aa5ea1c0f9
2. Add two new APIs to configure codec internal FEC 3. Add a test and listened to results. This is based modifying EncodeDecodeTest and deriving a new class from it. New ACM gives good result. Old ACM does not use NetEq 4, so FEC won't be decoded. BUG= R=tina.legrand@webrtc.org, turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/11759004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6233 4adac7df-926f-26a2-2b94-8c16560cd09d
119 lines
3.2 KiB
C++
119 lines
3.2 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ENCODEDECODETEST_H_
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#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ENCODEDECODETEST_H_
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#include <stdio.h>
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#include <string.h>
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#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
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#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
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#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
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#include "webrtc/modules/audio_coding/main/test/RTPFile.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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#define MAX_INCOMING_PAYLOAD 8096
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// TestPacketization callback which writes the encoded payloads to file
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class TestPacketization : public AudioPacketizationCallback {
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public:
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TestPacketization(RTPStream *rtpStream, uint16_t frequency);
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~TestPacketization();
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virtual int32_t SendData(const FrameType frameType, const uint8_t payloadType,
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const uint32_t timeStamp, const uint8_t* payloadData,
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const uint16_t payloadSize,
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const RTPFragmentationHeader* fragmentation);
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private:
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static void MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType,
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int16_t seqNo, uint32_t timeStamp, uint32_t ssrc);
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RTPStream* _rtpStream;
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int32_t _frequency;
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int16_t _seqNo;
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};
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class Sender {
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public:
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Sender();
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void Setup(AudioCodingModule *acm, RTPStream *rtpStream,
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std::string in_file_name, int sample_rate, int channels);
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void Teardown();
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void Run();
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bool Add10MsData();
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//for auto_test and logging
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uint8_t testMode;
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uint8_t codeId;
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protected:
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AudioCodingModule* _acm;
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private:
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PCMFile _pcmFile;
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AudioFrame _audioFrame;
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TestPacketization* _packetization;
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};
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class Receiver {
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public:
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Receiver();
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virtual ~Receiver() {};
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void Setup(AudioCodingModule *acm, RTPStream *rtpStream,
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std::string out_file_name, int channels);
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void Teardown();
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void Run();
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virtual bool IncomingPacket();
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bool PlayoutData();
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//for auto_test and logging
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uint8_t codeId;
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uint8_t testMode;
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private:
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PCMFile _pcmFile;
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int16_t* _playoutBuffer;
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uint16_t _playoutLengthSmpls;
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int32_t _frequency;
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bool _firstTime;
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protected:
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AudioCodingModule* _acm;
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uint8_t _incomingPayload[MAX_INCOMING_PAYLOAD];
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RTPStream* _rtpStream;
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WebRtcRTPHeader _rtpInfo;
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uint16_t _realPayloadSizeBytes;
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uint16_t _payloadSizeBytes;
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uint32_t _nextTime;
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};
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class EncodeDecodeTest : public ACMTest {
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public:
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EncodeDecodeTest();
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explicit EncodeDecodeTest(int testMode);
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virtual void Perform();
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uint16_t _playoutFreq;
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uint8_t _testMode;
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private:
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void EncodeToFile(int fileType, int codeId, int* codePars, int testMode);
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protected:
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Sender _sender;
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Receiver _receiver;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ENCODEDECODETEST_H_
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