2011-07-07 08:21:25 +00:00
|
|
|
/*
|
2012-04-26 07:54:30 +00:00
|
|
|
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
|
2011-07-07 08:21:25 +00:00
|
|
|
*
|
|
|
|
|
* 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.
|
|
|
|
|
*/
|
|
|
|
|
|
2015-11-26 04:44:54 -08:00
|
|
|
#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_PCMFILE_H_
|
|
|
|
|
#define WEBRTC_MODULES_AUDIO_CODING_TEST_PCMFILE_H_
|
2011-07-07 08:21:25 +00:00
|
|
|
|
2013-08-05 16:22:53 +00:00
|
|
|
#include <stdio.h>
|
|
|
|
|
#include <stdlib.h>
|
|
|
|
|
|
2012-08-17 10:38:28 +00:00
|
|
|
#include <string>
|
2011-07-07 08:21:25 +00:00
|
|
|
|
2015-12-10 16:24:39 +01:00
|
|
|
#include "webrtc/base/optional.h"
|
2015-11-04 08:31:52 +01:00
|
|
|
#include "webrtc/modules/include/module_common_types.h"
|
2013-10-02 21:44:33 +00:00
|
|
|
#include "webrtc/typedefs.h"
|
2012-04-26 07:54:30 +00:00
|
|
|
|
2011-12-16 10:09:04 +00:00
|
|
|
namespace webrtc {
|
2011-07-07 08:21:25 +00:00
|
|
|
|
2012-04-26 07:54:30 +00:00
|
|
|
class PCMFile {
|
|
|
|
|
public:
|
|
|
|
|
PCMFile();
|
2013-04-09 00:28:06 +00:00
|
|
|
PCMFile(uint32_t timestamp);
|
2016-08-29 10:05:24 -07:00
|
|
|
~PCMFile();
|
2012-08-17 10:38:28 +00:00
|
|
|
|
2013-05-03 07:34:12 +00:00
|
|
|
void Open(const std::string& filename, uint16_t frequency, const char* mode,
|
|
|
|
|
bool auto_rewind = false);
|
2012-04-26 07:54:30 +00:00
|
|
|
|
2013-04-09 00:28:06 +00:00
|
|
|
int32_t Read10MsData(AudioFrame& audio_frame);
|
2012-04-26 07:54:30 +00:00
|
|
|
|
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);
|
2012-04-26 07:54:30 +00:00
|
|
|
void Write10MsData(AudioFrame& audio_frame);
|
2011-07-07 08:21:25 +00:00
|
|
|
|
2013-04-09 00:28:06 +00:00
|
|
|
uint16_t PayloadLength10Ms() const;
|
|
|
|
|
int32_t SamplingFrequency() const;
|
2012-04-26 07:54:30 +00:00
|
|
|
void Close();
|
|
|
|
|
bool EndOfFile() const {
|
|
|
|
|
return end_of_file_;
|
|
|
|
|
}
|
2015-12-10 16:24:39 +01:00
|
|
|
// 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);
|
2012-04-26 07:54:30 +00:00
|
|
|
void Rewind();
|
2013-05-03 07:34:12 +00:00
|
|
|
static int16_t ChooseFile(std::string* file_name, int16_t max_len,
|
2013-04-09 00:28:06 +00:00
|
|
|
uint16_t* frequency_hz);
|
2012-04-26 07:54:30 +00:00
|
|
|
bool Rewinded();
|
|
|
|
|
void SaveStereo(bool is_stereo = true);
|
|
|
|
|
void ReadStereo(bool is_stereo = true);
|
2015-12-10 16:24:39 +01:00
|
|
|
// 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);
|
|
|
|
|
|
2012-04-26 07:54:30 +00:00
|
|
|
private:
|
|
|
|
|
FILE* pcm_file_;
|
2013-04-09 00:28:06 +00:00
|
|
|
uint16_t samples_10ms_;
|
|
|
|
|
int32_t frequency_;
|
2012-04-26 07:54:30 +00:00
|
|
|
bool end_of_file_;
|
|
|
|
|
bool auto_rewind_;
|
|
|
|
|
bool rewinded_;
|
2013-04-09 00:28:06 +00:00
|
|
|
uint32_t timestamp_;
|
2012-04-26 07:54:30 +00:00
|
|
|
bool read_stereo_;
|
|
|
|
|
bool save_stereo_;
|
2015-12-10 16:24:39 +01:00
|
|
|
rtc::Optional<int> num_10ms_blocks_to_read_;
|
|
|
|
|
int blocks_read_ = 0;
|
2011-07-07 08:21:25 +00:00
|
|
|
};
|
|
|
|
|
|
2012-04-26 07:54:30 +00:00
|
|
|
} // namespace webrtc
|
2011-12-16 10:09:04 +00:00
|
|
|
|
2015-11-26 04:44:54 -08:00
|
|
|
#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_PCMFILE_H_
|