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
84 lines
2.5 KiB
C++
84 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);
|
|
|
|
virtual 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) OVERRIDE;
|
|
|
|
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();
|
|
|
|
virtual void Perform() OVERRIDE;
|
|
|
|
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_
|