LCOV - code coverage report
Current view: top level - src/media/audio - audio_rtp_session.h (source / functions) Hit Total Coverage
Test: jami-coverage-filtered.info Lines: 3 3 100.0 %
Date: 2024-11-15 09:04:49 Functions: 2 2 100.0 %

          Line data    Source code
       1             : /*
       2             :  *  Copyright (C) 2004-2024 Savoir-faire Linux Inc.
       3             :  *
       4             :  *  This program is free software: you can redistribute it and/or modify
       5             :  *  it under the terms of the GNU General Public License as published by
       6             :  *  the Free Software Foundation, either version 3 of the License, or
       7             :  *  (at your option) any later version.
       8             :  *
       9             :  *  This program is distributed in the hope that it will be useful,
      10             :  *  but WITHOUT ANY WARRANTY; without even the implied warranty of
      11             :  *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
      12             :  *  GNU General Public License for more details.
      13             :  *
      14             :  *  You should have received a copy of the GNU General Public License
      15             :  *  along with this program. If not, see <https://www.gnu.org/licenses/>.
      16             :  */
      17             : #pragma once
      18             : 
      19             : #include "media/media_device.h"
      20             : #include "media/rtp_session.h"
      21             : #include "media/media_stream.h"
      22             : 
      23             : #include "threadloop.h"
      24             : 
      25             : #include <string>
      26             : #include <memory>
      27             : 
      28             : namespace jami {
      29             : 
      30             : class AudioInput;
      31             : class AudioReceiveThread;
      32             : class AudioSender;
      33             : class IceSocket;
      34             : class MediaRecorder;
      35             : class RingBuffer;
      36             : 
      37             : struct RTCPInfo
      38             : {
      39             :     float packetLoss;
      40             :     unsigned int jitter;
      41             :     unsigned int nb_sample;
      42             :     float latency;
      43             : };
      44             : 
      45             : class AudioRtpSession : public RtpSession, public std::enable_shared_from_this<AudioRtpSession>
      46             : {
      47             : public:
      48             :     AudioRtpSession(const std::string& callId,
      49             :                     const std::string& streamId,
      50             :                     const std::shared_ptr<MediaRecorder>& rec);
      51             :     virtual ~AudioRtpSession();
      52             : 
      53             :     void start(std::unique_ptr<dhtnet::IceSocket> rtp_sock, std::unique_ptr<dhtnet::IceSocket> rtcp_sock) override;
      54             :     void restartSender() override;
      55             :     void stop() override;
      56             :     void setMuted(bool muted, Direction dir = Direction::SEND) override;
      57             : 
      58             :     void initRecorder() override;
      59             :     void deinitRecorder() override;
      60             : 
      61         226 :     std::shared_ptr<AudioInput>& getAudioLocal() { return audioInput_; }
      62         234 :     std::unique_ptr<AudioReceiveThread>& getAudioReceive() { return receiveThread_; }
      63             : 
      64             :     void setVoiceCallback(std::function<void(bool)> cb);
      65             : 
      66             : private:
      67             :     void startSender();
      68             :     void startReceiver();
      69             :     bool check_RCTP_Info_RR(RTCPInfo& rtcpi);
      70             :     void adaptQualityAndBitrate();
      71             :     void dropProcessing(RTCPInfo* rtcpi);
      72             :     void setNewPacketLoss(unsigned int newPL);
      73             :     float getPonderateLoss(float lastLoss);
      74             : 
      75             :     std::unique_ptr<AudioSender> sender_;
      76             :     std::unique_ptr<AudioReceiveThread> receiveThread_;
      77             :     std::shared_ptr<AudioInput> audioInput_;
      78             :     std::shared_ptr<RingBuffer> ringbuffer_;
      79             :     uint16_t initSeqVal_ {0};
      80             :     bool muteState_ {false};
      81             :     unsigned packetLoss_ {10};
      82             :     DeviceParams localAudioParams_;
      83             : 
      84             :     InterruptedThreadLoop rtcpCheckerThread_;
      85             :     void processRtcpChecker();
      86             : 
      87             :     // Interval in seconds between RTCP checking
      88         397 :     std::chrono::seconds rtcp_checking_interval {4};
      89             : 
      90             :     std::function<void(bool)> voiceCallback_;
      91             : 
      92             :     void attachRemoteRecorder(const MediaStream& ms);
      93             :     void attachLocalRecorder(const MediaStream& ms);
      94             : };
      95             : 
      96             : } // namespace jami

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