LCOV - code coverage report
Current view: top level - src/media/audio - audio_rtp_session.h (source / functions) Coverage Total Hit
Test: jami-coverage-filtered.info Lines: 100.0 % 3 3
Test Date: 2026-06-13 09:18:46 Functions: 100.0 % 2 2

            Line data    Source code
       1              : /*
       2              :  *  Copyright (C) 2004-2026 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, const std::string& streamId, const std::shared_ptr<MediaRecorder>& rec);
      49              :     virtual ~AudioRtpSession();
      50              : 
      51              :     void start(std::unique_ptr<dhtnet::IceSocket> rtp_sock, std::unique_ptr<dhtnet::IceSocket> rtcp_sock) override;
      52              :     void restartSender() override;
      53              :     void stop() override;
      54              :     void setMuted(bool muted, Direction dir = Direction::SEND) override;
      55              : 
      56              :     void initRecorder() override;
      57              :     void deinitRecorder() override;
      58              : 
      59          329 :     std::shared_ptr<AudioInput>& getAudioLocal() { return audioInput_; }
      60          337 :     std::unique_ptr<AudioReceiveThread>& getAudioReceive() { return receiveThread_; }
      61              : 
      62              :     void setVoiceCallback(std::function<void(bool)> cb);
      63              : 
      64              : private:
      65              :     void startSender();
      66              :     void startReceiver();
      67              :     bool check_RCTP_Info_RR(RTCPInfo& rtcpi);
      68              :     void adaptQualityAndBitrate();
      69              :     void dropProcessing(RTCPInfo* rtcpi);
      70              :     void setNewPacketLoss(unsigned int newPL);
      71              :     float getPonderateLoss(float lastLoss);
      72              : 
      73              :     std::unique_ptr<AudioSender> sender_;
      74              :     std::unique_ptr<AudioReceiveThread> receiveThread_;
      75              :     std::shared_ptr<AudioInput> audioInput_;
      76              :     std::shared_ptr<RingBuffer> ringbuffer_;
      77              :     uint16_t initSeqVal_ {0};
      78              :     bool muteState_ {false};
      79              :     unsigned packetLoss_ {10};
      80              :     DeviceParams localAudioParams_;
      81              : 
      82              :     InterruptedThreadLoop rtcpCheckerThread_;
      83              :     void processRtcpChecker();
      84              : 
      85              :     // Interval in seconds between RTCP checking
      86          373 :     std::chrono::seconds rtcp_checking_interval {4};
      87              : 
      88              :     std::function<void(bool)> voiceCallback_;
      89              : 
      90              :     void attachRemoteRecorder(const MediaStream& ms);
      91              :     void attachLocalRecorder(const MediaStream& ms);
      92              : };
      93              : 
      94              : } // namespace jami
        

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