56 lines
1.7 KiB
C++
56 lines
1.7 KiB
C++
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/*
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* Copyright (c) 2015 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 "webrtc/modules/audio_processing/test/audio_buffer_tools.h"
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namespace webrtc {
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namespace test {
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void SetupFrame(StreamConfig stream_config,
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std::vector<float*>* frame,
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std::vector<float>* frame_samples) {
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frame_samples->resize(stream_config.num_channels() *
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stream_config.num_frames());
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frame->resize(stream_config.num_channels());
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for (size_t ch = 0; ch < stream_config.num_channels(); ++ch) {
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(*frame)[ch] = &(*frame_samples)[ch * stream_config.num_frames()];
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}
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}
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void CopyVectorToAudioBuffer(const StreamConfig& stream_config,
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const std::vector<float>& source,
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AudioBuffer* destination) {
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std::vector<float*> input;
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std::vector<float> input_samples;
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SetupFrame(stream_config, &input, &input_samples);
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RTC_DCHECK_EQ(input_samples.size(), source.size());
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input_samples = source;
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destination->CopyFrom(&input[0], stream_config);
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}
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std::vector<float> ExtractVectorFromAudioBuffer(
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const StreamConfig& stream_config,
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AudioBuffer* source) {
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std::vector<float*> output;
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std::vector<float> output_samples;
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SetupFrame(stream_config, &output, &output_samples);
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source->CopyTo(stream_config, &output[0]);
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return output_samples;
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}
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} // namespace test
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} // namespace webrtc
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