- initial import

2018-06-05 11:05:37 +03:00
commit e1a4931375
4673 changed files with 1383093 additions and 0 deletions
--- a/src/engine/media/MT_AudioStream.h
+++ b/src/engine/media/MT_AudioStream.h
@@ -0,0 +1,110 @@
+/* Copyright(C) 2007-2017 VoIPobjects (voipobjects.com)
+ * This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef __MT_AUDIOSTREAM_H
+#define __MT_AUDIOSTREAM_H
+
+#include "../config.h"
+#include "MT_Stream.h"
+#include "MT_NativeRtpSender.h"
+#include "MT_SingleAudioStream.h"
+#include "MT_Dtmf.h"
+#include "../helper/HL_VariantMap.h"
+#include "../helper/HL_ByteBuffer.h"
+#include "../helper/HL_NetworkSocket.h"
+#include "../helper/HL_Rtp.h"
+#include "../audio/Audio_DataWindow.h"
+#include "../audio/Audio_Mixer.h"
+#include "../audio/Audio_Resampler.h"
+#include "ice/ICESync.h"
+#include "jrtplib/src/rtpsession.h"
+#include "jrtplib/src/rtpexternaltransmitter.h"
+#include "audio/Audio_WavFile.h"
+
+namespace MT
+{
+  
+  class AudioStream: public Stream
+  {
+  public:
+    AudioStream(const CodecList::Settings& codecSettings);
+    ~AudioStream();
+
+    void setDestination(const RtpPair<InternetAddress>& dest) override;
+    //void setPacketTime(int packetTime);
+    
+    void setTransmittingCodec(Codec::Factory& factory, int payloadType) override;
+    PCodec transmittingCodec();
+
+    // Called to queue data captured from microphone. 
+    // Buffer holds 16bits PCM data with AUDIO_SAMPLERATE rate and AUDIO_CHANNELS channels.
+    void addData(const void* buffer, int length);
+    
+    // Called to get data to speaker (or mixer)
+    void copyDataTo(Audio::Mixer& mixer, int needed);
+
+    // Called to process incoming rtp packet
+    void dataArrived(PDatagramSocket s, const void* buffer, int length, InternetAddress& source) override;
+    void setSocket(const RtpPair<PDatagramSocket>& socket) override;
+    void setState(unsigned state) override;
+    
+    void setTelephoneCodec(int payloadType);
+    DtmfContext& queueOfDtmf();    
+
+    void readFile(const Audio::PWavFileReader& stream, MediaDirection direction) override;
+    void writeFile(const Audio::PWavFileWriter& writer, MediaDirection direction) override;
+    void setupMirror(bool enable) override;
+
+    void setFinalStatisticsOutput(Statistics* stats);
+
+  protected:
+    Audio::DataWindow mCapturedAudio;     // Data from microphone
+    Audio::DataWindow mStereoCapturedAudio;
+    char mIncomingPcmBuffer[AUDIO_MIC_BUFFER_SIZE]; // Temporary buffer to allow reading from file
+    char mResampleBuffer[AUDIO_MIC_BUFFER_SIZE*8];  // Temporary buffer to hold data
+    char mStereoBuffer[AUDIO_MIC_BUFFER_SIZE*8];    // Temporary buffer to hold data converted to stereo
+    PCodec mTransmittingCodec;                      // Current encoding codec
+    int mTransmittingPayloadType;                   // Payload type to mark outgoing packets
+    int mPacketTime;                                // Required packet time
+    char mFrameBuffer[MT_MAXAUDIOFRAME];            // Temporary buffer to hold results of encoder
+    ByteBuffer mEncodedAudio;                       // Encoded frame(s)
+    int mEncodedTime;                               // Time length of encoded audio
+    const CodecList::Settings& mCodecSettings;      // Configuration for stream
+    Mutex mMutex;                      	            // Mutex
+    int mRemoteTelephoneCodec;                      // Payload for remote telephone codec
+    jrtplib::RTPSession mRtpSession;                // Rtp session
+    jrtplib::RTPSession mRtpDtmfSession;            // Rtp dtmf session
+    NativeRtpSender mRtpSender;
+    AudioStreamMap mStreamMap;                      // Map of media streams. Key is RTP's SSRC value.
+    Audio::DataWindow mOutputBuffer;
+    RtpDump* mRtpDump;
+    Audio::Resampler  mCaptureResampler8,
+                      mCaptureResampler16,
+                      mCaptureResampler32,
+                      mCaptureResampler48;
+    DtmfContext mDtmfContext;  
+    char mReceiveBuffer[MAX_VALID_UDPPACKET_SIZE];
+
+    struct
+    {
+      Audio::PWavFileWriter  mStreamForRecordingIncoming,
+                      mStreamForRecordingOutgoing;
+      Audio::PWavFileReader  mStreamForReadingIncoming,
+                      mStreamForReadingOutgoing;
+    } mDumpStreams;
+
+    Audio::PWavFileWriter mSendingDump;
+
+    bool mMirror = false;
+    bool mMirrorPrebuffered = false;
+    Audio::DataWindow mMirrorBuffer;
+
+    Statistics* mFinalStatistics = nullptr;
+
+    bool decryptSrtp(void* data, int* len);
+  };
+};
+
+#endif