mirror of
https://github.com/oxen-io/session-android.git
synced 2024-12-24 16:57:50 +00:00
83 lines
2.5 KiB
C++
83 lines
2.5 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_TESTALLCODECS_H_
|
|
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTALLCODECS_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/system_wrappers/interface/scoped_ptr.h"
|
|
#include "webrtc/typedefs.h"
|
|
|
|
namespace webrtc {
|
|
|
|
class Config;
|
|
|
|
class TestPack : public AudioPacketizationCallback {
|
|
public:
|
|
TestPack();
|
|
~TestPack();
|
|
|
|
void RegisterReceiverACM(AudioCodingModule* acm);
|
|
|
|
int32_t SendData(FrameType frame_type, uint8_t payload_type,
|
|
uint32_t timestamp, const uint8_t* payload_data,
|
|
uint16_t payload_size,
|
|
const RTPFragmentationHeader* fragmentation);
|
|
|
|
uint16_t payload_size();
|
|
uint32_t timestamp_diff();
|
|
void reset_payload_size();
|
|
|
|
private:
|
|
AudioCodingModule* receiver_acm_;
|
|
uint16_t sequence_number_;
|
|
uint8_t payload_data_[60 * 32 * 2 * 2];
|
|
uint32_t timestamp_diff_;
|
|
uint32_t last_in_timestamp_;
|
|
uint64_t total_bytes_;
|
|
uint16_t payload_size_;
|
|
};
|
|
|
|
class TestAllCodecs : public ACMTest {
|
|
public:
|
|
explicit TestAllCodecs(int test_mode);
|
|
~TestAllCodecs();
|
|
|
|
void Perform();
|
|
|
|
private:
|
|
// The default value of '-1' indicates that the registration is based only on
|
|
// codec name, and a sampling frequency matching is not required.
|
|
// This is useful for codecs which support several sampling frequency.
|
|
// Note! Only mono mode is tested in this test.
|
|
void RegisterSendCodec(char side, char* codec_name, int32_t sampling_freq_hz,
|
|
int rate, int packet_size, int extra_byte);
|
|
|
|
void Run(TestPack* channel);
|
|
void OpenOutFile(int test_number);
|
|
void DisplaySendReceiveCodec();
|
|
|
|
int test_mode_;
|
|
scoped_ptr<AudioCodingModule> acm_a_;
|
|
scoped_ptr<AudioCodingModule> acm_b_;
|
|
TestPack* channel_a_to_b_;
|
|
PCMFile infile_a_;
|
|
PCMFile outfile_b_;
|
|
int test_count_;
|
|
int packet_size_samples_;
|
|
int packet_size_bytes_;
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTALLCODECS_H_
|