderDelayBuffer(int sample_rate_hz, size_t num_channels);
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virtual ~MockRenderDelayBuffer();
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MOCK_METHOD(void, Reset, (), (override));
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MOCK_METHOD(RenderDelayBuffer::BufferingEvent,
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Insert,
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(const std::vector<std::vector<std::vector<float>>>& block),
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(override));
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MOCK_METHOD(void, HandleSkippedCaptureProcessing, (), (override));
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MOCK_METHOD(RenderDelayBuffer::BufferingEvent,
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PrepareCaptureProcessing,
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(),
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(override));
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MOCK_METHOD(bool, AlignFromDelay, (size_t delay), (override));
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MOCK_METHOD(void, AlignFromExternalDelay, (), (override));
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MOCK_METHOD(size_t, Delay, (), (const, override));
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MOCK_METHOD(size_t, MaxDelay, (), (const, override));
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MOCK_METHOD(RenderBuffer*, GetRenderBuffer, (), (override));
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MOCK_METHOD(const DownsampledRenderBuffer&,
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GetDownsampledRenderBuffer,
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(),
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(const, override));
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MOCK_METHOD(void, SetAudioBufferDelay, (int delay_ms), (override));
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MOCK_METHOD(bool, HasReceivedBufferDelay, (), (override));
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private:
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RenderBuffer* FakeGetRenderBuffer() { return &render_buffer_; }
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const DownsampledRenderBuffer& FakeGetDownsampledRenderBuffer() const {
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return downsampled_render_buffer_;
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}
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BlockBuffer block_buffer_;
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SpectrumBuffer spectrum_buffer_;
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FftBuffer fft_buffer_;
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RenderBuffer render_buffer_;
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DownsampledRenderBuffer downsampled_render_buffer_;
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};
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} // namespace test
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_RENDER_DELAY_BUFFER_H_
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