TheArchive/session-android
1
0
Fork 0
mirror of https://github.com/oxen-io/session-android.git synced 2025-12-23 15:46:48 +00:00
Code Issues Packages Projects Releases Wiki Activity

Support for Signal calls.

Browse Source 
Merge in RedPhone

// FREEBIE
This commit is contained in:
Moxie Marlinspike
2015-09-09 13:54:29 -07:00
parent 3d4ae60d81
commit d83a3d71bc
2585 changed files with 803492 additions and 45 deletions
Download Patch File Download Diff File Expand all files Collapse all files

81
jni/webrtc/modules/audio_coding/main/test/iSACTest.h Normal file
View File

@@ -0,0 +1,81 @@
/*
* 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_ISACTEST_H_
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ISACTEST_H_
#include <string.h>
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
#include "webrtc/modules/audio_coding/main/test/Channel.h"
#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
#include "webrtc/modules/audio_coding/main/test/utility.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#define MAX_FILE_NAME_LENGTH_BYTE 500
#define NO_OF_CLIENTS 15
namespace webrtc {
struct ACMTestISACConfig {
int32_t currentRateBitPerSec;
int16_t currentFrameSizeMsec;
uint32_t maxRateBitPerSec;
int16_t maxPayloadSizeByte;
int16_t encodingMode;
uint32_t initRateBitPerSec;
int16_t initFrameSizeInMsec;
bool enforceFrameSize;
};
class ISACTest : public ACMTest {
public:
explicit ISACTest(int testMode);
~ISACTest();
void Perform();
private:
void Setup();
void Run10ms();
void EncodeDecode(int testNr, ACMTestISACConfig& wbISACConfig,
ACMTestISACConfig& swbISACConfig);
void SwitchingSamplingRate(int testNr, int maxSampRateChange);
scoped_ptr<AudioCodingModule> _acmA;
scoped_ptr<AudioCodingModule> _acmB;
scoped_ptr<Channel> _channel_A2B;
scoped_ptr<Channel> _channel_B2A;
PCMFile _inFileA;
PCMFile _inFileB;
PCMFile _outFileA;
PCMFile _outFileB;
uint8_t _idISAC16kHz;
uint8_t _idISAC32kHz;
CodecInst _paramISAC16kHz;
CodecInst _paramISAC32kHz;
std::string file_name_swb_;
ACMTestTimer _myTimer;
int _testMode;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ISACTEST_H_