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Bug: webrtc:13579 Change-Id: Ifec66fb6ba9724d18539de7245a358c2d13c7939 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268547 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Ali Tofigh <alito@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37573}
55 lines
2 KiB
C++
55 lines
2 KiB
C++
/*
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* Copyright (c) 2014 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_CODING_NETEQ_TOOLS_RESAMPLE_INPUT_AUDIO_FILE_H_
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#define MODULES_AUDIO_CODING_NETEQ_TOOLS_RESAMPLE_INPUT_AUDIO_FILE_H_
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#include <string>
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#include "absl/strings/string_view.h"
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#include "common_audio/resampler/include/resampler.h"
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#include "modules/audio_coding/neteq/tools/input_audio_file.h"
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namespace webrtc {
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namespace test {
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// Class for handling a looping input audio file with resampling.
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class ResampleInputAudioFile : public InputAudioFile {
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public:
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ResampleInputAudioFile(absl::string_view file_name,
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int file_rate_hz,
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bool loop_at_end = true)
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: InputAudioFile(file_name, loop_at_end),
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file_rate_hz_(file_rate_hz),
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output_rate_hz_(-1) {}
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ResampleInputAudioFile(absl::string_view file_name,
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int file_rate_hz,
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int output_rate_hz,
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bool loop_at_end = true)
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: InputAudioFile(file_name, loop_at_end),
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file_rate_hz_(file_rate_hz),
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output_rate_hz_(output_rate_hz) {}
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ResampleInputAudioFile(const ResampleInputAudioFile&) = delete;
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ResampleInputAudioFile& operator=(const ResampleInputAudioFile&) = delete;
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bool Read(size_t samples, int output_rate_hz, int16_t* destination);
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bool Read(size_t samples, int16_t* destination) override;
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void set_output_rate_hz(int rate_hz);
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private:
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const int file_rate_hz_;
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int output_rate_hz_;
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Resampler resampler_;
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};
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} // namespace test
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_RESAMPLE_INPUT_AUDIO_FILE_H_
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