ff761fba82
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
69 lines
1.8 KiB
C++
69 lines
1.8 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 WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_PCMFILE_H_
|
|
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_PCMFILE_H_
|
|
|
|
#include <stdio.h>
|
|
#include <stdlib.h>
|
|
|
|
#include <string>
|
|
|
|
#include "webrtc/modules/include/module_common_types.h"
|
|
#include "webrtc/typedefs.h"
|
|
|
|
namespace webrtc {
|
|
|
|
class PCMFile {
|
|
public:
|
|
PCMFile();
|
|
PCMFile(uint32_t timestamp);
|
|
~PCMFile() {
|
|
if (pcm_file_ != NULL) {
|
|
fclose(pcm_file_);
|
|
}
|
|
}
|
|
|
|
void Open(const std::string& filename, uint16_t frequency, const char* mode,
|
|
bool auto_rewind = false);
|
|
|
|
int32_t Read10MsData(AudioFrame& audio_frame);
|
|
|
|
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_;
|
|
}
|
|
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);
|
|
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_;
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_PCMFILE_H_
|