Merge branch 'dev'

build_linux.py

@@ -19,7 +19,8 @@ def make_build() -> Path:
     os.mkdir(DIR_BUILD)
     os.chdir(DIR_BUILD)
 
-    cmd = f'cmake ../src -G Ninja'
+    # OPUS_X86_MAY_HAVE_SSE4_1 is for clang builds
+    cmd = f'cmake ../src -G Ninja -D OPUS_X86_MAY_HAVE_SSE4_1=ON'
     retcode = os.system(cmd)
     if retcode != 0:
         raise RuntimeError('Problem when configuring the project')

src/CMakeLists.txt

@@ -339,6 +339,8 @@ set (LIBS_STATIC ${LIBS_STATIC} jrtplib g729_codec gsm_codec opus
 
 if (USE_AMR_CODEC)
     #include (${LIB_PLATFORM}/platform_libs.cmake)
+    set (OPENCORE_AMRNB opencore-amrnb)
+    set (OPENCORE_AMRWB opencore-amrwb)
     message("Media: AMR NB and WB codecs will be included.")
     set (DEFINES ${DEFINES} -DUSE_AMR_CODEC)
     set (LIBS_STATIC ${LIBS_STATIC} ${OPENCORE_AMRNB} ${OPENCORE_AMRWB})

src/engine/agent/Agent_Impl.cpp

@@ -16,6 +16,7 @@ const std::string Status_FailedToOpenFile = "failed to open file";
 const std::string Status_NoActiveProvider = "no active provider";
 const std::string Status_NoMediaAction = "no valid media action";
 const std::string Status_NoCommand = "no valid command";
+const std::string Status_NoAudioManager = "no audio manager";
 
 #define LOG_SUBSYSTEM "Agent"
 
@@ -336,7 +337,7 @@ void AgentImpl::processStartSession(JsonCpp::Value& request, JsonCpp::Value& ans
     {
         // Agent was not started
         ICELogError(<< "No audio manager installed.");
-        answer["status"] = "Audio manager not started. Most probably agent is not started.";
+        answer["status"] = Status_NoAudioManager;
         return;
     }
 
@@ -431,28 +432,35 @@ void AgentImpl::processAcceptSession(JsonCpp::Value& request, JsonCpp::Value& an
     auto sessionIter = mSessionMap.find(request["session_id"].asInt());
     if (sessionIter != mSessionMap.end())
     {
-        // Ensure audio manager is here
-        mAudioManager->start(mUseNativeAudio ? AudioManager::atReceiver : AudioManager::atNull);
+        if (!mAudioManager)
+        {
+            ICELogError(<< "No audio manager installed.");
+            answer["status"] = Status_NoAudioManager;
+        }
+        else
+        {
+            // Ensure audio manager is here
+            mAudioManager->start(mUseNativeAudio ? AudioManager::atReceiver : AudioManager::atNull);
 
-        // Accept session on SIP level
-        PSession session = sessionIter->second;
+            // Accept session on SIP level
+            PSession session = sessionIter->second;
 
-        // Get user headers
-        Session::UserHeaders info;
-        JsonCpp::Value& arg = request["userinfo"];
-        std::vector<std::string> keys = arg.getMemberNames();
-        for (const std::string& k: keys)
-            info[k] = arg[k].asString();
-        session->setUserHeaders(info);
+            // Get user headers
+            Session::UserHeaders info;
+            JsonCpp::Value& arg = request["userinfo"];
+            std::vector<std::string> keys = arg.getMemberNames();
+            for (const std::string& k: keys)
+                info[k] = arg[k].asString();
+            session->setUserHeaders(info);
 
-        // Accept finally
-        session->accept();
+            // Accept finally
+            session->accept();
 
-        answer["status"] = Status_Ok;
+            answer["status"] = Status_Ok;
+        }
     }
     else
         answer["status"] = Status_SessionNotFound;
-
 }
 
 void AgentImpl::processDestroySession(JsonCpp::Value& request, JsonCpp::Value& answer)
@@ -526,6 +534,8 @@ void AgentImpl::processGetMediaStats(JsonCpp::Value& request, JsonCpp::Value& an
             answer["rtt"] = result[SessionInfo_Rtt].asFloat();
         if (result.exists(SessionInfo_BitrateSwitchCounter))
             answer["bitrate_switch_counter"] = result[SessionInfo_BitrateSwitchCounter].asInt();
+        if (result.exists(SessionInfo_CngCounter))
+            answer["cng_counter"] = result[SessionInfo_CngCounter].asInt();
         if (result.exists(SessionInfo_SSRC))
             answer["rtp_ssrc"] = result[SessionInfo_SSRC].asInt();
         if (result.exists(SessionInfo_RemotePeer))
@@ -685,7 +695,7 @@ void AgentImpl::processUseStreamForSession(JsonCpp::Value& request, JsonCpp::Val
                 answer["status"] = Status_Ok;
             }
             else
-                answer["status"] = Status_AccountNotFound;
+                answer["status"] = Status_NoCommand;
         }
         else
             answer["status"] = Status_NoMediaAction;
@@ -707,11 +717,13 @@ void AgentImpl::onMedia(const void* data, int length, MT::Stream::MediaDirection
 
 PDataProvider AgentImpl::onProviderNeeded(const std::string& name)
 {
+    assert(mTerminal);
+
     EVENT_WITH_NAME("provider_needed");
     v["provider_name"] = name;
     addEvent(v);
 
-    return PDataProvider(new AudioProvider(*this, *mTerminal));
+    return std::make_shared<AudioProvider>(*this, *mTerminal);
 }
 
 // Called on new session offer

src/engine/audio/Audio_DataWindow.cpp

@@ -11,21 +11,31 @@ using namespace Audio;
 DataWindow::DataWindow()
 {
     mFilled = 0;
-    mData = NULL;
+    mData = nullptr;
     mCapacity = 0;
 }
 
 DataWindow::~DataWindow()
 {
     if (mData)
+    {
         free(mData);
+        mData = nullptr;
+    }
 }
 
 void DataWindow::setCapacity(int capacity)
 {
     Lock l(mMutex);
     int tail  = capacity - mCapacity;
+    char* buffer = mData;
     mData = (char*)realloc(mData, capacity);
+    if (!mData)
+    {
+        // Realloc failed
+        mData = buffer;
+        throw std::bad_alloc();
+    }
     if (tail > 0)
         memset(mData + mCapacity, 0, tail);
     mCapacity = capacity;
@@ -166,6 +176,25 @@ void DataWindow::zero(int length)
     memset(mData, 0, mFilled);
 }
 
+size_t DataWindow::moveTo(DataWindow& dst, size_t size)
+{
+    Lock l(mMutex);
+
+    size_t avail = std::min(size, (size_t)filled());
+    if (avail != 0)
+    {
+        dst.add(mData, avail);
+        erase(avail);
+    }
+    return avail;
+}
+
+std::chrono::milliseconds DataWindow::getTimeLength(int samplerate, int channels) const
+{
+    Lock l(mMutex);
+    return std::chrono::milliseconds(mFilled / sizeof(short) / channels / (samplerate / 1000));
+}
+
 void DataWindow::makeStereoFromMono(DataWindow& dst, DataWindow& src)
 {
     Lock lockDst(dst.mMutex), lockSrc(src.mMutex);

src/engine/audio/Audio_DataWindow.h

@@ -11,37 +11,40 @@
 
 namespace Audio
 {
-  class DataWindow
-  {
-  public:
+class DataWindow
+{
+public:
     DataWindow();
     ~DataWindow();
 
-    void setCapacity(int capacity);
-    int capacity() const;
+    void        setCapacity(int capacity);
+    int         capacity() const;
 
-    void addZero(int length);
-    void add(const void* data, int length);
-    void add(short sample);
-    int read(void* buffer, int length);
-    void erase(int length = -1);
+    void        addZero(int length);
+    void        add(const void* data, int length);
+    void        add(short sample);
+    int         read(void* buffer, int length);
+    void        erase(int length = -1);
     const char* data() const;
-    char* mutableData();
-    int filled() const;
-    void setFilled(int filled);
-    void clear();
+    char*       mutableData();
+    int         filled() const;
+    void        setFilled(int filled);
+    void        clear();
 
-    short shortAt(int index) const;
-    void setShortAt(short value, int index);
-    void zero(int length);
+    short       shortAt(int index) const;
+    void        setShortAt(short value, int index);
+    void        zero(int length);
+    size_t      moveTo(DataWindow& dst, size_t size);
+
+    std::chrono::milliseconds getTimeLength(int samplerate, int channels) const;
 
     static void makeStereoFromMono(DataWindow& dst, DataWindow& src);
 
-  protected:
+protected:
     mutable Mutex mMutex;
     char* mData;
     int mFilled;
     int mCapacity;
-  };
+};
 }
 #endif

src/engine/audio/Audio_Interface.h

@@ -51,6 +51,11 @@ struct Format
         return float((milliseconds * mRate) / 500.0 * mChannels);
     }
 
+    size_t sizeFromTime(std::chrono::milliseconds ms) const
+    {
+        return sizeFromTime(ms.count());
+    }
+
     std::string toString()
     {
         char buffer[64];

src/engine/endpoint/EP_Session.cpp

@@ -483,6 +483,7 @@ void Session::getSessionInfo(Session::InfoOptions options, VariantMap& info)
         info[SessionInfo_Rtt] = static_cast<float>(stat.mRttDelay * 1000);
 #if defined(USE_AMR_CODEC)
     info[SessionInfo_BitrateSwitchCounter] = stat.mBitrateSwitchCounter;
+    info[SessionInfo_CngCounter] = stat.mCng;
 #endif
     info[SessionInfo_SSRC] = stat.mSsrc;
     info[SessionInfo_RemotePeer] = stat.mRemotePeer.toStdString();
@@ -882,7 +883,7 @@ void Session::refreshMediaPath()
 
         // Bring new socket to provider and stream
         RtpPair<PDatagramSocket> s4 = SocketHeap::instance().allocSocketPair(AF_INET, this, IS_MULTIPLEX() ),
-                s6 = SocketHeap::instance().allocSocketPair(AF_INET, this, IS_MULTIPLEX());
+                                 s6 = SocketHeap::instance().allocSocketPair(AF_INET6, this, IS_MULTIPLEX());
 
         p->setSocket(s4, s6);
         s.setSocket4(s4);

src/engine/endpoint/EP_Session.h

@@ -72,6 +72,7 @@ enum SessionInfo
   SessionInfo_BitrateSwitchCounter, // It is for AMR codecs only
   SessionInfo_RemotePeer,
   SessionInfo_SSRC,
+  SessionInfo_CngCounter            // For AMR codecs only
 };
 
 

src/engine/engine_config.h

@@ -38,11 +38,11 @@
 //#define AUDIO_DUMPOUTPUT
 
 
-#define UA_REGISTRATION_TIME 3600
-#define UA_MEDIA_PORT_START  20000
-#define UA_MEDIA_PORT_FINISH 30000
-#define UA_MAX_UDP_PACKET_SIZE 576
-#define UA_PUBLICATION_ID "314"
+#define UA_REGISTRATION_TIME    3600
+#define UA_MEDIA_PORT_START     20000
+#define UA_MEDIA_PORT_FINISH    30000
+#define UA_MAX_UDP_PACKET_SIZE  576
+#define UA_PUBLICATION_ID       "314"
 
 #define MT_SAMPLERATE AUDIO_SAMPLERATE
 
@@ -50,13 +50,11 @@
 #define MT_MAXRTPPACKET      1500
 #define MT_DTMF_END_PACKETS  3
 
-#define RTP_BUFFER_HIGH      0
-#define RTP_BUFFER_LOW       0
-#define RTP_BUFFER_PREBUFFER 0
+// Milliseconds before
+#define RTP_BUFFER_HIGH      (2000)
+#define RTP_BUFFER_LOW       (0)
+#define RTP_BUFFER_PREBUFFER (100)
 
-// #define RTP_BUFFER_HIGH      160
-// #define RTP_BUFFER_LOW       10
-// #define RTP_BUFFER_PREBUFFER 160
 #define RTP_DECODED_CAPACITY 2048
 
 #define DEFAULT_SUBSCRIPTION_TIME 1200

src/engine/helper/HL_NetworkSocket.cpp

@@ -1,4 +1,4 @@
-/* Copyright(C) 2007-2017 VoIP objects (voipobjects.com)
+/* Copyright(C) 2007-2026 VoIP objects (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/. */
@@ -9,6 +9,7 @@
 
 #include "../engine_config.h"
 #include "HL_NetworkSocket.h"
+#include "HL_Log.h"
 
 #if defined(TARGET_OSX) || defined(TARGET_LINUX)
 # include <fcntl.h>
@@ -19,11 +20,11 @@
 #endif
 #include <assert.h>
 
+#define LOG_SUBSYSTEM "network"
+
 DatagramSocket::DatagramSocket()
     :mFamily(AF_INET), mHandle(INVALID_SOCKET), mLocalPort(0)
-{
-
-}
+{}
 
 DatagramSocket::~DatagramSocket()
 {
@@ -160,6 +161,12 @@ void DatagramAgreggator::addSocket(PDatagramSocket socket)
     if (socket->mHandle == INVALID_SOCKET)
         return;
 
+    if (mSocketVector.size() >= 62)
+    {
+        ICELogError(<< "fd_set overflow; too much sockets");
+        return;
+    }
+
     FD_SET(socket->mHandle, &mReadSet);
     if (socket->mHandle > mMaxHandle)
         mMaxHandle = socket->mHandle;

src/engine/helper/HL_Rtp.cpp

@@ -1,4 +1,4 @@
-/* Copyright(C) 2007-2017 VoIPobjects (voipobjects.com)
+/* Copyright(C) 2007-2026 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/. */
@@ -8,70 +8,95 @@
 # include <Windows.h>
 #endif
 
-#if defined(TARGET_LINUX) || defined(TARGET_ANDROID)
+#if defined(TARGET_LINUX) || defined(TARGET_ANDROID) || defined(TARGET_OSX)
 # include <arpa/inet.h>
 #endif
 
 #include "HL_Rtp.h"
 #include "HL_Exception.h"
-#include "HL_String.h"
+#include "HL_Log.h"
 
-#if defined(USE_RTP_DUMP)
-# include "jrtplib/src/rtprawpacket.h"
-# include "jrtplib/src/rtpipv4address.h"
-#endif
+#include "jrtplib/src/rtprawpacket.h"
+#include "jrtplib/src/rtpipv4address.h"
 
-#if !defined(TARGET_WIN)
-# include <alloca.h>
-#endif
+#include <stdexcept>
+#include <fstream>
+#include <cstring>
+#include <cstdio>
+#include <chrono>
 
-#include <sstream>
-#include <tuple>
+#define LOG_SUBSYSTEM "RtpDump"
 
+static constexpr size_t MAX_RTP_PACKET_SIZE = 65535;
+static const char RTPDUMP_SHEBANG[] = "#!rtpplay1.0";
+
+// RTP fixed header (little-endian bit-field layout)
 struct RtpHeader
 {
     unsigned char cc:4;         /* CSRC count             */
     unsigned char x:1;          /* header extension flag  */
     unsigned char p:1;          /* padding flag           */
-    unsigned char version:2;	 /* protocol version       */
-    unsigned char pt:7;	     /* payload type           */
-    unsigned char m:1;		     /* marker bit             */
-    unsigned short seq;		 /* sequence number        */
-    unsigned int ts;		     /* timestamp              */
-    unsigned int ssrc;	         /* synchronization source */
+    unsigned char version:2;    /* protocol version       */
+    unsigned char pt:7;         /* payload type           */
+    unsigned char m:1;          /* marker bit             */
+    unsigned short seq;         /* sequence number        */
+    unsigned int ts;            /* timestamp              */
+    unsigned int ssrc;          /* synchronization source */
 };
 
 struct RtcpHeader
 {
-    unsigned char rc:5;		  /* reception report count */
+    unsigned char rc:5;         /* reception report count */
     unsigned char p:1;          /* padding flag           */
     unsigned char version:2;    /* protocol version       */
-    unsigned char pt:8;         /* payload type           */
+    unsigned char pt;           /* payload type           */
     uint16_t len;               /* length                 */
-    uint32_t ssrc;	       	     /* synchronization source */
+    uint32_t ssrc;              /* synchronization source */
 };
 
+// --- IPv4 address helpers ---
+
+static std::string ipToString(uint32_t ip)
+{
+    // ip in host byte order → dotted-decimal
+    return std::to_string((ip >> 24) & 0xFF) + "." +
+           std::to_string((ip >> 16) & 0xFF) + "." +
+           std::to_string((ip >>  8) & 0xFF) + "." +
+           std::to_string( ip        & 0xFF);
+}
+
+static uint32_t stringToIp(const std::string& s)
+{
+    unsigned a = 0, b = 0, c = 0, d = 0;
+    if (std::sscanf(s.c_str(), "%u.%u.%u.%u", &a, &b, &c, &d) != 4)
+        return 0;
+    if (a > 255 || b > 255 || c > 255 || d > 255)
+        return 0;
+    return (a << 24) | (b << 16) | (c << 8) | d;
+}
+
+// --- RtpHelper implementation ---
+
 bool RtpHelper::isRtp(const void* buffer, size_t length)
 {
     if (length < 12)
         return false;
 
     const RtpHeader* h = reinterpret_cast<const RtpHeader*>(buffer);
-    if (h->version != 0b10)
+    if (h->version != 2)
         return false;
 
     unsigned char pt = h->pt;
-    bool rtp = ( (pt & 0x7F) >= 96 && (pt & 0x7F) <= 127) || ((pt & 0x7F) < 35);
+    bool rtp = (pt >= 96 && pt <= 127) || (pt < 35);
     return rtp;
 }
 
-
 bool RtpHelper::isRtpOrRtcp(const void* buffer, size_t length)
 {
     if (length < 12)
         return false;
     const RtcpHeader* h = reinterpret_cast<const RtcpHeader*>(buffer);
-    return h->version == 0b10;
+    return h->version == 2;
 }
 
 bool RtpHelper::isRtcp(const void* buffer, size_t length)
@@ -83,15 +108,16 @@ unsigned RtpHelper::findSsrc(const void* buffer, size_t length)
 {
     if (isRtp(buffer, length))
         return ntohl(reinterpret_cast<const RtpHeader*>(buffer)->ssrc);
-    else
+    else if (isRtpOrRtcp(buffer, length))
         return ntohl(reinterpret_cast<const RtcpHeader*>(buffer)->ssrc);
+    return 0;
 }
 
 void RtpHelper::setSsrc(void* buffer, size_t length, uint32_t ssrc)
 {
     if (isRtp(buffer, length))
         reinterpret_cast<RtpHeader*>(buffer)->ssrc = htonl(ssrc);
-    else
+    else if (isRtpOrRtcp(buffer, length))
         reinterpret_cast<RtcpHeader*>(buffer)->ssrc = htonl(ssrc);
 }
 
@@ -113,47 +139,191 @@ int RtpHelper::findPacketNo(const void *buffer, size_t length)
 
 int RtpHelper::findPayloadLength(const void* buffer, size_t length)
 {
-    if (isRtp(buffer, length))
-    {
-        return length - 12;
-    }
-    else
+    if (!isRtp(buffer, length))
         return -1;
+
+    const RtpHeader* h = reinterpret_cast<const RtpHeader*>(buffer);
+    const uint8_t* p = static_cast<const uint8_t*>(buffer);
+
+    // Fixed header (12 bytes) + CSRC list (4 * CC bytes)
+    size_t offset = 12 + 4u * h->cc;
+    if (offset > length)
+        return -1;
+
+    // Header extension
+    if (h->x) {
+        if (offset + 4 > length)
+            return -1;
+        uint16_t extWords = (static_cast<uint16_t>(p[offset + 2]) << 8) | p[offset + 3];
+        offset += 4 + 4u * extWords;
+        if (offset > length)
+            return -1;
+    }
+
+    size_t payloadLen = length - offset;
+
+    // Padding
+    if (h->p && payloadLen > 0) {
+        uint8_t padBytes = p[length - 1];
+        if (padBytes > payloadLen)
+            return -1;
+        payloadLen -= padBytes;
+    }
+
+    return static_cast<int>(payloadLen);
 }
 
-#if defined(USE_RTPDUMP)
-RtpDump::RtpDump(const char *filename)
-    :mFilename(filename)
-{}
-
-RtpDump::~RtpDump()
+std::chrono::microseconds RtpHelper::toMicroseconds(const jrtplib::RTPTime& t)
 {
-    flush();
-    for (PacketList::iterator packetIter=mPacketList.begin(); packetIter!=mPacketList.end(); ++packetIter)
-    {
-        //free(packetIter->mData);
-        delete packetIter->mPacket;
+    return std::chrono::microseconds(uint64_t(t.GetDouble() * 1000000));
+}
+
+// --- RtpDump implementation ---
+
+std::shared_ptr<jrtplib::RTPPacket> RtpDump::parseRtpData(const uint8_t* data, size_t len)
+{
+    if (!data || len < 12 || !RtpHelper::isRtp(data, len))
+        return nullptr;
+
+    try {
+        // Both are heap-allocated; RTPRawPacket takes ownership and deletes them
+        auto* addr = new jrtplib::RTPIPv4Address(uint32_t(0), uint16_t(0));
+        uint8_t* dataCopy = new uint8_t[len];
+        std::memcpy(dataCopy, data, len);
+
+        jrtplib::RTPRawPacket raw(dataCopy, len, addr, jrtplib::RTPTime(0), true);
+        auto packet = std::make_shared<jrtplib::RTPPacket>(raw);
+
+        if (packet->GetCreationError() != 0)
+            return nullptr;
+
+        return packet;
+    } catch (const std::exception& e) {
+        ICELogInfo(<< "Failed to parse RTP packet: " << e.what());
+        return nullptr;
     }
 }
 
+RtpDump::RtpDump(const char* filename)
+    : mFilename(filename ? filename : "")
+{
+}
+
+RtpDump::~RtpDump() = default;
+
+void RtpDump::setSource(uint32_t ip, uint16_t port)
+{
+    mSourceIp = ip;
+    mSourcePort = port;
+}
+
 void RtpDump::load()
 {
-    FILE* f = fopen(mFilename.c_str(), "rb");
-    if (!f)
-        throw Exception(ERR_WAVFILE_FAILED);
+    if (mFilename.empty())
+        throw std::runtime_error("No filename specified");
 
-    while (!feof(f))
-    {
-        RtpData data;
-        fread(&data.mLength, sizeof data.mLength, 1, f);
-        data.mData = new char[data.mLength];
-        fread(data.mData, 1, data.mLength, f);
-        jrtplib::RTPIPv4Address addr(jrtplib::RTPAddress::IPv4Address);
-        jrtplib::RTPTime t(0);
-        jrtplib::RTPRawPacket* raw = new jrtplib::RTPRawPacket((unsigned char*)data.mData, data.mLength, &addr, t, true);
-        data.mPacket = new jrtplib::RTPPacket(*raw);
-        mPacketList.push_back(data);
+    std::ifstream input(mFilename, std::ios::binary);
+    if (!input.is_open())
+        throw std::runtime_error("Failed to open RTP dump file: " + mFilename);
+
+    mPacketList.clear();
+
+    // --- 1. Text header: "#!rtpplay1.0 <ip>/<port>\n" ---
+    std::string textLine;
+    std::getline(input, textLine);
+    if (textLine.compare(0, sizeof(RTPDUMP_SHEBANG) - 1, RTPDUMP_SHEBANG) != 0)
+        throw std::runtime_error("Invalid rtpdump header: expected " + std::string(RTPDUMP_SHEBANG));
+
+    // Parse source address from the text line
+    size_t spacePos = textLine.find(' ');
+    if (spacePos != std::string::npos) {
+        std::string addrPart = textLine.substr(spacePos + 1);
+        size_t slashPos = addrPart.find('/');
+        if (slashPos != std::string::npos) {
+            mSourceIp = stringToIp(addrPart.substr(0, slashPos));
+            try {
+                mSourcePort = static_cast<uint16_t>(std::stoi(addrPart.substr(slashPos + 1)));
+            } catch (...) {
+                mSourcePort = 0;
+            }
+        }
     }
+
+    // --- 2. Binary file header (RD_hdr_t, 16 bytes) ---
+    uint32_t buf32;
+    uint16_t buf16;
+
+    input.read(reinterpret_cast<char*>(&buf32), 4);
+    mStartSec = ntohl(buf32);
+
+    input.read(reinterpret_cast<char*>(&buf32), 4);
+    mStartUsec = ntohl(buf32);
+
+    input.read(reinterpret_cast<char*>(&buf32), 4);  // source IP (already NBO in file)
+    // The binary header stores IP in network byte order; convert to host
+    mSourceIp = ntohl(buf32);
+
+    input.read(reinterpret_cast<char*>(&buf16), 2);
+    mSourcePort = ntohs(buf16);
+
+    input.read(reinterpret_cast<char*>(&buf16), 2);  // padding — discard
+
+    if (!input.good())
+        throw std::runtime_error("Failed to read rtpdump binary header");
+
+    // --- 3. Packet records ---
+    size_t packetCount = 0;
+
+    while (input.good() && input.peek() != EOF) {
+        // Packet header: length(2) + plen(2) + offset(4) = 8 bytes
+        uint16_t recLength, plen;
+        uint32_t offsetMs;
+
+        input.read(reinterpret_cast<char*>(&recLength), 2);
+        if (input.gcount() != 2) break;
+        recLength = ntohs(recLength);
+
+        input.read(reinterpret_cast<char*>(&plen), 2);
+        if (input.gcount() != 2) break;
+        plen = ntohs(plen);
+
+        input.read(reinterpret_cast<char*>(&offsetMs), 4);
+        if (input.gcount() != 4) break;
+        offsetMs = ntohl(offsetMs);
+
+        // All-zeros record signals end of file in some implementations
+        if (recLength == 0 && plen == 0 && offsetMs == 0)
+            break;
+
+        if (plen == 0 || plen > MAX_RTP_PACKET_SIZE)
+            throw std::runtime_error("Invalid packet payload length: " + std::to_string(plen));
+
+        if (recLength < plen + 8)
+            throw std::runtime_error("Record length (" + std::to_string(recLength) +
+                                     ") smaller than payload + header (" + std::to_string(plen + 8) + ")");
+
+        // Read body
+        std::vector<uint8_t> body(plen);
+        input.read(reinterpret_cast<char*>(body.data()), plen);
+        if (static_cast<size_t>(input.gcount()) != plen)
+            throw std::runtime_error("Incomplete packet data in rtpdump file");
+
+        // Skip any padding between plen and recLength-8
+        size_t pad = static_cast<size_t>(recLength) - 8 - plen;
+        if (pad > 0)
+            input.seekg(static_cast<std::streamoff>(pad), std::ios::cur);
+
+        RtpData entry;
+        entry.mRawData = std::move(body);
+        entry.mOffsetMs = offsetMs;
+        entry.mPacket = parseRtpData(entry.mRawData.data(), entry.mRawData.size());
+
+        mPacketList.push_back(std::move(entry));
+        packetCount++;
+    }
+
+    ICELogInfo(<< "Loaded " << packetCount << " packets from " << mFilename);
+    mLoaded = true;
 }
 
 size_t RtpDump::count() const
@@ -163,39 +333,142 @@ size_t RtpDump::count() const
 
 jrtplib::RTPPacket& RtpDump::packetAt(size_t index)
 {
+    if (index >= mPacketList.size())
+        throw std::out_of_range("Packet index out of range: " + std::to_string(index));
+
+    if (!mPacketList[index].mPacket)
+        throw std::runtime_error("No parsed RTP data at index " + std::to_string(index));
+
     return *mPacketList[index].mPacket;
 }
 
+const std::vector<uint8_t>& RtpDump::rawDataAt(size_t index) const
+{
+    if (index >= mPacketList.size())
+        throw std::out_of_range("Packet index out of range: " + std::to_string(index));
+
+    return mPacketList[index].mRawData;
+}
+
+uint32_t RtpDump::offsetAt(size_t index) const
+{
+    if (index >= mPacketList.size())
+        throw std::out_of_range("Packet index out of range: " + std::to_string(index));
+
+    return mPacketList[index].mOffsetMs;
+}
+
 void RtpDump::add(const void* buffer, size_t len)
 {
-    RtpData data;
-    data.mData = malloc(len);
-    memcpy(data.mData, buffer, len);
-    data.mLength = len;
+    if (!buffer || len == 0)
+        return;
 
-    jrtplib::RTPIPv4Address addr(jrtplib::RTPAddress::IPv4Address);
-    jrtplib::RTPTime t(0);
-    jrtplib::RTPRawPacket* raw = new jrtplib::RTPRawPacket((unsigned char*)const_cast<void*>(data.mData), data.mLength, &addr, t, true);
-    data.mPacket = new jrtplib::RTPPacket(*raw);
-    //delete raw;
-    mPacketList.push_back(data);
+    uint32_t offsetMs = 0;
+    auto now = std::chrono::steady_clock::now();
+
+    if (!mRecording) {
+        mRecording = true;
+        mRecordStart = now;
+
+        // Capture wall-clock start time
+        auto wallNow = std::chrono::system_clock::now();
+        auto epoch = wallNow.time_since_epoch();
+        auto sec = std::chrono::duration_cast<std::chrono::seconds>(epoch);
+        auto usec = std::chrono::duration_cast<std::chrono::microseconds>(epoch - sec);
+        mStartSec  = static_cast<uint32_t>(sec.count());
+        mStartUsec = static_cast<uint32_t>(usec.count());
+    } else {
+        auto elapsed = std::chrono::duration_cast<std::chrono::milliseconds>(now - mRecordStart);
+        offsetMs = static_cast<uint32_t>(elapsed.count());
+    }
+
+    add(buffer, len, offsetMs);
+}
+
+void RtpDump::add(const void* buffer, size_t len, uint32_t offsetMs)
+{
+    if (!buffer || len == 0)
+        return;
+
+    if (len > MAX_RTP_PACKET_SIZE)
+        throw std::runtime_error("Packet too large: " + std::to_string(len));
+
+    RtpData entry;
+    entry.mRawData.assign(static_cast<const uint8_t*>(buffer),
+                          static_cast<const uint8_t*>(buffer) + len);
+    entry.mOffsetMs = offsetMs;
+    entry.mPacket = parseRtpData(entry.mRawData.data(), entry.mRawData.size());
+
+    mPacketList.push_back(std::move(entry));
 }
 
 void RtpDump::flush()
 {
-    FILE* f = fopen(mFilename.c_str(), "wb");
-    if (!f)
-        throw Exception(ERR_WAVFILE_FAILED);
+    if (mFilename.empty())
+        throw std::runtime_error("No filename specified");
 
-    PacketList::iterator packetIter = mPacketList.begin();
-    for (;packetIter != mPacketList.end(); ++packetIter)
-    {
-        RtpData& data = *packetIter;
-        // Disabled for debugging only
-        //fwrite(&data.mLength, sizeof data.mLength, 1, f);
-        fwrite(data.mData, data.mLength, 1, f);
+    std::ofstream output(mFilename, std::ios::binary);
+    if (!output.is_open())
+        throw std::runtime_error("Failed to open file for writing: " + mFilename);
+
+    // --- 1. Text header ---
+    std::string textLine = std::string(RTPDUMP_SHEBANG) + " " +
+                           ipToString(mSourceIp) + "/" +
+                           std::to_string(mSourcePort) + "\n";
+    output.write(textLine.data(), static_cast<std::streamsize>(textLine.size()));
+
+    // --- 2. Binary file header (16 bytes) ---
+    uint32_t buf32;
+    uint16_t buf16;
+
+    buf32 = htonl(mStartSec);
+    output.write(reinterpret_cast<const char*>(&buf32), 4);
+
+    buf32 = htonl(mStartUsec);
+    output.write(reinterpret_cast<const char*>(&buf32), 4);
+
+    buf32 = htonl(mSourceIp);
+    output.write(reinterpret_cast<const char*>(&buf32), 4);
+
+    buf16 = htons(mSourcePort);
+    output.write(reinterpret_cast<const char*>(&buf16), 2);
+
+    buf16 = 0;  // padding
+    output.write(reinterpret_cast<const char*>(&buf16), 2);
+
+    // --- 3. Packet records ---
+    size_t written = 0;
+
+    for (const auto& pkt : mPacketList) {
+        if (pkt.mRawData.empty())
+            continue;
+
+        uint16_t plen = static_cast<uint16_t>(pkt.mRawData.size());
+        uint16_t recLength = static_cast<uint16_t>(plen + 8);
+
+        buf16 = htons(recLength);
+        output.write(reinterpret_cast<const char*>(&buf16), 2);
+
+        buf16 = htons(plen);
+        output.write(reinterpret_cast<const char*>(&buf16), 2);
+
+        buf32 = htonl(pkt.mOffsetMs);
+        output.write(reinterpret_cast<const char*>(&buf32), 4);
+
+        output.write(reinterpret_cast<const char*>(pkt.mRawData.data()), plen);
+
+        written++;
     }
-    fclose(f);
-}
-#endif
 
+    if (!output.good())
+        throw std::runtime_error("Failed to write rtpdump file: " + mFilename);
+
+    ICELogInfo(<< "Wrote " << written << " packets to " << mFilename);
+}
+
+void RtpDump::clear()
+{
+    mPacketList.clear();
+    mLoaded = false;
+    mRecording = false;
+}

src/engine/helper/HL_Rtp.h

@@ -1,4 +1,4 @@
-/* Copyright(C) 2007-2025 VoIPobjects (voipobjects.com)
+/* Copyright(C) 2007-2026 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/. */
@@ -6,12 +6,14 @@
 #ifndef __HL_RTP_H
 #define __HL_RTP_H
 
-#if defined(USE_RTPDUMP)
-# include "jrtplib/src/rtppacket.h"
-#endif
+#include "jrtplib/src/rtppacket.h"
 
 #include <cstdint>
-#include <stdlib.h>
+#include <cstdlib>
+#include <vector>
+#include <string>
+#include <memory>
+#include <chrono>
 
 // Class to carry rtp/rtcp socket pair
 template<class T>
@@ -27,7 +29,7 @@ struct RtpPair
         :mRtp(rtp), mRtcp(rtcp)
     {}
 
-    bool multiplexed() { return mRtp == mRtcp; }
+    bool multiplexed() const { return mRtp == mRtcp; }
 };
 
 class RtpHelper
@@ -35,39 +37,114 @@ class RtpHelper
 public:
     static bool     isRtp(const void* buffer, size_t length);
     static int      findPtype(const void* buffer, size_t length);
-    static int      findPacketNo(const void* buffer, size_t length);
+    static int      findPacketNo(const void *buffer, size_t length);
     static bool     isRtpOrRtcp(const void* buffer, size_t length);
     static bool     isRtcp(const void* buffer, size_t length);
     static unsigned findSsrc(const void* buffer, size_t length);
     static void     setSsrc(void* buffer, size_t length, uint32_t ssrc);
     static int      findPayloadLength(const void* buffer, size_t length);
+
+    static std::chrono::microseconds toMicroseconds(const jrtplib::RTPTime& t);
 };
 
-#if defined(USE_RTPDUMP)
+/**
+ * @brief Standard rtpdump file format (rtptools / Wireshark compatible)
+ *
+ * Conforms to the rtpdump format defined by rtptools:
+ *   https://formats.kaitai.io/rtpdump/
+ *
+ * File layout:
+ *   1. Text header line:
+ *        "#!rtpplay1.0 <source_ip>/<source_port>\n"
+ *
+ *   2. Binary file header (RD_hdr_t, 16 bytes, all big-endian):
+ *        uint32_t start_sec    - recording start time, seconds since epoch
+ *        uint32_t start_usec   - recording start time, microseconds
+ *        uint32_t source_ip    - source IP address (network byte order)
+ *        uint16_t source_port  - source port
+ *        uint16_t padding      - always 0
+ *
+ *   3. Packet records (repeated until EOF):
+ *      Per-packet header (RD_packet_t, 8 bytes, all big-endian):
+ *        uint16_t length       - total record length (this 8-byte header + plen)
+ *        uint16_t plen         - RTP/RTCP payload length in bytes
+ *        uint32_t offset       - milliseconds since recording start
+ *      Followed by plen bytes of RTP/RTCP packet data.
+ *
+ * Maximum single packet payload: 65535 bytes (enforced for safety).
+ */
 class RtpDump
 {
 protected:
     struct RtpData
     {
-        jrtplib::RTPPacket* mPacket;
-        void* mData;
-        size_t mLength;
+        std::shared_ptr<jrtplib::RTPPacket> mPacket;
+        std::vector<uint8_t> mRawData;
+        uint32_t mOffsetMs = 0;
     };
 
     typedef std::vector<RtpData> PacketList;
     PacketList mPacketList;
     std::string mFilename;
+    bool mLoaded = false;
+
+    // File header fields
+    uint32_t mSourceIp = 0;
+    uint16_t mSourcePort = 0;
+    uint32_t mStartSec = 0;
+    uint32_t mStartUsec = 0;
+
+    // Auto-compute packet offsets during recording
+    bool mRecording = false;
+    std::chrono::steady_clock::time_point mRecordStart;
+
+    std::shared_ptr<jrtplib::RTPPacket> parseRtpData(const uint8_t* data, size_t len);
 
 public:
-    RtpDump(const char* filename);
+    explicit RtpDump(const char* filename);
     ~RtpDump();
 
+    /** Set source address for the file header (host byte order). */
+    void setSource(uint32_t ip, uint16_t port);
+    uint32_t sourceIp() const { return mSourceIp; }
+    uint16_t sourcePort() const { return mSourcePort; }
+
+    /**
+     * @brief Load packets from an rtpdump file
+     * @throws std::runtime_error on file/format error
+     */
     void load();
+    bool isLoaded() const { return mLoaded; }
+
     size_t count() const;
+
+    /**
+     * @brief Get parsed RTP packet at index
+     * @throws std::out_of_range if index is invalid
+     * @throws std::runtime_error if packet could not be parsed as RTP
+     */
     jrtplib::RTPPacket& packetAt(size_t index);
+
+    /** @brief Get raw packet bytes at index */
+    const std::vector<uint8_t>& rawDataAt(size_t index) const;
+
+    /** @brief Get packet time offset in milliseconds */
+    uint32_t offsetAt(size_t index) const;
+
+    /** @brief Add a packet; time offset is auto-computed from first add() call */
     void add(const void* data, size_t len);
+
+    /** @brief Add a packet with an explicit millisecond offset */
+    void add(const void* data, size_t len, uint32_t offsetMs);
+
+    /**
+     * @brief Write all packets to file in rtpdump format
+     * @throws std::runtime_error on file error
+     */
     void flush();
+
+    void clear();
+    const std::string& filename() const { return mFilename; }
 };
-#endif
 
 #endif

src/engine/helper/HL_Sync.cpp

@@ -280,7 +280,8 @@ size_t TimerQueue::cancel(uint64_t id) {
 //! Cancels all timers
 // \return
 //  The number of timers cancelled
-size_t TimerQueue::cancelAll() {
+size_t TimerQueue::cancelAll()
+{
     // Setting all "end" to 0 (for immediate execution) is ok,
     // since it maintains the heap integrity
     std::unique_lock<std::mutex> lk(m_mtx);

src/engine/helper/HL_ThreadPool.cpp

@@ -65,6 +65,7 @@ void thread_pool::run_worker()
                 tasks.pop();
             }
         }
-        t(); // function<void()> type
+        if (t)
+            t(); // function<void()> type
     }
 }

src/engine/helper/HL_VariantMap.cpp

@@ -240,7 +240,7 @@ int64_t Variant::asInt64() const
     if (mType != VTYPE_INT64)
         throw Exception(ERR_BAD_VARIANT_TYPE);
 
-    return mInt;
+    return mInt64;
 }
 
 bool Variant::asBool() const

src/engine/media/MT_AmrCodec.cpp

@@ -3,11 +3,8 @@
 
 #include "MT_AmrCodec.h"
 #include "../helper/HL_ByteBuffer.h"
-#include "../helper/HL_Log.h"
 #include "../helper/HL_IuUP.h"
-#include "../helper/HL_Exception.h"
-
-#include <iostream>
+#include "../helper/HL_Log.h"
 
 #define LOG_SUBSYSTEM "AmrCodec"
 using namespace MT;
@@ -31,37 +28,37 @@ const uint16_t amrwb_framelenbits[10] =
 
 struct AmrPayloadInfo
 {
-    const uint8_t*  mPayload;
-    int             mPayloadLength;
-    bool            mOctetAligned;
-    bool            mInterleaving;
-    bool            mWideband;
-    uint64_t        mCurrentTimestamp;
+    const uint8_t*  mPayload = nullptr;
+    int             mPayloadLength = 0;
+    bool            mOctetAligned = false;
+    bool            mInterleaving = false;
+    bool            mWideband = false;
+    uint64_t        mCurrentTimestamp = 0;
 };
 
 
 struct AmrFrame
 {
-    uint8_t               mFrameType;
-    uint8_t               mMode;
-    bool                  mGoodQuality;
-    uint64_t              mTimestamp;
+    uint8_t               mFrameType = 0;
+    uint8_t               mMode = 0;
+    bool                  mGoodQuality = false;
+    uint64_t              mTimestamp = 0;
     std::shared_ptr<ByteBuffer> mData;
-    uint8_t               mSTI;
+    uint8_t               mSTI = 0;
 };
 
 struct AmrPayload
 {
-    uint8_t               mCodeModeRequest;
+    uint8_t               mCodeModeRequest = 0;
     std::vector<AmrFrame> mFrames;
-    bool                  mDiscardPacket;
+    bool                  mDiscardPacket = false;
 };
 
 // ARM RTP payload has next structure
 //   Header
 //   Table of Contents
 //   Frames
-static AmrPayload parseAmrPayload(AmrPayloadInfo& input)
+static AmrPayload parseAmrPayload(AmrPayloadInfo& input, size_t& cngCounter)
 {
     AmrPayload result;
 
@@ -128,6 +125,8 @@ static AmrPayload parseAmrPayload(AmrPayloadInfo& input)
         frame.mTimestamp = input.mCurrentTimestamp;
         result.mFrames.push_back(frame);
         input.mCurrentTimestamp += input.mWideband ? 320 : 160;
+        if (FT == SID_FT)
+            cngCounter++;
     }
     while (F != 0);
 
@@ -140,13 +139,17 @@ static AmrPayload parseAmrPayload(AmrPayloadInfo& input)
         //   avoid the loss of data synchronization in the depacketization
         //   process, which can result in a huge degradation in speech quality.
         bool discard = input.mWideband ? (f.mFrameType >= 10 && f.mFrameType <= 13) : (f.mFrameType >= 9 && f.mFrameType <= 14);
-        // discard |= input.mWideband ? f.mFrameType >= 14 : f.mFrameType >= 15;
         if (discard)
         {
             result.mDiscardPacket = true;
             continue;
         }
 
+        // if (input.mWideband && f.mMode == 0xFF /* CNG */)
+        // {
+        //     int a = 1;
+        // }`
+
         if (input.mWideband && f.mFrameType == 15)
         {
             // DTX, no sense to decode the data
@@ -165,8 +168,8 @@ static AmrPayload parseAmrPayload(AmrPayloadInfo& input)
             continue;
         }
 
-        size_t bitsLength = input.mWideband ? amrwb_framelenbits[f.mFrameType] : amrnb_framelenbits[f.mFrameType];
-        size_t byteLength = input.mWideband ? amrwb_framelen[f.mFrameType] : amrnb_framelen[f.mFrameType];
+        size_t bitsLength = input.mWideband ? amrwb_framelenbits[f.mFrameType]  : amrnb_framelenbits[f.mFrameType];
+        size_t byteLength = input.mWideband ? amrwb_framelen[f.mFrameType]      : amrnb_framelen[f.mFrameType];
 
         if (bitsLength > 0)
         {
@@ -260,8 +263,7 @@ PCodec AmrNbCodec::CodecFactory::create()
 
 
 AmrNbCodec::AmrNbCodec(const AmrCodecConfig& config)
-    :mEncoderCtx(nullptr), mDecoderCtx(nullptr), mConfig(config), mCurrentDecoderTimestamp(0),
-      mSwitchCounter(0), mPreviousPacketLength(0)
+    :mConfig(config)
 {
     mEncoderCtx = Encoder_Interface_init(1);
     mDecoderCtx = Decoder_Interface_init();
@@ -282,44 +284,31 @@ AmrNbCodec::~AmrNbCodec()
     }
 }
 
-const char* AmrNbCodec::name()
+Codec::Info AmrNbCodec::info()
 {
-    return MT_AMRNB_CODECNAME;
+    return {
+        .mName = MT_AMRNB_CODECNAME,
+        .mSamplerate = 8000,
+        .mChannels = 1,
+        .mPcmLength = 20 * 16,
+        .mFrameTime = 20,
+        .mRtpLength = 0
+    };
 }
 
-int AmrNbCodec::pcmLength()
+Codec::EncodeResult AmrNbCodec::encode(std::span<const uint8_t> input, std::span<uint8_t> output)
 {
-    return 20 * 16;
-}
-
-int AmrNbCodec::rtpLength()
-{
-    return 0;
-}
-
-int AmrNbCodec::frameTime()
-{
-    return 20;
-}
-
-int AmrNbCodec::samplerate()
-{
-    return 8000;
-}
-
-int AmrNbCodec::encode(const void* input, int inputBytes, void* output, int outputCapacity)
-{
-    if (inputBytes % pcmLength())
-        return 0;
+    if (input.size_bytes() % pcmLength())
+        return {.mEncoded = 0};
 
     // Declare the data input pointer
-    auto *dataIn = (const short *)input;
+    auto *dataIn = (const short *)input.data();
 
     // Declare the data output pointer
-    auto *dataOut = (unsigned char *)output;
+    auto *dataOut = (unsigned char *)output.data();
 
     // Find how much RTP frames will be generated
-    unsigned int frames = inputBytes / pcmLength();
+    unsigned int frames = input.size_bytes() / pcmLength();
 
     // Generate frames
     for (unsigned int i = 0; i < frames; i++)
@@ -328,28 +317,28 @@ int AmrNbCodec::encode(const void* input, int inputBytes, void* output, int outp
         dataIn += pcmLength() / 2;
     }
 
-    return dataOut - (unsigned char*)output;
+    return {.mEncoded = (size_t)(dataOut - (unsigned char*)output.data())};
 }
 
 #define L_FRAME 160
 #define AMR_BITRATE_DTX 15
-int AmrNbCodec::decode(const void* input, int inputBytes, void* output, int outputCapacity)
+Codec::DecodeResult AmrNbCodec::decode(std::span<const uint8_t> input, std::span<uint8_t> output)
 {
     if (mConfig.mOctetAligned)
-        return 0;
+        return {.mDecoded = 0};
 
     if (mConfig.mIuUP)
     {
         // Try to parse IuUP frame
         IuUP::Frame frame;
-        if (!IuUP::parse2((const uint8_t*)input, inputBytes, frame))
-            return 0;
+        if (!IuUP::parse2((const uint8_t*)input.data(), input.size_bytes(), frame))
+            return {0};
 
         // Check if CRC failed - it is check from IuUP data
         if (!frame.mHeaderCrcOk || !frame.mPayloadCrcOk)
         {
             ICELogInfo(<< "CRC check failed.");
-            return 0;
+            return {0};
         }
 
         // Build NB frame to decode
@@ -366,61 +355,61 @@ int AmrNbCodec::decode(const void* input, int inputBytes, void* output, int outp
 
         // Check if frameType comparing is correct
         if (frameType == 0xFF)
-            return 0;
+            return {0};
 
         dataToDecode.mutableData()[0] = (frameType << 3) | (1 << 2);
 
-        Decoder_Interface_Decode(mDecoderCtx, (const unsigned char*)dataToDecode.data(), (short*)output, 0);
-        return pcmLength();
+        Decoder_Interface_Decode(mDecoderCtx, (const unsigned char*)dataToDecode.data(), (short*)output.data(), 0);
+        return {.mDecoded = (size_t)pcmLength()};
     }
     else
     {
-        if (outputCapacity < pcmLength())
-            return 0;
+        if (output.size_bytes() < pcmLength())
+            return {.mDecoded = 0};
 
-        if (inputBytes == 0)
+        if (input.size_bytes() == 0)
         { // PLC part
             unsigned char buffer[32];
             buffer[0] = (AMR_BITRATE_DTX << 3)|4;
-            Decoder_Interface_Decode(mDecoderCtx, buffer, (short*)output, 0); // Handle missing data
-            return pcmLength();
+            Decoder_Interface_Decode(mDecoderCtx, buffer, (short*)output.data(), 0); // Handle missing data
+            return {.mDecoded = (size_t)pcmLength()};
         }
 
         AmrPayloadInfo info;
         info.mCurrentTimestamp = mCurrentDecoderTimestamp;
         info.mOctetAligned = mConfig.mOctetAligned;
-        info.mPayload = (const uint8_t*)input;
-        info.mPayloadLength = inputBytes;
+        info.mPayload = input.data();
+        info.mPayloadLength = input.size_bytes();
         info.mWideband = false;
         info.mInterleaving = false;
 
         AmrPayload ap;
         try
         {
-            ap = parseAmrPayload(info);
+            ap = parseAmrPayload(info, mCngCounter);
         }
         catch(...)
         {
             ICELogDebug(<< "Failed to decode AMR payload.");
-            return 0;
+            return {.mDecoded = 0};
         }
         // Save current timestamp
         mCurrentDecoderTimestamp = info.mCurrentTimestamp;
 
         // Check if packet is corrupted
         if (ap.mDiscardPacket)
-            return 0;
+            return {.mDecoded = 0};
 
 
         // Check for output buffer capacity
-        if (outputCapacity < (int)ap.mFrames.size() * pcmLength())
-            return 0;
+        if (output.size_bytes() < (int)ap.mFrames.size() * pcmLength())
+            return {.mDecoded = 0};
 
         if (ap.mFrames.empty())
         {
             ICELogError(<< "No AMR frames");
         }
-        short* dataOut = (short*)output;
+        short* dataOut = (short*)output.data();
         for (AmrFrame& frame: ap.mFrames)
         {
             if (frame.mData)
@@ -430,18 +419,18 @@ int AmrNbCodec::decode(const void* input, int inputBytes, void* output, int outp
                 dataOut += pcmLength() / 2;
             }
         }
-        return pcmLength() * ap.mFrames.size();
+        return {.mDecoded = pcmLength() * ap.mFrames.size()};
     }
 
-    return pcmLength();
+    return {.mDecoded = (size_t)pcmLength()};
 }
 
-int AmrNbCodec::plc(int lostFrames, void* output, int outputCapacity)
+size_t AmrNbCodec::plc(int lostFrames, std::span<uint8_t> output)
 {
-    if (outputCapacity < lostFrames * pcmLength())
+    if (output.size_bytes() < lostFrames * pcmLength())
         return 0;
 
-    short* dataOut = (short*)output;
+    short* dataOut = (short*)output.data();
 
     for (int i=0; i < lostFrames; i++)
     {
@@ -459,6 +448,11 @@ int AmrNbCodec::getSwitchCounter() const
     return mSwitchCounter;
 }
 
+int AmrNbCodec::getCngCounter() const
+{
+    return mCngCounter;
+}
+
 // -------- AMR WB codec
 AmrWbCodec::CodecFactory::CodecFactory(const AmrCodecConfig& config)
     :mConfig(config)
@@ -500,11 +494,9 @@ PCodec AmrWbCodec::CodecFactory::create()
 AmrWbStatistics MT::GAmrWbStatistics;
 
 AmrWbCodec::AmrWbCodec(const AmrCodecConfig& config)
-    :mEncoderCtx(nullptr), mDecoderCtx(nullptr), mConfig(config),
-      mSwitchCounter(0), mPreviousPacketLength(0)
+    :mConfig(config)
 {
     mDecoderCtx = D_IF_init();
-    mCurrentDecoderTimestamp = 0;
 }
 
 AmrWbCodec::~AmrWbCodec()
@@ -522,49 +514,37 @@ AmrWbCodec::~AmrWbCodec()
     }
 }
 
-const char* AmrWbCodec::name()
-{
-    return MT_AMRWB_CODECNAME;
+Codec::Info AmrWbCodec::info() {
+    return {
+        .mName = MT_AMRWB_CODECNAME,
+        .mSamplerate = 16000,
+        .mChannels = 1,
+        .mPcmLength = 20 * 16 * 2,
+        .mFrameTime = 20,
+        .mRtpLength = 0   /* There is complex structure inside AMR packet which may include multilple frames with various length. */
+    };
 }
 
-int AmrWbCodec::pcmLength()
-{
-    return 20 * 16 * 2;
-}
 
-int AmrWbCodec::rtpLength()
+Codec::EncodeResult AmrWbCodec::encode(std::span<const uint8_t> input, std::span<uint8_t> output)
 {
-    return 0; // VBR
-}
-
-int AmrWbCodec::frameTime()
-{
-    return 20;
-}
-
-int AmrWbCodec::samplerate()
-{
-    return 16000;
-}
-
-int AmrWbCodec::encode(const void* input, int inputBytes, void* output, int outputCapacity)
-{
-    throw Exception(ERR_NOT_IMPLEMENTED);
+    // Still no support for encoding - emit silence instead
+    return {.mEncoded = 0};
 }
 
 #define L_FRAME 160
 #define AMR_BITRATE_DTX 15
 
-int AmrWbCodec::decodeIuup(std::span<const uint8_t> input, std::span<uint8_t> output)
+Codec::DecodeResult AmrWbCodec::decodeIuup(std::span<const uint8_t> input, std::span<uint8_t> output)
 {
     IuUP::Frame frame;
     if (!IuUP::parse2(input.data(), input.size(), frame))
-        return 0;
+        return {.mDecoded = 0};
 
     if (!frame.mHeaderCrcOk || !frame.mPayloadCrcOk)
     {
         ICELogInfo(<< "CRC check failed.");
-        return 0;
+        return {.mDecoded = 0};
     }
 
     // Reserve space
@@ -579,34 +559,34 @@ int AmrWbCodec::decodeIuup(std::span<const uint8_t> input, std::span<uint8_t> ou
             frameType = ftIndex;
 
     if (frameType == 0xFF)
-        return 0;
+        return {.mDecoded = 0, .mIsCng = true};
 
     dataToDecode.mutableData()[0] = (frameType << 3) | (1 << 2);
 
     D_IF_decode(mDecoderCtx, (const unsigned char*)dataToDecode.data(), (short*)output.data(), 0);
-    return pcmLength();
+    return {.mDecoded = (size_t)pcmLength()};
 }
 
-int AmrWbCodec::decodePlain(std::span<const uint8_t> input, std::span<uint8_t> output)
+Codec::DecodeResult AmrWbCodec::decodePlain(std::span<const uint8_t> input, std::span<uint8_t> output)
 {
     AmrPayloadInfo info;
-    info.mCurrentTimestamp = mCurrentDecoderTimestamp;
-    info.mOctetAligned = mConfig.mOctetAligned;
-    info.mPayload = input.data();
-    info.mPayloadLength = input.size();
-    info.mWideband = true;
-    info.mInterleaving = false;
+    info.mCurrentTimestamp  = mCurrentDecoderTimestamp;
+    info.mOctetAligned      = mConfig.mOctetAligned;
+    info.mPayload           = input.data();
+    info.mPayloadLength     = input.size();
+    info.mWideband          = true;
+    info.mInterleaving      = false;
 
     AmrPayload ap;
     try
     {
-        ap = parseAmrPayload(info);
+        ap = parseAmrPayload(info, mCngCounter);
     }
     catch(...)
     {
         GAmrWbStatistics.mNonParsed++;
         ICELogDebug(<< "Failed to decode AMR payload");
-        return 0;
+        return {.mDecoded = 0};
     }
     // Save current timestamp
     mCurrentDecoderTimestamp = info.mCurrentTimestamp;
@@ -615,58 +595,67 @@ int AmrWbCodec::decodePlain(std::span<const uint8_t> input, std::span<uint8_t> o
     if (ap.mDiscardPacket)
     {
         GAmrWbStatistics.mDiscarded++;
-        return 0;
+        return {.mDecoded = 0};
     }
 
-    // Check for output buffer capacity
-    if (output.size() < (int)ap.mFrames.size() * pcmLength())
-        return 0;
+    // Find the required output capacity
+    size_t capacity = 0;
+    for (AmrFrame& frame: ap.mFrames)
+        capacity += frame.mMode == 0xFF /* CNG */ ? pcmLength() : pcmLength();
+
+    if (output.size() < capacity)
+        return {.mDecoded = 0};
 
     short* dataOut = (short*)output.data();
     size_t dataOutSizeInBytes = 0;
     for (AmrFrame& frame: ap.mFrames)
     {
-        memset(dataOut, 0, static_cast<size_t>(output.size()));
+        size_t frameOutputSize = frame.mMode == 0xFF ? pcmLength() : pcmLength();
+        memset(dataOut, 0, frameOutputSize);
 
         if (frame.mData)
         {
+            if (frame.mMode == 0xFF)
+            {
+                // int bp = 1;
+            }
             D_IF_decode(mDecoderCtx, (const unsigned char*)frame.mData->data(), (short*)dataOut, 0);
-            dataOut += pcmLength() / 2;
-            dataOutSizeInBytes += pcmLength();
+            dataOut += frameOutputSize / 2;
+            dataOutSizeInBytes += frameOutputSize;
         }
     }
-    return dataOutSizeInBytes;
+    return {.mDecoded = dataOutSizeInBytes,
+            .mIsCng = ap.mFrames.size() == 1 ? (ap.mFrames.front().mMode == 0xFF) : false};
 }
 
-int AmrWbCodec::decode(const void* input, int inputBytes, void* output, int outputCapacity)
+Codec::DecodeResult AmrWbCodec::decode(std::span<const uint8_t> input, std::span<uint8_t> output)
 {
-    auto inputBuffer = std::span<const uint8_t>((uint8_t*)input, (size_t)inputBytes);
-    auto outputBuffer = std::span<uint8_t>((uint8_t*)output, (size_t)outputCapacity);
-
     if (mConfig.mIuUP)
-        return decodeIuup(inputBuffer, outputBuffer);
+        return decodeIuup(input, output);
     else
-        return decodePlain(inputBuffer, outputBuffer);
-
-    return 0;
+        return decodePlain(input, output);
 }
 
-int AmrWbCodec::plc(int lostFrames, void* output, int outputCapacity)
+size_t AmrWbCodec::plc(int lostFrames, std::span<uint8_t> output)
 {
-    /*  if (outputCapacity < lostFrames * pcmLength())
-    return 0;
-
-  short* dataOut = (short*)output;
-
-  for (int i=0; i < lostFrames; i++)
-  {
-    unsigned char buffer[32];
-    buffer[0] = (AMR_BITRATE_DTX << 3)|4;
-    Decoder_Interface_Decode(mDecoderCtx, buffer, dataOut, 0); // Handle missing data
-    dataOut += L_FRAME;
-  }
-  */
+    // ToDo: Check again if PLC works for AMR-WB
+    // For now return the silence
+    memset(output.data(), 0, output.size_bytes());
     return lostFrames * pcmLength();
+    /*
+    if (outputCapacity < lostFrames * pcmLength())
+        return 0;
+
+    short* dataOut = (short*)output;
+
+    for (int i=0; i < lostFrames; i++)
+    {
+        unsigned char buffer[32];
+        buffer[0] = (AMR_BITRATE_DTX << 3)|4;
+        Decoder_Interface_Decode(mDecoderCtx, buffer, dataOut, 0); // Handle missing data
+        dataOut += L_FRAME;
+    }
+    */
 }
 
 int AmrWbCodec::getSwitchCounter() const
@@ -674,14 +663,16 @@ int AmrWbCodec::getSwitchCounter() const
     return mSwitchCounter;
 }
 
+int AmrWbCodec::getCngCounter() const
+{
+    return mCngCounter;
+}
 
 // ------------- GSM EFR -----------------
 
 GsmEfrCodec::GsmEfrFactory::GsmEfrFactory(bool iuup, int ptype)
     :mIuUP(iuup), mPayloadType(ptype)
-{
-
-}
+{}
 
 const char* GsmEfrCodec::GsmEfrFactory::name()
 {
@@ -699,9 +690,7 @@ int GsmEfrCodec::GsmEfrFactory::payloadType()
 }
 
 void GsmEfrCodec::GsmEfrFactory::updateSdp(resip::SdpContents::Session::Medium::CodecContainer& codecs, SdpDirection direction)
-{
-
-}
+{}
 
 int GsmEfrCodec::GsmEfrFactory::processSdp(const resip::SdpContents::Session::Medium::CodecContainer& codecs, SdpDirection direction)
 {
@@ -719,7 +708,7 @@ PCodec GsmEfrCodec::GsmEfrFactory::create()
 }
 
 GsmEfrCodec::GsmEfrCodec(bool iuup)
-    :mEncoderCtx(nullptr), mDecoderCtx(nullptr), mIuUP(iuup)
+    :mIuUP(iuup)
 {
     mEncoderCtx = Encoder_Interface_init(1);
     mDecoderCtx = Decoder_Interface_init();
@@ -740,44 +729,31 @@ GsmEfrCodec::~GsmEfrCodec()
     }
 }
 
-const char* GsmEfrCodec::name()
+Codec::Info GsmEfrCodec::info()
 {
-    return MT_GSMEFR_CODECNAME;
+    return {
+        .mName = MT_GSMEFR_CODECNAME,
+        .mSamplerate = 8000,
+        .mChannels = 1,
+        .mPcmLength = 20 * 16,
+        .mFrameTime = 20,
+        .mRtpLength = 0
+    };
 }
 
-int GsmEfrCodec::pcmLength()
+Codec::EncodeResult GsmEfrCodec::encode(std::span<const uint8_t> input, std::span<uint8_t> output)
 {
-    return 20 * 16;
-}
-
-int GsmEfrCodec::rtpLength()
-{
-    return 0;
-}
-
-int GsmEfrCodec::frameTime()
-{
-    return 20;
-}
-
-int GsmEfrCodec::samplerate()
-{
-    return 8000;
-}
-
-int GsmEfrCodec::encode(const void* input, int inputBytes, void* output, int outputCapacity)
-{
-    if (inputBytes % pcmLength())
-        return 0;
+    if (input.size_bytes() % pcmLength())
+        return {.mEncoded = 0};
 
     // Declare the data input pointer
-    const short *dataIn = (const short *)input;
+    const short *dataIn = (const short *)input.data();
 
     // Declare the data output pointer
-    unsigned char *dataOut = (unsigned char *)output;
+    unsigned char *dataOut = (unsigned char *)output.data();
 
     // Find how much RTP frames will be generated
-    unsigned int frames = inputBytes / pcmLength();
+    unsigned int frames = input.size_bytes() / pcmLength();
 
     // Generate frames
     for (unsigned int i = 0; i < frames; i++)
@@ -786,7 +762,7 @@ int GsmEfrCodec::encode(const void* input, int inputBytes, void* output, int out
         dataIn += pcmLength() / 2;
     }
 
-    return frames * rtpLength();
+    return {.mEncoded = frames * rtpLength()};
 }
 
 #define L_FRAME 160
@@ -843,100 +819,61 @@ const uint16_t gsm690_12_2_bitorder[244] = {
     237, 236,  96, 199,
 };
 
-int GsmEfrCodec::decode(const void* input, int inputBytes, void* output, int outputCapacity)
+Codec::DecodeResult GsmEfrCodec::decode(std::span<const uint8_t> input, std::span<uint8_t> output)
 {
-    /*  if (mIuUP)
-  {
-    // Try to parse IuUP frame
-    IuUP::Frame frame;
-    if (!IuUP::parse2((const uint8_t*)input, inputBytes, frame))
-      return 0;
+    if (output.size_bytes() < pcmLength())
+        return {.mDecoded = 0};
 
-    // Check if CRC failed - it is check from IuUP data
-    if (!frame.mHeaderCrcOk || !frame.mPayloadCrcOk)
-    {
-      ICELogInfo(<< "CRC check failed.");
-      return 0;
+    if (input.size_bytes() == 0)
+    { // PLC part
+        unsigned char buffer[32];
+        buffer[0] = (AMR_BITRATE_DTX << 3)|4;
+        Decoder_Interface_Decode(mDecoderCtx, buffer, (short*)output.data(), 0); // Handle missing data
     }
-
-    // Build NB frame to decode
-    ByteBuffer dataToDecode;
-    dataToDecode.resize(1 + frame.mPayloadSize); // Reserve place
-
-    // Copy AMR data
-    memmove(dataToDecode.mutableData() + 1, frame.mPayload, frame.mPayloadSize);
-
-    uint8_t frameType = 0xFF;
-    for (uint8_t ftIndex = 0; ftIndex <= 9 && frameType == 0xFF; ftIndex++)
-      if (amrnb_framelen[ftIndex] == frame.mPayloadSize)
-        frameType = ftIndex;
-
-    // Check if frameType comparing is correct
-    if (frameType == 0xFF)
-      return 0;
-
-    dataToDecode.mutableData()[0] = (frameType << 3) | (1 << 2);
-
-    Decoder_Interface_Decode(mDecoderCtx, (const unsigned char*)dataToDecode.data(), (short*)output, 0);
-    return pcmLength();
-  }
-  else */
+    else
     {
-        if (outputCapacity < pcmLength())
-            return 0;
+        // Reorder bytes from input to dst
+        uint8_t dst[GSM_EFR_FRAME_LEN];
+        const uint8_t* src = input.data();
+        for (int i=0; i<(GSM_EFR_FRAME_LEN-1); i++)
+            dst[i] = (src[i] << 4) | (src[i+1] >> 4);
+        dst[GSM_EFR_FRAME_LEN-1] = src[GSM_EFR_FRAME_LEN-1] << 4;
 
-        if (inputBytes == 0)
-        { // PLC part
-            unsigned char buffer[32];
-            buffer[0] = (AMR_BITRATE_DTX << 3)|4;
-            Decoder_Interface_Decode(mDecoderCtx, buffer, (short*)output, 0); // Handle missing data
-        }
-        else
+        unsigned char in[GSM_EFR_FRAME_LEN + 1];
+
+        // Reorder bits
+        in[0] = 0x3c; /* AMR mode 7 = GSM-EFR, Quality bit is set */
+        in[GSM_EFR_FRAME_LEN] = 0x0;
+
+        for (int i=0; i<244; i++)
         {
-            // Reorder bytes from input to dst
-            uint8_t dst[GSM_EFR_FRAME_LEN];
-            const uint8_t* src = (const uint8_t*)input;
-            for (int i=0; i<(GSM_EFR_FRAME_LEN-1); i++)
-                dst[i] = (src[i] << 4) | (src[i+1] >> 4);
-            dst[GSM_EFR_FRAME_LEN-1] = src[GSM_EFR_FRAME_LEN-1] << 4;
+            int si = gsm690_12_2_bitorder[i];
+            int di = i;
+            msb_put_bit(in + 1, di, msb_get_bit(dst, si));
+        }
 
-            unsigned char in[GSM_EFR_FRAME_LEN + 1];
+        // Decode
+        memset(output.data(), 0, pcmLength());
+        Decoder_Interface_Decode(mDecoderCtx, in, (short*)output.data(), 0);
 
-            // Reorder bits
-            in[0] = 0x3c; /* AMR mode 7 = GSM-EFR, Quality bit is set */
-            in[GSM_EFR_FRAME_LEN] = 0x0;
-
-            for (int i=0; i<244; i++)
-            {
-                int si = gsm690_12_2_bitorder[i];
-                int di = i;
-                msb_put_bit(in + 1, di, msb_get_bit(dst, si));
-            }
-
-            // Decode
-            memset(output, 0, pcmLength());
-            Decoder_Interface_Decode(mDecoderCtx, in, (short*)output, 0);
-
-
-            uint8_t* pcm = (uint8_t*)output;
-            for (int i=0; i<160; i++)
-            {
-                uint16_t w = ((uint16_t*)output)[i];
-                pcm[(i<<1)  ] =  w       & 0xff;
-                pcm[(i<<1)+1] = (w >> 8) & 0xff;
-            }
+        uint8_t* pcm = (uint8_t*)output.data();
+        for (int i=0; i<160; i++)
+        {
+            uint16_t w = ((uint16_t*)output.data())[i];
+            pcm[(i<<1)  ] =  w       & 0xff;
+            pcm[(i<<1)+1] = (w >> 8) & 0xff;
         }
     }
 
-    return pcmLength();
+    return {.mDecoded = (size_t)pcmLength()};
 }
 
-int GsmEfrCodec::plc(int lostFrames, void* output, int outputCapacity)
+size_t GsmEfrCodec::plc(int lostFrames, std::span<uint8_t> output)
 {
-    if (outputCapacity < lostFrames * pcmLength())
+    if (output.size_bytes() < lostFrames * pcmLength())
         return 0;
 
-    short* dataOut = (short*)output;
+    short* dataOut = (short*)output.data();
 
     for (int i=0; i < lostFrames; i++)
     {

src/engine/media/MT_AmrCodec.h

@@ -26,13 +26,13 @@ struct AmrCodecConfig
 class AmrNbCodec : public Codec
 {
 protected:
-    void* mEncoderCtx;
-    void* mDecoderCtx;
+    void* mEncoderCtx = nullptr;
+    void* mDecoderCtx = nullptr;
     AmrCodecConfig mConfig;
-    unsigned mCurrentDecoderTimestamp;
-    int mSwitchCounter;
-    int mPreviousPacketLength;
-
+    unsigned mCurrentDecoderTimestamp = 0;
+    int mPreviousPacketLength = 0;
+    size_t mCngCounter = 0;
+    size_t mSwitchCounter = 0;
 public:
     class CodecFactory: public Factory
     {
@@ -54,17 +54,16 @@ public:
     };
 
     AmrNbCodec(const AmrCodecConfig& config);
+    ~AmrNbCodec();
+
+    Info info() override;
+
+    EncodeResult encode(std::span<const uint8_t> input, std::span<uint8_t> output) override;
+    DecodeResult decode(std::span<const uint8_t> input, std::span<uint8_t> output) override;
+    size_t plc(int lostFrames, std::span<uint8_t> output) override;
 
-    virtual ~AmrNbCodec();
-    const char* name() override;
-    int pcmLength() override;
-    int rtpLength() override;
-    int frameTime() override;
-    int samplerate() override;
-    int encode(const void* input, int inputBytes, void* output, int outputCapacity) override;
-    int decode(const void* input, int inputBytes, void* output, int outputCapacity) override;
-    int plc(int lostFrames, void* output, int outputCapacity) override;
     int getSwitchCounter() const;
+    int getCngCounter() const;
 };
 
 struct AmrWbStatistics
@@ -77,15 +76,17 @@ extern AmrWbStatistics GAmrWbStatistics;
 class AmrWbCodec : public Codec
 {
 protected:
-    void* mEncoderCtx;
-    void* mDecoderCtx;
+    void* mEncoderCtx = nullptr;
+    void* mDecoderCtx = nullptr;
     AmrCodecConfig mConfig;
-    uint64_t mCurrentDecoderTimestamp;
-    int mSwitchCounter;
+    uint64_t mCurrentDecoderTimestamp = 0;
+    size_t mSwitchCounter = 0;
+    size_t mCngCounter = 0;
+
     int mPreviousPacketLength;
 
-    int decodeIuup(std::span<const uint8_t> input, std::span<uint8_t> output);
-    int decodePlain(std::span<const uint8_t> input, std::span<uint8_t> output);
+    DecodeResult decodeIuup(std::span<const uint8_t> input, std::span<uint8_t> output);
+    DecodeResult decodePlain(std::span<const uint8_t> input, std::span<uint8_t> output);
 
 public:
     class CodecFactory: public Factory
@@ -110,23 +111,21 @@ public:
     AmrWbCodec(const AmrCodecConfig& config);
     virtual ~AmrWbCodec();
 
-    const char* name() override;
-    int pcmLength() override;
-    int rtpLength() override;
-    int frameTime() override;
-    int samplerate() override;
-    int encode(const void* input, int inputBytes, void* output, int outputCapacity) override;
-    int decode(const void* input, int inputBytes, void* output, int outputCapacity) override;
-    int plc(int lostFrames, void* output, int outputCapacity) override;
+    Info info() override;
+
+    EncodeResult encode(std::span<const uint8_t> input, std::span<uint8_t> output) override;
+    DecodeResult decode(std::span<const uint8_t> input, std::span<uint8_t> output) override;
+    size_t       plc(int lostFrames, std::span<uint8_t> output) override;
     int getSwitchCounter() const;
+    int getCngCounter() const;
 };
 
 class GsmEfrCodec : public Codec
 {
 protected:
-    void* mEncoderCtx;
-    void* mDecoderCtx;
-    bool mIuUP;
+    void* mEncoderCtx = nullptr;
+    void* mDecoderCtx = nullptr;
+    bool mIuUP = false;
 
 public:
     class GsmEfrFactory: public Factory
@@ -143,23 +142,19 @@ public:
         void create(CodecMap& codecs) override;
 
         PCodec create() override;
-
     protected:
         bool mIuUP;
         int mPayloadType;
     };
 
     GsmEfrCodec(bool iuup = false);
+    ~GsmEfrCodec();
 
-    virtual ~GsmEfrCodec();
-    const char* name() override;
-    int pcmLength() override;
-    int rtpLength() override;
-    int frameTime() override;
-    int samplerate() override;
-    int encode(const void* input, int inputBytes, void* output, int outputCapacity) override;
-    int decode(const void* input, int inputBytes, void* output, int outputCapacity) override;
-    int plc(int lostFrames, void* output, int outputCapacity) override;
+    Info info() override;
+
+    EncodeResult encode(std::span<const uint8_t> input, std::span<uint8_t> output) override;
+    DecodeResult decode(std::span<const uint8_t> input, std::span<uint8_t> output) override;
+    size_t       plc(int lostFrames, std::span<uint8_t> output) override;
 };
 
 } // End of MT namespace

src/engine/media/MT_AudioCodec.cpp

@@ -91,38 +91,20 @@ G729Codec::~G729Codec()
     }
 }
 
-const char* G729Codec::name()
+Codec::Info G729Codec::info()
 {
-    return "G729";
-}
-
-int G729Codec::pcmLength()
-{
-    return 10 * 8 * 2;
-}
-
-int G729Codec::rtpLength()
-{
-    return 10;
-}
-
-int G729Codec::frameTime()
-{
-    return 10;
-}
-
-int G729Codec::samplerate()
-{
-    return 8000;
-}
-
-int G729Codec::channels()
-{
-    return 1;
+    return {
+        .mName = "G729",
+        .mSamplerate = 8000,
+        .mChannels = 1,
+        .mPcmLength = 10 * 8 * 2,
+        .mFrameTime = 10,
+        .mRtpLength = 10
+    };
 }
 
 // static const int SamplesPerFrame = 80;
-int G729Codec::encode(const void* input, int inputBytes, void* output, int outputCapacity)
+Codec::EncodeResult G729Codec::encode(std::span<const uint8_t> input, std::span<uint8_t> output)
 {
     // Create encoder if it is not done yet
     if (!mEncoder)
@@ -131,24 +113,24 @@ int G729Codec::encode(const void* input, int inputBytes, void* output, int outpu
         if (mEncoder)
             Init_Pre_Process(mEncoder);
     }
-    int result = 0;
+    size_t result = 0;
     if (mEncoder)
     {
-        int nrOfFrames = inputBytes / 160; // 10ms frames
-        Word16 parm[PRM_SIZE]; // ITU's service buffer
+        int nrOfFrames = input.size_bytes() / 160;  // 10ms frames
+        Word16 parm[PRM_SIZE];                      // ITU's service buffer
         for (int frameIndex = 0; frameIndex < nrOfFrames; frameIndex++)
         {
-            Copy((int16_t*)input + frameIndex * pcmLength() / 2, mEncoder->new_speech, pcmLength() / 2);
-            Pre_Process(mEncoder, mEncoder->new_speech, pcmLength() / 2);
+            Copy((int16_t*)input.data() + frameIndex * info().mPcmLength / 2, mEncoder->new_speech, info().mPcmLength / 2);
+            Pre_Process(mEncoder, mEncoder->new_speech, info().mPcmLength / 2);
             Coder_ld8a(mEncoder, parm);
-            Store_Params(parm, (uint8_t*)output + frameIndex * rtpLength());
-            result += rtpLength();
+            Store_Params(parm, output.data() + frameIndex * info().mRtpLength);
+            result += info().mRtpLength;
         }
     }
-    return result;
+    return {result};
 }
 
-int G729Codec::decode(const void* input, int inputBytes, void* output, int outputCapacity)
+Codec::DecodeResult G729Codec::decode(std::span<const uint8_t> input, std::span<uint8_t> output)
 {
     if (!mDecoder)
     {
@@ -160,29 +142,29 @@ int G729Codec::decode(const void* input, int inputBytes, void* output, int outpu
         }
     }
 
-    int result = 0;
+    size_t result = 0;
     if (mDecoder)
     {
         // See if there are silence bytes in the end
-        bool isSilence = (inputBytes % rtpLength()) / 2 != 0;
+        bool isSilence = (input.size_bytes() % info().mRtpLength) / 2 != 0;
 
         // Find number of frames
-        int nrOfFrames = inputBytes / rtpLength();
-        nrOfFrames = std::min(outputCapacity / pcmLength(), nrOfFrames);
+        int nrOfFrames = input.size_bytes() / info().mRtpLength;
+        nrOfFrames = std::min(output.size_bytes() / info().mPcmLength, (size_t)nrOfFrames);
 
         for (int frameIndex = 0; frameIndex < nrOfFrames; frameIndex++)
-            decodeFrame((const uint8_t*)input + frameIndex * rtpLength(), (int16_t*)output + frameIndex * pcmLength());
+            decodeFrame(input.data() + frameIndex * info().mRtpLength, (int16_t*)output.data() + frameIndex * info().mPcmLength);
 
-        result += nrOfFrames * pcmLength();
+        result += nrOfFrames * info().mPcmLength;
 
-        if (isSilence && nrOfFrames < outputCapacity / pcmLength())
+        if (isSilence && nrOfFrames < output.size_bytes() / info().mPcmLength)
         {
-            memset((uint8_t*)output + nrOfFrames * pcmLength(), 0, pcmLength());
-            result += pcmLength();
+            memset(output.data() + nrOfFrames * info().mPcmLength, 0, info().mPcmLength);
+            result += info().mPcmLength;
         }
     }
 
-    return result;
+    return {.mDecoded = result, .mIsCng = false};
 }
 
 void G729Codec::decodeFrame(const uint8_t* rtp, int16_t* pcm)
@@ -217,7 +199,7 @@ void G729Codec::decodeFrame(const uint8_t* rtp, int16_t* pcm)
 }
 
 
-int G729Codec::plc(int lostFrames, void* output, int outputCapacity)
+size_t G729Codec::plc(int lostFrames, std::span<uint8_t> output)
 {
     return 0;
 }
@@ -390,7 +372,7 @@ int OpusCodec::OpusFactory::processSdp(const resip::SdpContents::Session::Medium
 
 PCodec OpusCodec::OpusFactory::create()
 {
-    OpusCodec* result = new OpusCodec(mSamplerate, mChannels, mParams.mPtime);
+    OpusCodec* result = new OpusCodec(Audio::Format(mSamplerate, mChannels), mParams.mPtime);
     result->applyParams(mParams);
     PCodec c(result);
     mCodecList.push_back(c);
@@ -398,8 +380,8 @@ PCodec OpusCodec::OpusFactory::create()
     return c;
 }
 
-OpusCodec::OpusCodec(int samplerate, int channels, int ptime)
-    :mEncoderCtx(nullptr), mDecoderCtx(nullptr), mChannels(channels), mPTime(ptime), mSamplerate(samplerate), mDecoderChannels(0)
+OpusCodec::OpusCodec(Audio::Format fmt, int ptime)
+    :mEncoderCtx(nullptr), mDecoderCtx(nullptr), mChannels(fmt.channels()), mPTime(ptime), mSamplerate(fmt.rate()), mDecoderChannels(0)
 {
     int status;
     mEncoderCtx = opus_encoder_create(mSamplerate, mChannels, OPUS_APPLICATION_VOIP, &status);
@@ -441,52 +423,34 @@ OpusCodec::~OpusCodec()
     }
 }
 
-const char* OpusCodec::name()
-{
-    return OPUS_CODEC_NAME;
+Codec::Info OpusCodec::info() {
+    return {
+        .mName = OPUS_CODEC_NAME,
+        .mSamplerate = mSamplerate,
+        .mChannels = mChannels,
+        .mPcmLength = (int)(mSamplerate / 1000 * sizeof(short) * mChannels * mPTime),
+        .mFrameTime = mPTime,
+        .mRtpLength = 0 /* VBR */
+    };
 }
 
-int OpusCodec::pcmLength()
-{
-    return (samplerate() / 1000 ) * frameTime() * sizeof(short) * channels();
-}
-
-int OpusCodec::channels()
-{
-    return mChannels;
-}
-
-int OpusCodec::rtpLength()
-{
-    return 0; // VBR
-}
-
-int OpusCodec::frameTime()
-{
-    return mPTime;
-}
-
-int OpusCodec::samplerate()
-{
-    return mSamplerate;
-}
-
-int OpusCodec::encode(const void* input, int inputBytes, void* output, int outputCapacity)
+Codec::EncodeResult OpusCodec::encode(std::span<const uint8_t> input, std::span<uint8_t> output)
 {
     // Send number of samples for input and number of bytes for output
-    int written = opus_encode(mEncoderCtx, (const opus_int16*)input, inputBytes / (sizeof(short) * channels()), (unsigned char*)output, outputCapacity / (sizeof(short) * channels()));
+    int written = opus_encode(mEncoderCtx, (const opus_int16*)input.data(), input.size_bytes() / (sizeof(short) * channels()),
+                              output.data(), output.size_bytes() / (sizeof(short) * channels()));
     if (written < 0)
-        return 0;
+        return {.mEncoded = 0};
     else
-        return written;
+        return {.mEncoded = (size_t)written};
 }
 
-int OpusCodec::decode(const void* input, int inputBytes, void* output, int outputCapacity)
+Codec::DecodeResult OpusCodec::decode(std::span<const uint8_t> input, std::span<uint8_t> output)
 {
     int result = 0;
 
     // Examine the number of channels available in incoming packet
-    int nr_of_channels = opus_packet_get_nb_channels((const unsigned char *) input);
+    int nr_of_channels = opus_packet_get_nb_channels(input.data());
 
     // Recreate decoder if needed
     if (mDecoderChannels != nr_of_channels)
@@ -504,24 +468,22 @@ int OpusCodec::decode(const void* input, int inputBytes, void* output, int outpu
         int status = 0;
         mDecoderCtx = opus_decoder_create(mSamplerate, mDecoderChannels, &status);
         if (status)
-            return 0;
+            return {0};
     }
 
-    int nr_of_frames = opus_decoder_get_nb_samples(mDecoderCtx, (const unsigned char *) input,
-                                                   inputBytes);
+    int nr_of_frames = opus_decoder_get_nb_samples(mDecoderCtx, input.data(), input.size_bytes());
     if (nr_of_frames <= 0)
-        return 0;
+        return {0};
 
     // We support stereo and mono here.
     int buffer_capacity = nr_of_frames * sizeof(opus_int16) * nr_of_channels;
     opus_int16 *buffer_decode = (opus_int16 *)alloca(buffer_capacity);
-    int decoded = opus_decode(mDecoderCtx,
-                              reinterpret_cast<const unsigned char *>(input), inputBytes,
+    int decoded = opus_decode(mDecoderCtx, input.data(), input.size_bytes(),
                               buffer_decode, nr_of_frames, 0);
     if (decoded < 0)
     {
         ICELogCritical(<< "opus_decode() returned " << decoded);
-        return 0;
+        return {0};
     }
 
     opus_int16 *buffer_stereo = nullptr;
@@ -535,14 +497,14 @@ int OpusCodec::decode(const void* input, int inputBytes, void* output, int outpu
                 buffer_stereo[i * 2 + 1] = buffer_decode[i];
                 buffer_stereo[i * 2] = buffer_decode[i];
             }
-            assert(buffer_stereo_capacity <= outputCapacity);
-            memcpy(output, buffer_stereo, buffer_stereo_capacity);
+            assert(buffer_stereo_capacity <= output.size_bytes());
+            memcpy(output.data(), buffer_stereo, buffer_stereo_capacity);
             result = buffer_stereo_capacity;
             break;
 
         case 2:
-            assert(buffer_capacity <= outputCapacity);
-            memcpy(output, buffer_decode, buffer_capacity);
+            assert(buffer_capacity <= output.size_bytes());
+            memcpy(output.data(), buffer_decode, buffer_capacity);
             result = buffer_capacity;
             break;
 
@@ -550,17 +512,17 @@ int OpusCodec::decode(const void* input, int inputBytes, void* output, int outpu
             assert(0);
     }
 
-    return result;
+    return {.mDecoded = (size_t)result};
 }
 
-int OpusCodec::plc(int lostPackets, void* output, int outputCapacity)
+size_t OpusCodec::plc(int lostPackets, std::span<uint8_t> output)
 {
     // Find how much frames do we need to produce and prefill it with silence
     int frames_per_packet = (int)pcmLength() / (sizeof(opus_int16) * channels());
-    memset(output, 0, outputCapacity);
+    memset(output.data(), 0, output.size_bytes());
 
     // Use this pointer as output
-    opus_int16* data_output = reinterpret_cast<opus_int16*>(output);
+    opus_int16* data_output = reinterpret_cast<opus_int16*>(output.data());
 
     int nr_of_decoded_frames = 0;
 
@@ -575,10 +537,7 @@ int OpusCodec::plc(int lostPackets, void* output, int outputCapacity)
             case 1:
                 // Convert mono to stereo
                 for (int i=0; i < nr_of_decoded_frames; i++)
-                {
-                    data_output[i * 2] = buffer_plc[i];
-                    data_output[i * 2 + 1] = buffer_plc[i+1];
-                }
+                    data_output[i * 2] = data_output[i * 2 + 1] = buffer_plc[i];
                 data_output += frames_per_packet * mChannels;
                 break;
 
@@ -589,14 +548,14 @@ int OpusCodec::plc(int lostPackets, void* output, int outputCapacity)
                 break;
         }
     }
-    return ((char*)data_output - (char*)output) * sizeof(opus_int16);
+    return ((uint8_t*)data_output - output.data());
 }
 
 // -------------- ILBC -------------------
 #define ILBC_CODEC_NAME "ILBC"
 
 IlbcCodec::IlbcCodec(int packetTime)
-    :mPacketTime(packetTime), mEncoderCtx(nullptr), mDecoderCtx(nullptr)
+    :mPacketTime(packetTime)
 {
     WebRtcIlbcfix_EncoderCreate(&mEncoderCtx);
     WebRtcIlbcfix_DecoderCreate(&mDecoderCtx);
@@ -610,44 +569,31 @@ IlbcCodec::~IlbcCodec()
     WebRtcIlbcfix_EncoderFree(mEncoderCtx);
 }
 
-const char* IlbcCodec::name()
+Codec::Info IlbcCodec::info()
 {
-    return "ilbc";
+    return {
+        .mName = ILBC_CODEC_NAME,
+        .mSamplerate = 8000,
+        .mChannels = 1,
+        .mPcmLength = mPacketTime * 8 * (int)sizeof(short),
+        .mFrameTime = mPacketTime,
+        .mRtpLength = (mPacketTime == 20) ? 38 : 50
+    };
 }
 
-int IlbcCodec::rtpLength()
+Codec::EncodeResult IlbcCodec::encode(std::span<const uint8_t> input, std::span<uint8_t> output)
 {
-    return (mPacketTime == 20 ) ? 38 : 50;
-}
-
-int IlbcCodec::pcmLength()
-{
-    return mPacketTime * 16;
-}
-
-int IlbcCodec::frameTime()
-{
-    return mPacketTime;
-}
-
-int IlbcCodec::samplerate()
-{
-    return 8000;
-}
-
-int IlbcCodec::encode(const void *input, int inputBytes, void* outputBuffer, int outputCapacity)
-{
-    if (inputBytes % pcmLength())
-        return 0;
+    if (input.size_bytes() % pcmLength())
+        return {};
 
     // Declare the data input pointer
-    short *dataIn = (short *)input;
+    short *dataIn = (short *)input.data();
 
     // Declare the data output pointer
-    char *dataOut = (char *)outputBuffer;
+    char *dataOut = (char *)output.data();
 
     // Find how much RTP frames will be generated
-    unsigned int frames = inputBytes / pcmLength();
+    unsigned int frames = input.size_bytes() / pcmLength();
 
     // Generate frames
     for (unsigned int i=0; i<frames; i++)
@@ -657,15 +603,15 @@ int IlbcCodec::encode(const void *input, int inputBytes, void* outputBuffer, int
         dataOut += rtpLength();
     }
 
-    return frames * rtpLength();
+    return {frames * rtpLength()};
 }
 
-int IlbcCodec::decode(const void* input, int inputBytes, void* output, int outputCapacity)
+Codec::DecodeResult IlbcCodec::decode(std::span<const uint8_t> input, std::span<uint8_t> output)
 {
-    unsigned frames = inputBytes / rtpLength();
+    unsigned frames = input.size_bytes() / rtpLength();
 
-    char* dataIn = (char*)input;
-    short* dataOut = (short*)output;
+    char* dataIn = (char*)input.data();
+    short* dataOut = (short*)output.data();
 
     for (unsigned i=0; i < frames; ++i)
     {
@@ -675,12 +621,12 @@ int IlbcCodec::decode(const void* input, int inputBytes, void* output, int outpu
         dataOut += pcmLength() / 2;
     }
 
-    return frames * pcmLength();
+    return {frames * pcmLength()};
 }
 
-int IlbcCodec::plc(int lostFrames, void* output, int outputCapacity)
+size_t IlbcCodec::plc(int lostFrames, std::span<uint8_t> output)
 {
-    return 2 * WebRtcIlbcfix_DecodePlc(mDecoderCtx, (WebRtc_Word16*)output, lostFrames);
+    return sizeof(short) * WebRtcIlbcfix_DecodePlc(mDecoderCtx, (WebRtc_Word16*)output.data(), lostFrames);
 }
 
 // --- IlbcFactory ---
@@ -795,38 +741,24 @@ IsacCodec::~IsacCodec()
     WebRtcIsacfix_Free(mDecoderCtx); mDecoderCtx = NULL;
 }
 
-const char* IsacCodec::name()
-{
-    return "isac";
+Codec::Info IsacCodec::info() {
+    return {
+        .mName = "isac",
+        .mSamplerate = mSamplerate,
+        .mChannels = 1,
+        .mPcmLength = 60 * mSamplerate / 1000 * 2,
+        .mFrameTime = 60,
+        .mRtpLength = 0
+    };
 }
 
-int IsacCodec::frameTime()
+Codec::EncodeResult IsacCodec::encode(std::span<const uint8_t> input, std::span<uint8_t> output)
 {
-    return 60;
-}
-
-int IsacCodec::samplerate()
-{
-    return mSamplerate;
-}
-
-int IsacCodec::pcmLength()
-{
-    return frameTime() * samplerate() / 1000 * sizeof(short);
-}
-
-int IsacCodec::rtpLength()
-{
-    return 0;
-}
-
-int IsacCodec::encode(const void* input, int inputBytes, void* output, int outputCapacity)
-{
-    unsigned nrOfSamples = inputBytes / 2;
+    unsigned nrOfSamples = input.size_bytes() / 2;
     unsigned timeLength = nrOfSamples / (mSamplerate / 1000);
     int encoded = 0;
-    char* dataOut = (char*)output;
-    const WebRtc_Word16* dataIn = (const WebRtc_Word16*)input;
+    char* dataOut = (char*)output.data();
+    const WebRtc_Word16* dataIn = (const WebRtc_Word16*)input.data();
 
     // Iterate 10 milliseconds chunks
     for (unsigned i=0; i<timeLength/10; i++)
@@ -835,25 +767,25 @@ int IsacCodec::encode(const void* input, int inputBytes, void* output, int outpu
         if (encoded > 0)
             dataOut += encoded;
     }
-    return dataOut - (char*)output;
+    return {.mEncoded = (size_t)(dataOut - (char*)output.data())};
 }
 
-int IsacCodec::decode(const void* input, int inputBytes, void* output, int outputCapacity)
+Codec::DecodeResult IsacCodec::decode(std::span<const uint8_t> input, std::span<uint8_t> output)
 {
     WebRtc_Word16 speechType = 0;
-    unsigned produced = WebRtcIsacfix_Decode(mDecoderCtx, (const WebRtc_UWord16*)input, inputBytes, (WebRtc_Word16*)output, &speechType);
+    unsigned produced = WebRtcIsacfix_Decode(mDecoderCtx, (const WebRtc_UWord16*)input.data(), input.size_bytes(), (WebRtc_Word16*)output.data(), &speechType);
     if (produced == (unsigned)-1)
-        return 0;
+        return {.mDecoded = 0};
 
-    return produced * 2;
+    return {.mDecoded = produced * 2};
 }
 
-int IsacCodec::plc(int lostFrames, void* output, int outputCapacity)
+size_t IsacCodec::plc(int lostFrames, std::span<uint8_t> output)
 {
     // lostFrames are 30-milliseconds frames; but used encoding mode is 60 milliseconds.
     // So lostFrames * 2
     lostFrames *=2 ;
-    if (-1 == WebRtcIsacfix_DecodePlc(mDecoderCtx, (WebRtc_Word16*)output, lostFrames ))
+    if (-1 == WebRtcIsacfix_DecodePlc(mDecoderCtx, (WebRtc_Word16*)output.data(), lostFrames ))
         return 0;
 
     return lostFrames * 30 * (samplerate()/1000 * sizeof(short));
@@ -916,71 +848,55 @@ PCodec IsacCodec::IsacFactory32K::create()
 
 G711Codec::G711Codec(int type)
     :mType(type)
-{
-}
+{}
 
 G711Codec::~G711Codec()
-{
+{}
+
+Codec::Info G711Codec::info() {
+    return {
+        .mName = mType == ALaw ? "PCMA" : "PCMU",
+        .mSamplerate = 8000,
+        .mChannels = 1,
+        .mPcmLength = 10 * 16,
+        .mFrameTime = 10,
+        .mRtpLength = 10 * 8
+    };
 }
 
-const char* G711Codec::name()
-{
-    return "g711";
-}
-
-int G711Codec::pcmLength()
-{
-    return frameTime() * 16;
-}
-
-int G711Codec::rtpLength()
-{
-    return frameTime() * 8;
-}
-
-int G711Codec::frameTime()
-{
-    return 10;
-}
-
-int G711Codec::samplerate()
-{
-    return 8000;
-}
-
-int G711Codec::encode(const void* input, int inputBytes, void* output, int outputCapacity)
+Codec::EncodeResult G711Codec::encode(std::span<const uint8_t> input, std::span<uint8_t> output)
 {
     int result;
     if (mType == ALaw)
-        result = WebRtcG711_EncodeA(NULL, (WebRtc_Word16*)input, inputBytes/2, (WebRtc_Word16*)output);
+        result = WebRtcG711_EncodeA(nullptr, (WebRtc_Word16*)input.data(), input.size_bytes() / 2, (WebRtc_Word16*)output.data());
     else
-        result = WebRtcG711_EncodeU(NULL, (WebRtc_Word16*)input, inputBytes/2, (WebRtc_Word16*)output);
+        result = WebRtcG711_EncodeU(nullptr, (WebRtc_Word16*)input.data(), input.size_bytes() / 2, (WebRtc_Word16*)output.data());
 
-    if (result == -1)
-        throw Exception(ERR_WEBRTC, -1);
+    if (result < 0)
+        return {.mEncoded = 0};
 
-    return result;
+    return {.mEncoded = (size_t) result};
 }
 
-int G711Codec::decode(const void* input, int inputBytes, void* output, int outputCapacity)
+Codec::DecodeResult G711Codec::decode(std::span<const uint8_t> input, std::span<uint8_t> output)
 {
-    assert(outputCapacity >= inputBytes * 2);
+    assert(output.size_bytes() >= input.size_bytes() * 2);
 
     int result;
     WebRtc_Word16 speechType;
 
     if (mType == ALaw)
-        result = WebRtcG711_DecodeA(NULL, (WebRtc_Word16*)input, inputBytes, (WebRtc_Word16*)output, &speechType);
+        result = WebRtcG711_DecodeA(nullptr, (WebRtc_Word16*)input.data(), input.size_bytes(), (WebRtc_Word16*)output.data(), &speechType);
     else
-        result = WebRtcG711_DecodeU(NULL, (WebRtc_Word16*)input, inputBytes, (WebRtc_Word16*)output, &speechType);
+        result = WebRtcG711_DecodeU(nullptr, (WebRtc_Word16*)input.data(), input.size_bytes(), (WebRtc_Word16*)output.data(), &speechType);
 
-    if (result == -1)
-        throw Exception(ERR_WEBRTC, -1);
+    if (result < 0)
+        return {.mDecoded = 0};
 
-    return result * 2;
+    return {.mDecoded = (size_t)result * 2};
 }
 
-int G711Codec::plc(int lostSamples, void* output, int outputCapacity)
+size_t G711Codec::plc(int lostSamples, std::span<uint8_t> output)
 {
     return 0;
 }
@@ -1060,86 +976,64 @@ GsmCodec::GsmCodec(Type codecType)
 
 GsmCodec::~GsmCodec()
 {
-    gsm_destroy(mGSM);
+    gsm_destroy(mGSM); mGSM = nullptr;
 }
 
-const char* GsmCodec::name()
-{
-    return "GSM-06.10";
-}
-
-int GsmCodec::rtpLength()
-{
+Codec::Info GsmCodec::info() {
+    int rtpLength = 0;
     switch (mCodecType)
     {
-    case Type::Bytes_31:
-        return GSM_RTPFRAME_SIZE_31;
-        break;
-
-    case Type::Bytes_32:
-        return GSM_RTPFRAME_SIZE_32;
-
-    case Type::Bytes_33:
-        return GSM_RTPFRAME_SIZE_33;
-
-    case Type::Bytes_65:
-        return GSM_RTPFRAME_SIZE_32 + GSM_RTPFRAME_SIZE_33;
-
+    case Type::Bytes_31: rtpLength = GSM_RTPFRAME_SIZE_31; break;
+    case Type::Bytes_32: rtpLength = GSM_RTPFRAME_SIZE_32; break;
+    case Type::Bytes_33: rtpLength = GSM_RTPFRAME_SIZE_33; break;
+    case Type::Bytes_65: rtpLength = GSM_RTPFRAME_SIZE_32 + GSM_RTPFRAME_SIZE_33; break;
+    default:             rtpLength = GSM_RTPFRAME_SIZE_33;
     }
 
-    return GSM_RTPFRAME_SIZE_33;
+    return {
+        .mName = "GSM-06.10",
+        .mSamplerate = 8000,
+        .mChannels = 1,
+        .mPcmLength = GSM_AUDIOFRAME_TIME * 16,
+        .mFrameTime = GSM_AUDIOFRAME_TIME,
+        .mRtpLength = rtpLength
+    };
 }
 
-int GsmCodec::pcmLength()
-{
-    return GSM_AUDIOFRAME_TIME * 16;
-}
-
-int GsmCodec::frameTime()
-{
-    return GSM_AUDIOFRAME_TIME;
-}
-
-int GsmCodec::samplerate()
-{
-    return 8000;
-}
-
-int GsmCodec::encode(const void* input, int inputBytes, void* output, int outputCapacity)
+Codec::EncodeResult GsmCodec::encode(std::span<const uint8_t> input, std::span<uint8_t> output)
 {
     int outputBytes = 0;
+    char* outputBuffer = (char*)output.data();
 
-    char* outputBuffer = (char*)output;
-
-    for (int i = 0; i < inputBytes/pcmLength(); i++)
+    for (int i = 0; i < input.size_bytes() / pcmLength(); i++)
     {
-        gsm_encode(mGSM, (gsm_signal *)input+160*i, (gsm_byte*)outputBuffer);
+        gsm_encode(mGSM, (gsm_signal *)input.data()+160*i, (gsm_byte*)outputBuffer);
         outputBuffer += rtpLength();
         outputBytes += rtpLength();
     }
-    return outputBytes;
+    return {.mEncoded = (size_t)outputBytes};
 }
 
 
-int GsmCodec::decode(const void* input, int inputBytes, void* output, int outputCapacity)
+Codec::DecodeResult GsmCodec::decode(std::span<const uint8_t> input, std::span<uint8_t> output)
 {
-    if (inputBytes % rtpLength() != 0)
-        return 0;
+    if (input.size_bytes() % rtpLength() != 0)
+        return {.mDecoded = 0};
 
     int i=0;
-    for (i = 0; i < inputBytes/rtpLength(); i++)
-        gsm_decode(mGSM, (gsm_byte *)input + 33 * i, (gsm_signal *)output + 160 * i);
+    for (i = 0; i < input.size_bytes() / rtpLength(); i++)
+        gsm_decode(mGSM, (gsm_byte *)input.data() + 33 * i, (gsm_signal *)output.data() + 160 * i);
 
-    return i * 320;
+    return {.mDecoded = (size_t)i * 320};
 }
 
-int GsmCodec::plc(int lostFrames, void* output, int outputCapacity)
+size_t GsmCodec::plc(int lostFrames, std::span<uint8_t> output)
 {
-    if (outputCapacity < lostFrames * pcmLength())
+    if (output.size_bytes() < lostFrames * pcmLength())
         return 0;
 
     // Return silence frames
-    memset(output, 0, lostFrames * pcmLength());
+    memset(output.data(), 0, lostFrames * pcmLength());
     return lostFrames * pcmLength();
 }
 
@@ -1155,58 +1049,52 @@ G722Codec::G722Codec()
 
 G722Codec::~G722Codec()
 {
-    g722_decode_release((g722_decode_state_t*)mDecoder);
-    g722_encode_release((g722_encode_state_t*)mEncoder);
+    g722_decode_release((g722_decode_state_t*)mDecoder); mDecoder = nullptr;
+    g722_encode_release((g722_encode_state_t*)mEncoder); mEncoder = nullptr;
 }
 
-const char* G722Codec::name()
-{
-    return G722_MIME_NAME;
+Codec::Info G722Codec::info() {
+    // ToDo: double check the G722 calls - remember RFC has bug about samplerate
+    return {
+        .mName = G722_MIME_NAME,
+        .mSamplerate = 8000,
+        .mChannels = 1,
+        .mPcmLength = 640,
+        .mFrameTime = 20,
+        .mRtpLength = 160
+    };
 }
 
-int G722Codec::pcmLength()
+Codec::EncodeResult G722Codec::encode(std::span<const uint8_t> input, std::span<uint8_t> output)
 {
-    return 640;
+    if (output.size_bytes() < input.size_bytes() / 4)
+        return {.mEncoded = 0}; // Destination buffer not big enough
+
+    int r = g722_encode((g722_encode_state_t *)mEncoder, (unsigned char*)output.data(), ( short*)input.data(), input.size_bytes() / 2);
+    if (r < 0)
+        return {.mEncoded = 0};
+
+    return {.mEncoded = (size_t)r};
 }
 
-int G722Codec::frameTime()
+Codec::DecodeResult G722Codec::decode(std::span<const uint8_t> input, std::span<uint8_t> output)
 {
-    return 20;
+    if (output.size_bytes() < input.size_bytes() * 4)
+        return {.mDecoded = 0}; // Destination buffer not big enough
+
+    int r = g722_decode((g722_decode_state_t *)mDecoder, (short*)output.data(), (unsigned char*)input.data(), input.size_bytes()) * 2;
+    if (r < 0)
+        return {.mDecoded = 0};
+    return {.mDecoded = (size_t)r};
 }
 
-int G722Codec::rtpLength()
+size_t G722Codec::plc(int lostFrames, std::span<uint8_t> output)
 {
-    return 160;
-}
-
-int G722Codec::samplerate()
-{
-    return 8000;
-}
-
-int G722Codec::encode(const void* input, int inputBytes, void* output, int outputCapacity)
-{
-    if (outputCapacity < inputBytes / 4)
-        return 0; // Destination buffer not big enough
-
-    return g722_encode((g722_encode_state_t *)mEncoder, (unsigned char*)output, ( short*)input, inputBytes / 2);
-}
-
-int G722Codec::decode(const void* input, int inputBytes, void* output, int outputCapacity)
-{
-    if (outputCapacity < inputBytes * 4)
-        return 0; // Destination buffer not big enough
-
-    return g722_decode((g722_decode_state_t *)mDecoder, ( short*)output, (unsigned char*)input, inputBytes) * 2;
-}
-
-int G722Codec::plc(int lostFrames, void* output, int outputCapacity)
-{
-    if (outputCapacity < lostFrames * pcmLength())
+    if (output.size_bytes() < lostFrames * pcmLength())
         return 0;
 
     // Return silence frames
-    memset(output, 0, lostFrames * pcmLength());
+    memset(output.data(), 0, lostFrames * pcmLength());
     return lostFrames * pcmLength();
 }
 
@@ -1318,7 +1206,6 @@ static bool repackHalfRate(BitReader& br, uint16_t frame[22], bool& lastItem)
 }
 
 GsmHrCodec::GsmHrCodec()
-    :mDecoder(nullptr)
 {
     mDecoder = new GsmHr::Codec();
 }
@@ -1329,34 +1216,21 @@ GsmHrCodec::~GsmHrCodec()
     mDecoder = nullptr;
 }
 
-const char* GsmHrCodec::name()
-{
-    return "GSM-HR-08";
+Codec::Info GsmHrCodec::info() {
+    return {
+        .mName = "GSM-HR-08",
+        .mSamplerate = 8000,
+        .mChannels = 1,
+        .mPcmLength = 20 * 8 * 2,
+        .mFrameTime = 20,
+        .mRtpLength = 0
+    };
 }
 
-int GsmHrCodec::pcmLength()
+Codec::EncodeResult GsmHrCodec::encode(std::span<const uint8_t> input, std::span<uint8_t> output)
 {
-    return frameTime() * 8 * 2;
-}
-
-int GsmHrCodec::rtpLength()
-{
-    return 0;
-}
-
-int GsmHrCodec::frameTime()
-{
-    return 20;
-}
-
-int GsmHrCodec::samplerate()
-{
-    return 8000;
-}
-
-int GsmHrCodec::encode(const void* input, int inputBytes, void* output, int outputCapacity)
-{
-    return 0;
+    // Not supported yet
+    return {.mEncoded = 0};
 }
 
 static const int params_unvoiced[] = {
@@ -1447,23 +1321,23 @@ hr_ref_from_canon(uint16_t *hr_ref, const uint8_t *canon)
 [+] PQ: Adding conversion from canon to rawpcm-s16le (for codec pcm)
 [+] PQ: Adding file output (blk_len=320)
 */
-int GsmHrCodec::decode(const void* input, int inputBytes, void* output, int outputCapacity)
+Codec::DecodeResult GsmHrCodec::decode(std::span<const uint8_t> input, std::span<uint8_t> output)
 {
-    ByteBuffer bb(input, inputBytes, ByteBuffer::CopyBehavior::UseExternal);
+    ByteBuffer bb(input, ByteBuffer::CopyBehavior::UseExternal);
     BitReader br(bb);
     uint16_t hr_ref[22];
 
-    hr_ref_from_canon(hr_ref, (const uint8_t*)input + 1);
+    hr_ref_from_canon(hr_ref, input.data() + 1);
     hr_ref[18] = 0;		/* BFI : 1 bit */
     hr_ref[19] = 0;		/* UFI : 1 bit */
     hr_ref[20] = 0;		/* SID : 2 bit */
     hr_ref[21] = 0;		/* TAF : 1 bit */
 
-    reinterpret_cast<GsmHr::Codec*>(mDecoder)->speechDecoder((int16_t*)hr_ref, (int16_t*)output);
-    return 320;
+    reinterpret_cast<GsmHr::Codec*>(mDecoder)->speechDecoder((int16_t*)hr_ref, (int16_t*)output.data());
+    return {.mDecoded = 320};
 }
 
-int GsmHrCodec::plc(int lostFrames, void* output, int outputCapacity)
+size_t GsmHrCodec::plc(int lostFrames, std::span<uint8_t> output)
 {
     return 0;
 }

src/engine/media/MT_AudioCodec.h

@@ -40,42 +40,41 @@ public:
     {
     public:
         const char* name() override;
-        int channels() override;
-        int samplerate() override;
-        int payloadType() override;
+        int  channels() override;
+        int  samplerate() override;
+        int  payloadType() override;
 
         void updateSdp(resip::SdpContents::Session::Medium::CodecContainer& codecs, SdpDirection direction) override;
-        int processSdp(const resip::SdpContents::Session::Medium::CodecContainer& codecs, SdpDirection direction) override;
+        int  processSdp(const resip::SdpContents::Session::Medium::CodecContainer& codecs, SdpDirection direction) override;
 
         PCodec create() override;
     };
     G729Codec();
     ~G729Codec() override;
 
-    const char* name() override;
-    int pcmLength() override;
-    int rtpLength() override;
-    int frameTime() override;
-    int samplerate() override;
-    int channels() override;
-    int encode(const void* input, int inputBytes, void* output, int outputCapacity) override;
-    int decode(const void* input, int inputBytes, void* output, int outputCapacity) override;
-    int plc(int lostFrames, void* output, int outputCapacity) override;
+    Info info() override;
+
+    EncodeResult    encode(std::span<const uint8_t> input, std::span<uint8_t> output) override;
+    DecodeResult    decode(std::span<const uint8_t> input, std::span<uint8_t> output) override;
+    size_t          plc(int lostFrames, std::span<uint8_t> output) override;
 };
 
 class OpusCodec: public Codec
 {
 protected:
-    OpusEncoder        *mEncoderCtx;
-    OpusDecoder        *mDecoderCtx;
-    int mPTime, mSamplerate, mChannels;
-    // Audio::SpeexResampler mDecodeResampler;
-    int mDecoderChannels;
+    OpusEncoder        *mEncoderCtx = nullptr;
+    OpusDecoder        *mDecoderCtx = nullptr;
+    int mPTime = 0, mSamplerate = 0, mChannels = 0;
+    int mDecoderChannels = 0;
 public:
     struct Params
     {
-        bool mUseDtx, mUseInbandFec, mStereo;
-        int mPtime, mTargetBitrate, mExpectedPacketLoss;
+        bool mUseDtx = false,
+             mUseInbandFec = false,
+             mStereo = false;
+        int mPtime = 0,
+            mTargetBitrate = 0,
+            mExpectedPacketLoss = 0;
 
         Params();
         resip::Data toString() const;
@@ -102,28 +101,24 @@ public:
         PCodec create() override;
     };
 
-    OpusCodec(int samplerate, int channels, int ptime);
+    OpusCodec(Audio::Format fmt, int ptime);
     ~OpusCodec();
     void applyParams(const Params& params);
 
-    const char* name();
-    int pcmLength();
-    int rtpLength();
-    int frameTime();
-    int samplerate();
-    int channels();
-    int encode(const void* input, int inputBytes, void* output, int outputCapacity);
-    int decode(const void* input, int inputBytes, void* output, int outputCapacity);
-    int plc(int lostFrames, void* output, int outputCapacity);
+    Info info() override;
+
+    EncodeResult    encode(std::span<const uint8_t> input, std::span<uint8_t> output) override;
+    DecodeResult    decode(std::span<const uint8_t> input, std::span<uint8_t> output) override;
+    size_t          plc(int lostFrames, std::span<uint8_t> output) override;
 };
 
 
 class IlbcCodec: public Codec
 {
 protected:
-    int                 mPacketTime;          /// Single frame time (20 or 30 ms)
-    iLBC_encinst_t*     mEncoderCtx;
-    iLBC_decinst_t*     mDecoderCtx;
+    int                 mPacketTime = 0;          /// Single frame time (20 or 30 ms)
+    iLBC_encinst_t*     mEncoderCtx = nullptr;
+    iLBC_decinst_t*     mDecoderCtx = nullptr;
     
 public:
     class IlbcFactory: public Factory
@@ -146,14 +141,11 @@ public:
 
     IlbcCodec(int packetTime);
     virtual ~IlbcCodec();
-    const char* name();
-    int pcmLength();
-    int rtpLength();
-    int frameTime();
-    int samplerate();
-    int encode(const void* input, int inputBytes, void* output, int outputCapacity);
-    int decode(const void* input, int inputBytes, void* output, int outputCapacity);
-    int plc(int lostFrames, void* output, int outputCapacity);
+    Info info() override;
+
+    EncodeResult    encode(std::span<const uint8_t> input, std::span<uint8_t> output) override;
+    DecodeResult    decode(std::span<const uint8_t> input, std::span<uint8_t> output) override;
+    size_t          plc(int lostFrames, std::span<uint8_t> output) override;
 };
 
 class G711Codec: public Codec
@@ -186,15 +178,11 @@ public:
     G711Codec(int type);
     ~G711Codec();
     
-    const char* name();
-    int    pcmLength();
-    int    frameTime();
-    int    rtpLength();
-    int    samplerate();
+    Info info() override;
 
-    int encode(const void* input, int inputBytes, void* output, int outputCapacity);
-    int decode(const void* input, int inputBytes, void* output, int outputCapacity);
-    int plc(int lostSamples, void* output, int outputCapacity);
+    EncodeResult encode(std::span<const uint8_t> input, std::span<uint8_t> output) override;
+    DecodeResult decode(std::span<const uint8_t> input, std::span<uint8_t> output) override;
+    size_t       plc(int lostSamples, std::span<uint8_t> output) override ;
 
 protected:
     int mType;    /// Determines if it is u-law or a-law codec. Its value is ALaw or ULaw.
@@ -203,9 +191,9 @@ protected:
 class IsacCodec: public Codec
 {
 protected:
-    int                       mSamplerate;
-    ISACFIX_MainStruct*       mEncoderCtx;
-    ISACFIX_MainStruct*       mDecoderCtx;
+    int                       mSamplerate = 0;
+    ISACFIX_MainStruct*       mEncoderCtx = nullptr;
+    ISACFIX_MainStruct*       mDecoderCtx = nullptr;
 public:
     class IsacFactory16K: public Factory
     {
@@ -237,15 +225,11 @@ public:
     IsacCodec(int sampleRate);
     ~IsacCodec();
     
-    const char* name();
-    int pcmLength();
-    int rtpLength();
-    int frameTime();
-    int samplerate();
+    Info info() override;
 
-    int encode(const void* input, int inputBytes, void* output, int outputCapacity);
-    int decode(const void* input, int inputBytes, void* output, int outputCapacity);
-    int plc(int lostFrames, void* output, int outputCapacity);
+    EncodeResult    encode(std::span<const uint8_t> input, std::span<uint8_t> output) override;
+    DecodeResult    decode(std::span<const uint8_t> input, std::span<uint8_t> output) override;
+    size_t          plc(int lostFrames, std::span<uint8_t> output) override;
 };
 
 
@@ -309,17 +293,13 @@ public:
     GsmCodec(Type codecType);
 
     /*! Destructor. */
-    virtual ~GsmCodec();
+    ~GsmCodec();
 
-    const char* name();
-    int pcmLength();
-    int rtpLength();
-    int frameTime();
-    int samplerate();
+    Info info() override;
 
-    int encode(const void* input, int inputBytes, void* output, int outputCapacity);
-    int decode(const void* input, int inputBytes, void* output, int outputCapacity);
-    int plc(int lostFrames, void* output, int outputCapacity);
+    EncodeResult encode(std::span<const uint8_t> input, std::span<uint8_t> output) override;
+    DecodeResult decode(std::span<const uint8_t> input, std::span<uint8_t> output) override;
+    size_t plc(int lostFrames, std::span<uint8_t> output) override;
 };
 
 /// GSM MIME name
@@ -358,25 +338,19 @@ public:
         PCodec create();
     };
     G722Codec();
-    virtual ~G722Codec();
+    ~G722Codec();
 
-    const char* name();
-    int pcmLength();
-    int rtpLength();
-    int frameTime();
-    int samplerate();
+    Info info() override;
 
-    int encode(const void* input, int inputBytes, void* output, int outputCapacity);
-    int decode(const void* input, int inputBytes, void* output, int outputCapacity);
-    int plc(int lostFrames, void* output, int outputCapacity);
-
-    //unsigned GetSamplerate() { return 16000; }
+    EncodeResult encode(std::span<const uint8_t> input, std::span<uint8_t> output) override;
+    DecodeResult decode(std::span<const uint8_t> input, std::span<uint8_t> output) override;
+    size_t plc(int lostFrames, std::span<uint8_t> output) override;
 };
 
 class GsmHrCodec: public Codec
 {
 protected:
-    void* mDecoder;
+    void* mDecoder = nullptr;
 
 public:
     class GsmHrFactory: public Factory
@@ -396,15 +370,11 @@ public:
     GsmHrCodec();
     ~GsmHrCodec() override;
 
-    const char* name() override;
-    int pcmLength() override;
-    int rtpLength() override;
-    int frameTime() override;
-    int samplerate() override;
+    Info info() override;
 
-    int encode(const void* input, int inputBytes, void* output, int outputCapacity) override;
-    int decode(const void* input, int inputBytes, void* output, int outputCapacity) override;
-    int plc(int lostFrames, void* output, int outputCapacity) override;
+    EncodeResult    encode(std::span<const uint8_t> input, std::span<uint8_t> output) override;
+    DecodeResult    decode(std::span<const uint8_t> input, std::span<uint8_t> output) override;
+    size_t          plc(int lostFrames, std::span<uint8_t> output) override;
 };
 }
 

src/engine/media/MT_AudioReceiver.cpp

@@ -11,6 +11,7 @@
 #include "MT_AudioReceiver.h"
 #include "MT_AudioCodec.h"
 #include "MT_CngHelper.h"
+#include "MT_Dtmf.h"
 #include "../helper/HL_Log.h"
 #include "../helper/HL_Time.h"
 #include "../audio/Audio_Interface.h"
@@ -115,7 +116,7 @@ bool SequenceSort(const std::shared_ptr<RtpBuffer::Packet>& p1, const std::share
     return p1->rtp()->GetExtendedSequenceNumber() < p2->rtp()->GetExtendedSequenceNumber();
 }
 
-std::shared_ptr<RtpBuffer::Packet> RtpBuffer::add(std::shared_ptr<jrtplib::RTPPacket> packet, std::chrono::milliseconds timelength, int rate)
+std::shared_ptr<RtpBuffer::Packet> RtpBuffer::add(const std::shared_ptr<jrtplib::RTPPacket>& packet, std::chrono::milliseconds timelength, int rate)
 {
     if (!packet)
         return std::shared_ptr<Packet>();
@@ -191,24 +192,24 @@ std::shared_ptr<RtpBuffer::Packet> RtpBuffer::add(std::shared_ptr<jrtplib::RTPPa
     return std::shared_ptr<Packet>();
 }
 
-RtpBuffer::FetchResult RtpBuffer::fetch(ResultList& rl)
+RtpBuffer::FetchResult RtpBuffer::fetch()
 {
     Lock l(mGuard);
 
-    FetchResult result = FetchResult::NoPacket;
-    rl.clear();
+    FetchResult result;
 
     // See if there is enough information in buffer
     auto total = findTimelength();
 
-    while (total > mHigh && mPacketList.size() && 0ms != mHigh)
+    while (total > mHigh && mPacketList.size() > 1 && 0ms != mHigh)
     {
         ICELogMedia( << "Dropping RTP packets from jitter buffer");
         total -= mPacketList.front()->timelength();
 
         // Save it as last packet however - to not confuse loss packet counter
         mFetchedPacket = mPacketList.front();
-        mLastSeqno = mPacketList.front()->rtp()->GetExtendedSequenceNumber();
+        mLastSeqno = mFetchedPacket->rtp()->GetExtendedSequenceNumber();
+        mLastReceiveTime = mFetchedPacket->rtp()->GetReceiveTime();
 
         // Erase from packet list
         mPacketList.erase(mPacketList.begin());
@@ -217,10 +218,10 @@ RtpBuffer::FetchResult RtpBuffer::fetch(ResultList& rl)
         mStat.mPacketDropped++;
     }
 
-    if (total < mLow)
+    if (total < mLow || total == 0ms)
     {
         // Still not prebuffered
-        result = FetchResult::NoPacket;
+        result = {FetchResult::Status::NoPacket};
     }
     else
     {
@@ -228,8 +229,8 @@ RtpBuffer::FetchResult RtpBuffer::fetch(ResultList& rl)
         {
             if (mPacketList.empty())
             {
-                result = FetchResult::NoPacket;
                 // Don't increase counter of lost packets here; maybe it is DTX
+                result = {FetchResult::Status::NoPacket};
             }
             else
             {
@@ -237,34 +238,39 @@ RtpBuffer::FetchResult RtpBuffer::fetch(ResultList& rl)
                 auto& packet = *mPacketList.front();
                 uint32_t seqno = packet.rtp()->GetExtendedSequenceNumber();
 
-
                 // Gap between new packet and previous on
                 int gap = (int64_t)seqno - (int64_t)*mLastSeqno - 1;
-                gap = std::min(gap, 127);
                 if (gap > 0)
                 {
                     // std::cout << "Increase the packet loss for SSRC " << std::hex << mSsrc << std::endl;
-                    mStat.mPacketLoss++;
-                    auto currentTimestamp = std::chrono::microseconds(uint64_t(packet.rtp()->GetReceiveTime().GetDouble() * 1000000));
+                    mStat.mPacketLoss += gap;
 
+                    // Report is the onetime; there is no many sequential 1-packet gap reports
                     if (mStat.mPacketLossTimeline.empty() || (mStat.mPacketLossTimeline.back().mEndSeqno != seqno))
-                        mStat.mPacketLossTimeline.push_back({.mStartSeqno = *mLastSeqno,
-                                                             .mEndSeqno = seqno,
-                                                             .mGap = gap,
-                                                             .mTimestamp = currentTimestamp});
+                    {
+                        auto gapStart = RtpHelper::toMicroseconds(*mLastReceiveTime);
+                        auto gapEnd = RtpHelper::toMicroseconds(packet.rtp()->GetReceiveTime());
+                        mStat.mPacketLossTimeline.emplace_back(PacketLossEvent{.mStartSeqno = *mLastSeqno,
+                                                                               .mEndSeqno = seqno,
+                                                                               .mGap = gap,
+                                                                               .mTimestampStart = gapStart,
+                                                                               .mTimestampEnd = gapEnd});
+                    }
 
-                    mLastSeqno = *mLastSeqno + 1; // As we deal with the audio gap - return the silence and increase last seqno
-
-                    result = FetchResult::Gap;
+                    // ToDo: here we should decide smth - 2-packet gap shoud report Status::Gap two times at least; but current implementation gives only one.
+                    // It is not big problem - as gap is detected when we have smth to return usually
+                    mLastSeqno = seqno;
+                    mLastReceiveTime = packet.rtp()->GetReceiveTime();
+                    result = {FetchResult::Status::Gap};
                 }
                 else
                 {
-                    result = FetchResult::RegularPacket;
-                    rl.push_back(mPacketList.front());
+                    result = {FetchResult::Status::RegularPacket, mPacketList.front()};
 
                     // Save last returned normal packet
-                    mFetchedPacket = mPacketList.front();
-                    mLastSeqno = mPacketList.front()->rtp()->GetExtendedSequenceNumber();
+                    mFetchedPacket = result.mPacket;
+                    mLastSeqno = result.mPacket->rtp()->GetExtendedSequenceNumber();
+                    mLastReceiveTime = result.mPacket->rtp()->GetReceiveTime();
 
                     // Remove returned packet from the list
                     mPacketList.erase(mPacketList.begin());
@@ -277,14 +283,12 @@ RtpBuffer::FetchResult RtpBuffer::fetch(ResultList& rl)
             if (findTimelength() >= mPrebuffer && !mPacketList.empty())
             {
                 // Normal packet will be returned
-                result = FetchResult::RegularPacket;
-
-                // Put it to output list
-                rl.push_back(mPacketList.front());
+                result = {FetchResult::Status::RegularPacket, mPacketList.front()};
 
                 // Remember returned packet
-                mFetchedPacket = mPacketList.front();
-                mLastSeqno = mPacketList.front()->rtp()->GetExtendedSequenceNumber();
+                mFetchedPacket = result.mPacket;
+                mLastSeqno = result.mPacket->rtp()->GetExtendedSequenceNumber();
+                mLastReceiveTime = result.mPacket->rtp()->GetReceiveTime();
 
                 // Remove returned packet from buffer list
                 mPacketList.erase(mPacketList.begin());
@@ -292,12 +296,12 @@ RtpBuffer::FetchResult RtpBuffer::fetch(ResultList& rl)
             else
             {
                 ICELogMedia(<< "Jitter buffer was not prebuffered yet; resulting no packet");
-                result = FetchResult::NoPacket;
+                result = {FetchResult::Status::NoPacket};
             }
         }
     }
 
-    if (result != FetchResult::NoPacket)
+    if (result.mStatus != FetchResult::Status::NoPacket)
         mReturnedCounter++;
 
     return result;
@@ -333,8 +337,7 @@ Receiver::~Receiver()
 
 //-------------- AudioReceiver ----------------
 AudioReceiver::AudioReceiver(const CodecList::Settings& settings, MT::Statistics &stat)
-    :Receiver(stat), mBuffer(stat), mCodecSettings(settings),
-      mCodecList(settings)
+    :Receiver(stat), mBuffer(stat), mDtmfBuffer(stat), mCodecSettings(settings), mCodecList(settings), mDtmfReceiver(stat)
 {
     // Init resamplers
     mResampler8.start(AUDIO_CHANNELS, 8000, AUDIO_SAMPLERATE);
@@ -346,6 +349,12 @@ AudioReceiver::AudioReceiver(const CodecList::Settings& settings, MT::Statistics
     mCodecList.setSettings(settings);
     mCodecList.fillCodecMap(mCodecMap);
 
+    mAvailable.setCapacity(AUDIO_SAMPLERATE * sizeof(short));
+
+    mDtmfBuffer.setPrebuffer(0ms);
+    mDtmfBuffer.setLow(0ms);
+    mDtmfBuffer.setHigh(1ms);
+
 #if defined(DUMP_DECODED)
     mDecodedDump = std::make_shared<Audio::WavFileWriter>();
     mDecodedDump->open("decoded.wav", 8000 /*G711*/, AUDIO_CHANNELS);
@@ -401,12 +410,10 @@ size_t decode_packet(Codec& codec, RTPPacket& p, void* output_buffer, size_t out
 
         for (int i=0; i < frame_count; i++)
         {
-            auto decoded_length = codec.decode(p.GetPayloadData() + i * codec.rtpLength(),
-                                               frame_length,
-                                               output_buffer,
-                                               output_capacity);
+            auto r = codec.decode({p.GetPayloadData() + i * codec.rtpLength(), (size_t)frame_length},
+                                  {(uint8_t*)output_buffer, output_capacity});
 
-            result += decoded_length;
+            result += r.mDecoded;
         }
     }
     else
@@ -428,66 +435,72 @@ bool AudioReceiver::add(const std::shared_ptr<jrtplib::RTPPacket>& p, Codec** de
     // Increase codec counter
     mStat.mCodecCount[ptype]++;
 
-    // Check if codec can be handled
-    Codec* codec = nullptr;
-    auto codecIter = mCodecMap.find(ptype);
-    if (codecIter == mCodecMap.end())
+    // Check if we deal with telephone-event
+    if (p->GetPayloadType() == mCodecSettings.mTelephoneEvent)
     {
-        // Well, there is no information about the codec; skip this packet
+        *detectedCodec = nullptr;
+        mDtmfBuffer.add(p, 10ms, 8000);
     }
     else
     {
-        // Check if codec is creating lazily
-        if (!codecIter->second)
+        // Look for codec
+        // Check if codec can be handled
+        Codec* codec = nullptr;
+        auto codecIter = mCodecMap.find(ptype);
+        if (codecIter != mCodecMap.end())
         {
-            codecIter->second = mCodecList.createCodecByPayloadType(ptype);
+            // Check if codec is creating lazily
+            if (!codecIter->second)
+            {
+                codecIter->second = mCodecList.createCodecByPayloadType(ptype);
+            }
+            codec = codecIter->second.get();
+
+            // Return pointer to codec if needed.get()
+            if (detectedCodec)
+                *detectedCodec = codec;
+
+            if (mStat.mCodecName.empty() && codec)
+                mStat.mCodecName = codec->name();
+
+
+            if (!codec)
+                time_length = 10;
+            else
+            if (!codec->rtpLength())
+                time_length = codec->frameTime();
+            else
+                time_length = lround(double(payloadLength) / codec->rtpLength() * codec->frameTime());
+
+            if (codec)
+                samplerate = codec->samplerate();
         }
-        codec = codecIter->second.get();
-
-        // Return pointer to codec if needed.get()
-        if (detectedCodec)
-            *detectedCodec = codec;
-
-        if (mStat.mCodecName.empty() && codec)
-            mStat.mCodecName = codec->name();
 
+        // Process jitter anyway - can we decode payload or not
+        mJitterStats.process(p.get(), samplerate);
+        mStat.mJitter = static_cast<float>(mJitterStats.get());
 
         if (!codec)
-            time_length = 10;
+            return false; // There is no sense to add this packet into jitter buffer - we can't decode this
+
+        // Check if packet is CNG
+        if (payloadLength >= 1 && payloadLength <= 6 && (ptype == 0 || ptype == 8))
+            time_length = mLastPacketTimeLength ? mLastPacketTimeLength : 20;
         else
-        if (!codec->rtpLength())
-            time_length = codec->frameTime();
-        else
-            time_length = lround(double(payloadLength) / codec->rtpLength() * codec->frameTime());
+        // Check if packet is too short from time length side
+        if (time_length < 2)
+        {
+            // It will cause statistics to report about bad RTP packet
+            // I have to replay last packet payload here to avoid report about lost packet
+            mBuffer.add(p, std::chrono::milliseconds(time_length), samplerate);
+            return false;
+        }
 
-        if (codec)
-            samplerate = codec->samplerate();
+        // Queue packet to buffer
+        auto packet = mBuffer.add(p, std::chrono::milliseconds(time_length), samplerate).get();
+        return packet;
     }
-
-    // Process jitter
-    mJitterStats.process(p.get(), samplerate);
-    mStat.mJitter = static_cast<float>(mJitterStats.get());
-
-    if (!codec)
-        return false; // There is no sense to add this packet into jitter buffer - we can't decode this
-
-    // Check if packet is CNG
-    if (payloadLength >= 1 && payloadLength <= 6 && (ptype == 0 || ptype == 8))
-        time_length = mLastPacketTimeLength ? mLastPacketTimeLength : 20;
-    else
-    // Check if packet is too short from time length side
-    if (time_length < 2)
-    {
-        // It will cause statistics to report about bad RTP packet
-        // I have to replay last packet payload here to avoid report about lost packet
-        mBuffer.add(p, std::chrono::milliseconds(time_length), samplerate);
-        return false;
-    }
-
-    // Queue packet to buffer
-    auto packet = mBuffer.add(p, std::chrono::milliseconds(time_length), samplerate).get();
-
-    return packet;
+    return {};
 }
 
 void AudioReceiver::processDecoded(Audio::DataWindow& output, DecodeOptions options)
@@ -553,17 +566,21 @@ void AudioReceiver::produceCNG(std::chrono::milliseconds length, Audio::DataWind
     }
 }
 
-AudioReceiver::DecodeResult AudioReceiver::decodeGap(Audio::DataWindow& output, DecodeOptions options)
+AudioReceiver::DecodeResult AudioReceiver::decodeGapTo(Audio::DataWindow& output, DecodeOptions options)
 {
     ICELogDebug(<< "Gap detected.");
 
     mDecodedLength = mResampledLength = 0;
     if (mCngPacket && mCodec)
     {
-        // Synthesize comfort noise. It will be done on AUDIO_SAMPLERATE rate directly to mResampledFrame buffer.
-        // Do not forget to send this noise to analysis
-        mDecodedLength = mCngDecoder.produce(mCodec->samplerate(), mLastPacketTimeLength,
-                                             reinterpret_cast<short*>(mDecodedFrame), false);
+        if (mCngPacket->rtp()->GetPayloadType() == 13)
+        {
+            // Synthesize comfort noise. It will be done on AUDIO_SAMPLERATE rate directly to mResampledFrame buffer.
+            // Do not forget to send this noise to analysis
+            mDecodedLength = mCngDecoder.produce(mCodec->samplerate(), mLastPacketTimeLength, reinterpret_cast<short*>(mDecodedFrame), false);
+        }
+        else
+            decodePacketTo(output, options, mCngPacket);
     }
     else
     if (mCodec && mFrameCount && !mCodecSettings.mSkipDecode)
@@ -573,7 +590,7 @@ AudioReceiver::DecodeResult AudioReceiver::decodeGap(Audio::DataWindow& output,
             mDecodedLength = 0;
         else
         {
-            mDecodedLength = mCodec->plc(mFrameCount, mDecodedFrame, sizeof mDecodedFrame);
+            mDecodedLength = mCodec->plc(mFrameCount, {(uint8_t*)mDecodedFrame, sizeof mDecodedFrame});
             if (!mDecodedLength)
             {
                 // PLC is not support or failed
@@ -588,158 +605,258 @@ AudioReceiver::DecodeResult AudioReceiver::decodeGap(Audio::DataWindow& output,
     if (mDecodedLength)
     {
         processDecoded(output, options);
-        return DecodeResult_Ok;
+        return {.mStatus = DecodeResult::Status::Ok, .mSamplerate = mCodec->samplerate(), .mChannels = mCodec->channels()};
     }
     else
-        return DecodeResult_Skip;
+        return {.mStatus = DecodeResult::Status::Skip};
 }
 
-AudioReceiver::DecodeResult AudioReceiver::decodePacket(const RtpBuffer::ResultList& rl, Audio::DataWindow& output, DecodeOptions options, int* rate)
+AudioReceiver::DecodeResult AudioReceiver::decodePacketTo(Audio::DataWindow& output, DecodeOptions options, const std::shared_ptr<RtpBuffer::Packet>& packet)
 {
-    DecodeResult result = DecodeResult_Skip;
+    if (!packet || !packet->rtp())
+        return {DecodeResult::Status::Skip};
+
+    DecodeResult result = {.mStatus = DecodeResult::Status::Skip};
+    auto& rtp = *packet->rtp(); // Syntax sugar
 
     mFailedCount = 0;
-    for (const std::shared_ptr<RtpBuffer::Packet>& p: rl)
+    // Check if we need to emit silence or CNG - previously CNG packet was detected. Emit CNG audio here if needed.
+    if (mLastPacketTimestamp && mLastPacketTimeLength && mCodec)
     {
-        assert(p);
-        // Check if we need to emit silence or CNG - previously CNG packet was detected. Emit CNG audio here if needed.
-        if (mLastPacketTimestamp && mLastPacketTimeLength && mCodec)
+        int units = rtp.GetTimestamp() - *mLastPacketTimestamp;
+        int milliseconds = units / (mCodec->samplerate() / 1000);
+        if (milliseconds > mLastPacketTimeLength)
         {
-            int units = p->rtp()->GetTimestamp() - *mLastPacketTimestamp;
-            int milliseconds = units / (mCodec->samplerate() / 1000);
-            if (milliseconds > mLastPacketTimeLength)
-            {
-                auto silenceLength = std::chrono::milliseconds(milliseconds - mLastPacketTimeLength);
+            auto silenceLength = std::chrono::milliseconds(milliseconds - mLastPacketTimeLength);
 
-                if (mCngPacket && options.mFillGapByCNG)
-                    produceCNG(silenceLength, output, options);
-                else
-                    produceSilence(silenceLength, output, options);
-            }
+            if (mCngPacket && options.mFillGapByCNG)
+                produceCNG(silenceLength, output, options);
+            else
+                produceSilence(silenceLength, output, options);
         }
+    }
 
-        mLastPacketTimestamp = p->rtp()->GetTimestamp();
+    mLastPacketTimestamp = rtp.GetTimestamp();
 
-        // Find codec by payload type
-        int ptype = p->rtp()->GetPayloadType();
+    // Find codec by payload type
+    int ptype = rtp.GetPayloadType();
 
-        // Look into mCodecMap if exists
-        auto codecIter = mCodecMap.find(ptype);
-        if (codecIter == mCodecMap.end())
-            return  {};
+    // Look into mCodecMap if exists
+    auto codecIter = mCodecMap.find(ptype);
+    if (codecIter == mCodecMap.end())
+        return  {};
 
+    if (!codecIter->second)
+        codecIter->second = mCodecList.createCodecByPayloadType(ptype);
 
-        if (!codecIter->second)
-            codecIter->second = mCodecList.createCodecByPayloadType(ptype);
+    mCodec = codecIter->second;
+    if (mCodec)
+    {
+        result.mChannels = mCodec->channels();
+        result.mSamplerate = mCodec->samplerate();
 
-        mCodec = codecIter->second;
-        if (mCodec)
+        // Check if it is CNG packet
+        if (((ptype == 0 || ptype == 8) && rtp.GetPayloadLength() >= 1 && rtp.GetPayloadLength() <= 6) || rtp.GetPayloadType() == 13)
         {
-            if (rate)
-                *rate = mCodec->samplerate();
-
-            // Check if it is CNG packet
-            if ((ptype == 0 || ptype == 8) && p->rtp()->GetPayloadLength() >= 1 && p->rtp()->GetPayloadLength() <= 6)
+            if (options.mSkipDecode)
+                mDecodedLength = 0;
+            else
             {
-                if (options.mSkipDecode)
-                    mDecodedLength = 0;
-                else
-                {
-                    mCngPacket = p->rtp();
-                    mCngDecoder.decode3389(p->rtp()->GetPayloadData(), p->rtp()->GetPayloadLength());
+                mCngPacket = packet;
+                mCngDecoder.decode3389(rtp.GetPayloadData(), rtp.GetPayloadLength());
 
-                    // Emit CNG mLastPacketLength milliseconds
-                    mDecodedLength = mCngDecoder.produce(mCodec->samplerate(), mLastPacketTimeLength,
-                                                         (short*)mDecodedFrame, true);
-                    if (mDecodedLength)
-                        processDecoded(output, options);
+                // Emit CNG mLastPacketLength milliseconds
+                mDecodedLength = mCngDecoder.produce(mCodec->samplerate(), mLastPacketTimeLength, (short*)mDecodedFrame, true);
+                if (mDecodedLength)
+                    processDecoded(output, options);
+            }
+            result.mStatus = DecodeResult::Status::Ok;
+        }
+        else
+        {
+            // Reset CNG packet as we get regular RTP packet
+            mCngPacket.reset();
+
+            // Handle here regular RTP packets
+            // Check if payload length is ok
+            size_t payload_length = rtp.GetPayloadLength();
+            size_t rtp_frame_length = mCodec->rtpLength();
+
+            int tail = rtp_frame_length ? payload_length % rtp_frame_length : 0;
+
+            if (!tail)
+            {
+                // Find number of frames
+                mFrameCount = mCodec->rtpLength() ? rtp.GetPayloadLength() / mCodec->rtpLength() : 1;
+                int frameLength = mCodec->rtpLength() ? mCodec->rtpLength() : (int)rtp.GetPayloadLength();
+
+                // Save last packet time length
+                mLastPacketTimeLength = mFrameCount * mCodec->frameTime();
+
+                // Decode
+                for (int i=0; i<mFrameCount && !mCodecSettings.mSkipDecode; i++)
+                {
+                    if (options.mSkipDecode)
+                        mDecodedLength = 0;
+                    else
+                    {
+                        // Decode frame by frame
+                        auto codecInput = std::span{rtp.GetPayloadData() + i * mCodec->rtpLength(), (size_t)frameLength};
+                        auto codecOutput = std::span{(uint8_t*)mDecodedFrame, sizeof mDecodedFrame};
+                        auto r = mCodec->decode(codecInput, codecOutput);
+                        mDecodedLength = r.mDecoded;
+                        if (mDecodedLength > 0)
+                            processDecoded(output, options);
+
+                        // What is important - here we may have packet marked as CNG
+                        if (r.mIsCng)
+                            mCngPacket = packet;
+                    }
                 }
-                result = DecodeResult_Ok;
+                result.mStatus = mFrameCount > 0 ? DecodeResult::Status::Ok : DecodeResult::Status::Skip;
+
+                // Check for bitrate counter
+                updateAmrCodecStats(mCodec.get());
             }
             else
             {
-                // Reset CNG packet as we get regular RTP packet
-                mCngPacket.reset();
-
-                // Handle here regular RTP packets
-                // Check if payload length is ok
-                size_t payload_length = p->rtp()->GetPayloadLength();
-                size_t rtp_frame_length = mCodec->rtpLength();
-
-                int tail = rtp_frame_length ? payload_length % rtp_frame_length : 0;
-
-                if (!tail)
-                {
-                    // Find number of frames
-                    mFrameCount = mCodec->rtpLength() ? p->rtp()->GetPayloadLength() / mCodec->rtpLength() : 1;
-                    int frameLength = mCodec->rtpLength() ? mCodec->rtpLength() : (int)p->rtp()->GetPayloadLength();
-
-                    // Save last packet time length
-                    mLastPacketTimeLength = mFrameCount * mCodec->frameTime();
-
-                    // Decode
-                    for (int i=0; i<mFrameCount && !mCodecSettings.mSkipDecode; i++)
-                    {
-                        if (options.mSkipDecode)
-                            mDecodedLength = 0;
-                        else
-                        {
-                            // Decode frame by frame
-                            mDecodedLength = mCodec->decode(p->rtp()->GetPayloadData() + i * mCodec->rtpLength(),
-                                                            frameLength, mDecodedFrame, sizeof mDecodedFrame);
-                            if (mDecodedLength > 0)
-                                processDecoded(output, options);
-                        }
-                    }
-                    result = mFrameCount > 0 ? DecodeResult_Ok : DecodeResult_Skip;
-
-                    // Check for bitrate counter
-                    updateAmrCodecStats(mCodec.get());
-                }
-                else
-                {
-                    // RTP packet with tail - it should not happen
-                    result = DecodeResult_BadPacket;
-                }
+                // RTP packet with tail - it should not happen
+                result.mStatus = DecodeResult::Status::BadPacket;
             }
         }
     }
     return result;
 }
 
-AudioReceiver::DecodeResult AudioReceiver::decodeNone(Audio::DataWindow& output, DecodeOptions options)
+AudioReceiver::DecodeResult AudioReceiver::decodeEmptyTo(Audio::DataWindow& output, DecodeOptions options)
 {
-    // ICELogDebug(<< "No packet available in jitter buffer");
-    mFailedCount++;
-    return DecodeResult_Skip;
-}
+    // There are two cases
+    // First is we have no ready time estimated how much audio should be emitted i.e. audio is decoded right after the next packet arrives.
+    // In this case we just skip the analysis - we should not be called in this situation
+    if (options.mElapsed == 0ms || !mCodec)
+        return {.mStatus = DecodeResult::Status::Skip};
 
-AudioReceiver::DecodeResult AudioReceiver::getAudio(Audio::DataWindow& output, DecodeOptions options, int* rate)
-{
-    DecodeResult result = DecodeResult_Skip;
-
-    // Get next packet from buffer
-    RtpBuffer::ResultList rl;
-    RtpBuffer::FetchResult fr = mBuffer.fetch(rl);
-    switch (fr)
+    // No packet available at all (and no previous CNG packet) - so return the silence
+    if (options.mElapsed != 0ms && mCodec)
     {
-    case RtpBuffer::FetchResult::Gap:           result = decodeGap(output, options);                break;
-    case RtpBuffer::FetchResult::NoPacket:      result = decodeNone(output, options);               break;
-    case RtpBuffer::FetchResult::RegularPacket: result = decodePacket(rl, output, options, rate);   break;
-    default:
-        assert(0);
+        Audio::Format fmt = options.mResampleToMainRate ? Audio::Format(AUDIO_SAMPLERATE, 1) : mCodec->getAudioFormat();
+        if (mCngPacket)
+        {
+            // Try to decode it - replay previous audio decoded or use CNG decoder (if payload type is 13)
+            if (mCngPacket->rtp()->GetPayloadType() == 13)
+            {
+                // Using latest CNG packet to produce comfort noise
+                auto produced = mCngDecoder.produce(fmt.rate(), options.mElapsed.count(), (short*)(output.data() + output.filled()), false);
+                output.setFilled(output.filled() + produced);
+                return {.mStatus = DecodeResult::Status::Ok, .mSamplerate = fmt.rate(), .mChannels = fmt.channels()};
+            }
+            else
+            {
+                // Here we have another packet marked as CNG - for another decoder
+                // Just decode it +1 time
+                return decodePacketTo(output, options, mCngPacket);
+            }
+        }
+        else
+        {
+            // Emit silence if codec information is available - it is to properly handle the gaps
+            auto avail = output.getTimeLength(fmt.rate(), fmt.channels());
+            if (options.mElapsed > avail)
+                output.addZero(fmt.sizeFromTime(options.mElapsed - avail));
+        }
     }
 
-    if (result == DecodeResult_Ok)
+    mFailedCount++;
+    return {.mStatus = DecodeResult::Status::Skip};
+}
+
+AudioReceiver::DecodeResult AudioReceiver::getAudioTo(Audio::DataWindow& output, DecodeOptions options)
+{
+    DecodeResult result = {.mStatus = DecodeResult::Status::Skip};
+
+    // Process RFC2833 here; it doesn't result in any audio - only callbacks and statistics
+    auto fr = mDtmfBuffer.fetch();
+    if (fr.mPacket && fr.mStatus == RtpBuffer::FetchResult::Status::RegularPacket)
+        mDtmfReceiver.add(fr.mPacket->rtp());
+
+
+    auto produced = 0ms;
+    if (mAvailable.filled() && mCodec && options.mElapsed != 0ms)
+    {
+        Audio::Format fmt = options.mResampleToMainRate ? Audio::Format(AUDIO_SAMPLERATE, 1) : mCodec->getAudioFormat();
+        auto initiallyAvailable = mCodec ? mAvailable.getTimeLength(fmt.rate(), fmt.channels()) : 0ms;
+        if (initiallyAvailable != 0ms)
+        {
+            std::chrono::milliseconds resultTime = std::min(initiallyAvailable, options.mElapsed);
+            auto resultLen = fmt.sizeFromTime(resultTime);
+            mAvailable.moveTo(output, resultLen);
+            produced += resultTime;
+
+            // Maybe request is satisfied ?
+            if (produced >= options.mElapsed)
+                return {.mStatus = DecodeResult::Status::Ok, .mSamplerate = fmt.rate(), .mChannels = fmt.channels()};
+        }
+    }
+
+    std::chrono::milliseconds decoded = 0ms;
+    do
+    {
+        // Get next packet from buffer
+        RtpBuffer::ResultList rl;
+        RtpBuffer::FetchResult fr = mBuffer.fetch();
+        // ICELogDebug(<< fr.toString() << " " << mBuffer.findTimelength());
+
+        switch (fr.mStatus)
+        {
+        case RtpBuffer::FetchResult::Status::Gap:           result = decodeGapTo(mAvailable, options);                  break;
+        case RtpBuffer::FetchResult::Status::NoPacket:      result = decodeEmptyTo(mAvailable, options);                break;
+        case RtpBuffer::FetchResult::Status::RegularPacket: result = decodePacketTo(mAvailable, options, fr.mPacket);   break;
+        default:
+            assert(0);
+        }
+
+        // Was there decoding at all ?
+        if (!mCodec)
+            break; // No sense to continue - we have no information at all
+
+        Audio::Format fmt = options.mResampleToMainRate ? Audio::Format(AUDIO_SAMPLERATE, 1) : mCodec->getAudioFormat();
+        result.mSamplerate = fmt.rate();
+        result.mChannels = fmt.channels();
+
+        // Have we anything interesting in the buffer ?
+        auto bufferAvailable = mAvailable.getTimeLength(fmt.rate(), fmt.channels());
+        if (bufferAvailable == 0ms)
+            break; // No sense to continue - decoding / CNG / PLC stopped totally
+
+        // How much data should be moved to result buffer ?
+        if (options.mElapsed != 0ms)
+        {
+            std::chrono::milliseconds resultTime = std::min(bufferAvailable, options.mElapsed - produced);
+            auto resultLen = fmt.sizeFromTime(resultTime);
+            mAvailable.moveTo(output, resultLen);
+            produced += resultTime;
+        }
+        else
+            mAvailable.moveTo(output, mAvailable.filled());
+
+        decoded += bufferAvailable;
+    }
+    while (produced < options.mElapsed);
+
+    if (produced != 0ms)
+        result.mStatus = DecodeResult::Status::Ok;
+
+    // Time statistics
+    if (result.mStatus == DecodeResult::Status::Ok)
     {
         // Decode statistics
-        if (!mLastDecodeTimestamp)
-            mLastDecodeTimestamp = std::chrono::steady_clock::now();
+        if (!mDecodeTimestamp)
+            mDecodeTimestamp = std::chrono::steady_clock::now();
         else
         {
             auto t = std::chrono::steady_clock::now();
-            mStat.mDecodingInterval.process(std::chrono::duration_cast<std::chrono::milliseconds>(t - *mLastDecodeTimestamp).count());
-            mLastDecodeTimestamp = t;
+            mStat.mDecodingInterval.process(std::chrono::duration_cast<std::chrono::milliseconds>(t - *mDecodeTimestamp).count());
+            mDecodeTimestamp = t;
         }
     }
     return result;
@@ -795,21 +912,26 @@ void AudioReceiver::updateAmrCodecStats(Codec* c)
     AmrWbCodec* wb = dynamic_cast<AmrWbCodec*>(c);
 
     if (nb != nullptr)
+    {
         mStat.mBitrateSwitchCounter = nb->getSwitchCounter();
+        mStat.mCng = nb->getCngCounter();
+    }
     else
     if (wb != nullptr)
+    {
         mStat.mBitrateSwitchCounter = wb->getSwitchCounter();
+        mStat.mCng = wb->getCngCounter();
+    }
 #endif
 }
 
 int AudioReceiver::getSize() const
 {
     int result = 0;
-    result += sizeof(*this) + mResampler8.getSize() + mResampler16.getSize() + mResampler32.getSize()
-            + mResampler48.getSize();
+    result += sizeof(*this) + mResampler8.getSize() + mResampler16.getSize() + mResampler32.getSize() + mResampler48.getSize();
 
     if (mCodec)
-        result += mCodec->getSize();
+        ; // ToDo: need the way to calculate size of codec instances
 
     return result;
 }
@@ -860,5 +982,24 @@ DtmfReceiver::DtmfReceiver(Statistics& stat)
 DtmfReceiver::~DtmfReceiver()
 {}
 
-void DtmfReceiver::add(std::shared_ptr<RTPPacket> /*p*/)
-{}
+void DtmfReceiver::add(const std::shared_ptr<RTPPacket>& p)
+{
+    auto ev = DtmfBuilder::parseRfc2833({p->GetPayloadData(), p->GetPayloadLength()});
+    if (ev.mTone != mEvent || ev.mEnd != mEventEnded)
+    {
+        if (!(mEvent == ev.mTone && !mEventEnded && ev.mEnd))
+        {
+            // New tone is here
+            if (mCallback)
+                mCallback(ev.mTone);
+
+            // Queue statistics item
+            mStat.mDtmf2833Timeline.emplace_back(Dtmf2833Event{.mTone = ev.mTone,
+                                                               .mTimestamp = RtpHelper::toMicroseconds(p->GetReceiveTime())});
+
+            // Store to avoid triggering on the packet
+            mEvent = ev.mTone;
+            mEventEnded = ev.mEnd;
+        }
+    }
+}

src/engine/media/MT_AudioReceiver.h

@@ -28,13 +28,6 @@ using jrtplib::RTPPacket;
 class RtpBuffer
 {
 public:
-    enum class FetchResult
-    {
-        RegularPacket,
-        Gap,
-        NoPacket
-    };
-
     // Owns rtp packet data
     class Packet
     {
@@ -59,6 +52,29 @@ public:
         std::chrono::microseconds mTimestamp = 0us;
     };
 
+    struct FetchResult
+    {
+        enum class Status
+        {
+            RegularPacket,
+            Gap,
+            NoPacket
+        };
+
+        Status mStatus = Status::NoPacket;
+        std::shared_ptr<Packet> mPacket;
+
+        std::string toString() const
+        {
+            switch (mStatus)
+            {
+                case Status::RegularPacket: return "packet";
+                case Status::Gap:           return "gap";
+                case Status::NoPacket:      return "empty";
+            }
+        }
+    };
+
     RtpBuffer(Statistics& stat);
     ~RtpBuffer();
 
@@ -81,12 +97,12 @@ public:
     int getCount() const;
 
     // Returns false if packet was not add - maybe too old or too new or duplicate
-    std::shared_ptr<Packet> add(std::shared_ptr<RTPPacket> packet, std::chrono::milliseconds timelength, int rate);
+    std::shared_ptr<Packet> add(const std::shared_ptr<RTPPacket>& packet, std::chrono::milliseconds timelength, int rate);
 
     typedef std::vector<std::shared_ptr<Packet>> ResultList;
     typedef std::shared_ptr<ResultList> PResultList;
 
-    FetchResult fetch(ResultList& rl);
+    FetchResult fetch();
     
 protected:
     unsigned    mSsrc = 0;
@@ -104,6 +120,7 @@ protected:
     jrtplib::RTPSourceStats mRtpStats;
     std::shared_ptr<Packet> mFetchedPacket;
     std::optional<uint32_t> mLastSeqno;
+    std::optional<jrtplib::RTPTime> mLastReceiveTime;
 
     // To calculate average interval between packet add. It is close to jitter but more useful in debugging.
     float mLastAddTime = 0.0f;
@@ -119,6 +136,23 @@ protected:
     Statistics& mStat;
 };
 
+class DtmfReceiver: public Receiver
+{
+private:
+    char mEvent = 0;
+    bool mEventEnded = false;
+    std::chrono::milliseconds mEventStart = 0ms;
+    std::function<void(char)> mCallback;
+
+public:
+    DtmfReceiver(Statistics& stat);
+    ~DtmfReceiver();
+
+    void add(const std::shared_ptr<RTPPacket>& p);
+    void setCallback(std::function<void(char tone)> callback);
+};
+
+
 class AudioReceiver: public Receiver
 {
 public:
@@ -133,34 +167,33 @@ public:
     // Lifetime of pointer to codec is limited by lifetime of AudioReceiver (it is container).
     bool add(const std::shared_ptr<jrtplib::RTPPacket>& p, Codec** codec = nullptr);
 
-    // Returns false when there is no rtp data from jitter
-    /*enum DecodeOptions
-    {
-        DecodeOptions_ResampleToMainRate = 0,
-        DecodeOptions_DontResample = 1,
-        DecodeOptions_FillCngGap = 2,
-        DecodeOptions_SkipDecode = 4
-    };*/
-
     struct DecodeOptions
     {
-        bool mResampleToMainRate = true;
-        bool mFillGapByCNG = false;
-        bool mSkipDecode = false;
+        bool mResampleToMainRate = true;                // Resample all decoded audio to AUDIO_SAMPLERATE
+        bool mFillGapByCNG = false;                     // Use CNG information if available
+        bool mSkipDecode = false;                       // Don't do decode, just dry run - fetch packets, remove them from the jitter buffer
+        std::chrono::milliseconds mElapsed = 0ms;       // How much milliseconds should be decoded; zero value means "decode just next packet from the buffer"
     };
 
-    enum DecodeResult
+    struct DecodeResult
     {
-        DecodeResult_Ok,        // Decoded ok
-        DecodeResult_Skip,      // Just no data - emit silence instead
-        DecodeResult_BadPacket  // Error happened during the decode
+        enum class Status
+        {
+            Ok,        // Decoded ok
+            Skip,      // Just no data - emit silence instead
+            BadPacket  // Error happened during the decode
+        };
+
+        Status          mStatus = Status::Ok;
+        int             mSamplerate = 0;
+        int             mChannels = 0;
     };
 
-    DecodeResult getAudio(Audio::DataWindow& output, DecodeOptions options = {.mResampleToMainRate = true, .mFillGapByCNG = false, .mSkipDecode = false}, int* rate = nullptr);
+    DecodeResult getAudioTo(Audio::DataWindow& output, DecodeOptions options);
 
     // Looks for codec by payload type
     Codec* findCodec(int payloadType);
-    RtpBuffer& getRtpBuffer() { return mBuffer; }
+    RtpBuffer&  getRtpBuffer() { return mBuffer; }
 
     // Returns size of AudioReceiver's instance in bytes (including size of all data + codecs + etc.)
     int getSize() const;
@@ -173,17 +206,23 @@ public:
 
 protected:
     RtpBuffer                           mBuffer;                // Jitter buffer itself
+    RtpBuffer                           mDtmfBuffer;            // These two (mDtmfBuffer / mDtmfReceiver) are for our analyzer stack only; in normal softphone logic DTMF packets goes via SingleAudioStream::mDtmfReceiver
+    DtmfReceiver                        mDtmfReceiver;
+
     CodecMap                            mCodecMap;
     PCodec                              mCodec;
     int                                 mFrameCount = 0;
     CodecList::Settings                 mCodecSettings;
     CodecList                           mCodecList;
     JitterStatistics                    mJitterStats;
-    std::shared_ptr<jrtplib::RTPPacket> mCngPacket;
+    std::shared_ptr<RtpBuffer::Packet>  mCngPacket;
     CngDecoder                          mCngDecoder;
     size_t                              mDTXSamplesToEmit = 0;   // How much silence (or CNG) should be emited before next RTP packet gets into the action
 
-    // Buffer to hold decoded data
+    // Already decoded data that can be retrieved without actual decoding - it may happen because of getAudioTo() may be limited by time interval
+    Audio::DataWindow mAvailable;
+
+    // Temporary buffer to hold decoded data (it is better than allocate data on stack)
     int16_t mDecodedFrame[MT_MAX_DECODEBUFFER];
     size_t mDecodedLength = 0;
 
@@ -200,11 +239,14 @@ protected:
     std::optional<uint32_t> mLastPacketTimestamp;
 
     int mFailedCount = 0;
-    Audio::Resampler  mResampler8, mResampler16, mResampler32, mResampler48;
+    Audio::Resampler  mResampler8,
+                      mResampler16,
+                      mResampler32,
+                      mResampler48;
 
     Audio::PWavFileWriter mDecodedDump;
 
-    std::optional<std::chrono::steady_clock::time_point> mLastDecodeTimestamp; // Time last call happened to codec->decode()
+    std::optional<std::chrono::steady_clock::time_point> mDecodeTimestamp; // Time last call happened to codec->decode()
 
     float mIntervalSum = 0.0f;
     int mIntervalCount = 0;
@@ -220,19 +262,11 @@ protected:
     // Calculate bitrate switch statistics for AMR codecs
     void updateAmrCodecStats(Codec* c);
 
-    DecodeResult decodeGap(Audio::DataWindow& output, DecodeOptions options);
-    DecodeResult decodePacket(const RtpBuffer::ResultList& rl, Audio::DataWindow& output, DecodeOptions options, int* rate = nullptr);
-    DecodeResult decodeNone(Audio::DataWindow& output, DecodeOptions options);
+    DecodeResult decodeGapTo(Audio::DataWindow& output, DecodeOptions options);
+    DecodeResult decodePacketTo(Audio::DataWindow& output, DecodeOptions options, const std::shared_ptr<RtpBuffer::Packet>& p);
+    DecodeResult decodeEmptyTo(Audio::DataWindow& output, DecodeOptions options);
 };
 
-class DtmfReceiver: public Receiver
-{
-public:
-    DtmfReceiver(Statistics& stat);
-    ~DtmfReceiver();
-
-    void add(std::shared_ptr<RTPPacket> p);
-};
 }
 
 #endif

src/engine/media/MT_AudioStream.cpp

@@ -210,18 +210,17 @@ void AudioStream::addData(const void* buffer, int bytes)
         if (mSendingDump)
             mSendingDump->write((const char*)mCapturedAudio.data() + codec->pcmLength() * i, codec->pcmLength());
 
-        int produced;
-        produced = codec->encode((const char*)mCapturedAudio.data() + codec->pcmLength()*i,
-                                 codec->pcmLength(), mFrameBuffer, MT_MAXAUDIOFRAME);
+        auto r = codec->encode({(const uint8_t*)mCapturedAudio.data() + codec->pcmLength()*i, (size_t)codec->pcmLength()},
+                                 {(uint8_t*)mFrameBuffer, MT_MAXAUDIOFRAME});
         
         // Counter of processed input bytes of raw pcm data from microphone
         processed += codec->pcmLength();
         encodedTime += codec->frameTime();
         mEncodedTime += codec->frameTime();
 
-        if (produced)
+        if (r.mEncoded)
         {
-            mEncodedAudio.appendBuffer(mFrameBuffer, produced);
+            mEncodedAudio.appendBuffer(mFrameBuffer, r.mEncoded);
             if (packetTime <= encodedTime)
             {
                 // Time to send packet

src/engine/media/MT_Codec.cpp

@@ -1,4 +1,4 @@
-/* Copyright(C) 2007-2014 VoIP objects (voipobjects.com)
+/* Copyright(C) 2007-2026 VoIP objects (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/. */
@@ -9,31 +9,31 @@ using namespace MT;
 
 int Codec::Factory::channels()
 {
-  return 1;
+    return 1;
 }
 void Codec::Factory::create(CodecMap& codecs)
 {
-  codecs[payloadType()] = std::shared_ptr<Codec>(create());
+    codecs[payloadType()] = std::shared_ptr<Codec>(create());
 }
 
 void Codec::Factory::updateSdp(resip::SdpContents::Session::Medium::CodecContainer& codecs, SdpDirection direction)
 {
-  codecs.push_back(resipCodec());
+    codecs.push_back(resipCodec());
 }
 
 resip::Codec Codec::Factory::resipCodec()
 {
-  resip::Codec c(this->name(), this->payloadType(), this->samplerate());
-  return c;
+    resip::Codec c(this->name(), this->payloadType(), this->samplerate());
+    return c;
 }
 
 int Codec::Factory::processSdp(const resip::SdpContents::Session::Medium::CodecContainer& codecs, SdpDirection direction)
 {
-  for (resip::SdpContents::Session::Medium::CodecContainer::const_iterator codecIter = codecs.begin(); codecIter != codecs.end(); ++codecIter)
-  {
-    if (resipCodec() == *codecIter)
-      return codecIter->payloadType();
-  }
-  return -1;
+    for (resip::SdpContents::Session::Medium::CodecContainer::const_iterator codecIter = codecs.begin(); codecIter != codecs.end(); ++codecIter)
+    {
+        if (resipCodec() == *codecIter)
+            return codecIter->payloadType();
+    }
+    return -1;
 }
 

src/engine/media/MT_Codec.h

@@ -1,4 +1,4 @@
-/* Copyright(C) 2007-2016 VoIP objects (voipobjects.com)
+/* Copyright(C) 2007-2026 VoIP objects (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/. */
@@ -6,11 +6,12 @@
 #ifndef __MT_CODEC_H
 #define __MT_CODEC_H
 
+#include <map>
+#include <span>
+
 #include "resiprocate/resip/stack/SdpContents.hxx"
 #include "../helper/HL_Types.h"
-#include <map>
-#include "../helper/HL_Pointer.h"
-
+#include "../audio/Audio_Interface.h"
 
 namespace MT
 {
@@ -18,12 +19,12 @@ class Codec;
 typedef std::shared_ptr<Codec> PCodec;
 
 class CodecMap: public std::map<int, PCodec>
-{
-};
+{};
 
 class Codec
 {
 public:
+
     class Factory
     {
     public:
@@ -42,34 +43,51 @@ public:
         resip::Codec resipCodec();
     };
     virtual ~Codec() {}
-    virtual const char* name() = 0;
-    virtual int samplerate() = 0;
-    virtual float timestampUnit() { return float(1.0 / samplerate()); }
 
-    // Size of decoded audio frame in bytes
-    virtual int pcmLength() = 0;
+    struct Info
+    {
+        std::string mName;
+        int mSamplerate = 0;    // Hz
+        int mChannels = 0;
+        int mPcmLength = 0;     // In bytes
+        int mFrameTime = 0;     // In milliseconds
+        int mRtpLength = 0;     // In bytes
+        float mTimestampUnit = 0.0f;
+    };
+    // Returns information about this codec instance
+    virtual Info info() = 0;
 
-    // Time length of single audio frame
-    virtual int frameTime() = 0;
-
-    // Size of RTP frame in bytes. Can be zero for variable sized codecs.
-    virtual int rtpLength() = 0;
-
-    // Number of audio channels
-    virtual int channels() { return 1; }
+    // Helper functions to return information - they are based on info() method
+    int pcmLength()         { return info().mPcmLength;  }
+    int rtpLength()         { return info().mRtpLength;  }
+    int channels()          { return info().mChannels;   }
+    int samplerate()        { return info().mSamplerate; }
+    int frameTime()         { return info().mFrameTime;  }
+    std::string name()      { return info().mName;       }
+    float timestampUnit()   { return info().mTimestampUnit == 0.0f ? 1.0f / info().mSamplerate : info().mTimestampUnit; }
 
+    Audio::Format getAudioFormat() {
+        return Audio::Format(this->info().mSamplerate, this->info().mChannels);
+    }
 
     // Returns size of encoded data (RTP) in bytes
-    virtual int encode(const void* input, int inputBytes, void* output, int outputCapacity) = 0;
+    struct EncodeResult
+    {
+        size_t mEncoded = 0; // Number of encoded bytes
+    };
+    virtual EncodeResult encode(std::span<const uint8_t> input, std::span<uint8_t> output) = 0;
 
     // Returns size of decoded data (PCM signed short) in bytes
-    virtual int decode(const void* input, int inputBytes, void* output, int outputCapacity) = 0;
+    struct DecodeResult
+    {
+        size_t  mDecoded = 0;    // Number of decoded bytes
+        bool    mIsCng = false;    // Should this packet to be used as CNG ? (used for AMR codecs)
+    };
+    virtual DecodeResult decode(std::span<const uint8_t> input, std::span<uint8_t> output) = 0;
 
     // Returns size of produced data (PCM signed short) in bytes
-    virtual int plc(int lostFrames, void* output, int outputCapacity) = 0;
+    virtual size_t plc(int lostFrames, std::span<uint8_t> output) = 0;
 
-    // Returns size of codec in memory
-    virtual int getSize() const { return 0; };
 };
 }
 #endif

src/engine/media/MT_CodecList.cpp

@@ -65,6 +65,8 @@ bool CodecList::Settings::contains(int ptype) const
 
     if (mGsmEfrPayloadType == ptype || mGsmFrPayloadType == ptype || mGsmHrPayloadType == ptype)
         return true;
+    if (mTelephoneEvent == ptype)
+        return true;
 
     return false;
 }
@@ -122,6 +124,9 @@ std::string CodecList::Settings::toString() const
     if (mGsmEfrPayloadType != -1)
         oss << "GSM EFR ptype: " << mGsmEfrPayloadType << " ";
 
+    if (mTelephoneEvent != -1)
+        oss << "RFC2833 DTMF ptype: " << mTelephoneEvent;
+
     for (auto& spec: mEvsSpec)
     {
         oss << "EVS ptype: " << spec.mPayloadType << ", bw: " << spec.mBandwidth << ", enc: " << (spec.mEncodingType == EvsSpec::Encoding_MIME ? "mime" : "g192") << " ";
@@ -132,6 +137,7 @@ std::string CodecList::Settings::toString() const
         oss << "OPUS ptype: " << spec.mPayloadType << ", rate: " << spec.mRate << ", channels: " << spec.mChannels << std::endl;
     }
 
+
     return oss.str();
 }
 
@@ -151,6 +157,7 @@ void CodecList::Settings::clear()
     mGsmEfrPayloadType = -1;
     mGsmFrPayloadType = -1;
     mGsmHrPayloadType = -1;
+    mTelephoneEvent = -1;
 }
 
 bool CodecList::Settings::EvsSpec::isValid() const
@@ -268,15 +275,16 @@ CodecList::Settings CodecList::Settings::parseSdp(const std::list<resip::Codec>&
             }
         } else if (codec_name == "EVS") {
             r.mEvsSpec.push_back({ptype});
-        }
+        } else if (codec_name == "TELEPHONE-EVENT")
+            r.mTelephoneEvent = ptype;
     }
     return r;
 }
 
 bool CodecList::Settings::operator == (const Settings& rhs) const
 {
-    if (std::tie(mWrapIuUP, mSkipDecode, mIsac16KPayloadType, mIsac32KPayloadType, mIlbc20PayloadType, mIlbc30PayloadType, mGsmFrPayloadType, mGsmFrPayloadLength, mGsmEfrPayloadType, mGsmHrPayloadType) !=
-        std::tie(rhs.mWrapIuUP, rhs.mSkipDecode, rhs.mIsac16KPayloadType, rhs.mIsac32KPayloadType, rhs.mIlbc20PayloadType, rhs.mIlbc30PayloadType, rhs.mGsmFrPayloadType, rhs.mGsmFrPayloadLength, rhs.mGsmEfrPayloadType, rhs.mGsmHrPayloadType))
+    if (std::tie(mWrapIuUP, mSkipDecode, mIsac16KPayloadType, mIsac32KPayloadType, mIlbc20PayloadType, mIlbc30PayloadType, mGsmFrPayloadType, mGsmFrPayloadLength, mGsmEfrPayloadType, mGsmHrPayloadType, mTelephoneEvent) !=
+        std::tie(rhs.mWrapIuUP, rhs.mSkipDecode, rhs.mIsac16KPayloadType, rhs.mIsac32KPayloadType, rhs.mIlbc20PayloadType, rhs.mIlbc30PayloadType, rhs.mGsmFrPayloadType, rhs.mGsmFrPayloadLength, rhs.mGsmEfrPayloadType, rhs.mGsmHrPayloadType, rhs.mTelephoneEvent))
         return false;
 
     if (mAmrNbOctetPayloadType != rhs.mAmrNbOctetPayloadType)
@@ -306,6 +314,9 @@ bool CodecList::Settings::operator == (const Settings& rhs) const
         if (mOpusSpec[i] != rhs.mOpusSpec[i])
             return false;
 
+    if (mTelephoneEvent != rhs.mTelephoneEvent)
+        return false;
+
     return true;
 }
 

src/engine/media/MT_CodecList.h

@@ -28,6 +28,9 @@ public:
         bool mWrapIuUP              = false;
         bool mSkipDecode            = false;
 
+        // RFC2833 DTMF
+        int mTelephoneEvent = -1;
+
         // AMR payload types
         std::set<int64_t> mAmrWbPayloadType       = { };
         std::set<int64_t> mAmrNbPayloadType       = { };

src/engine/media/MT_Dtmf.cpp

@@ -16,38 +16,68 @@
 
 using namespace MT;
 
-void DtmfBuilder::buildRfc2833(int tone, int duration, int volume, bool endOfEvent, void* output)
+void DtmfBuilder::buildRfc2833(const Rfc2833Event& ev, void* output)
 {
-  assert(duration);
-  assert(output);
-  assert(tone);
+    assert(ev.mDuration != 0);
+    assert(output);
+    assert(ev.mTone != 0);
 
-  unsigned char toneValue = 0;
-  if (tone >= '0' && tone <='9')
-    toneValue = tone - '0';
-  else
-  if (tone >= 'A' && tone <='D' )
-    toneValue = tone - 'A' + 12;
-  else
-  if (tone == '*')
-    toneValue = 10;
-  else
-  if (tone == '#')
-    toneValue = 11;
+    unsigned char toneValue = 0;
+    if (ev.mTone >= '0' && ev.mTone <='9')
+        toneValue = ev.mTone - '0';
+    else
+    if (ev.mTone >= 'A' && ev.mTone <='D' )
+        toneValue = ev.mTone - 'A' + 12;
+    else
+    if (ev.mTone == '*')
+        toneValue = 10;
+    else
+    if (ev.mTone == '#')
+        toneValue = 11;
 
-  char* packet = (char*)output;
+    char* packet = (char*)output;
 
-  packet[0] = toneValue;
-  packet[1] = 1 | (volume << 2);
-  if (endOfEvent)
-    packet[1] |= 128;
-  else
-    packet[1] &= 127;
+    packet[0] = toneValue;
+    packet[1] = 1 | (ev.mVolume << 2);
+    if (ev.mEnd)
+        packet[1] |= 128;
+    else
+        packet[1] &= 127;
 
-  unsigned short durationValue = htons(duration);
-  memcpy(packet + 2, &durationValue, 2);
+    unsigned short durationValue = htons(ev.mDuration);
+    memcpy(packet + 2, &durationValue, 2);
 }
 
+
+DtmfBuilder::Rfc2833Event DtmfBuilder::parseRfc2833(std::span<uint8_t> payload)
+{
+    Rfc2833Event r;
+    if (payload.size_bytes() < 4)
+        return r;
+
+    uint8_t b0 = payload[0];
+    uint8_t b1 = payload[1];
+
+    if (b0 >=0 && b0 <= 9)
+        r.mTone = '0' + b0;
+    else
+    if (b0 >= 12 && b0 <= 17)
+        r.mTone = 'A' + b0;
+    else
+    if (b0 == 10)
+        r.mTone = '*';
+    else
+    if (b0 == 11)
+        r.mTone = '#';
+
+    r.mEnd = (b1 & 128);
+    r.mVolume = (b1 & 127) >> 2;
+    r.mDuration = ntohs(*(uint16_t*)payload.data()+2);
+
+    return r;
+}
+
+
 #pragma region Inband DTMF support
 #include <math.h>
 #ifndef TARGET_WIN
@@ -62,7 +92,7 @@ static bool sineTabInit = false;
 static double sinetab[1 << 11];
 static inline double sine(unsigned int ptr)
 { 
-  return sinetab[ptr >> (32-11)]; 
+    return sinetab[ptr >> (32-11)];
 }
 
 #define TWOPI	      (2.0 * 3.14159265358979323846)
@@ -75,13 +105,13 @@ static inline double sine(unsigned int ptr)
 static double amptab[2] = { 8191.75, 16383.5 };
 static inline int ifix(double x) 
 { 
-  return (x >= 0.0) ? (int) (x+0.5) : (int) (x-0.5); 
+    return (x >= 0.0) ? (int) (x+0.5) : (int) (x-0.5);
 }
 
 // given frequency f, return corresponding phase increment 
 static inline int phinc(double f)
 { 
-  return ifix(TWO32 * f / (double) AUDIO_SAMPLERATE);
+    return ifix(TWO32 * f / (double) AUDIO_SAMPLERATE);
 }
 
 static char dtmfSymbols[16] = {
@@ -106,31 +136,31 @@ static char dtmfSymbols[16] = {
 char PDTMFEncoder_DtmfChar(int i)
 {
 
-  if (i < 16)
-    return dtmfSymbols[i];
-  else
-    return 0;
+    if (i < 16)
+        return dtmfSymbols[i];
+    else
+        return 0;
 }
 
 // DTMF frequencies as per http://www.commlinx.com.au/DTMF_frequencies.htm
 
 static double dtmfFreqs[16][2] = {
-  { 941.0, 1336.0 },  // 0
-  { 697.0, 1209.0 },  // 1
-  { 697.0, 1336.0 },  // 2
-  { 697.0, 1477.0 },  // 3
-  { 770.0, 1209.0 },  // 4
-  { 770.0, 1336.0 },  // 5
-  { 770.0, 1477.0 },  // 6
-  { 852.0, 1209.0 },  // 7
-  { 852.0, 1336.0 },  // 8
-  { 852.0, 1477.0 },  // 9
-  { 697.0, 1633.0 },  // A
-  { 770.0, 1633.0 },  // B
-  { 852.0, 1633.0 },  // C
-  { 941.0, 1633.0 },  // D
-  { 941.0, 1209.0 },  // *
-  { 941.0, 1477.0 }   // #
+    { 941.0, 1336.0 },  // 0
+    { 697.0, 1209.0 },  // 1
+    { 697.0, 1336.0 },  // 2
+    { 697.0, 1477.0 },  // 3
+    { 770.0, 1209.0 },  // 4
+    { 770.0, 1336.0 },  // 5
+    { 770.0, 1477.0 },  // 6
+    { 852.0, 1209.0 },  // 7
+    { 852.0, 1336.0 },  // 8
+    { 852.0, 1477.0 },  // 9
+    { 697.0, 1633.0 },  // A
+    { 770.0, 1633.0 },  // B
+    { 852.0, 1633.0 },  // C
+    { 941.0, 1633.0 },  // D
+    { 941.0, 1209.0 },  // *
+    { 941.0, 1477.0 }   // #
 };
 
 
@@ -138,81 +168,81 @@ static Mutex LocalDtmfMutex;
 
 void PDTMFEncoder_MakeSineTable()
 { 
-  Lock lock(LocalDtmfMutex);
+    Lock lock(LocalDtmfMutex);
 
-  if (!sineTabInit) {
-    for (int k = 0; k < SINELEN; k++) { 
-      double th = TWOPI * (double) k / (double) SINELEN;
-      double v = sin(th);
-      sinetab[k] = v;
+    if (!sineTabInit) {
+        for (int k = 0; k < SINELEN; k++) {
+            double th = TWOPI * (double) k / (double) SINELEN;
+            double v = sin(th);
+            sinetab[k] = v;
+        }
+        sineTabInit = true;
     }
-    sineTabInit = true;
-  }
 }
 
 void PDTMFEncoder_AddTone(double f1, double f2, unsigned ms1, unsigned ms2, unsigned rate, short* result)
 {
-  int ak = 0;
+    int ak = 0;
 
-  PDTMFEncoder_MakeSineTable();
+    PDTMFEncoder_MakeSineTable();
 
-  int dataPtr = 0;
+    int dataPtr = 0;
 
-  double amp = amptab[ak];
-  int phinc1 = phinc(f1), phinc2 = phinc(f2);
-  int ns1 = ms1 * (rate/1000);
-  int ns2 = ms2 * (rate/1000);
-  unsigned int ptr1 = 0, ptr2 = 0;
-  ptr1 += phinc1 * ns1;
-  ptr2 += phinc2 * ns1;
+    double amp = amptab[ak];
+    int phinc1 = phinc(f1), phinc2 = phinc(f2);
+    int ns1 = ms1 * (rate/1000);
+    int ns2 = ms2 * (rate/1000);
+    unsigned int ptr1 = 0, ptr2 = 0;
+    ptr1 += phinc1 * ns1;
+    ptr2 += phinc2 * ns1;
 
-  for (int n = ns1; n < ns2; n++) { 
+    for (int n = ns1; n < ns2; n++) {
 
-    double val = amp * (sine(ptr1) + sine(ptr2));
-    int ival = ifix(val);
-    if (ival < -32768)
-      ival = -32768;
-    else if (val > 32767) 
-      ival = 32767;
+        double val = amp * (sine(ptr1) + sine(ptr2));
+        int ival = ifix(val);
+        if (ival < -32768)
+            ival = -32768;
+        else if (val > 32767)
+            ival = 32767;
 
-    result[dataPtr++] = ival / 2;
+        result[dataPtr++] = ival / 2;
 
-    ptr1 += phinc1; 
-    ptr2 += phinc2;
-  }
+        ptr1 += phinc1;
+        ptr2 += phinc2;
+    }
 }
 
 void PDTMFEncoder_AddTone(char _digit, unsigned startTime, unsigned finishTime, unsigned rate, short* result)
 {
-  char digit = (char)toupper(_digit); 
-  if ('0' <= digit && digit <= '9')
-    digit = digit - '0';
+    char digit = (char)toupper(_digit);
+    if ('0' <= digit && digit <= '9')
+        digit = digit - '0';
 
-  else if ('A' <= digit && digit <= 'D')
-    digit = digit + 10 - 'A';
+    else if ('A' <= digit && digit <= 'D')
+        digit = digit + 10 - 'A';
 
-  else if (digit == '*')
-    digit = 14;
+    else if (digit == '*')
+        digit = 14;
 
-  else if (digit == '#')
-    digit = 15;
+    else if (digit == '#')
+        digit = 15;
 
-  else
-    return ;
+    else
+        return ;
 
-  PDTMFEncoder_AddTone(dtmfFreqs[(int)digit][0], dtmfFreqs[(int)digit][1], startTime, finishTime, rate, result);
+    PDTMFEncoder_AddTone(dtmfFreqs[(int)digit][0], dtmfFreqs[(int)digit][1], startTime, finishTime, rate, result);
 }
 
 #pragma endregion
 
 void DtmfBuilder::buildInband(int tone, int startTime, int finishTime, int rate, short* buf)
 {
-  PDTMFEncoder_AddTone(tone, startTime, finishTime, rate, buf);
+    PDTMFEncoder_AddTone(tone, startTime, finishTime, rate, buf);
 }
 
 #pragma region DtmfContext
 DtmfContext::DtmfContext()
-:mType(Dtmf_Rfc2833)
+    :mType(Dtmf_Rfc2833)
 {
 }
 
@@ -222,112 +252,112 @@ DtmfContext::~DtmfContext()
 
 void DtmfContext::setType(Type t)
 {
-  mType = t;
+    mType = t;
 }
 
 DtmfContext::Type DtmfContext::type()
 {
-  return mType;
+    return mType;
 }
 
 void DtmfContext::startTone(int tone, int volume)
 {
-  Lock l(mGuard);
+    Lock l(mGuard);
 
-  // Stop current tone if needed
-  if (mQueue.size())
-    stopTone();
-  mQueue.push_back(Dtmf(tone, volume, 0));
+    // Stop current tone if needed
+    if (mQueue.size())
+        stopTone();
+    mQueue.push_back(Dtmf(tone, volume, 0));
 }
 
 void DtmfContext::stopTone()
 {
-  Lock l(mGuard);
+    Lock l(mGuard);
 
-  // Switch to "emit 3 terminating packets" mode
-  if (mQueue.size())
-  {
-    switch (mType)
+    // Switch to "emit 3 terminating packets" mode
+    if (mQueue.size())
     {
-    case Dtmf_Rfc2833:
-      mQueue.front().mStopped = true;
-      mQueue.erase(mQueue.begin());      
-      break;
+        switch (mType)
+        {
+        case Dtmf_Rfc2833:
+            mQueue.front().mStopped = true;
+            mQueue.erase(mQueue.begin());
+            break;
 
-    case Dtmf_Inband:
-      if (!mQueue.front().mFinishCount)
-        mQueue.front().mFinishCount = MT_DTMF_END_PACKETS;
-      break;
+        case Dtmf_Inband:
+            if (!mQueue.front().mFinishCount)
+                mQueue.front().mFinishCount = MT_DTMF_END_PACKETS;
+            break;
+        }
     }
-  }
 }
 
 void DtmfContext::queueTone(int tone, int volume, int duration)
 {
-  Lock l(mGuard);
-  mQueue.push_back(Dtmf(tone, volume, duration));
+    Lock l(mGuard);
+    mQueue.push_back(Dtmf(tone, volume, duration));
 }
 
 void DtmfContext::clearAllTones()
 {
-  Lock l(mGuard);
-  mQueue.clear();
+    Lock l(mGuard);
+    mQueue.clear();
 }
 
 bool DtmfContext::getInband(int milliseconds, int rate, ByteBuffer& output)
 {
-  Lock l(mGuard);
+    Lock l(mGuard);
 
-  if (!mQueue.size() || mType != Dtmf_Inband)
-    return false;
+    if (!mQueue.size() || mType != Dtmf_Inband)
+        return false;
 
-  // 
-  Dtmf& d = mQueue.front();
+    //
+    Dtmf& d = mQueue.front();
 
-  output.resize(milliseconds * rate / 1000 * 2);
-  DtmfBuilder::buildInband(d.mTone, d.mCurrentTime, d.mCurrentTime + milliseconds, rate, (short*)output.mutableData());
-  d.mCurrentTime += milliseconds;
-  return true;
+    output.resize((uint64_t)milliseconds * rate / 1000 * 2);
+    DtmfBuilder::buildInband(d.mTone, d.mCurrentTime, d.mCurrentTime + milliseconds, rate, (short*)output.mutableData());
+    d.mCurrentTime += milliseconds;
+    return true;
 }
 
 bool DtmfContext::getRfc2833(int milliseconds, ByteBuffer& output, ByteBuffer& stopPacket)
 {
-  Lock l(mGuard);
-  if (!mQueue.size() || mType != Dtmf_Rfc2833)
-    return false;
+    Lock l(mGuard);
+    if (!mQueue.size() || mType != Dtmf_Rfc2833)
+        return false;
 
-  Dtmf& d = mQueue.front();
-  // See if tone has enough duration to produce another packet
-  if (d.mDuration > 0)
-  {
-    // Emit rfc2833 packet
-    output.resize(4);
-    DtmfBuilder::buildRfc2833(d.mTone, milliseconds, d.mVolume, false, output.mutableData());
-    d.mDuration -= milliseconds;
-    if(d.mDuration <= 0)
-      d.mStopped = true;
-  }
-  else
-  if (!d.mStopped)
-  {
-    output.resize(4);
-    DtmfBuilder::buildRfc2833(d.mTone, milliseconds, d.mVolume, false, output.mutableData());
-  }
-  else
-    output.clear();
+    Dtmf& d = mQueue.front();
+    // See if tone has enough duration to produce another packet
+    if (d.mDuration > 0)
+    {
+        // Emit rfc2833 packet
+        output.resize(4);
+        DtmfBuilder::buildRfc2833({.mTone = (char)d.mTone, .mDuration = milliseconds, .mVolume = d.mVolume, .mEnd = false}, output.mutableData());
+        d.mDuration -= milliseconds;
+        if(d.mDuration <= 0)
+            d.mStopped = true;
+    }
+    else
+    if (!d.mStopped)
+    {
+        output.resize(4);
+        DtmfBuilder::buildRfc2833({.mTone = (char)d.mTone, .mDuration = milliseconds, .mVolume = d.mVolume, .mEnd = false}, output.mutableData());
+    }
+    else
+        output.clear();
 
-  if (d.mStopped)
-  {
-    stopPacket.resize(4);
-    DtmfBuilder::buildRfc2833(d.mTone, milliseconds, d.mVolume, true, stopPacket.mutableData());
-  }
-  else
-    stopPacket.clear();
+    if (d.mStopped)
+    {
+        stopPacket.resize(4);
+        DtmfBuilder::buildRfc2833({.mTone = (char)d.mTone, .mDuration = milliseconds, .mVolume = d.mVolume, .mEnd = true}, stopPacket.mutableData());
+    }
+    else
+        stopPacket.clear();
 
-  if (d.mStopped)
-    mQueue.erase(mQueue.begin());
+    if (d.mStopped)
+        mQueue.erase(mQueue.begin());
 
-  return true;
+    return true;
 }
 
 typedef struct
@@ -375,32 +405,32 @@ void  zap_dtmf_detect_init(dtmf_detect_state_t *s);
 int   zap_dtmf_detect(dtmf_detect_state_t *s, int16_t amp[], int samples, int isradio);
 int   zap_dtmf_get(dtmf_detect_state_t *s, char *buf, int max);
 
-DTMFDetector::DTMFDetector()
-:mState(NULL)
+InbandDtmfDetector::InbandDtmfDetector()
+    :mState(NULL)
 {
-  mState = malloc(sizeof(dtmf_detect_state_t));
+    mState = malloc(sizeof(dtmf_detect_state_t));
 
-  memset(mState, 0, sizeof(dtmf_detect_state_t));
-  zap_dtmf_detect_init((dtmf_detect_state_t*)mState);
+    memset(mState, 0, sizeof(dtmf_detect_state_t));
+    zap_dtmf_detect_init((dtmf_detect_state_t*)mState);
 }
 
-DTMFDetector::~DTMFDetector()
+InbandDtmfDetector::~InbandDtmfDetector()
 {
-  if (mState)
-    free(mState);
+    if (mState)
+        free(mState);
 }
 
-std::string DTMFDetector::streamPut(unsigned char* samples, unsigned int size)
+std::string InbandDtmfDetector::streamPut(unsigned char* samples, unsigned int size)
 {
-  char buf[16]; buf[0] = 0;
-  if (zap_dtmf_detect((dtmf_detect_state_t*)mState, (int16_t*)samples, size/2, 0))
-    zap_dtmf_get((dtmf_detect_state_t*)mState, buf, 15);
-  return buf;
+    char buf[16]; buf[0] = 0;
+    if (zap_dtmf_detect((dtmf_detect_state_t*)mState, (int16_t*)samples, size/2, 0))
+        zap_dtmf_get((dtmf_detect_state_t*)mState, buf, 15);
+    return buf;
 }
 
-void DTMFDetector::resetState()
+void InbandDtmfDetector::resetState()
 {
-  zap_dtmf_detect_init((dtmf_detect_state_t*)mState);
+    zap_dtmf_detect_init((dtmf_detect_state_t*)mState);
 }
 
 #ifndef TRUE
@@ -448,12 +478,12 @@ static tone_detection_descriptor_t fax_detect;
 static tone_detection_descriptor_t fax_detect_2nd;
 
 static float dtmf_row[] =
-{
-     697.0,  770.0,  852.0,  941.0
+    {
+        697.0,  770.0,  852.0,  941.0
 };
 static float dtmf_col[] =
-{
-    1209.0, 1336.0, 1477.0, 1633.0
+    {
+        1209.0, 1336.0, 1477.0, 1633.0
 };
 
 static float fax_freq = 1100.0;
@@ -461,10 +491,10 @@ static float fax_freq = 1100.0;
 static char dtmf_positions[] = "123A" "456B" "789C" "*0#D";
 
 static void goertzel_init(goertzel_state_t *s,
-                   tone_detection_descriptor_t *t)
+                          tone_detection_descriptor_t *t)
 {
     s->v2 =
-    s->v3 = 0.0;
+        s->v3 = 0.0;
     s->fac = t->fac;
 }
 /*- End of function --------------------------------------------------------*/
@@ -497,50 +527,50 @@ static inline void _dtmf_goertzel_update(goertzel_state_t *s,
     //s->v3 = s->fac*s->v2 - v1 + x[0];
 
     __asm__ __volatile__ (
-         " femms;\n"
+        " femms;\n"
 
-             " movq        16(%%edx),%%mm2;\n"
-             " movq        24(%%edx),%%mm3;\n"
-             " movq        32(%%edx),%%mm4;\n"
-             " movq        40(%%edx),%%mm5;\n"
-             " movq        48(%%edx),%%mm6;\n"
-             " movq        56(%%edx),%%mm7;\n"
+        " movq        16(%%edx),%%mm2;\n"
+        " movq        24(%%edx),%%mm3;\n"
+        " movq        32(%%edx),%%mm4;\n"
+        " movq        40(%%edx),%%mm5;\n"
+        " movq        48(%%edx),%%mm6;\n"
+        " movq        56(%%edx),%%mm7;\n"
 
-             " jmp         1f;\n"
-             " .align 32;\n"
+        " jmp         1f;\n"
+        " .align 32;\n"
 
-         " 1: ;\n"
-             " prefetch    (%%eax);\n"
-             " movq        %%mm3,%%mm1;\n"
-             " movq        %%mm2,%%mm0;\n"
-             " movq        %%mm5,%%mm3;\n"
-             " movq        %%mm4,%%mm2;\n"
+        " 1: ;\n"
+        " prefetch    (%%eax);\n"
+        " movq        %%mm3,%%mm1;\n"
+        " movq        %%mm2,%%mm0;\n"
+        " movq        %%mm5,%%mm3;\n"
+        " movq        %%mm4,%%mm2;\n"
 
-             " pfmul       %%mm7,%%mm5;\n"
-             " pfmul       %%mm6,%%mm4;\n"
-             " pfsub       %%mm1,%%mm5;\n"
-             " pfsub       %%mm0,%%mm4;\n"
+        " pfmul       %%mm7,%%mm5;\n"
+        " pfmul       %%mm6,%%mm4;\n"
+        " pfsub       %%mm1,%%mm5;\n"
+        " pfsub       %%mm0,%%mm4;\n"
 
-             " movq        (%%eax),%%mm0;\n"
-             " movq        %%mm0,%%mm1;\n"
-             " punpckldq   %%mm0,%%mm1;\n"
-             " add         $4,%%eax;\n"
-             " pfadd       %%mm1,%%mm5;\n"
-             " pfadd       %%mm1,%%mm4;\n"
+        " movq        (%%eax),%%mm0;\n"
+        " movq        %%mm0,%%mm1;\n"
+        " punpckldq   %%mm0,%%mm1;\n"
+        " add         $4,%%eax;\n"
+        " pfadd       %%mm1,%%mm5;\n"
+        " pfadd       %%mm1,%%mm4;\n"
 
-             " dec         %%ecx;\n"
+        " dec         %%ecx;\n"
 
-             " jnz         1b;\n"
+        " jnz         1b;\n"
 
-             " movq        %%mm2,16(%%edx);\n"
-             " movq        %%mm3,24(%%edx);\n"
-             " movq        %%mm4,32(%%edx);\n"
-             " movq        %%mm5,40(%%edx);\n"
+        " movq        %%mm2,16(%%edx);\n"
+        " movq        %%mm3,24(%%edx);\n"
+        " movq        %%mm4,32(%%edx);\n"
+        " movq        %%mm5,40(%%edx);\n"
 
-             " femms;\n"
-     :
-             : "c" (samples), "a" (x), "d" (vv)
-             : "memory", "eax", "ecx");
+        " femms;\n"
+        :
+        : "c" (samples), "a" (x), "d" (vv)
+        : "memory", "eax", "ecx");
 
     s[0].v2 = vv[4];
     s[1].v2 = vv[5];
@@ -555,8 +585,8 @@ static inline void _dtmf_goertzel_update(goertzel_state_t *s,
 /*- End of function --------------------------------------------------------*/
 
 void zap_goertzel_update(goertzel_state_t *s,
-                     int16_t x[],
-                     int samples)
+                         int16_t x[],
+                         int samples)
 {
     int i;
     float v1;
@@ -582,7 +612,7 @@ void zap_dtmf_detect_init (dtmf_detect_state_t *s)
     float theta;
 
     s->hit1 = 
-    s->hit2 = 0;
+        s->hit2 = 0;
 
     for (i = 0;  i < 4;  i++)
     {
@@ -598,12 +628,12 @@ void zap_dtmf_detect_init (dtmf_detect_state_t *s)
         theta = float(2.0*M_PI*(dtmf_col[i]*2.0/SAMPLE_RATE));
         dtmf_detect_col_2nd[i].fac = float(2.0*cos(theta));
 
-   goertzel_init (&s->row_out[i], &dtmf_detect_row[i]);
-    goertzel_init (&s->col_out[i], &dtmf_detect_col[i]);
-    goertzel_init (&s->row_out2nd[i], &dtmf_detect_row_2nd[i]);
-    goertzel_init (&s->col_out2nd[i], &dtmf_detect_col_2nd[i]);
+        goertzel_init (&s->row_out[i], &dtmf_detect_row[i]);
+        goertzel_init (&s->col_out[i], &dtmf_detect_col[i]);
+        goertzel_init (&s->row_out2nd[i], &dtmf_detect_row_2nd[i]);
+        goertzel_init (&s->col_out2nd[i], &dtmf_detect_col_2nd[i]);
 
-s->energy = 0.0;
+        s->energy = 0.0;
     }
 
     /* Same for the fax dector */
@@ -625,9 +655,9 @@ s->energy = 0.0;
 /*- End of function --------------------------------------------------------*/
 
 int zap_dtmf_detect (dtmf_detect_state_t *s,
-                 int16_t amp[],
-                 int samples, 
- int isradio)
+                    int16_t amp[],
+                    int samples,
+                    int isradio)
 {
 
     float row_energy[4];
@@ -666,7 +696,7 @@ int zap_dtmf_detect (dtmf_detect_state_t *s,
         {
             famp = amp[j];
 
-    s->energy += famp*famp;
+            s->energy += famp*famp;
 
             /* With GCC 2.95, the following unrolled code seems to take about 35%
                (rough estimate) as long as a neat little 0-3 loop */
@@ -734,7 +764,7 @@ int zap_dtmf_detect (dtmf_detect_state_t *s,
             s->row_out2nd[3].v2 = s->row_out2nd[3].v3;
             s->row_out2nd[3].v3 = s->row_out2nd[3].fac*s->row_out2nd[3].v2 - v1 + famp;
 
-/* Update fax tone */
+            /* Update fax tone */
             v1 = s->fax_tone.v2;
             s->fax_tone.v2 = s->fax_tone.v3;
             s->fax_tone.v3 = s->fax_tone.fac*s->fax_tone.v2 - v1 + famp;
@@ -748,28 +778,28 @@ int zap_dtmf_detect (dtmf_detect_state_t *s,
         if (s->current_sample < 102)
             continue;
 
-/* Detect the fax energy, too */
-fax_energy = zap_goertzel_result(&s->fax_tone);
+        /* Detect the fax energy, too */
+        fax_energy = zap_goertzel_result(&s->fax_tone);
 
         /* We are at the end of a DTMF detection block */
         /* Find the peak row and the peak column */
         row_energy[0] = zap_goertzel_result (&s->row_out[0]);
         col_energy[0] = zap_goertzel_result (&s->col_out[0]);
 
-for (best_row = best_col = 0, i = 1;  i < 4;  i++)
-{
-        row_energy[i] = zap_goertzel_result (&s->row_out[i]);
+        for (best_row = best_col = 0, i = 1;  i < 4;  i++)
+        {
+            row_energy[i] = zap_goertzel_result (&s->row_out[i]);
             if (row_energy[i] > row_energy[best_row])
                 best_row = i;
-        col_energy[i] = zap_goertzel_result (&s->col_out[i]);
+            col_energy[i] = zap_goertzel_result (&s->col_out[i]);
             if (col_energy[i] > col_energy[best_col])
                 best_col = i;
-    }
+        }
         hit = 0;
         /* Basic signal level test and the twist test */
         if (row_energy[best_row] >= DTMF_THRESHOLD
-    &&
-    col_energy[best_col] >= DTMF_THRESHOLD
+            &&
+            col_energy[best_col] >= DTMF_THRESHOLD
             &&
             col_energy[best_col] < row_energy[best_row]*DTMF_REVERSE_TWIST
             &&
@@ -787,8 +817,8 @@ for (best_row = best_col = 0, i = 1;  i < 4;  i++)
             }
             /* ... and second harmonic test */
             if (i >= 4
-        &&
-(row_energy[best_row] + col_energy[best_col]) > 42.0*s->energy
+                &&
+                (row_energy[best_row] + col_energy[best_col]) > 42.0*s->energy
                 &&
                 zap_goertzel_result (&s->col_out2nd[best_col])*DTMF_2ND_HARMONIC_COL < col_energy[best_col]
                 &&
@@ -804,7 +834,7 @@ for (best_row = best_col = 0, i = 1;  i < 4;  i++)
    to a digit. */
                 if (hit == s->hit3  &&  s->hit3 != s->hit2)
                 {
-    s->mhit = hit;
+                    s->mhit = hit;
                     s->digit_hits[(best_row << 2) + best_col]++;
                     s->detected_digits++;
                     if (s->current_digits < MAX_DTMF_DIGITS)
@@ -819,60 +849,60 @@ for (best_row = best_col = 0, i = 1;  i < 4;  i++)
                 }
             }
         } 
-if (!hit && (fax_energy >= FAX_THRESHOLD) && (fax_energy > s->energy * 21.0)) {
-fax_energy_2nd = zap_goertzel_result(&s->fax_tone2nd);
-if (fax_energy_2nd * FAX_2ND_HARMONIC < fax_energy) {
+        if (!hit && (fax_energy >= FAX_THRESHOLD) && (fax_energy > s->energy * 21.0)) {
+            fax_energy_2nd = zap_goertzel_result(&s->fax_tone2nd);
+            if (fax_energy_2nd * FAX_2ND_HARMONIC < fax_energy) {
 #if 0
 printf("Fax energy/Second Harmonic: %f/%f\n", fax_energy, fax_energy_2nd);
 #endif
-/* XXX Probably need better checking than just this the energy XXX */
-hit = 'f';
-s->fax_hits++;
-} /* Don't reset fax hits counter */
-} else {
-if (s->fax_hits > 5) {
- s->mhit = 'f';
-             s->detected_digits++;
-             if (s->current_digits < MAX_DTMF_DIGITS)
-             {
-                  s->digits[s->current_digits++] = hit;
-                  s->digits[s->current_digits] = '\0';
-             }
-             else
-             {
-                   s->lost_digits++;
-             }
-}
-s->fax_hits = 0;
-}
+                /* XXX Probably need better checking than just this the energy XXX */
+                hit = 'f';
+                s->fax_hits++;
+            } /* Don't reset fax hits counter */
+        } else {
+            if (s->fax_hits > 5) {
+                s->mhit = 'f';
+                s->detected_digits++;
+                if (s->current_digits < MAX_DTMF_DIGITS)
+                {
+                    s->digits[s->current_digits++] = hit;
+                    s->digits[s->current_digits] = '\0';
+                }
+                else
+                {
+                    s->lost_digits++;
+                }
+            }
+            s->fax_hits = 0;
+        }
         s->hit1 = s->hit2;
         s->hit2 = s->hit3;
         s->hit3 = hit;
         /* Reinitialise the detector for the next block */
         for (i = 0;  i < 4;  i++)
         {
-           goertzel_init (&s->row_out[i], &dtmf_detect_row[i]);
+            goertzel_init (&s->row_out[i], &dtmf_detect_row[i]);
             goertzel_init (&s->col_out[i], &dtmf_detect_col[i]);
-        goertzel_init (&s->row_out2nd[i], &dtmf_detect_row_2nd[i]);
-        goertzel_init (&s->col_out2nd[i], &dtmf_detect_col_2nd[i]);
+            goertzel_init (&s->row_out2nd[i], &dtmf_detect_row_2nd[i]);
+            goertzel_init (&s->col_out2nd[i], &dtmf_detect_col_2nd[i]);
         }
-    goertzel_init (&s->fax_tone, &fax_detect);
-    goertzel_init (&s->fax_tone2nd, &fax_detect_2nd);
-s->energy = 0.0;
+        goertzel_init (&s->fax_tone, &fax_detect);
+        goertzel_init (&s->fax_tone2nd, &fax_detect_2nd);
+        s->energy = 0.0;
         s->current_sample = 0;
     }
     if ((!s->mhit) || (s->mhit != hit))
     {
-s->mhit = 0;
-return(0);
+        s->mhit = 0;
+        return(0);
     }
     return (hit);
 }
 /*- End of function --------------------------------------------------------*/
 
 int zap_dtmf_get (dtmf_detect_state_t *s,
-              char *buf,
-              int max)
+                 char *buf,
+                 int max)
 {
     if (max > s->current_digits)
         max = s->current_digits;

src/engine/media/MT_Dtmf.h

@@ -15,18 +15,32 @@
 namespace MT
 {
 
-  class DtmfBuilder
-  {
-  public:
+class DtmfBuilder
+{
+public:
+    struct Rfc2833Event
+    {
+        char mTone = 0;
+        int mDuration = 0;
+        int mVolume = 0;
+        bool mEnd = false;
+
+        bool isValid() const {
+            return mTone != 0;
+        }
+    };
+
     // Output should be 4 bytes length
-    static void buildRfc2833(int tone, int duration, int volume, bool endOfEvent, void* output);
+    static void buildRfc2833(const Rfc2833Event& ev, void* output);
+
+    static Rfc2833Event parseRfc2833(std::span<uint8_t> payload);
 
     // Buf receives PCM audio
     static void buildInband(int tone, int startTime, int finishTime, int rate, short* buf);
-  };
+};
 
-  struct Dtmf
-  {
+struct Dtmf
+{
     Dtmf(): mTone(0), mDuration(0), mVolume(0), mFinishCount(3), mCurrentTime(0), mStopped(false) {}
     Dtmf(int tone, int volume, int duration): mTone(tone), mVolume(volume), mDuration(duration), mFinishCount(3), mCurrentTime(0), mStopped(false) {}
 
@@ -36,17 +50,17 @@ namespace MT
     int mVolume;
     int mFinishCount;
     int mCurrentTime;
-  };
+};
 
-  typedef std::vector<Dtmf> DtmfQueue;
+typedef std::vector<Dtmf> DtmfQueue;
 
-  class DtmfContext
-  {
-  public:
+class DtmfContext
+{
+public:
     enum Type
     {
-      Dtmf_Inband,
-      Dtmf_Rfc2833
+        Dtmf_Inband,
+        Dtmf_Rfc2833
     };
 
     DtmfContext();
@@ -65,33 +79,34 @@ namespace MT
     bool getInband(int milliseconds, int rate, ByteBuffer& output);
     bool getRfc2833(int milliseconds, ByteBuffer& output, ByteBuffer& stopPacket);
 
-  protected:
+protected:
     Mutex mGuard;
     Type mType;
     DtmfQueue mQueue;
-  };
+};
 
-class DTMFDetector 
+class InbandDtmfDetector
 {
 public:
-  /*! The default constructor. Allocates space for detector context. */
-  DTMFDetector();
+    /*! The default constructor. Allocates space for detector context. */
+    InbandDtmfDetector();
 
-  /*! The destructor. Free the detector context's memory. */
-  ~DTMFDetector();
+    /*! The destructor. Free the detector context's memory. */
+    ~InbandDtmfDetector();
 
-  /*! This method receives the input PCM 16-bit data and returns found DTMF event(s) in string representation.
+    /*! This method receives the input PCM 16-bit data and returns found DTMF event(s) in string representation.
    *  @param samples Input PCM buffer pointer.
    *  @param size Size of input buffer in bytes
    *  @return Found DTMF event(s) in string representation. The returned value has variable length.
    */
-  std::string streamPut(unsigned char* samples, unsigned int size);
+    std::string streamPut(unsigned char* samples, unsigned int size);
 
-  void resetState();
+    void resetState();
 
 protected:
-  void* mState; /// DTMF detector context
+    void* mState; /// DTMF detector context
 };
+
 }
 
 #endif

src/engine/media/MT_EvsCodec.cpp

@@ -1,6 +1,6 @@
 #include "MT_EvsCodec.h"
-
-
+#include <set>
+#include <map>
 
 /*-------------------------------------------------------------------*
 * rate2AMRWB_IOmode()
@@ -167,67 +167,58 @@ EVSCodec::~EVSCodec()
     }
 }
 
-int EVSCodec::samplerate()
-{
-    return st_dec->output_Fs;
+Codec::Info EVSCodec::info() {
+    return {
+        .mName = MT_EVS_CODECNAME,
+        .mSamplerate = st_dec->output_Fs,
+        .mChannels = 1,
+        .mPcmLength = st_dec->output_Fs / 1000 * sp.ptime * 2,
+        .mFrameTime = sp.ptime,
+        .mRtpLength = 0
+    };
 }
 
-int EVSCodec::pcmLength()
-{
-    return samplerate() / 50 * 2;
-}
 
-int EVSCodec::frameTime()
-{
-	return sp.ptime;
-}
-
-int EVSCodec::rtpLength()
-{
-    // Variable sized codec - bitrate can be changed during the call
-	return 0;
-}
-
-int EVSCodec::encode(const void* input, int inputBytes, void* output, int outputCapacity)
+Codec::EncodeResult EVSCodec::encode(std::span<const uint8_t> input, std::span<uint8_t> output)
 {
     // Encoding is not supported yet.
-	return 0;
+    return {.mEncoded = 0};
 }
 
-int EVSCodec::decode(const void* input, int input_length, void* output, int outputCapacity)
+Codec::DecodeResult EVSCodec::decode(std::span<const uint8_t> input, std::span<uint8_t> output)
 {
-    if (outputCapacity < pcmLength())
-        return 0;
+    if (output.size_bytes() < pcmLength())
+        return {.mDecoded = 0};
 
     std::string buffer;
 
     // Check if we get payload with CMR
-    auto payload_iter = FixedPayload_EVSPrimary.find((input_length - 2) * 8);
+    auto payload_iter = FixedPayload_EVSPrimary.find((input.size_bytes() - 2) * 8);
     if (payload_iter == FixedPayload_EVSPrimary.end())
     {
         // Check if we get payload with ToC and without CMR
-        payload_iter = FixedPayload_EVSPrimary.find((input_length - 1) * 8);
+        payload_iter = FixedPayload_EVSPrimary.find((input.size_bytes() - 1) * 8);
         if (payload_iter == FixedPayload_EVSPrimary.end())
         {
             // Maybe there is no ToC ?
-            payload_iter = FixedPayload_EVSPrimary.find(input_length * 8);
+            payload_iter = FixedPayload_EVSPrimary.find(input.size_bytes() * 8);
             if (payload_iter == FixedPayload_EVSPrimary.end())
             {
                 // Bad payload size at all
-                return 0;
+                return {.mDecoded = 0};
             }
             /* Add ToC byte.
              * WARNING maybe it will be work incorrect with 56bit payload,
              * see 3GPP TS 26.445 Annex A, A.2.1.3 */
-            char c = evs::rate2EVSmode(FixedPayload_EVSPrimary.find(input_length * 8)->second);
+            char c = evs::rate2EVSmode(FixedPayload_EVSPrimary.find(input.size_bytes() * 8)->second);
             buffer += c;
-            buffer += std::string(reinterpret_cast<const char*>(input), input_length);
+            buffer += std::string(reinterpret_cast<const char*>(input.data()), input.size_bytes());
         }
         else
-            buffer = std::string(reinterpret_cast<const char*>(input), input_length);
+            buffer = std::string(reinterpret_cast<const char*>(input.data()), input.size_bytes());
     }
     else // Skip CMR byte
-        buffer = std::string(reinterpret_cast<const char*>(input) + 1, input_length-1);
+        buffer = std::string(reinterpret_cast<const char*>(input.data()) + 1, input.size_bytes()-1);
 
 
     // Output buffer for 48 KHz
@@ -263,7 +254,7 @@ int EVSCodec::decode(const void* input, int input_length, void* output, int outp
 		}
 
 		/* convert 'float' output data to 'short' */
-        evs::syn_output(data, static_cast<short>(pcmLength() / 2), static_cast<short*>(output) + offset);
+        evs::syn_output(data, static_cast<short>(pcmLength() / 2), reinterpret_cast<short*>(output.data()) + offset);
         offset += pcmLength() / 2;
         if (st_dec->ini_frame < MAX_FRAME_COUNTER)
 		{
@@ -271,12 +262,12 @@ int EVSCodec::decode(const void* input, int input_length, void* output, int outp
 		}
 	}
 
-    return pcmLength();
+    return {.mDecoded = (size_t)pcmLength()};
 }
 
-int EVSCodec::plc(int lostFrames, void* output, int outputCapacity)
+size_t EVSCodec::plc(int lostFrames, std::span<uint8_t> output)
 {
-	return 0;
+    return 0;
 }
 
 void EVSCodec::initDecoder(const StreamParameters& sp)

src/engine/media/MT_EvsCodec.h

@@ -2,15 +2,11 @@
 #define __MT_EVS_CODEC_H
 
 #include "../engine_config.h"
-#include <set>
-#include <map>
-#include <memory>
 #include <stdio.h>
 #include <stdlib.h>
 #include <string.h>
 #include <time.h>
 #include <assert.h>
-#include <sstream>
 
 #include "MT_Codec.h"
 
@@ -52,18 +48,14 @@ public:
     EVSCodec(const StreamParameters& sp);
     ~EVSCodec() override;
 
-    const char* name() override { return MT_EVS_CODECNAME; }
-    int samplerate() override;
-    int pcmLength() override;
-    int frameTime() override;
-    int rtpLength() override;
-    int encode(const void* input, int inputBytes, void* output, int outputCapacity) override;
-    int decode(const void* input, int inputBytes, void* output, int outputCapacity) override;
-    int plc(int lostFrames, void* output, int outputCapacity) override;
+    Info info() override;
+
+    EncodeResult encode(std::span<const uint8_t> input, std::span<uint8_t> output) override;
+    DecodeResult decode(std::span<const uint8_t> input, std::span<uint8_t> output) override;
+    size_t       plc(int lostFrames, std::span<uint8_t> output) override;
 
 private:
-    evs::Decoder_State* st_dec;
-    //Encoder_State_fx* st_enc;
+    evs::Decoder_State* st_dec = nullptr;
     StreamParameters sp;
     void initDecoder(const StreamParameters& sp);
 };

src/engine/media/MT_SingleAudioStream.cpp

@@ -13,19 +13,17 @@
 using namespace MT;
 
 SingleAudioStream::SingleAudioStream(const CodecList::Settings& codecSettings, Statistics& stat)
-    :mReceiver(codecSettings, stat), mDtmfReceiver(stat)
-{
-}
+    :mReceiver(codecSettings, stat),
+     mDtmfReceiver(stat)
+{}
 
 SingleAudioStream::~SingleAudioStream()
-{
-
-}
+{}
 
 void SingleAudioStream::process(const std::shared_ptr<jrtplib::RTPPacket>& packet)
 {
     ICELogMedia(<< "Processing incoming RTP/RTCP packet");
-    if (packet->GetPayloadType() == 101/*resip::Codec::TelephoneEvent.payloadType()*/)
+    if (packet->GetPayloadType() == mReceiver.getCodecSettings().mTelephoneEvent)
         mDtmfReceiver.add(packet);
     else
         mReceiver.add(packet);
@@ -33,9 +31,10 @@ void SingleAudioStream::process(const std::shared_ptr<jrtplib::RTPPacket>& packe
 
 void SingleAudioStream::copyPcmTo(Audio::DataWindow& output, int needed)
 {
+    // Packet by packet
     while (output.filled() < needed)
     {
-        if (mReceiver.getAudio(output, {}) != AudioReceiver::DecodeResult_Ok)
+        if (mReceiver.getAudioTo(output, {}).mStatus != AudioReceiver::DecodeResult::Status::Ok)
             break;
     }
 

src/engine/media/MT_SingleAudioStream.h

@@ -13,19 +13,19 @@
 #include "MT_AudioReceiver.h"
 namespace MT
 {
-  class SingleAudioStream
-  {
-  public:
+class SingleAudioStream
+{
+public:
     SingleAudioStream(const CodecList::Settings& codecSettings, Statistics& stat);
     ~SingleAudioStream();
     void process(const std::shared_ptr<jrtplib::RTPPacket>& packet);
     void copyPcmTo(Audio::DataWindow& output, int needed);
 
-  protected:
+protected:
     DtmfReceiver mDtmfReceiver;
     AudioReceiver mReceiver;
-  };
+};
 
-  typedef std::map<unsigned, SingleAudioStream*> AudioStreamMap; 
+typedef std::map<unsigned, SingleAudioStream*> AudioStreamMap;
 }
 #endif

src/engine/media/MT_SrtpHelper.cpp

@@ -212,15 +212,13 @@ void SrtpSession::close()
 
 SrtpKeySalt& SrtpSession::outgoingKey(SrtpSuite suite)
 {
-    Lock l(mGuard);
     assert(suite > SRTP_NONE && suite <= SRTP_LAST);
-    return mOutgoingKey[int(suite)-1];
+    Lock l(mGuard);
+    return mOutgoingKey[int(suite)-1];  // The automated review sometimes give the hints about the possible underflow array index access
 }
 
 bool SrtpSession::protectRtp(void* buffer, int* length)
 {
-    // addSsrc(RtpHelper::findSsrc(buffer, *length), sdOutgoing);
-
     Lock l(mGuard);
     if (mOutboundSession)
     {

src/engine/media/MT_SrtpHelper.h

@@ -20,20 +20,17 @@
 
 enum SrtpSuite
 {
-    SRTP_NONE,
-    SRTP_AES_128_AUTH_80,
-    SRTP_AES_256_AUTH_80,
-    SRTP_AES_192_AUTH_80,
-    SRTP_AES_128_AUTH_32,
-    SRTP_AES_256_AUTH_32,
-    SRTP_AES_192_AUTH_32,
-    SRTP_AES_128_AUTH_NULL,
-    SRTP_AED_AES_256_GCM,
-    SRTP_AED_AES_128_GCM,
+    SRTP_NONE               = 0,
+    SRTP_AES_128_AUTH_80    = 1,
+    SRTP_AES_256_AUTH_80    = 2,
+    SRTP_AES_192_AUTH_80    = 3,
+    SRTP_AES_128_AUTH_32    = 4,
+    SRTP_AES_256_AUTH_32    = 5,
+    SRTP_AES_192_AUTH_32    = 6,
+    SRTP_AES_128_AUTH_NULL  = 7,
+    SRTP_AED_AES_256_GCM    = 8,
+    SRTP_AED_AES_128_GCM    = 9,
     SRTP_LAST = SRTP_AED_AES_128_GCM
-    // ToDo:
-    // a=crypto:1 AEAD_AES_256_GCM_8 inline:tN2A0vRjFBimpQsW2GasuJuPe7hKE26gki30APC8DVuySqCOYTs8lYBPR5I=
-    // a=crypto:3 AEAD_AES_128_GCM_8 inline:Ok7VL8SmBHSbZLw4dK6iQgpliYKGdY9BHLJcRw==
 };
 
 extern SrtpSuite        toSrtpSuite(const std::string_view& s);

src/engine/media/MT_Statistics.cpp

@@ -74,13 +74,16 @@ Statistics::~Statistics()
 
 void Statistics::calculateBurstr(double* burstr, double* lossr) const
 {
-    int lost = 0;
-    int bursts = 0;
-    for (int i = 0; i < 128; i++)
-    {
-        lost += i * mLoss[i];
-        bursts += mLoss[i];
-    }
+    int lost = 0; // Total packet lost
+    for (const auto& item: mPacketLossTimeline)
+        lost += item.mGap;
+    int bursts = mPacketLossTimeline.size(); // number of events
+
+    // for (const auto& entry: mLoss)
+    // {
+    //     lost += entry.first * entry.second;
+    //     bursts += entry.second;
+    // }
 
     if (lost < 5)
     {
@@ -109,14 +112,14 @@ void Statistics::calculateBurstr(double* burstr, double* lossr) const
 double Statistics::calculateMos(double maximalMos) const
 {
     // calculate lossrate and burst rate
-    double burstr, lossr;
+    double burstr = 0, lossr = 0;
     calculateBurstr(&burstr, &lossr);
 
-    double r;
+    double r = 0.0;
     double bpl = 8.47627; //mos = -4.23836 + 0.29873 * r - 0.00416744 * r * r + 0.0000209855 * r * r * r;
-    double mos;
+    double mos = 0.0;
 
-    if (mReceivedRtp < 100)
+    if (mReceivedRtp < 10)
         return 0.0;
 
     if (lossr == 0.0 || burstr == 0.0)

src/engine/media/MT_Statistics.h

@@ -4,16 +4,15 @@
 #include <chrono>
 #include <map>
 #include <optional>
-#include <array>
 
-#include "audio/Audio_DataWindow.h"
-#include "helper/HL_Optional.hpp"
 #include "helper/HL_Statistics.h"
 #include "helper/HL_Types.h"
+#include "helper/HL_InternetAddress.h"
 
 #include "jrtplib/src/rtptimeutilities.h"
 #include "jrtplib/src/rtppacket.h"
 
+using namespace std::chrono_literals;
 
 namespace MT
 {
@@ -56,46 +55,55 @@ struct PacketLossEvent
     uint32_t    mStartSeqno = 0,
                 mEndSeqno = 0;
     int         mGap = 0;
+    std::chrono::microseconds mTimestampStart = 0us,
+                              mTimestampEnd = 0us;
+};
+
+struct Dtmf2833Event
+{
+    char mTone;
     std::chrono::microseconds mTimestamp;
 };
 
 class Statistics
 {
 public:
-    size_t                      mReceived = 0,        // Received traffic in bytes
-                                mSent = 0,            // Sent traffic in bytes
-                                mReceivedRtp = 0,     // Number of received rtp packets
-                                mSentRtp = 0,         // Number of sent rtp packets
-                                mReceivedRtcp = 0,    // Number of received rtcp packets
-                                mSentRtcp = 0,        // Number of sent rtcp packets
-                                mDuplicatedRtp = 0,   // Number of received duplicated rtp packets
-                                mOldRtp = 0,          // Number of late rtp packets
-                                mPacketLoss = 0,      // Number of lost packets
-                                mPacketDropped = 0,   // Number of dropped packets (due to time unsync when playing)б
-                                mIllegalRtp = 0;      // Number of rtp packets with bad payload type
+    size_t                          mReceived = 0,        // Received traffic in bytes
+                                    mSent = 0,            // Sent traffic in bytes
+                                    mReceivedRtp = 0,     // Number of received rtp packets
+                                    mSentRtp = 0,         // Number of sent rtp packets
+                                    mReceivedRtcp = 0,    // Number of received rtcp packets
+                                    mSentRtcp = 0,        // Number of sent rtcp packets
+                                    mDuplicatedRtp = 0,   // Number of received duplicated rtp packets
+                                    mOldRtp = 0,          // Number of late rtp packets
+                                    mPacketLoss = 0,      // Number of lost packets
+                                    mPacketDropped = 0,   // Number of dropped packets (due to time unsync when playing)б
+                                    mIllegalRtp = 0;      // Number of rtp packets with bad payload type
 
-    TestResult<float>           mDecodingInterval,      // Average interval on call to packet decode
-                                mDecodeRequested,       // Average amount of requested audio frames to play
-                                mPacketInterval;        // Average interval between packet adding to jitter buffer
+    TestResult<float>               mDecodingInterval,      // Average interval on call to packet decode
+                                    mDecodeRequested,       // Average amount of requested audio frames to play
+                                    mPacketInterval;        // Average interval between packet adding to jitter buffer
 
-    std::array<float, 128>      mLoss = {0};            // Every item is number of loss of corresping length
-    size_t                      mAudioTime = 0;         // Decoded/found time in milliseconds
-    size_t                      mDecodedSize = 0;       // Number of decoded bytes
-    uint16_t                    mSsrc = 0;              // Last known SSRC ID in a RTP stream
-    ice::NetworkAddress         mRemotePeer;            // Last known remote RTP address
+    std::map<int,int>               mLoss;            // Every item is number of loss of corresping length
+    size_t                          mAudioTime = 0;         // Decoded/found time in milliseconds
+    size_t                          mDecodedSize = 0;       // Number of decoded bytes
+    uint16_t                        mSsrc = 0;              // Last known SSRC ID in a RTP stream
+    ice::NetworkAddress             mRemotePeer;            // Last known remote RTP address
 
     // AMR codec bitrate switch counter
-    int                         mBitrateSwitchCounter = 0;
-    std::string                 mCodecName;
-    float                       mJitter = 0.0f;         // Jitter
-    TestResult<float>           mRttDelay;              // RTT delay
+    int                             mBitrateSwitchCounter = 0;
+    int                             mCng = 0;
+    std::string                     mCodecName;
+    float                           mJitter = 0.0f;         // Jitter
+    TestResult<float>               mRttDelay;              // RTT delay
 
     // Timestamp when first RTP packet has arrived
-    std::optional<timepoint_t>  mFirstRtpTime;
+    std::optional<timepoint_t>      mFirstRtpTime;
 
-    std::map<int, int>          mCodecCount;            // Stats on used codecs
+    std::map<int, int>              mCodecCount;            // Stats on used codecs
 
-    std::vector<PacketLossEvent> mPacketLossTimeline;   // Packet loss timeline
+    std::vector<PacketLossEvent>    mPacketLossTimeline;   // Packet loss timeline
+    std::vector<Dtmf2833Event>      mDtmf2833Timeline;
 
     // It is to calculate network MOS
     void calculateBurstr(double* burstr, double* loss) const;

test/rtp_decode/CMakeLists.txt

@@ -0,0 +1,10 @@
+cmake_minimum_required(VERSION 3.20)
+project(rtp_decode)
+
+set (CMAKE_CXX_STANDARD 20)
+set (CMAKE_CXX_STANDARD_REQUIRED ON)
+
+add_subdirectory(../../src build_rtphone)
+
+add_executable(rtp_decode main.cpp)
+target_link_libraries(rtp_decode PRIVATE rtphone)

test/rtp_decode/main.cpp

@@ -0,0 +1,291 @@
+/* Copyright(C) 2007-2026 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/. */
+
+// rtp_decode — read an rtpdump file, decode RTP with a given codec, write WAV.
+//
+// Usage:
+//   rtp_decode <input.rtp> <output.wav> --codec <name> [--pt <N>] [--rate <N>] [--channels <N>]
+
+#include "helper/HL_Rtp.h"
+#include "media/MT_CodecList.h"
+#include "media/MT_Codec.h"
+#include "audio/Audio_WavFile.h"
+
+#include <cstdio>
+#include <cstdlib>
+#include <cstring>
+#include <string>
+#include <vector>
+#include <stdexcept>
+
+// ---------------------------------------------------------------------------
+// CLI helpers
+// ---------------------------------------------------------------------------
+
+static void usage(const char* progname)
+{
+    fprintf(stderr,
+        "Usage: %s <input.rtp> <output.wav> --codec <name> [--pt <N>] [--rate <N>] [--channels <N>]\n"
+        "\n"
+        "Codecs: pcmu pcma g722 g729 opus gsm gsmhr gsmefr\n"
+        "        amrnb amrwb amrnb-bwe amrwb-bwe evs ilbc20 ilbc30 isac16 isac32\n"
+        "\n"
+        "Options:\n"
+        "  --codec <name>    Codec name (required)\n"
+        "  --pt <N>          Override RTP payload type\n"
+        "  --rate <N>        Sample rate hint for Opus (default 48000)\n"
+        "  --channels <N>    Channel count hint for Opus (default 2)\n",
+        progname);
+}
+
+static const char* getOption(int argc, char* argv[], const char* name)
+{
+    for (int i = 1; i < argc - 1; ++i) {
+        if (strcmp(argv[i], name) == 0)
+            return argv[i + 1];
+    }
+    return nullptr;
+}
+
+// ---------------------------------------------------------------------------
+// Default payload types for codecs without a fixed standard PT
+// ---------------------------------------------------------------------------
+struct CodecDefaults
+{
+    const char* name;
+    int defaultPt;      // -1 = must be specified via --pt
+    bool needsPt;       // true if --pt is required when no default exists
+};
+
+static const CodecDefaults kCodecTable[] = {
+    { "pcmu",   0,   false },
+    { "pcma",   8,   false },
+    { "g722",   9,   false },
+    { "g729",   18,  false },
+    { "gsm",    3,   false },
+    { "opus",   106, false },
+    { "amrnb",     -1,  true  },
+    { "amrwb",     -1,  true  },
+    { "amrnb-bwe", -1,  true  },
+    { "amrwb-bwe", -1,  true  },
+    { "gsmhr",  -1,  true  },
+    { "gsmefr", 126, false },
+    { "evs",    127, false },
+    { "ilbc20", -1,  true  },
+    { "ilbc30", -1,  true  },
+    { "isac16", -1,  true  },
+    { "isac32", -1,  true  },
+};
+
+static const CodecDefaults* findCodecDefaults(const std::string& name)
+{
+    for (auto& c : kCodecTable)
+        if (name == c.name)
+            return &c;
+    return nullptr;
+}
+
+// ---------------------------------------------------------------------------
+// Build CodecList::Settings for the requested codec
+// ---------------------------------------------------------------------------
+static MT::CodecList::Settings buildSettings(const std::string& codecName, int pt,
+                                              int opusRate, int opusChannels)
+{
+    MT::CodecList::Settings s;
+
+    if (codecName == "opus") {
+        s.mOpusSpec.push_back(MT::CodecList::Settings::OpusSpec(pt, opusRate, opusChannels));
+    } else if (codecName == "gsm") {
+        s.mGsmFrPayloadType = pt;
+    } else if (codecName == "gsmhr") {
+        s.mGsmHrPayloadType = pt;
+    } else if (codecName == "gsmefr") {
+        s.mGsmEfrPayloadType = pt;
+    } else if (codecName == "amrnb") {
+        s.mAmrNbOctetPayloadType.insert(pt);
+    } else if (codecName == "amrwb") {
+        s.mAmrWbOctetPayloadType.insert(pt);
+    } else if (codecName == "amrnb-bwe") {
+        s.mAmrNbPayloadType.insert(pt);
+    } else if (codecName == "amrwb-bwe") {
+        s.mAmrWbPayloadType.insert(pt);
+    } else if (codecName == "evs") {
+        MT::CodecList::Settings::EvsSpec ev;
+        ev.mPayloadType = pt;
+        s.mEvsSpec.push_back(ev);
+    } else if (codecName == "ilbc20") {
+        s.mIlbc20PayloadType = pt;
+    } else if (codecName == "ilbc30") {
+        s.mIlbc30PayloadType = pt;
+    } else if (codecName == "isac16") {
+        s.mIsac16KPayloadType = pt;
+    } else if (codecName == "isac32") {
+        s.mIsac32KPayloadType = pt;
+    }
+    // pcmu, pcma, g722, g729 — fixed PT, auto-registered by CodecList::init()
+
+    return s;
+}
+
+// ---------------------------------------------------------------------------
+// main
+// ---------------------------------------------------------------------------
+int main(int argc, char* argv[])
+{
+    if (argc < 4) {
+        usage(argv[0]);
+        return 1;
+    }
+
+    const char* inputPath  = argv[1];
+    const char* outputPath = argv[2];
+
+    const char* codecArg = getOption(argc, argv, "--codec");
+    if (!codecArg) {
+        fprintf(stderr, "Error: --codec is required\n\n");
+        usage(argv[0]);
+        return 1;
+    }
+    std::string codecName = codecArg;
+
+    const auto* defaults = findCodecDefaults(codecName);
+    if (!defaults) {
+        fprintf(stderr, "Error: unknown codec '%s'\n\n", codecArg);
+        usage(argv[0]);
+        return 1;
+    }
+
+    // Resolve payload type
+    int pt = defaults->defaultPt;
+    const char* ptArg = getOption(argc, argv, "--pt");
+    if (ptArg) {
+        pt = atoi(ptArg);
+    } else if (defaults->needsPt) {
+        fprintf(stderr, "Error: --pt is required for codec '%s'\n\n", codecArg);
+        usage(argv[0]);
+        return 1;
+    }
+
+    int opusRate     = 48000;
+    int opusChannels = 2;
+    const char* rateArg = getOption(argc, argv, "--rate");
+    if (rateArg)
+        opusRate = atoi(rateArg);
+    const char* chArg = getOption(argc, argv, "--channels");
+    if (chArg)
+        opusChannels = atoi(chArg);
+
+    // -----------------------------------------------------------------------
+    // 1. Load rtpdump
+    // -----------------------------------------------------------------------
+    RtpDump dump(inputPath);
+    try {
+        dump.load();
+    } catch (const std::exception& e) {
+        fprintf(stderr, "Error loading rtpdump '%s': %s\n", inputPath, e.what());
+        return 1;
+    }
+
+    if (dump.count() == 0) {
+        fprintf(stderr, "No packets in '%s'\n", inputPath);
+        return 1;
+    }
+    fprintf(stderr, "Loaded %zu packets from '%s'\n", dump.count(), inputPath);
+
+    // -----------------------------------------------------------------------
+    // 2. Create codec
+    // -----------------------------------------------------------------------
+    auto settings = buildSettings(codecName, pt, opusRate, opusChannels);
+    MT::CodecList codecList(settings);
+    MT::PCodec codec = codecList.createCodecByPayloadType(pt);
+    if (!codec) {
+        fprintf(stderr, "Error: could not create codec for payload type %d\n", pt);
+        return 1;
+    }
+
+    auto codecInfo = codec->info();
+    fprintf(stderr, "Codec: %s  samplerate=%d  channels=%d  pcmLength=%d  frameTime=%dms\n",
+            codecInfo.mName.c_str(), codecInfo.mSamplerate, codecInfo.mChannels,
+            codecInfo.mPcmLength, codecInfo.mFrameTime);
+
+    // -----------------------------------------------------------------------
+    // 3. Open WAV writer
+    // -----------------------------------------------------------------------
+    Audio::WavFileWriter writer;
+    if (!writer.open(outputPath, codecInfo.mSamplerate, codecInfo.mChannels)) {
+        fprintf(stderr, "Error: could not open WAV file '%s' for writing\n", outputPath);
+        return 1;
+    }
+
+    // -----------------------------------------------------------------------
+    // 4. Decode loop
+    // -----------------------------------------------------------------------
+    std::vector<uint8_t> pcmBuffer(65536);
+    size_t totalDecodedBytes = 0;
+    size_t packetsDecoded = 0;
+    size_t packetsSkipped = 0;
+
+    for (size_t i = 0; i < dump.count(); ++i) {
+        const auto& rawData = dump.rawDataAt(i);
+
+        // Verify it's actually RTP
+        if (!RtpHelper::isRtp(rawData.data(), rawData.size())) {
+            ++packetsSkipped;
+            continue;
+        }
+
+        // Parse RTP to get payload
+        jrtplib::RTPPacket& rtpPacket = dump.packetAt(i);
+
+        // Check payload type matches what we expect
+        int pktPt = rtpPacket.GetPayloadType();
+        if (pktPt != pt) {
+            ++packetsSkipped;
+            continue;
+        }
+
+        uint8_t* payloadData = rtpPacket.GetPayloadData();
+        size_t   payloadLen  = rtpPacket.GetPayloadLength();
+
+        if (!payloadData || payloadLen == 0) {
+            ++packetsSkipped;
+            continue;
+        }
+
+        std::span<const uint8_t> input(payloadData, payloadLen);
+        std::span<uint8_t> output(pcmBuffer.data(), pcmBuffer.size());
+
+        try {
+            auto result = codec->decode(input, output);
+            if (result.mDecoded > 0) {
+                writer.write(pcmBuffer.data(), result.mDecoded);
+                totalDecodedBytes += result.mDecoded;
+                ++packetsDecoded;
+            }
+        } catch (const std::exception& e) {
+            fprintf(stderr, "Warning: decode error at packet %zu: %s\n", i, e.what());
+            ++packetsSkipped;
+        }
+    }
+
+    // -----------------------------------------------------------------------
+    // 5. Close WAV and print summary
+    // -----------------------------------------------------------------------
+    writer.close();
+
+    size_t totalSamples = totalDecodedBytes / (sizeof(int16_t) * codecInfo.mChannels);
+    double durationSec  = (codecInfo.mSamplerate > 0)
+                          ? static_cast<double>(totalSamples) / codecInfo.mSamplerate
+                          : 0.0;
+
+    fprintf(stderr, "\nDone.\n");
+    fprintf(stderr, "  Packets decoded: %zu\n", packetsDecoded);
+    fprintf(stderr, "  Packets skipped: %zu\n", packetsSkipped);
+    fprintf(stderr, "  Decoded PCM:     %zu bytes\n", totalDecodedBytes);
+    fprintf(stderr, "  Duration:        %.3f seconds\n", durationSec);
+    fprintf(stderr, "  Output:          %s\n", outputPath);
+
+    return 0;
+}