Danil Chapovalov b602123a5a Replace rtc::Optional with absl::optional in modules/audio_coding
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}
2018-06-19 12:46:20 +00:00

77 lines
2.2 KiB
C++

/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef MODULES_AUDIO_CODING_TEST_PCMFILE_H_
#define MODULES_AUDIO_CODING_TEST_PCMFILE_H_
#include <stdio.h>
#include <stdlib.h>
#include <string>
#include "absl/types/optional.h"
#include "api/audio/audio_frame.h"
#include "typedefs.h" // NOLINT(build/include)
namespace webrtc {
class PCMFile {
public:
PCMFile();
PCMFile(uint32_t timestamp);
~PCMFile();
void Open(const std::string& filename, uint16_t frequency, const char* mode,
bool auto_rewind = false);
int32_t Read10MsData(AudioFrame& audio_frame);
void Write10MsData(const int16_t *playout_buffer, size_t length_smpls);
void Write10MsData(const AudioFrame& audio_frame);
uint16_t PayloadLength10Ms() const;
int32_t SamplingFrequency() const;
void Close();
bool EndOfFile() const {
return end_of_file_;
}
// Moves forward the specified number of 10 ms blocks. If a limit has been set
// with SetNum10MsBlocksToRead, fast-forwarding does not count towards this
// limit.
void FastForward(int num_10ms_blocks);
void Rewind();
static int16_t ChooseFile(std::string* file_name, int16_t max_len,
uint16_t* frequency_hz);
bool Rewinded();
void SaveStereo(bool is_stereo = true);
void ReadStereo(bool is_stereo = true);
// If set, the reading will stop after the specified number of blocks have
// been read. When that has happened, EndOfFile() will return true. Calling
// Rewind() will reset the counter and start over.
void SetNum10MsBlocksToRead(int value);
private:
FILE* pcm_file_;
uint16_t samples_10ms_;
int32_t frequency_;
bool end_of_file_;
bool auto_rewind_;
bool rewinded_;
uint32_t timestamp_;
bool read_stereo_;
bool save_stereo_;
absl::optional<int> num_10ms_blocks_to_read_;
int blocks_read_ = 0;
};
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_TEST_PCMFILE_H_