d7a71d0719
- This roll brings in VS2010 by default. The buildbots need updating (issue710). - We'll roll to 149181 later (past current Canary) to fix a Mac gyp issue: https://chromiumcodereview.appspot.com/10824105 - Chromium is now using a later libvpx than us. We should investigate rolling our standalone build. - Fix set-but-unused-warning - Fix -Wunused-private-field warnings on Mac. TBR=kjellander@webrtc.org BUG=issue709,issue710 TEST=trybots Review URL: https://webrtc-codereview.appspot.com/709007 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2544 4adac7df-926f-26a2-2b94-8c16560cd09d
115 lines
3.0 KiB
C++
115 lines
3.0 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 "ACMTest.h"
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#include "audio_coding_module.h"
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#include "RTPFile.h"
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#include "PCMFile.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, WebRtc_UWord16 frequency);
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~TestPacketization();
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virtual WebRtc_Word32 SendData(const FrameType frameType,
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const WebRtc_UWord8 payloadType,
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const WebRtc_UWord32 timeStamp,
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const WebRtc_UWord8* payloadData,
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const WebRtc_UWord16 payloadSize,
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const RTPFragmentationHeader* fragmentation);
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private:
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static void MakeRTPheader(WebRtc_UWord8* rtpHeader, WebRtc_UWord8 payloadType,
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WebRtc_Word16 seqNo, WebRtc_UWord32 timeStamp,
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WebRtc_UWord32 ssrc);
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RTPStream* _rtpStream;
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WebRtc_Word32 _frequency;
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WebRtc_Word16 _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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void Teardown();
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void Run();
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bool Add10MsData();
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bool Process();
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//for auto_test and logging
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WebRtc_UWord8 testMode;
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WebRtc_UWord8 codeId;
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private:
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AudioCodingModule* _acm;
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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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void Setup(AudioCodingModule *acm, RTPStream *rtpStream);
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void Teardown();
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void Run();
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bool IncomingPacket();
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bool PlayoutData();
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//for auto_test and logging
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WebRtc_UWord8 codeId;
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WebRtc_UWord8 testMode;
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private:
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AudioCodingModule* _acm;
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RTPStream* _rtpStream;
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PCMFile _pcmFile;
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WebRtc_Word16* _playoutBuffer;
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WebRtc_UWord16 _playoutLengthSmpls;
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WebRtc_UWord8 _incomingPayload[MAX_INCOMING_PAYLOAD];
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WebRtc_UWord16 _payloadSizeBytes;
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WebRtc_UWord16 _realPayloadSizeBytes;
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WebRtc_Word32 _frequency;
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bool _firstTime;
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WebRtcRTPHeader _rtpInfo;
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WebRtc_UWord32 _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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EncodeDecodeTest(int testMode);
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virtual void Perform();
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WebRtc_UWord16 _playoutFreq;
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WebRtc_UWord8 _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
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