47658f1269
RTPStream, and NetEq as such. Also mark all other virtual overrides in the same files. This will make further changes to these classes safer by ensuring that the compile breaks if the base class changes and not all overrides are fixed. This also deletes ACMTest.cc, which existed solely to define ~ACMTest(), which was marked pure virtual in the header. (Pure virtual destructors still need a definition.) Because there is another pure virtual method in this class, the class is already abstract, so there's no benefit to making the desturctor pure. Making it non-pure allows removing the separate source file. BUG=none TEST=none R=henrik.lundin@webrtc.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/29389004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7144 4adac7df-926f-26a2-2b94-8c16560cd09d
119 lines
3.3 KiB
C++
119 lines
3.3 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) OVERRIDE;
|
|
|
|
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();
|
|
|
|
virtual void Perform() OVERRIDE;
|
|
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_
|