This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script with parameter 'modules/audio_coding'
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"[\./api]*:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: Ic980ee605148fdb160666d4aa03cc87175e48fe8
Reviewed-on: https://webrtc-review.googlesource.com/84130
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23659}
77 lines
2.2 KiB
C++
77 lines
2.2 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, uint16_t frequency, 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 {
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return end_of_file_;
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}
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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, 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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