Line data    Source code

       1             : /*
       2             :  *  Copyright (C) 2004-2024 Savoir-faire Linux Inc.
       3             :  *
       4             :  *  Author: Tristan Matthews <tristan.matthews@savoirfairelinux.com>
       5             :  *  Author: Adrien Béraud <adrien.beraud@savoirfairelinux.com>
       6             :  *
       7             :  *  This program is free software; you can redistribute it and/or modify
       8             :  *  it under the terms of the GNU General Public License as published by
       9             :  *  the Free Software Foundation; either version 3 of the License, or
      10             :  *  (at your option) any later version.
      11             :  *
      12             :  *  This program is distributed in the hope that it will be useful,
      13             :  *  but WITHOUT ANY WARRANTY; without even the implied warranty of
      14             :  *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
      15             :  *  GNU General Public License for more details.
      16             :  *
      17             :  *  You should have received a copy of the GNU General Public License
      18             :  *  along with this program; if not, write to the Free Software
      19             :  *  Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA  02110-1301 USA.
      20             :  */
      21             : 
      22             : #pragma once
      23             : 
      24             : #include "media/media_device.h"
      25             : #include "media/rtp_session.h"
      26             : #include "media/media_stream.h"
      27             : 
      28             : #include "threadloop.h"
      29             : 
      30             : #include <string>
      31             : #include <memory>
      32             : 
      33             : namespace jami {
      34             : 
      35             : class AudioInput;
      36             : class AudioReceiveThread;
      37             : class AudioSender;
      38             : class IceSocket;
      39             : class MediaRecorder;
      40             : class RingBuffer;
      41             : 
      42             : struct RTCPInfo
      43             : {
      44             :     float packetLoss;
      45             :     unsigned int jitter;
      46             :     unsigned int nb_sample;
      47             :     float latency;
      48             : };
      49             : 
      50             : class AudioRtpSession : public RtpSession, public std::enable_shared_from_this<AudioRtpSession>
      51             : {
      52             : public:
      53             :     AudioRtpSession(const std::string& callId,
      54             :                     const std::string& streamId,
      55             :                     const std::shared_ptr<MediaRecorder>& rec);
      56             :     virtual ~AudioRtpSession();
      57             : 
      58             :     void start(std::unique_ptr<dhtnet::IceSocket> rtp_sock, std::unique_ptr<dhtnet::IceSocket> rtcp_sock) override;
      59             :     void restartSender() override;
      60             :     void stop() override;
      61             :     void setMuted(bool muted, Direction dir = Direction::SEND) override;
      62             : 
      63             :     void initRecorder() override;
      64             :     void deinitRecorder() override;
      65             : 
      66         238 :     std::shared_ptr<AudioInput>& getAudioLocal() { return audioInput_; }
      67         246 :     std::unique_ptr<AudioReceiveThread>& getAudioReceive() { return receiveThread_; }
      68             : 
      69             :     void setVoiceCallback(std::function<void(bool)> cb);
      70             : 
      71             : private:
      72             :     void startSender();
      73             :     void startReceiver();
      74             :     bool check_RCTP_Info_RR(RTCPInfo& rtcpi);
      75             :     void adaptQualityAndBitrate();
      76             :     void dropProcessing(RTCPInfo* rtcpi);
      77             :     void setNewPacketLoss(unsigned int newPL);
      78             :     float getPonderateLoss(float lastLoss);
      79             : 
      80             :     std::unique_ptr<AudioSender> sender_;
      81             :     std::unique_ptr<AudioReceiveThread> receiveThread_;
      82             :     std::shared_ptr<AudioInput> audioInput_;
      83             :     std::shared_ptr<RingBuffer> ringbuffer_;
      84             :     uint16_t initSeqVal_ {0};
      85             :     bool muteState_ {false};
      86             :     unsigned packetLoss_ {10};
      87             :     DeviceParams localAudioParams_;
      88             : 
      89             :     InterruptedThreadLoop rtcpCheckerThread_;
      90             :     void processRtcpChecker();
      91             : 
      92             :     // Interval in seconds between RTCP checking
      93         408 :     std::chrono::seconds rtcp_checking_interval {4};
      94             : 
      95             :     std::function<void(bool)> voiceCallback_;
      96             : 
      97             :     void attachRemoteRecorder(const MediaStream& ms);
      98             :     void attachLocalRecorder(const MediaStream& ms);
      99             : };
     100             : 
     101             : } // namespace jami