mirror of
https://github.com/oxen-io/session-android.git
synced 2024-12-19 06:27:30 +00:00
118 lines
3.4 KiB
C++
118 lines
3.4 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_TESTSTEREO_H_
|
|
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTSTEREO_H_
|
|
|
|
#include <math.h>
|
|
|
|
#include "webrtc/system_wrappers/interface/scoped_ptr.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"
|
|
|
|
namespace webrtc {
|
|
|
|
enum StereoMonoMode {
|
|
kNotSet,
|
|
kMono,
|
|
kStereo
|
|
};
|
|
|
|
class TestPackStereo : public AudioPacketizationCallback {
|
|
public:
|
|
TestPackStereo();
|
|
~TestPackStereo();
|
|
|
|
void RegisterReceiverACM(AudioCodingModule* acm);
|
|
|
|
virtual int32_t SendData(const FrameType frame_type,
|
|
const uint8_t payload_type,
|
|
const uint32_t timestamp,
|
|
const uint8_t* payload_data,
|
|
const uint16_t payload_size,
|
|
const RTPFragmentationHeader* fragmentation);
|
|
|
|
uint16_t payload_size();
|
|
uint32_t timestamp_diff();
|
|
void reset_payload_size();
|
|
void set_codec_mode(StereoMonoMode mode);
|
|
void set_lost_packet(bool lost);
|
|
|
|
private:
|
|
AudioCodingModule* receiver_acm_;
|
|
int16_t seq_no_;
|
|
uint32_t timestamp_diff_;
|
|
uint32_t last_in_timestamp_;
|
|
uint64_t total_bytes_;
|
|
int payload_size_;
|
|
StereoMonoMode codec_mode_;
|
|
// Simulate packet losses
|
|
bool lost_packet_;
|
|
};
|
|
|
|
class TestStereo : public ACMTest {
|
|
public:
|
|
explicit TestStereo(int test_mode);
|
|
~TestStereo();
|
|
|
|
void Perform();
|
|
private:
|
|
// The default value of '-1' indicates that the registration is based only on
|
|
// codec name and a sampling frequncy matching is not required. This is useful
|
|
// for codecs which support several sampling frequency.
|
|
void RegisterSendCodec(char side, char* codec_name, int32_t samp_freq_hz,
|
|
int rate, int pack_size, int channels,
|
|
int payload_type);
|
|
|
|
void Run(TestPackStereo* channel, int in_channels, int out_channels,
|
|
int percent_loss = 0);
|
|
void OpenOutFile(int16_t test_number);
|
|
void DisplaySendReceiveCodec();
|
|
|
|
int32_t SendData(const FrameType frame_type, const uint8_t payload_type,
|
|
const uint32_t timestamp, const uint8_t* payload_data,
|
|
const uint16_t payload_size,
|
|
const RTPFragmentationHeader* fragmentation);
|
|
|
|
int test_mode_;
|
|
|
|
scoped_ptr<AudioCodingModule> acm_a_;
|
|
scoped_ptr<AudioCodingModule> acm_b_;
|
|
|
|
TestPackStereo* channel_a2b_;
|
|
|
|
PCMFile* in_file_stereo_;
|
|
PCMFile* in_file_mono_;
|
|
PCMFile out_file_;
|
|
int16_t test_cntr_;
|
|
uint16_t pack_size_samp_;
|
|
uint16_t pack_size_bytes_;
|
|
int counter_;
|
|
char* send_codec_name_;
|
|
|
|
// Payload types for stereo codecs and CNG
|
|
int g722_pltype_;
|
|
int l16_8khz_pltype_;
|
|
int l16_16khz_pltype_;
|
|
int l16_32khz_pltype_;
|
|
int pcma_pltype_;
|
|
int pcmu_pltype_;
|
|
int celt_pltype_;
|
|
int opus_pltype_;
|
|
int cn_8khz_pltype_;
|
|
int cn_16khz_pltype_;
|
|
int cn_32khz_pltype_;
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTSTEREO_H_
|