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Diffstat (limited to 'third_party/libwebrtc/modules/audio_processing/test/simulator_buffers.h')
-rw-r--r-- | third_party/libwebrtc/modules/audio_processing/test/simulator_buffers.h | 66 |
1 files changed, 66 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/test/simulator_buffers.h b/third_party/libwebrtc/modules/audio_processing/test/simulator_buffers.h new file mode 100644 index 0000000000..36dcf301a2 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/simulator_buffers.h @@ -0,0 +1,66 @@ +/* + * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef MODULES_AUDIO_PROCESSING_TEST_SIMULATOR_BUFFERS_H_ +#define MODULES_AUDIO_PROCESSING_TEST_SIMULATOR_BUFFERS_H_ + +#include <memory> +#include <vector> + +#include "modules/audio_processing/audio_buffer.h" +#include "modules/audio_processing/include/audio_processing.h" +#include "rtc_base/random.h" + +namespace webrtc { +namespace test { + +struct SimulatorBuffers { + SimulatorBuffers(int render_input_sample_rate_hz, + int capture_input_sample_rate_hz, + int render_output_sample_rate_hz, + int capture_output_sample_rate_hz, + size_t num_render_input_channels, + size_t num_capture_input_channels, + size_t num_render_output_channels, + size_t num_capture_output_channels); + ~SimulatorBuffers(); + + void CreateConfigAndBuffer(int sample_rate_hz, + size_t num_channels, + Random* rand_gen, + std::unique_ptr<AudioBuffer>* buffer, + StreamConfig* config, + std::vector<float*>* buffer_data, + std::vector<float>* buffer_data_samples); + + void UpdateInputBuffers(); + + std::unique_ptr<AudioBuffer> render_input_buffer; + std::unique_ptr<AudioBuffer> capture_input_buffer; + std::unique_ptr<AudioBuffer> render_output_buffer; + std::unique_ptr<AudioBuffer> capture_output_buffer; + StreamConfig render_input_config; + StreamConfig capture_input_config; + StreamConfig render_output_config; + StreamConfig capture_output_config; + std::vector<float*> render_input; + std::vector<float> render_input_samples; + std::vector<float*> capture_input; + std::vector<float> capture_input_samples; + std::vector<float*> render_output; + std::vector<float> render_output_samples; + std::vector<float*> capture_output; + std::vector<float> capture_output_samples; +}; + +} // namespace test +} // namespace webrtc + +#endif // MODULES_AUDIO_PROCESSING_TEST_SIMULATOR_BUFFERS_H_ |