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In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
66 lines
2.4 KiB
C++
66 lines
2.4 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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#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TEST_SIMULATOR_BUFFERS_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_SIMULATOR_BUFFERS_H_
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#include <memory>
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#include <vector>
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#include "webrtc/modules/audio_processing/audio_buffer.h"
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#include "webrtc/modules/audio_processing/include/audio_processing.h"
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#include "webrtc/rtc_base/random.h"
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namespace webrtc {
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namespace test {
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struct SimulatorBuffers {
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SimulatorBuffers(int render_input_sample_rate_hz,
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int capture_input_sample_rate_hz,
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int render_output_sample_rate_hz,
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int capture_output_sample_rate_hz,
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size_t num_render_input_channels,
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size_t num_capture_input_channels,
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size_t num_render_output_channels,
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size_t num_capture_output_channels);
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~SimulatorBuffers();
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void CreateConfigAndBuffer(int sample_rate_hz,
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size_t num_channels,
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Random* rand_gen,
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std::unique_ptr<AudioBuffer>* buffer,
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StreamConfig* config,
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std::vector<float*>* buffer_data,
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std::vector<float>* buffer_data_samples);
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void UpdateInputBuffers();
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std::unique_ptr<AudioBuffer> render_input_buffer;
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std::unique_ptr<AudioBuffer> capture_input_buffer;
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std::unique_ptr<AudioBuffer> render_output_buffer;
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std::unique_ptr<AudioBuffer> capture_output_buffer;
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StreamConfig render_input_config;
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StreamConfig capture_input_config;
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StreamConfig render_output_config;
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StreamConfig capture_output_config;
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std::vector<float*> render_input;
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std::vector<float> render_input_samples;
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std::vector<float*> capture_input;
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std::vector<float> capture_input_samples;
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std::vector<float*> render_output;
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std::vector<float> render_output_samples;
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std::vector<float*> capture_output;
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std::vector<float> capture_output_samples;
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};
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} // namespace test
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_SIMULATOR_BUFFERS_H_
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