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Running clang-format with chromium's style guide. The goal is n-fold: * providing consistency and readability (that's what code guidelines are for) * preventing noise with presubmit checks and git cl format * building on the previous point: making it easier to automatically fix format issues * you name it Please consider using git-hyper-blame to ignore this commit. Bug: webrtc:9340 Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87 Reviewed-on: https://webrtc-review.googlesource.com/81185 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23660}
77 lines
2.3 KiB
C++
77 lines
2.3 KiB
C++
/*
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* Copyright (c) 2012 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_TEST_PCMFILE_H_
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#define MODULES_AUDIO_CODING_TEST_PCMFILE_H_
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#include <stdio.h>
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#include <stdlib.h>
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#include <string>
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#include "absl/types/optional.h"
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#include "api/audio/audio_frame.h"
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#include "typedefs.h" // NOLINT(build/include)
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namespace webrtc {
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class PCMFile {
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public:
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PCMFile();
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PCMFile(uint32_t timestamp);
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~PCMFile();
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void Open(const std::string& filename,
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uint16_t frequency,
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const char* mode,
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bool auto_rewind = false);
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int32_t Read10MsData(AudioFrame& audio_frame);
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void Write10MsData(const int16_t* playout_buffer, size_t length_smpls);
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void Write10MsData(const AudioFrame& audio_frame);
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uint16_t PayloadLength10Ms() const;
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int32_t SamplingFrequency() const;
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void Close();
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bool EndOfFile() const { return end_of_file_; }
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// Moves forward the specified number of 10 ms blocks. If a limit has been set
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// with SetNum10MsBlocksToRead, fast-forwarding does not count towards this
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// limit.
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void FastForward(int num_10ms_blocks);
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void Rewind();
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static int16_t ChooseFile(std::string* file_name,
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int16_t max_len,
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uint16_t* frequency_hz);
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bool Rewinded();
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void SaveStereo(bool is_stereo = true);
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void ReadStereo(bool is_stereo = true);
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// If set, the reading will stop after the specified number of blocks have
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// been read. When that has happened, EndOfFile() will return true. Calling
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// Rewind() will reset the counter and start over.
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void SetNum10MsBlocksToRead(int value);
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private:
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FILE* pcm_file_;
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uint16_t samples_10ms_;
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int32_t frequency_;
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bool end_of_file_;
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bool auto_rewind_;
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bool rewinded_;
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uint32_t timestamp_;
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bool read_stereo_;
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bool save_stereo_;
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absl::optional<int> num_10ms_blocks_to_read_;
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int blocks_read_ = 0;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_TEST_PCMFILE_H_
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