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Mechanically generated by running this command: tools_webrtc/do-renames.sh update all-renames.txt && git cl format Then manually updating: tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc Bug: webrtc:10159 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833 Reviewed-on: https://webrtc-review.googlesource.com/c/115653 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26226}
50 lines
1.7 KiB
C++
50 lines
1.7 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 "common_audio/resampler/include/resampler.h"
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#include "modules/audio_coding/neteq/tools/input_audio_file.h"
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#include "rtc_base/constructor_magic.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(const std::string file_name, int file_rate_hz)
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: InputAudioFile(file_name),
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file_rate_hz_(file_rate_hz),
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output_rate_hz_(-1) {}
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ResampleInputAudioFile(const std::string file_name,
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int file_rate_hz,
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int output_rate_hz)
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: InputAudioFile(file_name),
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file_rate_hz_(file_rate_hz),
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output_rate_hz_(output_rate_hz) {}
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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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RTC_DISALLOW_COPY_AND_ASSIGN(ResampleInputAudioFile);
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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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