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Diffstat (limited to 'webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h')
-rw-r--r-- | webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h | 123 |
1 files changed, 0 insertions, 123 deletions
diff --git a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h b/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h deleted file mode 100644 index 4ad92cec15..0000000000 --- a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h +++ /dev/null @@ -1,123 +0,0 @@ -/* - * 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_ENCODEDECODETEST_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ENCODEDECODETEST_H_ - -#include <stdio.h> -#include <string.h> - -#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h" -#include "webrtc/modules/audio_coding/main/test/ACMTest.h" -#include "webrtc/modules/audio_coding/main/test/PCMFile.h" -#include "webrtc/modules/audio_coding/main/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, int 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, int 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_MAIN_TEST_ENCODEDECODETEST_H_ |