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/*
* 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 WEBRTC_MODULES_AUDIO_CODING_TEST_PCMFILE_H_
#define WEBRTC_MODULES_AUDIO_CODING_TEST_PCMFILE_H_
#include <stdio.h>
#include <stdlib.h>
#include <string>
#include "webrtc/base/optional.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/typedefs.h"
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);
Update a ton of audio code to use size_t more correctly and in general reduce use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 14:52:23 -07:00
void Write10MsData(int16_t *playout_buffer, size_t length_smpls);
void Write10MsData(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_;
rtc::Optional<int> num_10ms_blocks_to_read_;
int blocks_read_ = 0;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_PCMFILE_H_