92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
124 lines
3.2 KiB
C++
124 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 MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_
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#define MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_
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#include <stdio.h>
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#include <string.h>
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#include "modules/audio_coding/include/audio_coding_module.h"
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#include "modules/audio_coding/test/ACMTest.h"
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#include "modules/audio_coding/test/PCMFile.h"
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#include "modules/audio_coding/test/RTPFile.h"
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#include "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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int32_t SendData(const FrameType frameType,
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const uint8_t payloadType,
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const uint32_t timeStamp,
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const uint8_t* payloadData,
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const size_t payloadSize,
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const RTPFragmentationHeader* fragmentation) override;
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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, size_t 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, size_t 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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size_t _realPayloadSizeBytes;
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size_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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void Perform() override;
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uint16_t _playoutFreq;
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uint8_t _testMode;
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private:
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std::string EncodeToFile(int fileType,
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int codeId,
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int* codePars,
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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 // MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_
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