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Diffstat (limited to 'webrtc/modules/audio_coding/test/EncodeDecodeTest.h')
-rw-r--r-- | webrtc/modules/audio_coding/test/EncodeDecodeTest.h | 123 |
1 files changed, 123 insertions, 0 deletions
diff --git a/webrtc/modules/audio_coding/test/EncodeDecodeTest.h b/webrtc/modules/audio_coding/test/EncodeDecodeTest.h new file mode 100644 index 0000000000..f9a9a5bb52 --- /dev/null +++ b/webrtc/modules/audio_coding/test/EncodeDecodeTest.h @@ -0,0 +1,123 @@ +/* + * 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_TEST_ENCODEDECODETEST_H_ +#define WEBRTC_MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_ + +#include <stdio.h> +#include <string.h> + +#include "webrtc/modules/audio_coding/include/audio_coding_module.h" +#include "webrtc/modules/audio_coding/test/ACMTest.h" +#include "webrtc/modules/audio_coding/test/PCMFile.h" +#include "webrtc/modules/audio_coding/test/RTPFile.h" +#include "webrtc/typedefs.h" + +namespace webrtc { + +#define MAX_INCOMING_PAYLOAD 8096 + +// TestPacketization callback which writes the encoded payloads to file +class TestPacketization : public AudioPacketizationCallback { + public: + TestPacketization(RTPStream *rtpStream, uint16_t frequency); + ~TestPacketization(); + int32_t SendData(const FrameType frameType, + const uint8_t payloadType, + const uint32_t timeStamp, + const uint8_t* payloadData, + const size_t payloadSize, + const RTPFragmentationHeader* fragmentation) override; + + private: + static void MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType, + int16_t seqNo, uint32_t timeStamp, uint32_t ssrc); + RTPStream* _rtpStream; + int32_t _frequency; + int16_t _seqNo; +}; + +class Sender { + public: + Sender(); + void Setup(AudioCodingModule *acm, RTPStream *rtpStream, + std::string in_file_name, int sample_rate, size_t channels); + void Teardown(); + void Run(); + bool Add10MsData(); + + //for auto_test and logging + uint8_t testMode; + uint8_t codeId; + + protected: + AudioCodingModule* _acm; + + private: + PCMFile _pcmFile; + AudioFrame _audioFrame; + TestPacketization* _packetization; +}; + +class Receiver { + public: + Receiver(); + virtual ~Receiver() {}; + void Setup(AudioCodingModule *acm, RTPStream *rtpStream, + std::string out_file_name, size_t channels); + void Teardown(); + void Run(); + virtual bool IncomingPacket(); + bool PlayoutData(); + + //for auto_test and logging + uint8_t codeId; + uint8_t testMode; + + private: + PCMFile _pcmFile; + int16_t* _playoutBuffer; + uint16_t _playoutLengthSmpls; + int32_t _frequency; + bool _firstTime; + + protected: + AudioCodingModule* _acm; + uint8_t _incomingPayload[MAX_INCOMING_PAYLOAD]; + RTPStream* _rtpStream; + WebRtcRTPHeader _rtpInfo; + size_t _realPayloadSizeBytes; + size_t _payloadSizeBytes; + uint32_t _nextTime; +}; + +class EncodeDecodeTest : public ACMTest { + public: + EncodeDecodeTest(); + explicit EncodeDecodeTest(int testMode); + void Perform() override; + + uint16_t _playoutFreq; + uint8_t _testMode; + + private: + std::string EncodeToFile(int fileType, + int codeId, + int* codePars, + int testMode); + + protected: + Sender _sender; + Receiver _receiver; +}; + +} // namespace webrtc + +#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_ |