fix build errors

* support tts elevenlabs to mod_dub

Signed-off-by: Quan HL <quan.luuhoang8@gmail.com>

* wip

Signed-off-by: Quan HL <quan.luuhoang8@gmail.com>

* wip

* wip

* fix aws race condition when 2 start transcribes are sent at the same instant

* wip

* wip

* wip

* allow queue play on track

* wip

Signed-off-by: Quan HL <quan.luuhoang8@gmail.com>

* revert change for aws transcribe

Signed-off-by: Quan HL <quan.luuhoang8@gmail.com>

* fix type

* wip

* wip

* rename parameters

* rename paramters

* wip

* wip

* wip

* wip

* wip

* bug: there exists scenarios where callback is not defined

* wip

* wip

* revert unintended changes to mod_google_transcribe

* fix bugs w/ streaming tts simplified arg parsing to use freeswitch conventions

---------

Signed-off-by: Quan HL <quan.luuhoang8@gmail.com>
Co-authored-by: Quan HL <quan.luuhoang8@gmail.com>

mod_aws_lex/Makefile.am

@@ -7,4 +7,4 @@ mod_aws_lex_la_CFLAGS   = $(AM_CFLAGS)
 mod_aws_lex_la_CXXFLAGS = $(AM_CXXFLAGS) -std=c++11 -I${switch_srcdir}/libs/aws-sdk-cpp/aws-cpp-sdk-core/include -I${switch_srcdir}/libs/aws-sdk-cpp/aws-cpp-sdk-lexv2-runtime/include -I${switch_srcdir}/libs/aws-sdk-cpp/build/.deps/install/include
 
 mod_aws_lex_la_LIBADD   = $(switch_builddir)/libfreeswitch.la
-mod_aws_lex_la_LDFLAGS  = -avoid-version -module -no-undefined -L${switch_srcdir}/libs/aws-sdk-cpp/build/.deps/install/lib -L${switch_srcdir}/libs/aws-sdk-cpp/build/aws-cpp-sdk-core -L${switch_srcdir}/libs/aws-sdk-cpp/build/aws-cpp-sdk-lexv2-runtime -laws-cpp-sdk-lexv2-runtime -laws-cpp-sdk-core -laws-c-event-stream -laws-checksums -laws-c-common -lpthread -lcurl -lcrypto -lssl -lz
+mod_aws_lex_la_LDFLAGS  = -avoid-version -module -no-undefined -L/usr/local/lib -L${switch_srcdir}/libs/aws-sdk-cpp/build/.deps/install/lib -L${switch_srcdir}/libs/aws-sdk-cpp/build/aws-cpp-sdk-core -L${switch_srcdir}/libs/aws-sdk-cpp/build/aws-cpp-sdk-lexv2-runtime -laws-cpp-sdk-lexv2-runtime -laws-cpp-sdk-core -laws-c-event-stream -laws-checksums -laws-c-common -lpthread -lcurl -lcrypto -lssl -lz

mod_aws_transcribe/Makefile.am

@@ -7,4 +7,4 @@ mod_aws_transcribe_la_CFLAGS   = $(AM_CFLAGS)
 mod_aws_transcribe_la_CXXFLAGS = $(AM_CXXFLAGS) -std=c++11 -I${switch_srcdir}/libs/aws-sdk-cpp/aws-cpp-sdk-core/include -I${switch_srcdir}/libs/aws-sdk-cpp/aws-cpp-sdk-transcribestreaming/include -I${switch_srcdir}/libs/aws-sdk-cpp/build/.deps/install/include
 
 mod_aws_transcribe_la_LIBADD   = $(switch_builddir)/libfreeswitch.la
-mod_aws_transcribe_la_LDFLAGS  = -avoid-version -module -no-undefined -L${switch_srcdir}/libs/aws-sdk-cpp/build/.deps/install/lib -L${switch_srcdir}/libs/aws-sdk-cpp/build/aws-cpp-sdk-core -L${switch_srcdir}/libs/aws-sdk-cpp/build/aws-cpp-sdk-transcribestreaming -laws-cpp-sdk-transcribestreaming -laws-cpp-sdk-core -laws-c-event-stream -laws-checksums -laws-c-common -lpthread -lcurl -lcrypto -lssl -lz
+mod_aws_transcribe_la_LDFLAGS  = -avoid-version -module -no-undefined -L/usr/local/lib -L${switch_srcdir}/libs/aws-sdk-cpp/build/.deps/install/lib -L${switch_srcdir}/libs/aws-sdk-cpp/build/aws-cpp-sdk-core -L${switch_srcdir}/libs/aws-sdk-cpp/build/aws-cpp-sdk-transcribestreaming -laws-cpp-sdk-transcribestreaming -laws-cpp-sdk-core -laws-c-event-stream -laws-checksums -laws-c-common -laws-crt-cpp -lpthread -lcurl -lcrypto -lssl -lz

mod_dub/Makefile.am

@@ -2,9 +2,17 @@ include $(top_srcdir)/build/modmake.rulesam
 MODNAME=mod_dub
 
 mod_LTLIBRARIES = mod_dub.la
-mod_dub_la_SOURCES  = file_loader.cpp audio_downloader.cpp mod_dub.c dub_glue.cpp 
+mod_dub_la_SOURCES  = ap_file.cpp ap_http.cpp track.cpp dub_glue.cpp tts_vendor_parser.cpp mod_dub.c vector_math.cpp mpg_decode.cpp
 mod_dub_la_CFLAGS   = $(AM_CFLAGS)
 mod_dub_la_CXXFLAGS = $(AM_CXXFLAGS) -std=c++17
 
+if USE_AVX2
+mod_dub_la_CXXFLAGS += -mavx2 -DUSE_AVX2
+else
+if USE_SSE2
+mod_dub_la_CXXFLAGS += -msse2 -DUSE_SSE2
+endif
+endif
+
 mod_dub_la_LIBADD   = $(switch_builddir)/libfreeswitch.la
 mod_dub_la_LDFLAGS  = -avoid-version -module -no-undefined -shared `pkg-config --libs boost` -lstdc++ -lmpg123

mod_dub/ap.h

@@ -0,0 +1,41 @@
+#ifndef __AP_H__
+#define __AP_H__
+
+#include <mutex>
+#include <functional>
+#include "common.h"
+
+class AudioProducer {
+public:
+  AudioProducer(
+    std::mutex& mutex,
+    CircularBuffer_t& circularBuffer,
+    int sampleRate
+  ) : _mutex(mutex), _buffer(circularBuffer), _sampleRate(sampleRate), _notified(false), _loop(false), _gain(0) {}
+  virtual ~AudioProducer() {}
+
+  virtual void notifyDone(bool error, const std::string& errorMsg) {
+    if (!_notified) {
+      _notified = true;
+      if (_callback) _callback(error, errorMsg);
+    }
+  }
+  virtual void start(std::function<void(bool, const std::string&)> callback) = 0;
+  virtual void stop() = 0;
+
+  bool isLoopedAudio() const { return _loop; }
+
+protected:
+  std::mutex& _mutex;
+  CircularBuffer_t& _buffer;
+  int _sampleRate;
+  int _gain;
+  bool _loop;
+  std::function<void(bool, const std::string&)> _callback;
+  bool _notified;
+};
+
+
+
+
+#endif

mod_dub/ap_file.cpp

@@ -0,0 +1,196 @@
+#include "switch.h"
+#include "ap_file.h"
+#include "mpg_decode.h"
+
+#define INIT_BUFFER_SIZE (80000)
+#define BUFFER_GROW_SIZE (80000)
+#define BUFFER_THROTTLE_LOW (40000)
+#define BUFFER_THROTTLE_HIGH (160000)
+
+bool AudioProducerFile::initialized = false;
+boost::asio::io_service AudioProducerFile::io_service;
+std::thread AudioProducerFile::worker_thread;
+
+void AudioProducerFile::threadFunc() {      
+  io_service.reset() ;
+  boost::asio::io_service::work work(io_service);
+  
+  switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "file_loader threadFunc - starting\n");
+
+  for(;;) {
+      
+    try {
+      io_service.run() ;
+      break ;
+    }
+    catch( std::exception& e) {
+      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "file_loader threadFunc - Error: %s\n", e.what());
+    }
+  }
+  switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "file_loader threadFunc - ending\n");
+}
+
+void AudioProducerFile::_init() {
+  if (!initialized) {
+    initialized = true;
+    if (mpg123_init() != MPG123_OK) {
+      throw std::runtime_error("AudioProducerFile::AudioProducerFile: failed to initiate MPG123");
+      return ;
+    }
+
+    /* start worker thread */
+    std::thread t(threadFunc) ;
+    worker_thread.swap( t ) ;
+  }
+}
+
+void AudioProducerFile::_deinit() {
+  if (initialized) {
+    initialized = false;
+    io_service.stop();
+    if (worker_thread.joinable()) {
+      worker_thread.join();
+    }
+    mpg123_exit();
+  }
+}
+AudioProducerFile::AudioProducerFile(
+  std::mutex& mutex,
+  CircularBuffer_t& circularBuffer,
+  int sampleRate
+) : AudioProducer(mutex, circularBuffer, sampleRate), _timer(io_service), _mh(nullptr) {
+
+  AudioProducerFile::_init();
+}
+
+AudioProducerFile::~AudioProducerFile() {
+  reset();
+  switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "AudioProducerFile::~AudioProducerFile %p\n", (void *)this);
+}
+
+void AudioProducerFile::queueFileAudio(const std::string& path, int gain, bool loop) {
+    _path = path;
+    _gain = gain;
+    _loop = loop;
+
+    /* we only handle mp3 or r8 files atm */
+    size_t pos = path.find_last_of('.');
+    if (pos == std::string::npos) {
+      throw std::runtime_error("file " + path + " has no extension");
+    }
+    auto filetype = path.substr(pos + 1);
+    if (0 == filetype.compare("mp3")) _type = FILE_TYPE_MP3;
+    else if (0 == filetype.compare("r8")) _type = FILE_TYPE_R8;
+    else throw std::runtime_error("file " + path + " has unsupported extension " + filetype);
+}
+
+void AudioProducerFile::start(std::function<void(bool, const std::string&)> callback) {
+    int mhError = 0;
+
+    _callback = callback;
+
+    /* allocate handle for mpeg decoding */
+    _mh = mpg123_new("auto", &mhError);
+    if (!_mh) {
+      const char *mhErr = mpg123_plain_strerror(mhError);
+      throw std::runtime_error("Error allocating mpg123 handle! " + std::string(mhErr));
+    }
+
+    if (mpg123_open_feed(_mh) != MPG123_OK) throw std::runtime_error("Error mpg123_open_feed!");
+    if (mpg123_format_all(_mh) != MPG123_OK) throw std::runtime_error("Error mpg123_format_all!");
+    if (mpg123_param(_mh, MPG123_FORCE_RATE, _sampleRate, 0) != MPG123_OK) throw std::runtime_error("Error forcing resample to 8k!");
+    if (mpg123_param(_mh, MPG123_FLAGS, MPG123_MONO_MIX, 0) != MPG123_OK) throw std::runtime_error("Error forcing single channel!");
+    
+    _fp = fopen(_path.c_str(), "rb");
+    if (!_fp) throw std::runtime_error("Error opening file " + _path);
+
+    _status = Status_t::STATUS_AWAITING_RESTART;
+
+    /* do the initial read in the worker thread so we don't block here */
+    _timer.expires_from_now(boost::posix_time::millisec(1));
+    _timer.async_wait(boost::bind(&AudioProducerFile::read_cb, this, boost::placeholders::_1));
+}
+
+void AudioProducerFile::read_cb(const boost::system::error_code& error) {
+  if (_status == Status_t::STATUS_STOPPED) {
+    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "read_cb: session gone\n");
+    return;
+  }
+  if (_status == Status_t::STATUS_AWAITING_RESTART) {
+    _status = Status_t::STATUS_IN_PROGRESS;
+  }
+  if (!error) {
+    size_t size = _buffer.size();
+    if (size < BUFFER_THROTTLE_LOW) {
+      std::vector<int16_t> pcm_data;
+      int8_t buf[INIT_BUFFER_SIZE];
+
+      size_t bytesRead = ::fread(buf, sizeof(int8_t), INIT_BUFFER_SIZE, _fp);
+      if (bytesRead <= 0) {
+        if (::feof(_fp)) switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "read_cb: %p eof\n", (void *) this);
+        else if (::ferror(_fp)) switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "read_cb: %p error reading file\n", (void *) this);
+        else switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "read_cb: %p unknown error reading file\n", (void *) this);
+        _status = Status_t::STATUS_COMPLETE;
+        return;
+      }
+
+      if (_type == FileType_t::FILE_TYPE_MP3) pcm_data = convert_mp3_to_linear(_mh, _gain, buf, bytesRead);
+      else pcm_data = std::vector<int16_t>(reinterpret_cast<int16_t*>(buf), reinterpret_cast<int16_t*>(buf) + bytesRead / 2);
+
+      {
+        std::lock_guard<std::mutex> lock(_mutex); 
+
+        // Resize the buffer if necessary
+        size_t capacity = _buffer.capacity();
+        if (capacity - size < pcm_data.size()) {
+          //switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "write_cb %p growing buffer, size now %ld\n", (void *) this, size); 
+          _buffer.set_capacity(size + std::max(pcm_data.size(), (size_t)BUFFER_GROW_SIZE));
+        }
+        
+        /* Push the data into the buffer */
+        _buffer.insert(_buffer.end(), pcm_data.data(), pcm_data.data() + pcm_data.size());
+        //switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "read_cb: %p wrote data, buffer size is now %ld\n", (void *) this, _buffer.size());        
+      }
+
+      if (bytesRead < INIT_BUFFER_SIZE) {
+        _status = Status_t::STATUS_COMPLETE;
+        //switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "read_cb: %u reached end of file, status is %s\n", (void *) this, status2String(_status));
+      }
+    }
+
+    if (_status == Status_t::STATUS_COMPLETE) {
+      cleanup(Status_t::STATUS_COMPLETE);
+    }
+    else {
+      // read more in 2 seconds
+      _timer.expires_from_now(boost::posix_time::millisec(2000));
+      _timer.async_wait(boost::bind(&AudioProducerFile::read_cb, this, boost::placeholders::_1));
+    }
+  } else {
+    cleanup(Status_t::STATUS_FAILED, error.message());
+  }
+}
+
+void AudioProducerFile::stop() {
+  cleanup(Status_t::STATUS_STOPPED, "");
+}
+
+void AudioProducerFile::reset() {
+    if (_fp) {
+    fclose(_fp);
+    _fp = nullptr;
+  }
+  if (_mh) {
+    mpg123_close(_mh);
+    mpg123_delete(_mh);
+    _mh = nullptr;
+  }
+    _timer.cancel();
+  _status = Status_t::STATUS_NONE;
+}
+
+void AudioProducerFile::cleanup(Status_t status, std::string errMsg) {
+  reset();
+  _status = status;
+  notifyDone(status != Status_t::STATUS_COMPLETE, errMsg);
+}

mod_dub/ap_file.h

@@ -0,0 +1,89 @@
+#ifndef __AP_FILE_H__
+#define __AP_FILE_H__
+
+#include <thread>
+#include <boost/asio.hpp>
+#include <boost/bind/bind.hpp>
+
+#include <mpg123.h>
+
+#include "ap.h"
+
+class AudioProducerFile : public AudioProducer {
+public:
+
+  typedef enum
+  {
+    STATUS_NONE = 0,
+    STATUS_FAILED,
+    STATUS_IN_PROGRESS,
+    STATUS_PAUSED,
+    STATUS_COMPLETE,
+    STATUS_AWAITING_RESTART,
+    STATUS_STOPPED
+  } Status_t;
+
+  typedef enum {
+    FILE_TYPE_MP3 = 0,
+    FILE_TYPE_R8
+  } FileType_t;
+
+  const char* status2String(Status_t status)
+  {
+    static const char* statusStrings[] = {
+      "STATUS_NONE",
+      "STATUS_FAILED",
+      "STATUS_IN_PROGRESS",
+      "STATUS_PAUSED",
+      "STATUS_COMPLETE",
+      "STATUS_AWAITING_RESTART",
+      "STATUS_STOPPED"
+    };
+
+    if (status >= 0 && status < sizeof(statusStrings) / sizeof(statusStrings[0]))
+    {
+      return statusStrings[status];
+    }
+    else
+    {
+      return "UNKNOWN_STATUS";
+    }
+  }
+
+  AudioProducerFile(
+    std::mutex& mutex,
+    CircularBuffer_t& circularBuffer,
+    int sampleRate
+  );
+  virtual ~AudioProducerFile();
+  virtual void start(std::function<void(bool, const std::string&)> callback);
+  virtual void stop();
+  void cleanup(Status_t status, std::string errMsg = "");
+  void reset();
+
+  void queueFileAudio(const std::string& path, int gain = 0, bool loop = false);
+
+  void read_cb(const boost::system::error_code& error);
+
+  static bool initialized;
+  static std::thread worker_thread;
+  static boost::asio::io_service io_service;
+  static void threadFunc();
+
+private:
+
+  static void _init();
+  static void _deinit();
+
+  void stop_file_load();
+
+  std::string _path;
+  Status_t _status;
+  FileType_t _type;
+  mpg123_handle *_mh;
+  boost::asio::deadline_timer _timer;
+  FILE* _fp;
+
+};
+
+#endif

mod_dub/ap_http.cpp

@@ -0,0 +1,660 @@
+#include <functional>
+#include <mpg123.h>
+#include "switch.h"
+#include "ap_http.h"
+#include "mpg_decode.h"
+
+#include <boost/asio/ssl.hpp>
+#include <boost/pool/object_pool.hpp>
+#include <boost/bind/bind.hpp>
+#include <boost/tokenizer.hpp>
+#include <boost/algorithm/string.hpp>
+#include <boost/assign/list_of.hpp>
+
+#define BUFFER_GROW_SIZE (80000)
+#define BUFFER_THROTTLE_LOW (40000)
+#define BUFFER_THROTTLE_HIGH (160000)
+
+bool AudioProducerHttp::initialized = false;
+boost::asio::io_service AudioProducerHttp::io_service;
+std::thread AudioProducerHttp::worker_thread;
+GlobalInfo_t AudioProducerHttp::global;
+std::map<curl_socket_t, boost::asio::ip::tcp::socket *> AudioProducerHttp::socket_map;
+boost::asio::deadline_timer AudioProducerHttp::multi_timer(AudioProducerHttp::io_service);
+
+void AudioProducerHttp::threadFunc() {      
+  /* to make sure the event loop doesn't terminate when there is no work to do */
+  io_service.reset() ;
+  boost::asio::io_service::work work(io_service);
+  
+  switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "ap_http threadFunc - starting\n");
+
+  for(;;) {
+      
+    try {
+      io_service.run() ;
+      break ;
+    }
+    catch( std::exception& e) {
+      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "ap_http threadFunc - Error: %s\n", e.what());
+    }
+  }
+  switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "ap_http threadFunc - ending\n");
+}
+
+void AudioProducerHttp::_init() {
+  if (!initialized) {
+    initialized = true;
+    if (mpg123_init() != MPG123_OK) {
+      throw std::runtime_error("AudioProducerFile::AudioProducerFile: failed to initiate MPG123");
+      return ;
+    }
+
+    memset(&global, 0, sizeof(GlobalInfo_t));
+    global.multi = curl_multi_init();
+    if (!global.multi) {
+      throw std::runtime_error("AudioProducerHttp::_init: failed to initiate CURL multi");
+      return ;
+    }
+
+    curl_multi_setopt(global.multi, CURLMOPT_SOCKETFUNCTION, &AudioProducerHttp::sock_cb);
+    curl_multi_setopt(global.multi, CURLMOPT_SOCKETDATA, &global);
+    curl_multi_setopt(global.multi, CURLMOPT_TIMERFUNCTION, &AudioProducerHttp::multi_timer_cb);
+    curl_multi_setopt(global.multi, CURLMOPT_TIMERDATA, &global);
+    curl_multi_setopt(global.multi, CURLMOPT_PIPELINING, CURLPIPE_MULTIPLEX);
+
+    /* start worker thread */
+    std::thread t(threadFunc) ;
+    worker_thread.swap( t ) ;
+  }
+}
+
+void AudioProducerHttp::_deinit() {
+  if (initialized) {
+    initialized = false;
+    io_service.stop();
+    if (worker_thread.joinable()) {
+      worker_thread.join();
+    }
+    /* cleanup curl multi handle*/
+    curl_multi_cleanup(global.multi);
+
+    mpg123_exit();
+  }
+}
+
+AudioProducerHttp::AudioProducerHttp(
+    std::mutex& mutex,
+    CircularBuffer_t& circularBuffer,
+    int sampleRate
+) : AudioProducer(mutex, circularBuffer, sampleRate), _status(Status_t::STATUS_NONE), _mh(nullptr), _easy(nullptr), 
+    _error{0}, _response_code(0), _timer(io_service) {
+
+  AudioProducerHttp::_init();
+}
+
+AudioProducerHttp::~AudioProducerHttp() {
+  reset();
+  switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "AudioProducerFile::~AudioProducerFile %p\n", (void *)this);
+}
+
+void AudioProducerHttp::start(std::function<void(bool, const std::string&)> callback) {
+  int mhError = 0;
+
+  _callback = callback;
+  memset(_error, 0, sizeof(_error));
+
+  /* allocate handle for mpeg decoding */
+  _mh = mpg123_new("auto", &mhError);
+  if (!_mh) {
+    const char *mhErr = mpg123_plain_strerror(mhError);
+    throw std::runtime_error("Error allocating mpg123 handle! " + std::string(mhErr));
+  }
+
+  if (mpg123_open_feed(_mh) != MPG123_OK) throw std::runtime_error("Error mpg123_open_feed!\n");
+  if (mpg123_format_all(_mh) != MPG123_OK) throw std::runtime_error("Error mpg123_format_all!\n");
+  if (mpg123_param(_mh, MPG123_FORCE_RATE, _sampleRate, 0) != MPG123_OK) throw std::runtime_error("Error forcing resample to 8k!\n");
+  if (mpg123_param(_mh, MPG123_FLAGS, MPG123_MONO_MIX, 0) != MPG123_OK) throw std::runtime_error("Error forcing single channel!\n");
+
+  _easy = createEasyHandle();
+  if (!_easy) throw std::runtime_error("Error creating easy handle!\n");
+
+  curl_easy_setopt(_easy, CURLOPT_WRITEFUNCTION, &AudioProducerHttp::static_write_cb);
+  curl_easy_setopt(_easy, CURLOPT_URL, _url.c_str());
+  curl_easy_setopt(_easy, CURLOPT_HTTPGET, _method == HttpMethod_t::HTTP_METHOD_GET ? 1L : 0L);
+  curl_easy_setopt(_easy, CURLOPT_WRITEDATA, this);
+  curl_easy_setopt(_easy, CURLOPT_ERRORBUFFER, _error);
+  curl_easy_setopt(_easy, CURLOPT_PRIVATE, this);
+  curl_easy_setopt(_easy, CURLOPT_VERBOSE, 0L);
+  curl_easy_setopt(_easy, CURLOPT_NOPROGRESS, 1L);
+  curl_easy_setopt(_easy, CURLOPT_HEADERFUNCTION, &AudioProducerHttp::static_header_callback);
+  curl_easy_setopt(_easy, CURLOPT_HEADERDATA, this);
+  
+  /* call this function to get a socket */
+  curl_easy_setopt(_easy, CURLOPT_OPENSOCKETFUNCTION, &AudioProducerHttp::open_socket);
+
+  /* call this function to close a socket */
+  curl_easy_setopt(_easy, CURLOPT_CLOSESOCKETFUNCTION, &AudioProducerHttp::close_socket);
+
+  curl_easy_setopt(_easy, CURLOPT_HTTP_VERSION, CURL_HTTP_VERSION_2_0);
+
+  /* keep the speed down so we don't have to buffer large amounts*/
+  curl_easy_setopt(_easy, CURLOPT_MAX_RECV_SPEED_LARGE, (curl_off_t)31415);
+  /*Add request body*/
+  if (!_body.empty()) curl_easy_setopt(_easy, CURLOPT_POSTFIELDS, _body.c_str());
+
+  /*Add request headers*/
+  struct curl_slist *hdr_list = nullptr;
+  for(const auto& header : _headers) {
+    hdr_list = curl_slist_append(hdr_list, header.c_str());
+  }
+  if (hdr_list) curl_easy_setopt(_easy, CURLOPT_HTTPHEADER, hdr_list);
+
+  _status = Status_t::STATUS_AWAITING_RESTART;
+
+  _timer.expires_from_now(boost::posix_time::millisec(1));
+  _timer.async_wait(boost::bind(&AudioProducerHttp::addCurlHandle, this, boost::placeholders::_1));
+}
+void AudioProducerHttp::queueHttpGetAudio(const std::string& url, int gain, bool loop) {
+  _method = HttpMethod_t::HTTP_METHOD_GET;
+  _url = url;
+  _gain = gain;
+  _loop = loop;
+}
+void AudioProducerHttp::queueHttpPostAudio(const std::string& url, int gain, bool loop) {
+  _method = HttpMethod_t::HTTP_METHOD_POST;
+  _url = url;
+  _gain = gain;
+  _loop = loop;
+}
+void AudioProducerHttp::queueHttpPostAudio(const std::string& url, const std::string& body, std::vector<std::string>& headers, int gain, bool loop) {
+  _method = HttpMethod_t::HTTP_METHOD_POST;
+  _url = url;
+  _body = body;
+  _headers = headers;
+  _gain = gain;
+  _loop = loop;
+}
+
+void AudioProducerHttp::addCurlHandle(const boost::system::error_code& error) {
+  if (_status == Status_t::STATUS_AWAITING_RESTART) {
+    auto rc = curl_multi_add_handle(global.multi, _easy);
+    if (mcode_test("new_conn: curl_multi_add_handle", rc) < 0) {
+      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "AudioProducerHttp::addCurlHandle: Error adding easy handle to multi handle\n");
+    }
+    else {
+        switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "AudioProducerHttp::addCurlHandle retrieving from %s\n", _url.c_str());
+    }
+  }
+}
+
+CURL* AudioProducerHttp::createEasyHandle(void) {
+  CURL* easy = curl_easy_init();
+  if(!easy) {
+    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "curl_easy_init() failed!\n");
+    return nullptr ;
+  }  
+
+  curl_easy_setopt(easy, CURLOPT_FOLLOWLOCATION, 1L);
+  curl_easy_setopt(easy, CURLOPT_USERAGENT, "jambonz/0.8.5");
+
+  // set connect timeout to 3 seconds and no total timeout as files could be large
+  curl_easy_setopt(easy, CURLOPT_CONNECTTIMEOUT_MS, 3000L);
+  curl_easy_setopt(easy, CURLOPT_TIMEOUT, 0L); // no timeout
+
+  return easy ;    
+}
+
+size_t AudioProducerHttp::static_write_cb(char* ptr, size_t size, size_t nmemb, void* userdata) {
+  return static_cast<AudioProducerHttp*>(userdata)->write_cb(ptr, size, nmemb);
+}
+
+size_t AudioProducerHttp::write_cb(void *ptr, size_t size, size_t nmemb) {
+  int8_t *data = (int8_t *) ptr;
+  size_t bytes_received = size * nmemb;
+  std::vector<int16_t> pcm_data;
+  if (_status == Status_t::STATUS_STOPPING || _status == Status_t::STATUS_STOPPED) {
+    _timer.cancel();
+    //switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, 
+    //  "AudioProducerHttp::write_cb: aborting transfer, status %s, mutex %p, buffer %p\n", status2String(_status));
+    /* this will abort the transfer */
+    return 0;
+  }
+  if (_response_code > 0 && _response_code != 200) {
+    std::string body((char *) ptr, bytes_received);
+    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "AudioProducerHttp::write_cb: received body %s\n", body.c_str());
+    _err_msg = body;
+    _status = Status_t::STATUS_FAILED;
+    return 0;
+  }
+
+  /* throttle after reaching high water mark */
+  size_t bufSize = _buffer.size();
+  if (bufSize > BUFFER_THROTTLE_HIGH) {
+    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "AudioProducerHttp::write_cb: throttling download, buffer size is %ld\n", _buffer.size());
+    
+    // check back in 2 seconds
+    _timer.expires_from_now(boost::posix_time::millisec(2000));
+    _timer.async_wait(boost::bind(&AudioProducerHttp::throttling_cb, this, boost::placeholders::_1));
+
+    _status = Status_t::STATUS_DOWNLOAD_PAUSED;
+    return CURL_WRITEFUNC_PAUSE;
+  }
+
+  pcm_data = convert_mp3_to_linear(_mh, _gain, data, bytes_received);
+  size_t samples = pcm_data.size();
+  std::lock_guard<std::mutex> lock(_mutex); 
+
+  // Resize the buffer if necessary
+  if (_buffer.capacity() - bufSize < samples) {
+    //switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "AudioProducerHttp::write_cb growing buffer, size now %ld\n", _buffer.size()); 
+
+    //TODO: if buffer exceeds some max size, return CURL_WRITEFUNC_ERROR to abort the transfer
+    _buffer.set_capacity(_buffer.size() + std::max(samples, (size_t)BUFFER_GROW_SIZE));
+  }
+  //switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "AudioProducerHttp::write_cb: writing %ld samples to buffer\n", pcm_data.size());
+  
+  /* Push the data into the buffer */
+  _buffer.insert(_buffer.end(), pcm_data.data(), pcm_data.data() + pcm_data.size());
+  return bytes_received;
+
+}
+
+size_t AudioProducerHttp::static_header_callback(char *buffer, size_t size, size_t nitems, void* userdata) {
+  return static_cast<AudioProducerHttp*>(userdata)->header_callback(buffer, size, nitems);
+}
+
+size_t AudioProducerHttp::header_callback(char *buffer, size_t size, size_t nitems) {
+  size_t bytes_received = size * nitems;
+  const std::string prefix = "HTTP/";
+  std::string header, value;
+  std::string input(buffer, bytes_received);
+  if (parseHeader(input, header, value)) {
+    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "header_callback: %s with value %s\n", header.c_str(), value.c_str());
+  }
+  else {
+    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "header_callback: %s\n", input.c_str());
+    if (input.rfind(prefix, 0) == 0) {
+      try {
+        _response_code = extract_response_code(input);
+        switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "header_callback: parsed response code: %ld\n", _response_code);
+      } catch (const std::invalid_argument& e) {
+        switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "header_callback: invalid response code %s\n", input.substr(prefix.length()).c_str());
+      }
+    }
+  }
+  return bytes_received;
+}
+
+void AudioProducerHttp::throttling_cb(const boost::system::error_code& error) {
+  if (_status == Status_t::STATUS_STOPPING) {
+    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "throttling_cb: session gone, resume download so we can complete\n");
+    curl_easy_pause(_easy, CURLPAUSE_CONT);
+    return;
+  }
+  //switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "throttling_cb: status is %s\n", status2String(_status));
+
+  if (!error) {
+    auto size = _buffer.size();
+
+    //switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "throttling_cb: size is now %ld\n", size);
+    if (size < BUFFER_THROTTLE_LOW) {
+      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "throttling_cb: resuming download\n");
+      curl_easy_pause(_easy, CURLPAUSE_CONT);
+     return;
+    }
+
+    // check back in 2 seconds
+    _timer.expires_from_now(boost::posix_time::millisec(2000));
+    _timer.async_wait(boost::bind(&AudioProducerHttp::throttling_cb, this, boost::placeholders::_1));
+
+  } else if (125 == error.value()) {
+    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "throttling_cb: timer canceled\n");
+  }
+  else {
+    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "throttling_cb: error (%d): %s\n", error.value(), error.message().c_str());
+  }
+}
+
+int AudioProducerHttp::sock_cb(CURL *e, curl_socket_t s, int what, void *cbp, void *sockp) {
+  GlobalInfo_t *g = &global;
+
+  int *actionp = (int *) sockp;
+  static const char *whatstr[] = { "none", "IN", "OUT", "INOUT", "REMOVE"};
+
+  if(what == CURL_POLL_REMOVE) {
+    *actionp = what;
+  }
+  else {
+    if(!actionp) {
+      addsock(s, e, what, g);
+    }
+    else {
+      setsock(actionp, s, e, what, *actionp, g);
+    }
+  }
+  return 0;  
+}
+
+void AudioProducerHttp::timer_cb(const boost::system::error_code & error, GlobalInfo_t *g)
+{
+  //switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "timer_cb\n");
+
+  if(!error) {
+    CURLMcode rc = curl_multi_socket_action(g->multi, CURL_SOCKET_TIMEOUT, 0, &g->still_running);
+    mcode_test("timer_cb: curl_multi_socket_action", rc);
+    check_multi_info(g);
+  }
+}
+
+int AudioProducerHttp::multi_timer_cb(CURLM *multi, long timeout_ms, GlobalInfo_t *g) {
+
+  /* cancel running timer */
+  multi_timer.cancel();
+
+  if(timeout_ms >= 0) {
+    // from libcurl 7.88.1-10+deb12u4 does not allow call curl_multi_socket_action or curl_multi_perform in curl_multi callback directly
+    multi_timer.expires_from_now(boost::posix_time::millisec(timeout_ms ? timeout_ms : 1));
+    multi_timer.async_wait(boost::bind(&timer_cb, boost::placeholders::_1, g));
+  }
+
+  return 0;
+}
+
+curl_socket_t AudioProducerHttp::open_socket(void *clientp, curlsocktype purpose, struct curl_sockaddr *address) {
+  curl_socket_t sockfd = CURL_SOCKET_BAD;
+
+  /* restrict to IPv4 */
+  if(purpose == CURLSOCKTYPE_IPCXN && address->family == AF_INET) {
+    /* create a tcp socket object */
+    boost::asio::ip::tcp::socket *tcp_socket = new boost::asio::ip::tcp::socket(io_service);
+
+    /* open it and get the native handle*/
+    boost::system::error_code ec;
+    tcp_socket->open(boost::asio::ip::tcp::v4(), ec);
+
+    if(ec) {
+      /* An error occurred */
+      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Couldn't open socket [%ld][%s]\n", ec, ec.message().c_str());
+    }
+    else {
+      sockfd = tcp_socket->native_handle();
+
+      /* save it for monitoring */
+      socket_map.insert(std::pair<curl_socket_t, boost::asio::ip::tcp::socket *>(sockfd, tcp_socket));
+    }
+  }
+  return sockfd;
+}
+
+int AudioProducerHttp::close_socket(void *clientp, curl_socket_t item) {
+  std::map<curl_socket_t, boost::asio::ip::tcp::socket *>::iterator it = socket_map.find(item);
+  if(it != socket_map.end()) {
+    delete it->second;
+    socket_map.erase(it);
+  }
+  return 0;
+}
+
+int AudioProducerHttp::mcode_test(const char *where, CURLMcode code) {
+  if(CURLM_OK != code) {
+    const char *s;
+    switch(code) {
+    case CURLM_CALL_MULTI_PERFORM:
+      s = "CURLM_CALL_MULTI_PERFORM";
+      break;
+    case CURLM_BAD_HANDLE:
+      s = "CURLM_BAD_HANDLE";
+      break;
+    case CURLM_BAD_EASY_HANDLE:
+      s = "CURLM_BAD_EASY_HANDLE";
+      break;
+    case CURLM_OUT_OF_MEMORY:
+      s = "CURLM_OUT_OF_MEMORY";
+      break;
+    case CURLM_INTERNAL_ERROR:
+      s = "CURLM_INTERNAL_ERROR";
+      break;
+    case CURLM_UNKNOWN_OPTION:
+      s = "CURLM_UNKNOWN_OPTION";
+      break;
+    case CURLM_LAST:
+      s = "CURLM_LAST";
+      break;
+    default:
+      s = "CURLM_unknown";
+      break;
+    case CURLM_BAD_SOCKET:
+      s = "CURLM_BAD_SOCKET";
+      break;
+    }
+    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "mcode_test ERROR: %s returns %s:%d\n", where, s, code);
+
+    return -1;
+  }
+  return 0 ;
+}
+
+bool AudioProducerHttp::parseHeader(const std::string& str, std::string& header, std::string& value) {
+  std::vector<std::string> parts;
+  boost::split(parts, str, boost::is_any_of(":"), boost::token_compress_on);
+
+  if (parts.size() != 2)
+      return false;
+
+  header = boost::trim_copy(parts[0]);
+  value = boost::trim_copy(parts[1]);
+  return true;
+}
+
+int AudioProducerHttp::extract_response_code(const std::string& input) {
+  std::size_t space_pos = input.find(' ');
+  if (space_pos == std::string::npos) {
+    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Invalid HTTP response format %s\n", input.c_str());
+    return 0;
+  }
+
+  std::size_t code_start_pos = space_pos + 1;
+  std::size_t code_end_pos = input.find(' ', code_start_pos);
+  if (code_end_pos == std::string::npos) {
+    code_end_pos = input.length();
+  }
+
+  std::string code_str = input.substr(code_start_pos, code_end_pos - code_start_pos);
+  int response_code = std::stoi(code_str);
+  return response_code;
+}
+
+void AudioProducerHttp::setsock(int *fdp, curl_socket_t s, CURL *e, int act, int oldact, GlobalInfo_t *g) {
+  std::map<curl_socket_t, boost::asio::ip::tcp::socket *>::iterator it = socket_map.find(s);
+
+  if(it == socket_map.end()) {
+    return;
+  }
+
+  boost::asio::ip::tcp::socket * tcp_socket = it->second;
+
+  *fdp = act;
+
+  if(act == CURL_POLL_IN) {
+    if(oldact != CURL_POLL_IN && oldact != CURL_POLL_INOUT) {
+      tcp_socket->async_read_some(boost::asio::null_buffers(),
+                                  boost::bind(&event_cb, g, s,
+                                  CURL_POLL_IN, boost::placeholders::_1, fdp));
+    }
+  }
+  else if(act == CURL_POLL_OUT) {
+    if(oldact != CURL_POLL_OUT && oldact != CURL_POLL_INOUT) {
+      tcp_socket->async_write_some(boost::asio::null_buffers(),
+                                    boost::bind(&event_cb, g, s,
+                                    CURL_POLL_OUT, boost::placeholders::_1, fdp));
+    }
+  }
+  else if(act == CURL_POLL_INOUT) {
+    if(oldact != CURL_POLL_IN && oldact != CURL_POLL_INOUT) {
+      tcp_socket->async_read_some(boost::asio::null_buffers(),
+                                  boost::bind(&event_cb, g, s,
+                                  CURL_POLL_IN, boost::placeholders::_1, fdp));
+    }
+    if(oldact != CURL_POLL_OUT && oldact != CURL_POLL_INOUT) {
+      tcp_socket->async_write_some(boost::asio::null_buffers(),
+                                    boost::bind(&event_cb, g, s,
+                                    CURL_POLL_OUT, boost::placeholders::_1, fdp));
+    }
+  }
+}
+
+/* Called by asio when there is an action on a socket */
+void AudioProducerHttp::event_cb(GlobalInfo_t *g, curl_socket_t s, int action, const boost::system::error_code & error, int *fdp) {
+  int f = *fdp;
+
+
+  // Socket already POOL REMOVED.
+  if (f == CURL_POLL_REMOVE) {
+    remsock(fdp, g);
+    return;
+  }
+
+  if(socket_map.find(s) == socket_map.end()) {
+    return;
+  }
+
+  /* make sure the event matches what are wanted */
+  if(f == action || f == CURL_POLL_INOUT) {
+    if(error) {
+      action = CURL_CSELECT_ERR;
+    }
+    CURLMcode rc = curl_multi_socket_action(g->multi, s, action, &g->still_running);
+
+    mcode_test("event_cb: curl_multi_socket_action", rc);
+    check_multi_info(g);
+
+    if(g->still_running <= 0) {
+      multi_timer.cancel();
+    }
+
+    /* keep on watching.
+      * the socket may have been closed and/or fdp may have been changed
+      * in curl_multi_socket_action(), so check them both */
+    if(!error && socket_map.find(s) != socket_map.end() &&
+        (f == action || f == CURL_POLL_INOUT)) {
+      boost::asio::ip::tcp::socket *tcp_socket = socket_map.find(s)->second;
+
+      if(action == CURL_POLL_IN) {
+        tcp_socket->async_read_some(boost::asio::null_buffers(),
+                                    boost::bind(&event_cb, g, s,
+                                                action, boost::placeholders::_1, fdp));
+      }
+      if(action == CURL_POLL_OUT) {
+        tcp_socket->async_write_some(boost::asio::null_buffers(),
+                                      boost::bind(&event_cb, g, s,
+                                                  action, boost::placeholders::_1, fdp));
+      } 
+    }
+  }
+}
+
+void AudioProducerHttp::remsock(int *f, GlobalInfo_t *g) {
+  if(f) {
+    free(f);
+    f = NULL;
+  }
+}
+
+void AudioProducerHttp::addsock(curl_socket_t s, CURL *easy, int action, GlobalInfo_t *g) {
+  /* fdp is used to store current action */
+  int *fdp = (int *) calloc(sizeof(int), 1);
+
+  setsock(fdp, s, easy, action, 0, g);
+  curl_multi_assign(g->multi, s, fdp);
+}
+
+/* Check for completed transfers, and remove their easy handles */
+void AudioProducerHttp::check_multi_info(GlobalInfo_t *g) {
+  CURLMsg *msg;
+  int msgs_left;
+  AudioProducerHttp *ap;
+  CURL *easy;
+  CURLcode res;
+  
+  while((msg = curl_multi_info_read(g->multi, &msgs_left))) {
+    if(msg->msg == CURLMSG_DONE) {
+      long response_code;
+      double namelookup=0, connect=0, total=0 ;
+      char *ct = NULL ;
+
+      easy = msg->easy_handle;
+      res = msg->data.result;
+      curl_easy_getinfo(easy, CURLINFO_PRIVATE, &ap);
+      curl_easy_getinfo(easy, CURLINFO_RESPONSE_CODE, &response_code);
+      curl_easy_getinfo(easy, CURLINFO_CONTENT_TYPE, &ct);
+
+      curl_easy_getinfo(easy, CURLINFO_NAMELOOKUP_TIME, &namelookup);
+      curl_easy_getinfo(easy, CURLINFO_CONNECT_TIME, &connect);
+      curl_easy_getinfo(easy, CURLINFO_TOTAL_TIME, &total);
+
+      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "curl done, response code %d, status %s\n", response_code, status2String(ap->getStatus()));
+
+      bool restart = ap->isLoopedAudio() && ap->getStatus() != Status_t::STATUS_STOPPING && response_code == 200;
+      if (restart) {
+        switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "restarting looped audio\n");
+        ap->getTimer().expires_from_now(boost::posix_time::millisec(1000));
+        ap->getTimer().async_wait(boost::bind(&AudioProducerHttp::static_restart_cb, boost::placeholders::_1, ap));
+      }
+      else {
+        ap->cleanup(Status_t::STATUS_DOWNLOAD_COMPLETE, (int) response_code);
+      }
+    }
+  }
+}
+
+void AudioProducerHttp::static_restart_cb(const boost::system::error_code& error, void* userdata) {
+  static_cast<AudioProducerHttp*>(userdata)->restart_cb(error);
+}
+
+void AudioProducerHttp::restart_cb(const boost::system::error_code& error) {
+  if (_status == Status_t::STATUS_STOPPING) {
+    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "restart_cb: session gone\n");
+    return;
+  }
+  if (!error) {
+    reset();
+    start(_callback);
+  }
+  else if (error.value() == boost::asio::error::operation_aborted) {
+    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "restart_cb: %s, cancelling retrieve\n", error.message().c_str());
+  }
+  else {
+    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "restart_cb: error (%d): %s\n", error.value(), error.message().c_str());
+  }
+}
+
+
+void AudioProducerHttp::stop() {
+  cleanup(Status_t::STATUS_STOPPED, 0);
+  return;
+}
+
+void AudioProducerHttp::reset() {
+  if (_easy) {
+    curl_multi_remove_handle(global.multi, _easy);
+    curl_easy_cleanup(_easy);
+    _easy = nullptr;
+  }
+  if (_mh) {
+    mpg123_close(_mh);
+    mpg123_delete(_mh);
+    _mh = nullptr;
+  }
+  _err_msg.clear();
+  _response_code = 0;
+  _timer.cancel();
+  _status = Status_t::STATUS_NONE;
+}
+
+void AudioProducerHttp::cleanup(Status_t status, int response_code) {
+  std::string errMsg = response_code > 200 ? "http response: " + std::to_string(response_code) : "";
+  reset();
+  _status = status;
+  notifyDone(!errMsg.empty(), errMsg);
+}

mod_dub/ap_http.h

@@ -0,0 +1,134 @@
+#ifndef __AP_HTTP_H__
+#define __AP_HTTP_H__
+
+#include "ap.h"
+#include <curl/curl.h>
+#include <mpg123.h>
+#include <boost/asio.hpp>
+
+
+typedef struct
+{
+  CURLM *multi;
+  int still_running;
+} GlobalInfo_t;
+
+class AudioProducerHttp : public AudioProducer {
+public:
+
+  typedef enum
+  {
+    STATUS_NONE = 0,
+    STATUS_FAILED,
+    STATUS_DOWNLOAD_IN_PROGRESS,
+    STATUS_DOWNLOAD_PAUSED,
+    STATUS_DOWNLOAD_COMPLETE,
+    STATUS_AWAITING_RESTART,
+    STATUS_STOPPING,
+    STATUS_STOPPED
+  } Status_t;
+
+  typedef enum {
+    HTTP_METHOD_GET = 0,
+    HTTP_METHOD_POST
+  } HttpMethod_t;
+
+  static const char* status2String(Status_t status) {
+    static const char* statusStrings[] = {
+      "STATUS_NONE",
+      "STATUS_FAILED",
+      "STATUS_DOWNLOAD_IN_PROGRESS",
+      "STATUS_DOWNLOAD_PAUSED",
+      "STATUS_DOWNLOAD_COMPLETE",
+      "STATUS_AWAITING_RESTART",
+      "STATUS_STOPPED"
+    };
+
+    if (status >= 0 && status < sizeof(statusStrings) / sizeof(statusStrings[0]))
+    {
+      return statusStrings[status];
+    }
+    else
+    {
+      return "UNKNOWN_STATUS";
+    }
+  }
+  AudioProducerHttp(
+    std::mutex& mutex,
+    CircularBuffer_t& circularBuffer,
+    int sampleRate
+  );
+  virtual ~AudioProducerHttp();
+
+  virtual void start(std::function<void(bool, const std::string&)> callback);
+  void addCurlHandle(const boost::system::error_code& error);
+  virtual void stop();
+  void cleanup(Status_t status, int response_code);
+  void reset();
+
+  void queueHttpGetAudio(const std::string& url, int gain = 0, bool loop = false);
+  void queueHttpPostAudio(const std::string& url, int gain = 0, bool loop = false);
+  void queueHttpPostAudio(const std::string& url, const std::string& body, std::vector<std::string>& headers, int gain = 0, bool loop = false);
+
+  Status_t getStatus() const { return _status; }
+  void setStatus(Status_t status) { _status = status; }
+
+  boost::asio::deadline_timer& getTimer() { return _timer; }
+  
+  static bool initialized;
+  static std::thread worker_thread;
+  static boost::asio::io_service io_service;
+  static void threadFunc();
+  static std::map<curl_socket_t, boost::asio::ip::tcp::socket *> socket_map;
+  static boost::asio::deadline_timer multi_timer;
+
+
+  static GlobalInfo_t global;
+
+  static int sock_cb(CURL *e, curl_socket_t s, int what, void *cbp, void *sockp);
+  static int multi_timer_cb(CURLM *multi, long timeout_ms, GlobalInfo_t *g);
+  static void timer_cb(const boost::system::error_code & error, GlobalInfo_t *g);
+  static void addsock(curl_socket_t s, CURL *easy, int action, GlobalInfo_t *g);
+  static void setsock(int *fdp, curl_socket_t s, CURL *e, int act, int oldact, GlobalInfo_t *g);
+  static int mcode_test(const char *where, CURLMcode code);
+  static void check_multi_info(GlobalInfo_t *g);
+  static void event_cb(GlobalInfo_t *g, curl_socket_t s, int action, const boost::system::error_code & error, int *fdp);
+  static void remsock(int *f, GlobalInfo_t *g);
+  
+  static size_t static_write_cb(char* ptr, size_t size, size_t nmemb, void* userdata);
+  size_t write_cb(void *ptr, size_t size, size_t nmemb);
+
+  static size_t static_header_callback(char *buffer, size_t size, size_t nitems, void* userdata);
+  size_t header_callback(char *buffer, size_t size, size_t nitems);
+
+  void throttling_cb(const boost::system::error_code& error);
+
+  static void static_restart_cb(const boost::system::error_code& error, void* userdata);
+  void restart_cb(const boost::system::error_code& error);
+
+  static int close_socket(void *clientp, curl_socket_t item);
+  static curl_socket_t open_socket(void *clientp, curlsocktype purpose, struct curl_sockaddr *address);
+
+private:
+
+  static void _init();
+  static void _deinit();
+
+  CURL* createEasyHandle();
+  bool parseHeader(const std::string& str, std::string& header, std::string& value);
+  int extract_response_code(const std::string& input);
+
+  HttpMethod_t _method;
+  std::string _url;
+  std::string _body;
+  std::vector<std::string> _headers;
+  Status_t _status;
+  mpg123_handle *_mh;
+  CURL *_easy;
+  char _error[CURL_ERROR_SIZE]; // curl error buffer
+  std::string _err_msg;
+  int _response_code;
+  boost::asio::deadline_timer _timer;
+};
+
+#endif

mod_dub/audio_downloader.cpp

@@ -1,885 +0,0 @@
-#include "audio_downloader.h"
-
-#include <boost/thread.hpp>
-#include <boost/asio.hpp>
-#include <boost/asio/ssl.hpp>
-#include <boost/pool/object_pool.hpp>
-#include <boost/bind/bind.hpp>
-#include <boost/tokenizer.hpp>
-#include <boost/foreach.hpp>
-#include <boost/asio.hpp>
-#include <boost/assign/list_of.hpp>
-#include <boost/algorithm/string.hpp>
-
-#include <map>
-
-#include <mpg123.h>
-
-#include <curl/curl.h>
-
-#define BUFFER_GROW_SIZE (80000)
-#define BUFFER_THROTTLE_LOW (40000)
-#define BUFFER_THROTTLE_HIGH (160000)
-
-static uint16_t currDownloadId = 0;
-
-typedef struct
-{
-  CURLM *multi;
-  int still_running;
-} GlobalInfo_t;
-static GlobalInfo_t global;
-
-typedef enum
-{
-  STATUS_NONE = 0,
-  STATUS_FAILED,
-  STATUS_DOWNLOAD_IN_PROGRESS,
-  STATUS_DOWNLOAD_PAUSED,
-  STATUS_DOWNLOAD_COMPLETE,
-  STATUS_AWAITING_RESTART,
-  STATUS_STOPPING,
-  STATUS_STOPPED
-} Status_t;
-
-static const char* status2String(Status_t status)
-{
-  static const char* statusStrings[] = {
-    "STATUS_NONE",
-    "STATUS_FAILED",
-    "STATUS_DOWNLOAD_IN_PROGRESS",
-    "STATUS_DOWNLOAD_PAUSED",
-    "STATUS_DOWNLOAD_COMPLETE",
-    "STATUS_AWAITING_RESTART",
-    "STATUS_STOPPING",
-    "STATUS_STOPPED"
-  };
-
-  if (status >= 0 && status < sizeof(statusStrings) / sizeof(statusStrings[0]))
-  {
-    return statusStrings[status];
-  }
-  else
-  {
-    return "UNKNOWN_STATUS";
-  }
-}
-
-typedef struct
-{
-  GlobalInfo_t *global;
-  CURL *easy;
-  switch_mutex_t* mutex;
-  CircularBuffer_t* buffer;
-  mpg123_handle *mh;
-  char error[CURL_ERROR_SIZE]; // curl error buffer
-  char *err_msg; // http server error message
-  char* url;
-  bool loop;
-  int rate;
-  boost::asio::deadline_timer *timer;
-  Status_t status;
-  downloadId_t id;
-  int response_code;
-  int gain;
-} ConnInfo_t;
-
-typedef std::map<int32_t, ConnInfo_t *> Id2ConnMap_t;
-static Id2ConnMap_t id2ConnMap;
-
-static boost::object_pool<ConnInfo_t> pool ;
-static std::map<curl_socket_t, boost::asio::ip::tcp::socket *> socket_map;
-static boost::asio::io_service io_service;
-static boost::asio::deadline_timer timer(io_service);
-static std::string fullDirPath;
-static std::thread worker_thread;
-
-/* forward declarations */
-static ConnInfo_t* createDownloader(const char *url, int rate, int loop, int gain, mpg123_handle *mhm, switch_mutex_t *mutex, CircularBuffer_t *buffer);
-static CURL* createEasyHandle(void);
-static void destroyConnection(ConnInfo_t *conn);
-static void check_multi_info(GlobalInfo_t *g) ;
-static int mcode_test(const char *where, CURLMcode code);
-static void event_cb(GlobalInfo_t *g, curl_socket_t s, int action, const boost::system::error_code & error, int *fdp);
-static void setsock(int *fdp, curl_socket_t s, CURL *e, int act, int oldact, GlobalInfo_t *g);
-static void addsock(curl_socket_t s, CURL *easy, int action, GlobalInfo_t *g);
-static int sock_cb(CURL *e, curl_socket_t s, int what, void *cbp, void *sockp);
-static void threadFunc();
-static void timer_cb(const boost::system::error_code & error, GlobalInfo_t *g);
-static int multi_timer_cb(CURLM *multi, long timeout_ms, GlobalInfo_t *g);
-static std::vector<int16_t> convert_mp3_to_linear(ConnInfo_t *conn, int8_t *data, size_t len);
-static void throttling_cb(const boost::system::error_code& error, ConnInfo_t* conn) ;
-static void restart_cb(const boost::system::error_code& error, ConnInfo_t* conn);
-static size_t write_cb(void *ptr, size_t size, size_t nmemb, ConnInfo_t *conn);
-static bool parseHeader(const std::string& str, std::string& header, std::string& value) ;
-static int extract_response_code(const std::string& input) ;
-static size_t header_callback(char *buffer, size_t size, size_t nitems, ConnInfo_t *conn);
-static curl_socket_t opensocket(void *clientp, curlsocktype purpose, struct curl_sockaddr *address);
-static int close_socket(void *clientp, curl_socket_t item);
-
-/* apis */
-extern "C" {
-
-  switch_status_t init_audio_downloader() {
-    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "init_audio_downloader loading..\n");
-    memset(&global, 0, sizeof(GlobalInfo_t));
-    global.multi = curl_multi_init();
-
-    if (!global.multi) {
-      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "init_audio_downloader curl_multi_init() failed, exiting!\n");
-      return SWITCH_STATUS_FALSE;
-    }
-
-    curl_multi_setopt(global.multi, CURLMOPT_SOCKETFUNCTION, sock_cb);
-    curl_multi_setopt(global.multi, CURLMOPT_SOCKETDATA, &global);
-    curl_multi_setopt(global.multi, CURLMOPT_TIMERFUNCTION, multi_timer_cb);
-    curl_multi_setopt(global.multi, CURLMOPT_TIMERDATA, &global);
-    curl_multi_setopt(global.multi, CURLMOPT_PIPELINING, CURLPIPE_MULTIPLEX);
-
-    if (mpg123_init() != MPG123_OK) {
-      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "init_audio_downloader: failed to initiate MPG123");
-      return SWITCH_STATUS_FALSE;
-    }
-
-    /* start worker thread */
-    std::thread t(threadFunc) ;
-    worker_thread.swap( t ) ;
-
-    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "init_audio_downloader: loaded\n");
-
-    return SWITCH_STATUS_SUCCESS;
-
-  }
-
-  switch_status_t deinit_audio_downloader() {
-    /* stop the ASIO IO service */
-    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "deinit_audio_downloader: stopping io service\n");
-    io_service.stop();
-
-    /* Join the worker thread */
-    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "deinit_audio_downloader: wait for worker thread to complete\n");
-    if (worker_thread.joinable()) {
-        worker_thread.join();
-    }
-
-    /* cleanup curl multi handle*/
-    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "deinit_audio_downloader: release curl multi\n");
-    curl_multi_cleanup(global.multi);
-    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "deinit_audio_downloader: completed\n");
-
-    mpg123_exit();
-
-    return SWITCH_STATUS_SUCCESS;
-  }
-
-  downloadId_t start_audio_download(const char* url, int rate, int loop, int gain, switch_mutex_t* mutex, CircularBuffer_t* buffer) {
-    int mhError = 0;
-
-    /* allocate handle for mpeg decoding */
-    mpg123_handle *mh = mpg123_new("auto", &mhError);
-    if (!mh) {
-      const char *mhErr = mpg123_plain_strerror(mhError);
-      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error allocating mpg123 handle! %s\n", switch_str_nil(mhErr));
-      return INVALID_DOWNLOAD_ID;
-    }
-
-    if (mpg123_open_feed(mh) != MPG123_OK) {
-      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error mpg123_open_feed!\n");
-      return INVALID_DOWNLOAD_ID;
-    }
-
-    if (mpg123_format_all(mh) != MPG123_OK) {
-      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error mpg123_format_all!\n");
-      return INVALID_DOWNLOAD_ID;
-    }
-
-    if (mpg123_param(mh, MPG123_FORCE_RATE, rate, 0) != MPG123_OK) {
-      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error forcing resample to 8k!\n");
-      return INVALID_DOWNLOAD_ID;
-    }
-
-    if (mpg123_param(mh, MPG123_FLAGS, MPG123_MONO_MIX, 0) != MPG123_OK) {
-      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error forcing single channel!\n");
-      return INVALID_DOWNLOAD_ID;
-    }
-
-    ConnInfo_t* conn = createDownloader(url, rate, loop, gain, mh, mutex, buffer);
-    if (!conn) {
-      return INVALID_DOWNLOAD_ID;
-    }
-
-    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, 
-      "start_audio_download: starting download %d\n", conn->id);
-
-
-    return conn->id;
-  }
-
-  switch_status_t stop_audio_download(int id) {
-    auto it = id2ConnMap.find(id);
-    if (it == id2ConnMap.end()) {
-      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "stop_audio_download: id %d has already completed\n", id);
-      return SWITCH_STATUS_FALSE;
-    }
-    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, 
-      "stop_audio_download: stopping download %d, status %s\n", id, status2String(it->second->status));
-
-    ConnInfo_t *conn = it->second;
-    auto status = conn->status;
-
-    /* past this point I shall not access either the mutex or the buffer provided */
-    conn->mutex = nullptr;
-    conn->buffer = nullptr;
-
-    /* if download is in progress set status to cancel it during next call back */
-    if (status == Status_t::STATUS_DOWNLOAD_PAUSED) {
-      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "stop_audio_download: resuming download %d so we can cancel it\n", id);
-      conn->status = Status_t::STATUS_STOPPING;
-      curl_easy_pause(conn->easy, CURLPAUSE_CONT);
-    }
-    if (status != Status_t::STATUS_DOWNLOAD_IN_PROGRESS) {
-      destroyConnection(conn);
-    }
-    conn->status = Status_t::STATUS_STOPPING;
-    return SWITCH_STATUS_SUCCESS;
-  }
-}
-
-/* internal */
-ConnInfo_t* createDownloader(const char *url, int rate, int loop, int gain, mpg123_handle *mh, switch_mutex_t *mutex, CircularBuffer_t *buffer) {
-  ConnInfo_t *conn = pool.malloc() ;
-  CURL* easy = createEasyHandle();
-
-  if (!easy || !conn) {
-    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "createDownloader: failed to allocate memory\n");
-    return nullptr;
-  }
-
-  memset(conn, 0, sizeof(ConnInfo_t));
-  conn->easy = easy;
-  conn->mutex = mutex;
-  conn->buffer = buffer;
-  conn->mh = mh;
-  conn->loop = loop;
-  conn->gain = gain;
-  conn->rate = rate;
-  conn->url = strdup(url);
-  conn->global = &global;
-  conn->status = Status_t::STATUS_NONE; 
-  conn->timer = new boost::asio::deadline_timer(io_service);
-
-  downloadId_t id = ++currDownloadId;
-  if (id == 0) id++;
-
-  id2ConnMap[id] = conn;
-  conn->id = id;
-
-  curl_easy_setopt(easy, CURLOPT_URL, url);
-  curl_easy_setopt(easy, CURLOPT_HTTPGET, 1L);
-  curl_easy_setopt(easy, CURLOPT_WRITEFUNCTION, write_cb);
-  curl_easy_setopt(easy, CURLOPT_WRITEDATA, conn);
-  curl_easy_setopt(easy, CURLOPT_ERRORBUFFER, conn->error);
-  curl_easy_setopt(easy, CURLOPT_PRIVATE, conn);
-  curl_easy_setopt(easy, CURLOPT_VERBOSE, 0L);
-  curl_easy_setopt(easy, CURLOPT_NOPROGRESS, 1L);
-  curl_easy_setopt(easy, CURLOPT_HEADERFUNCTION, header_callback);
-  curl_easy_setopt(easy, CURLOPT_HEADERDATA, conn);
-  
-  /* call this function to get a socket */
-  curl_easy_setopt(easy, CURLOPT_OPENSOCKETFUNCTION, opensocket);
-
-  /* call this function to close a socket */
-  curl_easy_setopt(easy, CURLOPT_CLOSESOCKETFUNCTION, close_socket);
-
-  curl_easy_setopt(easy, CURLOPT_HTTP_VERSION, CURL_HTTP_VERSION_2_0);
-
-  /* keep the speed down so we don't have to buffer large amounts*/
-  curl_easy_setopt(easy, CURLOPT_MAX_RECV_SPEED_LARGE, (curl_off_t)31415);
-
-  auto rc = curl_multi_add_handle(global.multi, conn->easy);
-  if (mcode_test("new_conn: curl_multi_add_handle", rc) < 0) {
-    return nullptr;
-  }
-  conn->status = Status_t::STATUS_DOWNLOAD_IN_PROGRESS;
-  switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "createDownloader: launched request, gain %d\n", conn->gain);
-  return conn;
-}
-
-void destroyConnection(ConnInfo_t *conn) {
-  switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "destroyConnection\n");
-
-  /* clean up the curl handle*/
-  curl_multi_remove_handle(conn->global, conn->easy);
-  curl_easy_cleanup(conn->easy);
-
-  /* clear asio resources and free resources */
-  if (conn->timer) {
-    conn->timer->cancel();
-    delete conn->timer;
-  }
-  if (conn->err_msg) {
-    free(conn->err_msg);
-  }
-
-  /* free mp3 decoder */
-  if (conn->mh) {
-    mpg123_close(conn->mh);
-    mpg123_delete(conn->mh);
-  }
-
-  if (conn->url) {
-    free(conn->url);
-  }
-
-  if (conn->mutex) switch_mutex_lock(conn->mutex);
-  id2ConnMap.erase(conn->id);
-  if (conn->mutex) switch_mutex_unlock(conn->mutex);
-
-  memset(conn, 0, sizeof(ConnInfo_t));
-  pool.destroy(conn) ;
-}
-
-CURL* createEasyHandle(void) {
-  CURL* easy = curl_easy_init();
-  if(!easy) {
-    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "curl_easy_init() failed!\n");
-    return nullptr ;
-  }  
-
-  curl_easy_setopt(easy, CURLOPT_FOLLOWLOCATION, 1L);
-  curl_easy_setopt(easy, CURLOPT_USERAGENT, "jambonz/0.8.5");
-
-  // set connect timeout to 3 seconds and no total timeout as files could be large
-  curl_easy_setopt(easy, CURLOPT_CONNECTTIMEOUT_MS, 3000L);
-  curl_easy_setopt(easy, CURLOPT_TIMEOUT, 0L); // no timeout
-
-  return easy ;    
-}
-
-/* Check for completed transfers, and remove their easy handles */
-void check_multi_info(GlobalInfo_t *g) {
-  CURLMsg *msg;
-  int msgs_left;
-  ConnInfo_t *conn;
-  CURL *easy;
-  CURLcode res;
-  
-  while((msg = curl_multi_info_read(g->multi, &msgs_left))) {
-    if(msg->msg == CURLMSG_DONE) {
-      long response_code;
-      double namelookup=0, connect=0, total=0 ;
-      char *ct = NULL ;
-
-      easy = msg->easy_handle;
-      res = msg->data.result;
-      curl_easy_getinfo(easy, CURLINFO_PRIVATE, &conn);
-      curl_easy_getinfo(easy, CURLINFO_RESPONSE_CODE, &response_code);
-      curl_easy_getinfo(easy, CURLINFO_CONTENT_TYPE, &ct);
-
-      curl_easy_getinfo(easy, CURLINFO_NAMELOOKUP_TIME, &namelookup);
-      curl_easy_getinfo(easy, CURLINFO_CONNECT_TIME, &connect);
-      curl_easy_getinfo(easy, CURLINFO_TOTAL_TIME, &total);
-
-      downloadId_t id = conn->id;
-      auto mutex = conn->mutex;
-      auto buffer = conn->buffer;
-      auto rate = conn->rate;
-      auto loop = conn->loop;
-      auto gain = conn->gain;
-      auto oldId = conn->id;
-      bool restart = conn->loop && conn->status != Status_t::STATUS_STOPPING && response_code == 200;
-
-      conn->response_code = response_code;
-      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "curl done, response code %d, status %s\n", response_code, status2String(conn->status));
-      conn->status = Status_t::STATUS_DOWNLOAD_COMPLETE;
-
-      curl_multi_remove_handle(g->multi, easy);
-
-      if (restart) {
-        conn->status = Status_t::STATUS_AWAITING_RESTART;
-        conn->timer->expires_from_now(boost::posix_time::millisec(1000));
-        conn->timer->async_wait(boost::bind(&restart_cb, boost::placeholders::_1, conn));
-
-        //TODO: this seems to not be working from this callback; maybe start it from a timer callback?
-      }
-      else {
-        destroyConnection(conn);
-      }
-    }
-  }
-}
-
-int mcode_test(const char *where, CURLMcode code) {
-  if(CURLM_OK != code) {
-    const char *s;
-    switch(code) {
-    case CURLM_CALL_MULTI_PERFORM:
-      s = "CURLM_CALL_MULTI_PERFORM";
-      break;
-    case CURLM_BAD_HANDLE:
-      s = "CURLM_BAD_HANDLE";
-      break;
-    case CURLM_BAD_EASY_HANDLE:
-      s = "CURLM_BAD_EASY_HANDLE";
-      break;
-    case CURLM_OUT_OF_MEMORY:
-      s = "CURLM_OUT_OF_MEMORY";
-      break;
-    case CURLM_INTERNAL_ERROR:
-      s = "CURLM_INTERNAL_ERROR";
-      break;
-    case CURLM_UNKNOWN_OPTION:
-      s = "CURLM_UNKNOWN_OPTION";
-      break;
-    case CURLM_LAST:
-      s = "CURLM_LAST";
-      break;
-    default:
-      s = "CURLM_unknown";
-      break;
-    case CURLM_BAD_SOCKET:
-      s = "CURLM_BAD_SOCKET";
-      break;
-    }
-    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "mcode_test ERROR: %s returns %s:%d\n", where, s, code);
-
-    return -1;
-  }
-  return 0 ;
-}
-
-void remsock(int *f, GlobalInfo_t *g) {
-  if(f) {
-    free(f);
-    f = NULL;
-  }
-}
-
-/* Called by asio when there is an action on a socket */
-void event_cb(GlobalInfo_t *g, curl_socket_t s, int action, const boost::system::error_code & error, int *fdp) {
-  int f = *fdp;
-
-  //switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "event_cb socket %#X has action %d\n", s, action) ; 
-
-  // Socket already POOL REMOVED.
-  if (f == CURL_POLL_REMOVE) {
-    //switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "event_cb socket %#X removed\n", s); 
-    remsock(fdp, g);
-    return;
-  }
-
-  if(socket_map.find(s) == socket_map.end()) {
-    //switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "event_cb: socket  %#X already closed\n, s");
-    return;
-  }
-
-  /* make sure the event matches what are wanted */
-  if(f == action || f == CURL_POLL_INOUT) {
-    if(error) {
-      action = CURL_CSELECT_ERR;
-    }
-    CURLMcode rc = curl_multi_socket_action(g->multi, s, action, &g->still_running);
-
-    mcode_test("event_cb: curl_multi_socket_action", rc);
-    check_multi_info(g);
-
-    if(g->still_running <= 0) {
-      timer.cancel();
-    }
-
-    /* keep on watching.
-      * the socket may have been closed and/or fdp may have been changed
-      * in curl_multi_socket_action(), so check them both */
-    if(!error && socket_map.find(s) != socket_map.end() &&
-        (f == action || f == CURL_POLL_INOUT)) {
-      boost::asio::ip::tcp::socket *tcp_socket = socket_map.find(s)->second;
-
-      if(action == CURL_POLL_IN) {
-        tcp_socket->async_read_some(boost::asio::null_buffers(),
-                                    boost::bind(&event_cb, g, s,
-                                                action, boost::placeholders::_1, fdp));
-      }
-      if(action == CURL_POLL_OUT) {
-        tcp_socket->async_write_some(boost::asio::null_buffers(),
-                                      boost::bind(&event_cb, g, s,
-                                                  action, boost::placeholders::_1, fdp));
-      } 
-    }
-  }
-}
-
-/* socket functions */
-void setsock(int *fdp, curl_socket_t s, CURL *e, int act, int oldact, GlobalInfo_t *g) {
-  std::map<curl_socket_t, boost::asio::ip::tcp::socket *>::iterator it = socket_map.find(s);
-
-  if(it == socket_map.end()) {
-    //switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "setsock: socket  %#X not found\n, s");
-    return;
-  }
-
-  boost::asio::ip::tcp::socket * tcp_socket = it->second;
-
-  *fdp = act;
-
-  if(act == CURL_POLL_IN) {
-    if(oldact != CURL_POLL_IN && oldact != CURL_POLL_INOUT) {
-      tcp_socket->async_read_some(boost::asio::null_buffers(),
-                                  boost::bind(&event_cb, g, s,
-                                  CURL_POLL_IN, boost::placeholders::_1, fdp));
-    }
-  }
-  else if(act == CURL_POLL_OUT) {
-    if(oldact != CURL_POLL_OUT && oldact != CURL_POLL_INOUT) {
-      tcp_socket->async_write_some(boost::asio::null_buffers(),
-                                    boost::bind(&event_cb, g, s,
-                                    CURL_POLL_OUT, boost::placeholders::_1, fdp));
-    }
-  }
-  else if(act == CURL_POLL_INOUT) {
-    if(oldact != CURL_POLL_IN && oldact != CURL_POLL_INOUT) {
-      tcp_socket->async_read_some(boost::asio::null_buffers(),
-                                  boost::bind(&event_cb, g, s,
-                                  CURL_POLL_IN, boost::placeholders::_1, fdp));
-    }
-    if(oldact != CURL_POLL_OUT && oldact != CURL_POLL_INOUT) {
-      tcp_socket->async_write_some(boost::asio::null_buffers(),
-                                    boost::bind(&event_cb, g, s,
-                                    CURL_POLL_OUT, boost::placeholders::_1, fdp));
-    }
-  }
-}
-
-void addsock(curl_socket_t s, CURL *easy, int action, GlobalInfo_t *g) {
-  /* fdp is used to store current action */
-  int *fdp = (int *) calloc(sizeof(int), 1);
-
-  setsock(fdp, s, easy, action, 0, g);
-  curl_multi_assign(g->multi, s, fdp);
-}
-
-int sock_cb(CURL *e, curl_socket_t s, int what, void *cbp, void *sockp) {
-  GlobalInfo_t *g = &global;
-
-  int *actionp = (int *) sockp;
-  static const char *whatstr[] = { "none", "IN", "OUT", "INOUT", "REMOVE"};
-
-  if(what == CURL_POLL_REMOVE) {
-    *actionp = what;
-  }
-  else {
-    if(!actionp) {
-      addsock(s, e, what, g);
-    }
-    else {
-      setsock(actionp, s, e, what, *actionp, g);
-    }
-  }
-  return 0;  
-}
-
-void threadFunc() {      
-  /* to make sure the event loop doesn't terminate when there is no work to do */
-  io_service.reset() ;
-  boost::asio::io_service::work work(io_service);
-  
-  switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "mod_dub threadFunc - starting\n");
-
-  for(;;) {
-      
-    try {
-      io_service.run() ;
-      break ;
-    }
-    catch( std::exception& e) {
-      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "mod_dub threadFunc - Error: %s\n", e.what());
-    }
-  }
-  switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "mod_dub threadFunc - ending\n");
-}
-
-
-/* Called by asio when our timeout expires */
-void timer_cb(const boost::system::error_code & error, GlobalInfo_t *g)
-{
-  //switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "timer_cb\n");
-
-  if(!error) {
-    CURLMcode rc = curl_multi_socket_action(g->multi, CURL_SOCKET_TIMEOUT, 0, &g->still_running);
-    mcode_test("timer_cb: curl_multi_socket_action", rc);
-    check_multi_info(g);
-  }
-}
-
-int multi_timer_cb(CURLM *multi, long timeout_ms, GlobalInfo_t *g) {
-
-  /* cancel running timer */
-  timer.cancel();
-
-  if(timeout_ms >= 0) {
-    // from libcurl 7.88.1-10+deb12u4 does not allow call curl_multi_socket_action or curl_multi_perform in curl_multi callback directly
-    timer.expires_from_now(boost::posix_time::millisec(timeout_ms ? timeout_ms : 1));
-    timer.async_wait(boost::bind(&timer_cb, boost::placeholders::_1, g));
-  }
-
-  return 0;
-}
-
-std::vector<int16_t> convert_mp3_to_linear(ConnInfo_t *conn, int8_t *data, size_t len) {
-  std::vector<int16_t> linear_data;
-  int eof = 0;
-  int mp3err = 0;
-
-  if(mpg123_feed(conn->mh, (const unsigned char*) data, len) == MPG123_OK) {
-    while(!eof) {
-      size_t usedlen = 0;
-      off_t frame_offset;
-      unsigned char* audio;
-
-      int decode_status = mpg123_decode_frame(conn->mh, &frame_offset, &audio, &usedlen);
-
-      switch(decode_status) {
-        case MPG123_NEW_FORMAT:
-          continue;
-
-        case MPG123_OK:
-          {
-            size_t samples = usedlen / sizeof(int16_t);
-            linear_data.insert(linear_data.end(), reinterpret_cast<int16_t*>(audio), reinterpret_cast<int16_t*>(audio) + samples);
-          }
-          break;
-
-        case MPG123_DONE:
-        case MPG123_NEED_MORE:
-          eof = 1;
-          break;
-
-        case MPG123_ERR:
-        default:
-          if(++mp3err >= 5) {
-            switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Decoder Error!\n");
-            eof = 1;
-          }
-      }
-
-      if (eof)
-        break;
-
-      mp3err = 0;
-    }
-
-    if (conn->gain != 0) {
-      switch_change_sln_volume_granular(linear_data.data(), linear_data.size(), conn->gain);
-    }
-  }
-
-  return linear_data;
-}
-
-void restart_cb(const boost::system::error_code& error, ConnInfo_t* conn) {
-  switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "restart_cb status is %s\n", status2String(conn->status));
-  if (conn->status == Status_t::STATUS_AWAITING_RESTART) {
-    auto url = strdup(conn->url);
-    auto rate = conn->rate;
-    auto loop = conn->loop;
-    auto gain = conn->gain;
-    auto mutex = conn->mutex;
-    auto buffer = conn->buffer;
-    auto oldId = conn->id;
-
-    destroyConnection(conn);
-
-    downloadId_t id = start_audio_download(url, rate, loop, gain, mutex, buffer);
-
-    /* re-use id since caller is tracking that id */
-    auto * newConnection = id2ConnMap[id];
-    id2ConnMap[oldId] = newConnection;
-    id2ConnMap.erase(id);
-
-    free(url);
-    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "restarted looped download\n");
-  }
-}
-
-void throttling_cb(const boost::system::error_code& error, ConnInfo_t* conn) {
-  if (conn->status == Status_t::STATUS_STOPPING || !conn->mutex || !conn->buffer) {
-    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "throttling_cb: session gone, resume download so we can complete\n");
-    curl_easy_pause(conn->easy, CURLPAUSE_CONT);
-    return;
-  }
-  switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "throttling_cb: status is %s\n", status2String(conn->status));
-
-  switch_mutex_lock(conn->mutex);
-  if (!error) {
-    auto size = conn->buffer->size();
-
-    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "throttling_cb: size is now %ld\n", size);
-    if (size < BUFFER_THROTTLE_LOW) {
-      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "throttling_cb: resuming download\n");
-      curl_easy_pause(conn->easy, CURLPAUSE_CONT);
-      switch_mutex_unlock(conn->mutex);
-     return;
-    }
-
-    // check back in 2 seconds
-    conn->timer->expires_from_now(boost::posix_time::millisec(2000));
-    conn->timer->async_wait(boost::bind(&throttling_cb, boost::placeholders::_1, conn));
-
-  } else {
-    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "throttling_cb: error (%d): %s\n", error.value(), error.message().c_str());
-
-    // Handle any errors
-  }
-  switch_mutex_unlock(conn->mutex);
-}
-
-
-/* CURLOPT_WRITEFUNCTION - here is where we receive the data */
-size_t write_cb(void *ptr, size_t size, size_t nmemb, ConnInfo_t *conn) {
-  int8_t *data = (int8_t *) ptr;
-  size_t bytes_received = size * nmemb;
-  std::vector<int16_t> pcm_data;
-  
-  if (conn->status == Status_t::STATUS_STOPPING || conn->status == Status_t::STATUS_STOPPED || !conn->mutex || !conn->buffer) {
-    if (conn->timer) conn->timer->cancel();
-      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, 
-      "write_cb: aborting transfer, status %s, mutex %p, buffer %p\n", status2String(conn->status), conn->mutex, conn->buffer);
-    /* this will abort the transfer */
-    return 0;
-  }
-  {
-    switch_mutex_lock(conn->mutex);
-
-    if (conn->response_code > 0 && conn->response_code != 200) {
-      std::string body((char *) ptr, bytes_received);
-      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "write_cb: received body %s\n", body.c_str());
-      conn->err_msg = strdup(body.c_str());
-      conn->status = Status_t::STATUS_FAILED;
-      switch_mutex_unlock(conn->mutex);
-      return 0;
-    }
-
-    /* throttle after reaching high water mark */
-    if (conn->buffer->size() > BUFFER_THROTTLE_HIGH) {
-      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "write_cb: throttling download, buffer size is %ld\n", conn->buffer->size());
-      
-      // check back in 2 seconds
-      conn->timer->expires_from_now(boost::posix_time::millisec(2000));
-      conn->timer->async_wait(boost::bind(&throttling_cb, boost::placeholders::_1, conn));
-
-      conn->status = Status_t::STATUS_DOWNLOAD_PAUSED;
-      switch_mutex_unlock(conn->mutex);
-      return CURL_WRITEFUNC_PAUSE;
-    }
-
-    pcm_data = convert_mp3_to_linear(conn, data, bytes_received);
-    size_t bytesResampled = pcm_data.size() * sizeof(int16_t);
-
-    // Resize the buffer if necessary
-    if (conn->buffer->capacity() - conn->buffer->size() < (bytesResampled / sizeof(int16_t))) {
-      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "write_cb growing buffer, size now %ld\n", conn->buffer->size()); 
-
-      //TODO: if buffer exceeds some max size, return CURL_WRITEFUNC_ERROR to abort the transfer
-      conn->buffer->set_capacity(conn->buffer->size() + std::max((bytesResampled / sizeof(int16_t)), (size_t)BUFFER_GROW_SIZE));
-    }
-    
-    /* Push the data into the buffer */
-    conn->buffer->insert(conn->buffer->end(), pcm_data.data(), pcm_data.data() + pcm_data.size());
-    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "write_cb: wrote data, buffer size is now %ld\n", conn->buffer->size());
-
-    switch_mutex_unlock(conn->mutex);
-  }
-  return bytes_received;
-}
-
-bool parseHeader(const std::string& str, std::string& header, std::string& value) {
-    std::vector<std::string> parts;
-    boost::split(parts, str, boost::is_any_of(":"), boost::token_compress_on);
-
-    if (parts.size() != 2)
-        return false;
-
-    header = boost::trim_copy(parts[0]);
-    value = boost::trim_copy(parts[1]);
-    return true;
-}
-
-int extract_response_code(const std::string& input) {
-  std::size_t space_pos = input.find(' ');
-  if (space_pos == std::string::npos) {
-    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Invalid HTTP response format %s\n", input.c_str());
-    return 0;
-  }
-
-  std::size_t code_start_pos = space_pos + 1;
-  std::size_t code_end_pos = input.find(' ', code_start_pos);
-  if (code_end_pos == std::string::npos) {
-    code_end_pos = input.length();
-  }
-
-  std::string code_str = input.substr(code_start_pos, code_end_pos - code_start_pos);
-  int response_code = std::stoi(code_str);
-  return response_code;
-}
-
-size_t header_callback(char *buffer, size_t size, size_t nitems, ConnInfo_t *conn) {
-  size_t bytes_received = size * nitems;
-  const std::string prefix = "HTTP/";
-  std::string header, value;
-  std::string input(buffer, bytes_received);
-  if (parseHeader(input, header, value)) {
-    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "recv header: %s with value %s\n", header.c_str(), value.c_str());
-  }
-  else {
-    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "recv header: %s\n", input.c_str());
-    if (input.rfind(prefix, 0) == 0) {
-      try {
-        conn->response_code = extract_response_code(input);
-        switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "parsed response code: %ld\n", conn->response_code);
-      } catch (const std::invalid_argument& e) {
-        switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "header_callback: invalid response code %s\n", input.substr(prefix.length()).c_str());
-      }
-    }
-  }
-  return bytes_received;
-}
-
-/* CURLOPT_OPENSOCKETFUNCTION */
-curl_socket_t opensocket(void *clientp, curlsocktype purpose, struct curl_sockaddr *address) {
-  curl_socket_t sockfd = CURL_SOCKET_BAD;
-
-  //switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "opensocket: %d\n", purpose);
-  /* restrict to IPv4 */
-  if(purpose == CURLSOCKTYPE_IPCXN && address->family == AF_INET) {
-    /* create a tcp socket object */
-    boost::asio::ip::tcp::socket *tcp_socket = new boost::asio::ip::tcp::socket(io_service);
-
-    /* open it and get the native handle*/
-    boost::system::error_code ec;
-    tcp_socket->open(boost::asio::ip::tcp::v4(), ec);
-
-    if(ec) {
-      /* An error occurred */
-      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Couldn't open socket [%ld][%s]\n", ec, ec.message().c_str());
-    }
-    else {
-      sockfd = tcp_socket->native_handle();
-
-      /* save it for monitoring */
-      socket_map.insert(std::pair<curl_socket_t, boost::asio::ip::tcp::socket *>(sockfd, tcp_socket));
-    }
-  }
-  return sockfd;
-}
-
-/* CURLOPT_CLOSESOCKETFUNCTION */
-int close_socket(void *clientp, curl_socket_t item) {
-  //switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "close_socket : %#X\n", item);
-
-  std::map<curl_socket_t, boost::asio::ip::tcp::socket *>::iterator it = socket_map.find(item);
-  if(it != socket_map.end()) {
-    delete it->second;
-    socket_map.erase(it);
-  }
-  return 0;
-}
-

mod_dub/audio_downloader.h

@@ -1,18 +0,0 @@
-#ifndef __AUDIO_DOWNLOADER_H__
-#define __AUDIO_DOWNLOADER_H__
-
-#include <switch.h>
-#include "common.h"
-
-extern "C" {
-
-switch_status_t init_audio_downloader();
-switch_status_t deinit_audio_downloader();
-
-int start_audio_download(const char* url, int rate, int loop, int gain, switch_mutex_t* mutex, CircularBuffer_t* buffer);
-switch_status_t stop_audio_download(int id);
-
-}
-
-#endif
-

mod_dub/common.h

@@ -3,10 +3,7 @@
 
 #include <boost/circular_buffer.hpp>
 
-
 typedef boost::circular_buffer<int16_t> CircularBuffer_t;
-typedef int32_t downloadId_t;
 
-#define INVALID_DOWNLOAD_ID (-1)
 
 #endif

mod_dub/dub_glue.cpp

@@ -1,8 +1,9 @@
 #include "mod_dub.h"
-#include "audio_downloader.h"
-#include "file_loader.h"
-
+#include "tts_vendor_parser.h"
+#include "track.h"
+#include "vector_math.h"
 #include <string>
+#include <queue>
 
 #include <switch.h>
 
@@ -15,124 +16,111 @@ typedef boost::circular_buffer<int16_t> CircularBuffer_t;
 
 extern "C" {
 
-  void init_dub_track(dub_track_t *track, char* trackName, int sampleRate) {
-    track->state = DUB_TRACK_STATE_READY;
-    track->trackName = strdup(trackName);
-    track->sampleRate = sampleRate;
-    track->circularBuffer = new CircularBuffer_t(INIT_BUFFER_SIZE);
+  Track* find_track_by_name(void** tracks, const std::string& trackName) {
+    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "find_track_by_name: searching for %s\n", trackName.c_str());
+
+    for (int i = 0; i < MAX_DUB_TRACKS; i++) {
+      Track* track = static_cast<Track*>(tracks[i]);
+      std::string name = track ? track->getTrackName() : "null";
+      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "find_track_by_name: offset %d: %s\n", i, name.c_str());
+      if (track && 0 == track->getTrackName().compare(trackName)) {
+        return track;
+      }
+    }
+    return nullptr;
   }
 
-  switch_status_t silence_dub_track(dub_track_t *track) {
-    assert(track);
-    switch (track->generator) {
-      case DUB_GENERATOR_TYPE_HTTP:
-        stop_audio_download(track->generatorId);
-        break;
-      case DUB_GENERATOR_TYPE_FILE:
-        stop_file_load(track->generatorId);
-        break;
-      case DUB_GENERATOR_TYPE_TTS:
-        //TODO
-        break;
-    }
-    CircularBuffer_t* buffer = reinterpret_cast<CircularBuffer_t*>(track->circularBuffer);
-    buffer->clear();
-    track->state = DUB_TRACK_STATE_READY;
-    track->generator = DUB_GENERATOR_TYPE_UNKNOWN;
-    track->generatorId = 0;
-
-    return SWITCH_STATUS_SUCCESS;
-  }
-
-  switch_status_t remove_dub_track(dub_track_t *track) {
-    assert(track);
-    switch (track->generator) {
-      case DUB_GENERATOR_TYPE_HTTP:
-        stop_audio_download(track->generatorId);
-        break;
-      case DUB_GENERATOR_TYPE_FILE:
-        stop_file_load(track->generatorId);
-        break;
-      case DUB_GENERATOR_TYPE_TTS:
-        //TODO
-        break;
-    }
-    CircularBuffer_t* buffer = reinterpret_cast<CircularBuffer_t*>(track->circularBuffer);
-    if (buffer) {
-      delete buffer;
-    }
-    if (track->trackName) {
-      free(track->trackName);
-    }
-    memset(track, 0, sizeof(dub_track_t));
-
-    return SWITCH_STATUS_SUCCESS;
-  }
-
-  switch_status_t play_dub_track(dub_track_t *track, switch_mutex_t *mutex, char* url, int loop, int gain) {
-    bool isHttp = strncmp(url, "http", 4) == 0;
-    if (track->state != DUB_TRACK_STATE_READY) {
-      silence_dub_track(track);
-    }
-    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "play_dub_track: starting %s download: %s\n", (isHttp ? "HTTP" : "file"), url);
-    int id = isHttp ?
-      start_audio_download(url, track->sampleRate, loop, gain, mutex, (CircularBuffer_t*) track->circularBuffer) :
-      start_file_load(url, track->sampleRate, loop, gain, mutex, (CircularBuffer_t*) track->circularBuffer);
-
-    if (id == INVALID_DOWNLOAD_ID) {
-      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "play_dub_track: failed to start audio download\n");
+  switch_status_t add_track(struct cap_cb* cb, char* trackName, int sampleRate) {
+    Track* existingTrack = find_track_by_name(cb->tracks, trackName);
+    if (existingTrack) {
+      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "add_track: track %s already exists\n", trackName);
       return SWITCH_STATUS_FALSE;
     }
-    track->state = DUB_TRACK_STATE_ACTIVE;
-    track->generatorId = id;
-    track->generator = isHttp ? DUB_GENERATOR_TYPE_HTTP : DUB_GENERATOR_TYPE_FILE;
-    track->gain = gain;
+
+    for (int i = 0; i < MAX_DUB_TRACKS; i++) {
+      if (!cb->tracks[i]) {
+        cb->tracks[i] = new Track(trackName, sampleRate);
+        switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "add_track: added track %s at offset %d\n", trackName, i);
+        return SWITCH_STATUS_SUCCESS;
+      }
+    }
+
+    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "add_track: no room for track %s\n", trackName);
+    return SWITCH_STATUS_FALSE;
+  }
+
+  switch_status_t silence_dub_track(struct cap_cb* cb, char* trackName) {
+    Track* track = find_track_by_name(cb->tracks, trackName);
+    if (track) {
+      track->removeAllAudio();
+      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "silence_dub_track: silenced track %s\n", trackName);
+      return SWITCH_STATUS_SUCCESS;
+    }
+
+    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "silence_dub_track: track %s not found\n", trackName);
+    return SWITCH_STATUS_FALSE;
+  }
+
+  switch_status_t remove_dub_track(struct cap_cb* cb, char* trackName) {
+    for (int i = 0; i < MAX_DUB_TRACKS; i++) {
+      Track* track = static_cast<Track*>(cb->tracks[i]);
+      if (track && track->getTrackName() == trackName) {
+        track->removeAllAudio();
+        delete track;
+        cb->tracks[i] = nullptr;
+        switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "remove_dub_track: removed track %s\n", trackName);
+        return SWITCH_STATUS_SUCCESS;
+      }
+    }
+
+    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "remove_dub_track: track %s not found\n", trackName);
+    return SWITCH_STATUS_FALSE;
+  }
+
+  switch_status_t play_dub_track(struct cap_cb* cb, char* trackName, char* url, int loop, int gain) {
+    bool isHttp = strncmp(url, "http", 4) == 0;
+    Track* track = find_track_by_name(cb->tracks, trackName);
+
+    if (!track) {
+      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "play_dub_track: track %s not found\n", trackName);
+      return SWITCH_STATUS_FALSE;
+    }
+
+    if (isHttp) {
+      track->queueHttpGetAudio(url, gain, loop);
+    }
+    else {
+      track->queueFileAudio(url, gain, loop);
+    }
 
     return SWITCH_STATUS_SUCCESS;
   }
 
-  switch_status_t say_dub_track(dub_track_t *track, switch_mutex_t *mutex, char* text, int gain) {
-    if (track->state != DUB_TRACK_STATE_READY) {
-      silence_dub_track(track); // wait...shouldnt we queue says?
-    }
-    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "say_dub_track: starting TTS\n");
-    
-    /**
-     * TODO:
-     * This is not implemented yet.  We can play TTS using using the playOnSay function
-     * because jambonz can generate local audio files using TTS vendors.
-     * However, we should probably at least implement support for elevenlabs streaming api
-     * here because it is so much faster.
-     * 
-     */
+  switch_status_t say_dub_track(struct cap_cb* cb, char* trackName, char* text, int gain) {
+    std::vector<std::string> headers;
+    std::string url, body;
+    Track* track = find_track_by_name(cb->tracks, trackName);
 
-    /*
-    track->state = DUB_TRACK_STATE_ACTIVE;
-    track->generatorId = id;
-    track->generator = DUB_GENERATOR_TYPE_TTS;
-    track->gain = gain;
-    */
+    if (!track) {
+      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "play_dub_track: track %s not found\n", trackName);
+      return SWITCH_STATUS_FALSE;
+    }
+    if (tts_vendor_parse_text(text, url, body, headers) != SWITCH_STATUS_SUCCESS) {
+      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "say_dub_track: failed to parse text\n");
+      return SWITCH_STATUS_FALSE;
+    }
+    track->queueHttpPostAudio(url, body, headers, gain);
     return SWITCH_STATUS_SUCCESS;
   }
 
 
   /* module load and unload */
   switch_status_t dub_init() {
-    switch_status_t status;
-    status = init_audio_downloader();
-    if (status == SWITCH_STATUS_SUCCESS) {
-      status = init_file_loader();
-    }
-    return status;
+    return SWITCH_STATUS_SUCCESS;
   }
 
   switch_status_t dub_cleanup() {
-    switch_status_t status;
-    status = deinit_audio_downloader();
-    if (status == SWITCH_STATUS_SUCCESS) {
-      status = deinit_file_loader();
-    }
-    return status;
+    return SWITCH_STATUS_SUCCESS;
   }
 
   switch_status_t dub_session_cleanup(switch_core_session_t *session, int channelIsClosing, switch_media_bug_t *bug) {
@@ -144,17 +132,18 @@ extern "C" {
 
       if (!switch_channel_get_private(channel, MY_BUG_NAME)) {
         // race condition
-        switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_INFO, "%s Bug is not attached (race).\n", switch_channel_get_name(channel));
+        switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "%s Bug is not attached (race).\n", switch_channel_get_name(channel));
         switch_mutex_unlock(cb->mutex);
         return SWITCH_STATUS_FALSE;
       }
       switch_channel_set_private(channel, MY_BUG_NAME, NULL);
 
       for (int i = 0; i < MAX_DUB_TRACKS; i++) {
-        dub_track_t* track = &cb->tracks[i];
-        if (track->state != DUB_TRACK_STATE_INACTIVE) {
-    			switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "dub_session_cleanup: cleared track %d:%s\n", i, track->trackName);
-          remove_dub_track(track);
+        Track* track = static_cast<Track*>(cb->tracks[i]);
+        if (track) {
+          track->removeAllAudio();
+          delete track;
+          cb->tracks[i] = nullptr;
         }
       }
 
@@ -162,7 +151,7 @@ extern "C" {
         switch_core_media_bug_remove(session, &bug);
       }
 
-			switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_INFO, "dub_session_cleanup: removed bug and cleared tracks\n");
+			switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "dub_session_cleanup: removed bug and cleared tracks\n");
 			switch_mutex_unlock(cb->mutex);
     }
     switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_INFO, "%s dub_session_cleanup: Bug is not attached.\n", switch_channel_get_name(channel));
@@ -175,12 +164,17 @@ extern "C" {
 
     if (switch_mutex_trylock(cb->mutex) == SWITCH_STATUS_SUCCESS) {
 
-      /* check if any tracks are actively pushing audio */
-      int trackCount = 0;
+      /* check if any tracks have audio to contribute */
+      std::vector<Track*> activeTracks;
+      activeTracks.reserve(MAX_DUB_TRACKS);
       for (int i = 0; i < MAX_DUB_TRACKS; i++) {
-        if (cb->tracks[i].state == DUB_TRACK_STATE_ACTIVE) trackCount++;
+        if (cb->tracks[i]) {
+          auto track = static_cast<Track*>(cb->tracks[i]);
+          if (track->hasAudio_NoLock()) activeTracks.push_back(static_cast<Track*>(cb->tracks[i]));
+        }
       }
-      if (trackCount == 0 && cb->gain == 0) {
+
+      if (activeTracks.size() == 0 && cb->gain == 0) {
         switch_mutex_unlock(cb->mutex);
         return SWITCH_TRUE;
       }
@@ -189,31 +183,24 @@ extern "C" {
       int16_t *fp = reinterpret_cast<int16_t*>(rframe->data);
 
       rframe->channels = 1;
-      rframe->datalen = rframe->samples * rframe->channels * sizeof(int16_t);
+      rframe->datalen = rframe->samples * sizeof(int16_t);
 
       /* apply gain to audio in main channel if requested*/
       if (cb->gain != 0) {
-        switch_change_sln_volume_granular(fp, rframe->samples, cb->gain);
+        vector_change_sln_volume_granular(fp, rframe->samples, cb->gain);
       }
 
       /* now mux in the data from tracks */
-      for (int i = 0; i < rframe->samples; i++) {
-        int16_t input = fp[i];
-        int16_t value = input;
-        for (int j = 0; j < MAX_DUB_TRACKS; j++) {
-          dub_track_t* track = &cb->tracks[j];
-          if (track->state == DUB_TRACK_STATE_ACTIVE) {
-            CircularBuffer_t* buffer = reinterpret_cast<CircularBuffer_t*>(track->circularBuffer);
-            if (buffer && !buffer->empty()) {
-              int16_t sample = buffer->front();
-              buffer->pop_front();
-              value += sample;
-            }
-          }
+      for (auto track : activeTracks) {
+        int16_t data[SWITCH_RECOMMENDED_BUFFER_SIZE];
+        memset(data, 0, sizeof(data));
+        auto samples = track->retrieveAndClearAudio(data, rframe->samples);
+        if (samples > 0) {
+          vector_add(fp, data, rframe->samples);
         }
-        switch_normalize_to_16bit(value);
-        fp[i] = (int16_t) value;
       }
+      vector_normalize(fp, rframe->samples);
+
       switch_core_media_bug_set_write_replace_frame(bug, rframe);
       switch_mutex_unlock(cb->mutex);
     }

mod_dub/dub_glue.h

@@ -4,11 +4,11 @@
 switch_status_t dub_init();
 switch_status_t dub_cleanup();
 
-void init_dub_track(dub_track_t *track, char* trackName, int sampleRate);
-switch_status_t silence_dub_track(dub_track_t *track);
-switch_status_t remove_dub_track(dub_track_t *track);
-switch_status_t play_dub_track(dub_track_t *track, switch_mutex_t *mutex, char* url, int loop, int gain);
-switch_status_t say_dub_track(dub_track_t *track, switch_mutex_t *mutex, char* text, int gain);
+switch_status_t add_track(struct cap_cb* cb, char* trackName, int sampleRate);
+switch_status_t silence_dub_track(struct cap_cb* cb, char* trackName);
+switch_status_t remove_dub_track(struct cap_cb* cb, char* trackName);
+switch_status_t play_dub_track(struct cap_cb* cb, char* trackName, char* url, int loop, int gain);
+switch_status_t say_dub_track(struct cap_cb* cb, char* trackName, char* text, int gain);
 
 switch_status_t dub_session_cleanup(switch_core_session_t *session, int channelIsClosing, switch_media_bug_t *bug);
 switch_bool_t dub_speech_frame(switch_media_bug_t *bug, void* user_data);

mod_dub/file_loader.cpp

@@ -1,437 +0,0 @@
-#include "file_loader.h"
-
-#include <boost/thread.hpp>
-#include <boost/asio.hpp>
-#include <boost/asio/ssl.hpp>
-#include <boost/pool/object_pool.hpp>
-#include <boost/bind/bind.hpp>
-#include <boost/tokenizer.hpp>
-#include <boost/foreach.hpp>
-#include <boost/asio.hpp>
-#include <boost/assign/list_of.hpp>
-#include <boost/algorithm/string.hpp>
-
-#include <map>
-
-#include <mpg123.h>
-
-#define INIT_BUFFER_SIZE (80000)
-#define BUFFER_GROW_SIZE (80000)
-#define BUFFER_THROTTLE_LOW (40000)
-#define BUFFER_THROTTLE_HIGH (160000)
-
-static uint16_t currDownloadId = 0;
-
-typedef enum
-{
-  STATUS_NONE = 0,
-  STATUS_FAILED,
-  STATUS_FILE_IN_PROGRESS,
-  STATUS_FILE_PAUSED,
-  STATUS_FILE_COMPLETE,
-  STATUS_AWAITING_RESTART,
-  STATUS_STOPPING,
-  STATUS_STOPPED
-} Status_t;
-
-typedef enum {
-  FILE_TYPE_MP3 = 0,
-  FILE_TYPE_R8
-} FileType_t;
-
-static const char* status2String(Status_t status)
-{
-  static const char* statusStrings[] = {
-    "STATUS_NONE",
-    "STATUS_FAILED",
-    "STATUS_FILE_IN_PROGRESS",
-    "STATUS_FILE_PAUSED",
-    "STATUS_FILE_COMPLETE",
-    "STATUS_AWAITING_RESTART",
-    "STATUS_STOPPING",
-    "STATUS_STOPPED"
-  };
-
-  if (status >= 0 && status < sizeof(statusStrings) / sizeof(statusStrings[0]))
-  {
-    return statusStrings[status];
-  }
-  else
-  {
-    return "UNKNOWN_STATUS";
-  }
-}
-
-typedef struct
-{
-  switch_mutex_t* mutex;
-  CircularBuffer_t* buffer;
-  mpg123_handle *mh;
-  FILE* fp;
-  char* path;
-  bool loop;
-  int rate;
-  boost::asio::deadline_timer *timer;
-  Status_t status;
-  FileType_t type;
-  downloadId_t id;
-  int gain;
-} FileInfo_t;
-
-typedef std::map<int32_t, FileInfo_t *> Id2FileMap_t;
-static Id2FileMap_t id2FileMap;
-
-static boost::object_pool<FileInfo_t> pool ;
-static boost::asio::io_service io_service;
-static std::thread worker_thread;
-
-
-/* forward declarations */
-static FileInfo_t* createFileLoader(const char *path, int rate, int loop, int gain, mpg123_handle *mhm, switch_mutex_t *mutex, CircularBuffer_t *buffer);
-static void destroyFileInfo(FileInfo_t *finfo);
-static void threadFunc();
-static std::vector<int16_t> convert_mp3_to_linear(FileInfo_t *file, int8_t *data, size_t len);
-static void read_cb(const boost::system::error_code& error, FileInfo_t* finfo) ;
-
-/* apis */
-extern "C" {
-
-  switch_status_t init_file_loader() {
-    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "init_file_loader loading..\n");
-
-    if (mpg123_init() != MPG123_OK) {
-      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "init_file_loader: failed to initiate MPG123");
-      return SWITCH_STATUS_FALSE;
-    }
-
-    /* start worker thread */
-    std::thread t(threadFunc) ;
-    worker_thread.swap( t ) ;
-
-    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "init_file_loader: loaded\n");
-
-    return SWITCH_STATUS_SUCCESS;
-
-  }
-
-  switch_status_t deinit_file_loader() {
-    /* stop the ASIO IO service */
-    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "deinit_file_loader: stopping io service\n");
-    io_service.stop();
-
-    /* Join the worker thread */
-    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "deinit_file_loader: wait for worker thread to complete\n");
-    if (worker_thread.joinable()) {
-        worker_thread.join();
-    }
-
-    mpg123_exit();
-
-    return SWITCH_STATUS_SUCCESS;
-  }
-
-  downloadId_t start_file_load(const char* path, int rate, int loop, int gain, switch_mutex_t* mutex, CircularBuffer_t* buffer) {
-    int mhError = 0;
-
-    /* we only handle mp3 or r8 files atm */
-    const char *ext = strrchr(path, '.');
-    if (!ext) {
-      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "start_file_load: file %s has no extension\n", path);
-      return INVALID_DOWNLOAD_ID;
-    }
-    ext++;
-    if (0 != strcmp(ext, "mp3") && 0 != strcmp(ext, "r8")) {
-      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "start_file_load: file %s has unsupported extension %s\n", path, ext);
-      return INVALID_DOWNLOAD_ID;
-    }
-
-    /* allocate handle for mpeg decoding */
-    mpg123_handle *mh = mpg123_new("auto", &mhError);
-    if (!mh) {
-      const char *mhErr = mpg123_plain_strerror(mhError);
-      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error allocating mpg123 handle! %s\n", switch_str_nil(mhErr));
-      return INVALID_DOWNLOAD_ID;
-    }
-
-    if (mpg123_open_feed(mh) != MPG123_OK) {
-      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error mpg123_open_feed!\n");
-      return INVALID_DOWNLOAD_ID;
-    }
-
-    if (mpg123_format_all(mh) != MPG123_OK) {
-      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error mpg123_format_all!\n");
-      return INVALID_DOWNLOAD_ID;
-    }
-
-    if (mpg123_param(mh, MPG123_FORCE_RATE, rate, 0) != MPG123_OK) {
-      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error forcing resample to 8k!\n");
-      return INVALID_DOWNLOAD_ID;
-    }
-
-    if (mpg123_param(mh, MPG123_FLAGS, MPG123_MONO_MIX, 0) != MPG123_OK) {
-      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error forcing single channel!\n");
-      return INVALID_DOWNLOAD_ID;
-    }
-
-    FileInfo_t* finfo = createFileLoader(path, rate, loop, gain, mh, mutex, buffer);
-    if (!finfo) {
-      return INVALID_DOWNLOAD_ID;
-    }
-
-    /* do the initial read in the worker thread so we don't block here */
-    finfo->timer->expires_from_now(boost::posix_time::millisec(1));
-    finfo->timer->async_wait(boost::bind(&read_cb, boost::placeholders::_1, finfo));
-
-    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, 
-      "start_file_load: starting load %d\n", finfo->id);
-
-    return finfo->id;
-  }
-
-  switch_status_t stop_file_load(int id) {
-    auto it = id2FileMap.find(id);
-    if (it == id2FileMap.end()) {
-      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "stop_file_load: id %d has already completed\n", id);
-      return SWITCH_STATUS_FALSE;
-    }
-    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, 
-      "stop_audio_download: stopping download %d, status %s\n", id, status2String(it->second->status));
-
-    FileInfo_t *finfo = it->second;
-    auto status = finfo->status;
-
-    /* past this point I shall not access either the mutex or the buffer provided */
-    finfo->mutex = nullptr;
-    finfo->buffer = nullptr;
-
-    destroyFileInfo(finfo);
-
-    finfo->status = Status_t::STATUS_STOPPED;
-    return SWITCH_STATUS_SUCCESS;
-  }
-}
-
-/* internal */
-FileInfo_t* createFileLoader(const char *path, int rate, int loop, int gain, mpg123_handle *mh, switch_mutex_t *mutex, CircularBuffer_t *buffer) {
-  FileInfo_t *finfo = pool.malloc() ;
-  const char *ext = strrchr(path, '.');
-
-  memset(finfo, 0, sizeof(FileInfo_t));
-  finfo->mutex = mutex;
-  finfo->buffer = buffer;
-  finfo->mh = mh;
-  finfo->loop = loop;
-  finfo->gain = gain;
-  finfo->rate = rate;
-  finfo->path = strdup(path);
-  finfo->status = Status_t::STATUS_NONE; 
-  finfo->timer = new boost::asio::deadline_timer(io_service);
-
-  if (0 == strcmp(ext, "mp3")) finfo->type = FileType_t::FILE_TYPE_MP3;
-  else if (0 == strcmp(ext, "r8")) finfo->type = FileType_t::FILE_TYPE_R8;
-
-  downloadId_t id = ++currDownloadId;
-  if (id == 0) id++;
-
-  id2FileMap[id] = finfo;
-  finfo->id = id;
-
-  finfo->status = Status_t::STATUS_AWAITING_RESTART;
-
-  finfo->fp = fopen(finfo->path, "rb");
-  if (finfo->fp == NULL) {
-    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "createFileLoader: failed to open file %s\n", finfo->path);
-    destroyFileInfo(finfo);
-    return nullptr;
-  }
-
-  switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG,
-    "createFileLoader: launched request, loop %s, gain %d\n", (finfo->loop ? "yes": "no"), finfo->gain);
-  return finfo;
-}
-
-void destroyFileInfo(FileInfo_t *finfo) {
-  switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "destroyFileInfo\n");
-
-  /* clear asio resources and free resources */
-  if (finfo->timer) {
-    finfo->timer->cancel();
-    delete finfo->timer;
-  }
-
-  /* free mp3 decoder */
-  if (finfo->mh) {
-    mpg123_close(finfo->mh);
-    mpg123_delete(finfo->mh);
-  }
-
-  if (finfo->path) {
-    free(finfo->path);
-  }
-
-  if (finfo->mutex) switch_mutex_lock(finfo->mutex);
-  id2FileMap.erase(finfo->id);
-  if (finfo->mutex) switch_mutex_unlock(finfo->mutex);
-
-  memset(finfo, 0, sizeof(FileInfo_t));
-  pool.destroy(finfo) ;
-}
-
-void threadFunc() {      
-  /* to make sure the event loop doesn't terminate when there is no work to do */
-  io_service.reset() ;
-  boost::asio::io_service::work work(io_service);
-  
-  switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "file_loader threadFunc - starting\n");
-
-  for(;;) {
-      
-    try {
-      io_service.run() ;
-      break ;
-    }
-    catch( std::exception& e) {
-      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "file_loader threadFunc - Error: %s\n", e.what());
-    }
-  }
-  switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "file_loader threadFunc - ending\n");
-}
-
-std::vector<int16_t> convert_mp3_to_linear(FileInfo_t *finfo, int8_t *data, size_t len) {
-  std::vector<int16_t> linear_data;
-  int eof = 0;
-  int mp3err = 0;
-
-  if(mpg123_feed(finfo->mh, (const unsigned char*) data, len) == MPG123_OK) {
-    while(!eof) {
-      size_t usedlen = 0;
-      off_t frame_offset;
-      unsigned char* audio;
-
-      int decode_status = mpg123_decode_frame(finfo->mh, &frame_offset, &audio, &usedlen);
-
-      switch(decode_status) {
-        case MPG123_NEW_FORMAT:
-          continue;
-
-        case MPG123_OK:
-          {
-            size_t samples = usedlen / sizeof(int16_t);
-            linear_data.insert(linear_data.end(), reinterpret_cast<int16_t*>(audio), reinterpret_cast<int16_t*>(audio) + samples);
-          }
-          break;
-
-        case MPG123_DONE:
-        case MPG123_NEED_MORE:
-          eof = 1;
-          break;
-
-        case MPG123_ERR:
-        default:
-          if(++mp3err >= 5) {
-            switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Decoder Error!\n");
-            eof = 1;
-          }
-      }
-
-      if (eof)
-        break;
-
-      mp3err = 0;
-    }
-
-    if (finfo->gain != 0) {
-      switch_change_sln_volume_granular(linear_data.data(), linear_data.size(), finfo->gain);
-    }
-  }
-
-  return linear_data;
-}
-
-void read_cb(const boost::system::error_code& error, FileInfo_t* finfo) {
-  if (finfo->status == Status_t::STATUS_STOPPING || !finfo->mutex || !finfo->buffer) {
-    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "read_cb: %u session gone\n", finfo->id);
-    return;
-  }
-  switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "read_cb: %u status is %s\n", finfo->id, status2String(finfo->status));
-  if (finfo->status == Status_t::STATUS_AWAITING_RESTART) {
-    finfo->status = Status_t::STATUS_FILE_IN_PROGRESS;
-    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "read_cb: %u starting initial read of file\n", finfo->id);
-  }
-
-  if (!error) {
-    size_t size = 0;
-    
-    switch_mutex_lock(finfo->mutex);
-    size = finfo->buffer->size();
-    switch_mutex_unlock(finfo->mutex);
-
-    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "read_cb: %u size is now %ld\n", finfo->id, size);
-    if (size < BUFFER_THROTTLE_LOW) {
-      std::vector<int16_t> pcm_data;
-      int8_t buf[INIT_BUFFER_SIZE];
-
-      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "read_cb: %u reading data\n", finfo->id);
-
-      size_t bytesRead = ::fread(buf, sizeof(int8_t), INIT_BUFFER_SIZE, finfo->fp);
-      if (bytesRead <= 0) {
-        if (::feof(finfo->fp)) {
-          switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "read_cb: %u eof\n", finfo->id);
-        }
-        else if (::ferror(finfo->fp)) {
-          switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "read_cb: %u error reading file\n", finfo->id);
-        }
-        else {
-          switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "read_cb: %u unknown error reading file\n", finfo->id);
-        }
-        finfo->status = Status_t::STATUS_FILE_COMPLETE;
-        return;
-      }
-
-      if (finfo->type == FileType_t::FILE_TYPE_MP3) {
-        pcm_data = convert_mp3_to_linear(finfo, buf, bytesRead);
-      } else {
-        pcm_data = std::vector<int16_t>(reinterpret_cast<int16_t*>(buf), reinterpret_cast<int16_t*>(buf) + bytesRead / 2);
-      }
-
-      switch_mutex_lock(finfo->mutex);
-
-      // Resize the buffer if necessary
-      if (finfo->buffer->capacity() - finfo->buffer->size() < pcm_data.size()) {
-        switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "write_cb %u growing buffer, size now %ld\n", finfo->id, finfo->buffer->size()); 
-        finfo->buffer->set_capacity(finfo->buffer->size() + std::max(pcm_data.size(), (size_t)BUFFER_GROW_SIZE));
-      }
-      
-      /* Push the data into the buffer */
-      finfo->buffer->insert(finfo->buffer->end(), pcm_data.data(), pcm_data.data() + pcm_data.size());
-      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "read_cb: %u wrote data, buffer size is now %ld\n", finfo->id, finfo->buffer->size());
-
-      switch_mutex_unlock(finfo->mutex);
-
-      if (bytesRead < INIT_BUFFER_SIZE) {
-        finfo->status = Status_t::STATUS_FILE_COMPLETE;
-        switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "read_cb: %u reached end of file, status is %s\n", finfo->id, status2String(finfo->status));
-      }
-    }
-
-    if (finfo->status == Status_t::STATUS_FILE_COMPLETE && finfo->loop) {
-      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "read_cb: %u looping\n", finfo->id);
-      ::fseek(finfo->fp, 0, SEEK_SET);
-      finfo->status = Status_t::STATUS_AWAITING_RESTART;
-    }
-
-    if (finfo->status != Status_t::STATUS_FILE_COMPLETE) {
-      // read more in 2 seconds
-      finfo->timer->expires_from_now(boost::posix_time::millisec(2000));
-      finfo->timer->async_wait(boost::bind(&read_cb, boost::placeholders::_1, finfo));
-    }
-    else {
-      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "read_cb: %u file complete, status %s loop %s\n", 
-        finfo->id, status2String(finfo->status), (finfo->loop ? "yes" : "no"));
-    }
-  } else {
-    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "read_cb: %u error (%d): %s\n", finfo->id, error.value(), error.message().c_str());
-
-    // Handle any errors
-  }
-}

mod_dub/file_loader.h

@@ -1,18 +0,0 @@
-#ifndef __FILE_LOADER_H__
-#define __FILE_LOADER_H__
-
-#include <switch.h>
-#include "common.h"
-
-extern "C" {
-
-switch_status_t init_file_loader();
-switch_status_t deinit_file_loader();
-
-int start_file_load(const char* path, int rate, int loop, int gain, switch_mutex_t* mutex, CircularBuffer_t* buffer);
-switch_status_t stop_file_load(int id);
-
-}
-
-#endif
-

mod_dub/mod_dub.c

@@ -8,6 +8,7 @@
 #include <switch.h>
 #include <switch_curl.h>
 #include "dub_glue.h"
+#include <string.h>
 
 /* Prototypes */
 SWITCH_MODULE_SHUTDOWN_FUNCTION(mod_dub_shutdown);
@@ -88,7 +89,6 @@ static switch_status_t dub_add_track(switch_core_session_t *session, char* track
   switch_channel_t *channel = switch_core_session_get_channel(session);
   switch_media_bug_t *bug = NULL;
   struct cap_cb *cb = NULL;
-  int offset = 0;
   int samples_per_second;
 
   switch_codec_implementation_t write_impl = { 0 };
@@ -102,36 +102,32 @@ static switch_status_t dub_add_track(switch_core_session_t *session, char* track
     cb =(struct cap_cb *) switch_core_session_alloc(session, sizeof(struct cap_cb));
     memset(cb, 0, sizeof(struct cap_cb));
     switch_mutex_init(&cb->mutex, SWITCH_MUTEX_NESTED, switch_core_session_get_pool(session));    
-    offset = 0;
   }
   else {
     /* retrieve the bug and search for an empty track */
     cb = (struct cap_cb *) switch_core_media_bug_get_user_data(bug);
-    while (offset < MAX_DUB_TRACKS && cb->tracks[offset].state != DUB_TRACK_STATE_INACTIVE) {
-      offset++;
-    }
-    if (offset == MAX_DUB_TRACKS) {
-      /* all tracks are in use */
-      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "dub_add_track: no available tracks\n");
-      return SWITCH_STATUS_FALSE;
-    }
   }
 
   switch_core_session_get_write_impl(session, &write_impl);
 	samples_per_second = !strcasecmp(write_impl.iananame, "g722") ? write_impl.actual_samples_per_second : write_impl.samples_per_second;
 
-  init_dub_track(&cb->tracks[offset], trackName, samples_per_second);
+  switch_mutex_lock(cb->mutex);
+  if (add_track(cb, trackName, samples_per_second) != SWITCH_STATUS_SUCCESS) {
+    switch_mutex_unlock(cb->mutex);
+    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "dub_add_track: error adding track %s\n", trackName);
+    return SWITCH_STATUS_FALSE;
+  }
 
   if (!bug) {
     switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "dub_add_track: adding bug for track %s\n", trackName);
     if (switch_core_media_bug_add(session, MY_BUG_NAME, NULL, capture_callback, (void *) cb, 0, SMBF_WRITE_REPLACE, &bug) != SWITCH_STATUS_SUCCESS) {
+      switch_mutex_unlock(cb->mutex);
       switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "dub_add_track: error adding bug!\n");
       return SWITCH_STATUS_FALSE;
     }
     switch_channel_set_private(channel, MY_BUG_NAME, bug);
   }
-
-  switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "dub_add_track: added track %s at offset %d\n", trackName, offset);
+  switch_mutex_unlock(cb->mutex);
 
   return SWITCH_STATUS_SUCCESS;
 }
@@ -140,39 +136,17 @@ static switch_status_t dub_remove_track(switch_core_session_t *session, char* tr
   switch_channel_t *channel = switch_core_session_get_channel(session);
   switch_media_bug_t *bug = (switch_media_bug_t*) switch_channel_get_private(channel, MY_BUG_NAME);
   switch_status_t status = SWITCH_STATUS_FALSE;
-  dub_track_t *track = NULL;
   
   if (bug) {
     struct cap_cb *cb =(struct cap_cb *) switch_core_media_bug_get_user_data(bug);
+
     switch_mutex_lock(cb->mutex);
-    for (int i = 0; i < MAX_DUB_TRACKS && track == NULL; i++) {
-      if (cb->tracks[i].state != DUB_TRACK_STATE_INACTIVE && strcmp(cb->tracks[i].trackName, trackName) == 0) {
-        track = &cb->tracks[i];
-        switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "dub_remove_track: removing track %s at offset %d\n", trackName, i);
-        break;
-      }
-    }
-
-    if (track) {
-      int count = 0;
-
-      remove_dub_track(track);
-
-      /* check if this is the last bug */
-      for (int i = 0; i < MAX_DUB_TRACKS; i++) {
-        if (cb->tracks[i].state != DUB_TRACK_STATE_INACTIVE) count++;
-      }
- 
-      if (count == 0) {
-        switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "dub_remove_track: removing bug after removing last track\n");
-        dub_session_cleanup(session, 0, bug);
-      }
-      status = SWITCH_STATUS_SUCCESS;
-    }
-    else {
-      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "dub_remove_track: track %s not found\n", trackName);
-    }
+    status = remove_dub_track(cb, trackName);
     switch_mutex_unlock(cb->mutex);
+    if (status != SWITCH_STATUS_SUCCESS) {
+      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "dub_remove_track: error removing track %s\n", trackName);
+      return SWITCH_STATUS_FALSE;
+    }
   }
   else {
     switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "dub_remove_track: bug not found\n");
@@ -185,24 +159,14 @@ static switch_status_t dub_silence_track(switch_core_session_t *session, char* t
   switch_channel_t *channel = switch_core_session_get_channel(session);
   switch_media_bug_t *bug = (switch_media_bug_t*) switch_channel_get_private(channel, MY_BUG_NAME);
   switch_status_t status = SWITCH_STATUS_FALSE;
-  dub_track_t *track = NULL;
   
   if (bug) {
     struct cap_cb *cb =(struct cap_cb *) switch_core_media_bug_get_user_data(bug);
-    switch_mutex_lock(cb->mutex);
-    for (int i = 0; i < MAX_DUB_TRACKS; i++) {
-      if (cb->tracks[i].state != DUB_TRACK_STATE_INACTIVE && strcmp(cb->tracks[i].trackName, trackName) == 0) {
-        track = &cb->tracks[i];
-        switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "dub_silence_track: silencing track %s at offset %d\n", trackName, i);
-        break;
-      }
+    status = silence_dub_track(cb, trackName);
+    if (status != SWITCH_STATUS_SUCCESS) {
+      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "dub_silence_track: error silencing track %s\n", trackName);
+      return SWITCH_STATUS_FALSE;
     }
-
-    if (track) {
-      silence_dub_track(track);
-      status = SWITCH_STATUS_SUCCESS;
-    }
-    switch_mutex_unlock(cb->mutex);
   }
   else {
     switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "dub_silence_track: bug not found\n");
@@ -215,24 +179,14 @@ static switch_status_t dub_play_on_track(switch_core_session_t *session, char* t
   switch_channel_t *channel = switch_core_session_get_channel(session);
   switch_media_bug_t *bug = (switch_media_bug_t*) switch_channel_get_private(channel, MY_BUG_NAME);
   switch_status_t status = SWITCH_STATUS_FALSE;
-  dub_track_t *track = NULL;
   
   if (bug) {
     struct cap_cb *cb =(struct cap_cb *) switch_core_media_bug_get_user_data(bug);
-    switch_mutex_lock(cb->mutex);
-    for (int i = 0; i < MAX_DUB_TRACKS; i++) {
-      if (cb->tracks[i].state != DUB_TRACK_STATE_INACTIVE && strcmp(cb->tracks[i].trackName, trackName) == 0) {
-        track = &cb->tracks[i];
-        switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, 
-          "dub_play_on_track: playing %s on track %s at offset %d with gain %d\n", url, trackName, i, gain);
-        break;
-      }
+    status = play_dub_track(cb, trackName, url, loop, gain);
+    if (status != SWITCH_STATUS_SUCCESS) {
+      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "dub_play_on_track: error playing %s on track %s\n", url, trackName);
+      return SWITCH_STATUS_FALSE;
     }
-
-    if (track) {
-      status = play_dub_track(track, cb->mutex, url, loop, gain);
-    }
-    switch_mutex_unlock(cb->mutex);
   }
   return status;
 }
@@ -241,37 +195,30 @@ static switch_status_t dub_say_on_track(switch_core_session_t *session, char* tr
   switch_channel_t *channel = switch_core_session_get_channel(session);
   switch_media_bug_t *bug = (switch_media_bug_t*) switch_channel_get_private(channel, MY_BUG_NAME);
   switch_status_t status = SWITCH_STATUS_FALSE;
-  dub_track_t *track = NULL;
   
   if (bug) {
     struct cap_cb *cb =(struct cap_cb *) switch_core_media_bug_get_user_data(bug);
-    switch_mutex_lock(cb->mutex);
-    for (int i = 0; i < MAX_DUB_TRACKS; i++) {
-      if (cb->tracks[i].state != DUB_TRACK_STATE_INACTIVE && strcmp(cb->tracks[i].trackName, trackName) == 0) {
-        track = &cb->tracks[i];
-        break;
-      }
+    status = say_dub_track(cb, trackName, text, gain);
+    if (status != SWITCH_STATUS_SUCCESS) {
+      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "dub_say_on_track: error saying %s on track %s\n", text, trackName);
+      return SWITCH_STATUS_FALSE;
     }
-
-    if (track) {
-      status = say_dub_track(track, cb->mutex, text, gain);
-    }
-    switch_mutex_unlock(cb->mutex);
   }
   return status;
 }
 
 #define DUB_API_SYNTAX "<uuid> [addTrack|removeTrack|silenceTrack|playOnTrack|sayOnTrack|setGain] track [url|text|gain] [gain] [loop]"
+#define MAX_PARAMS 6
 SWITCH_STANDARD_API(dub_function)
 {
-	char *mycmd = NULL, *argv[6] = { 0 };
+	char *mycmd = NULL, *argv[MAX_PARAMS] = { 0 };
 	int argc = 0;
   int error_written = 0;
 	switch_status_t status = SWITCH_STATUS_FALSE;
 
 	if (!zstr(cmd) && (mycmd = strdup(cmd))) {
     switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "dub_function: %s\n", mycmd);
-		argc = switch_separate_string(mycmd, ' ', argv, (sizeof(argv) / sizeof(argv[0])));
+    argc = switch_separate_string(mycmd, ' ', argv, (sizeof(argv) / sizeof(argv[0])));
 	}
 
   if (zstr(cmd) || argc < 3 || zstr(argv[0]) || zstr(argv[1]) || zstr(argv[2])) {
@@ -302,6 +249,7 @@ SWITCH_STANDARD_API(dub_function)
       }
       else if (0 == strcmp(action, "playOnTrack")) {
         if (argc < 4) {
+          switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_INFO, "playOnTrack requires a url\n");
           stream->write_function(stream, "-USAGE: %s\n", DUB_API_SYNTAX);
           error_written = 1;
         }

mod_dub/mod_dub.h

@@ -17,42 +17,11 @@
 /* per-channel data */
 typedef void (*responseHandler_t)(switch_core_session_t* session, const char* json, const char* bugname, const char* details);
 
-typedef enum {
-  DUB_TRACK_STATE_INACTIVE = 0,
-  DUB_TRACK_STATE_READY,
-  DUB_TRACK_STATE_ACTIVE,
-  DUB_TRACK_STATE_PAUSED
-} dub_state_t;
-
-typedef enum {
-  DUB_GENERATOR_TYPE_UNKNOWN = 0,
-  DUB_GENERATOR_TYPE_HTTP,
-  DUB_GENERATOR_TYPE_FILE,
-  DUB_GENERATOR_TYPE_TTS
-} dub_generator_t;
-
-typedef enum {
-  DUB_TRACK_EVENT_PLAY = 0,
-  DUB_TRACK_EVENT_STOP,
-  DUB_TRACK_EVENT_PAUSE,
-  DUB_TRACK_EVENT_RESUME
-} dub_event_t;
-
-
-typedef struct dub_track {
-  dub_state_t state;
-  dub_generator_t generator;
-  char* trackName;
-  int sampleRate;
-  int gain;
-  void* circularBuffer;
-  int generatorId;
-} dub_track_t;
 
 struct cap_cb {
 	switch_mutex_t *mutex;
   int gain;
-  dub_track_t tracks[MAX_DUB_TRACKS];
+  void *tracks[MAX_DUB_TRACKS];
 };
 
 

mod_dub/mpg_decode.cpp

@@ -0,0 +1,58 @@
+
+#include "mpg_decode.h"
+#include "vector_math.h"
+
+std::vector<int16_t> convert_mp3_to_linear(mpg123_handle *mh, int gain, int8_t *data, size_t len) {
+  std::vector<int16_t> linear_data;
+  int eof = 0;
+  int mp3err = 0;
+
+  if(mpg123_feed(mh, (const unsigned char*) data, len) == MPG123_OK) {
+    while(!eof) {
+      size_t usedlen = 0;
+      off_t frame_offset;
+      unsigned char* audio;
+
+      int decode_status = mpg123_decode_frame(mh, &frame_offset, &audio, &usedlen);
+
+      switch(decode_status) {
+        case MPG123_NEW_FORMAT:
+          continue;
+
+        case MPG123_OK:
+          {
+            size_t samples = usedlen / sizeof(int16_t);
+            linear_data.insert(linear_data.end(), reinterpret_cast<int16_t*>(audio), reinterpret_cast<int16_t*>(audio) + samples);
+          }
+          break;
+
+        case MPG123_DONE:
+        case MPG123_NEED_MORE:
+          eof = 1;
+          break;
+
+        case MPG123_ERR:
+        default:
+          if(++mp3err >= 5) {
+            switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Decoder Error!\n");
+            eof = 1;
+          }
+      }
+
+      if (eof)
+        break;
+
+      mp3err = 0;
+    }
+
+    if (gain != 0) {
+      vector_change_sln_volume_granular(linear_data.data(), linear_data.size(), gain);
+    }
+  }
+  else {
+    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error feeding data to mpg123\n");
+  }
+
+
+  return linear_data;
+}

mod_dub/mpg_decode.h

@@ -0,0 +1,11 @@
+
+#ifndef MPG_DECODE_H
+#define MPG_DECODE_H
+
+#include <vector>
+#include <mpg123.h>
+#include "switch.h"
+
+std::vector<int16_t> convert_mp3_to_linear(mpg123_handle *mh, int gain, int8_t *data, size_t len);
+
+#endif

mod_dub/track.cpp

@@ -0,0 +1,143 @@
+#include "track.h"
+#include "ap_file.h"
+#include "ap_http.h"
+#include "switch.h"
+
+#define INIT_BUFFER_SIZE (80000)
+
+Track::Track(const std::string& trackName, int sampleRate) : _trackName(trackName), _sampleRate(sampleRate), 
+  _buffer(INIT_BUFFER_SIZE), _stopping(false)
+{
+}
+
+Track::~Track() {
+  removeAllAudio();
+  switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Track::~Track: track %s\n", _trackName.c_str());
+}
+
+/**
+ * @brief called when an audio producer has finished retrieving the audio.
+ * If we have another producer queued, then start it.
+ * 
+ * @param hasError 
+ * @param errMsg 
+ */
+void Track::onPlayDone(bool hasError, const std::string& errMsg) {
+  switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Track::onPlayDone %s\n", _trackName.c_str());
+
+  if (!_stopping) {
+    if (hasError) switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "onPlayDone: error: %s\n", errMsg.c_str());
+    std::lock_guard<std::mutex> lock(_mutex);
+    _apQueue.pop();
+    if (!_apQueue.empty()) {
+      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "onPlayDone: starting queued audio on track %s\n", _trackName.c_str());
+      _apQueue.front()->start(std::bind(&Track::onPlayDone, this, std::placeholders::_1, std::placeholders::_2));
+    }
+  }
+  else {
+    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "onPlayDone: track %s stopping\n", _trackName.c_str());
+  }
+}
+
+void Track::queueFileAudio(const std::string& path, int gain, bool loop) {
+  bool startIt = false;
+  if (_stopping) return;
+
+  auto ap = std::make_shared<AudioProducerFile>(_mutex, _buffer, _sampleRate);
+  ap->queueFileAudio(path, gain, loop);
+  {
+    std::lock_guard<std::mutex> lock(_mutex);
+    _apQueue.push(ap);
+    startIt = _apQueue.size() == 1;
+  }
+
+  if (startIt) {
+    try {
+      ap->start(std::bind(&Track::onPlayDone, this, std::placeholders::_1, std::placeholders::_2));
+    } catch (std::exception& e) {
+      onPlayDone(true, e.what());
+    }
+  }
+}
+
+void Track::queueHttpGetAudio(const std::string& url, int gain, bool loop) {
+  bool startIt = false;
+  if (_stopping) return;
+  auto ap = std::make_shared<AudioProducerHttp>(_mutex, _buffer, _sampleRate);
+  ap->queueHttpGetAudio(url, gain, loop);
+  {
+    std::lock_guard<std::mutex> lock(_mutex);
+    _apQueue.push(ap);
+    startIt = _apQueue.size() == 1;
+  }
+
+  if (startIt) {
+    try {
+      ap->start(std::bind(&Track::onPlayDone, this, std::placeholders::_1, std::placeholders::_2));
+    } catch (std::exception& e) {
+      onPlayDone(true, e.what());
+    }
+  }
+}
+
+void Track::queueHttpPostAudio(const std::string& url, int gain, bool loop) {
+  bool startIt = false;
+  if (_stopping) return;
+  auto ap = std::make_shared<AudioProducerHttp>(_mutex, _buffer, _sampleRate);
+  ap->queueHttpPostAudio(url, gain, loop);
+  {
+    std::lock_guard<std::mutex> lock(_mutex);
+    _apQueue.push(ap);
+    startIt = _apQueue.size() == 1;
+  }
+
+  if (startIt) {
+    try {
+      ap->start(std::bind(&Track::onPlayDone, this, std::placeholders::_1, std::placeholders::_2));
+    } catch (std::exception& e) {
+      onPlayDone(true, e.what());
+    }
+  }
+}
+
+void Track::queueHttpPostAudio(const std::string& url, const std::string& body, std::vector<std::string>& headers, int gain, bool loop) {
+  bool startIt = false;
+  if (_stopping) return;
+  auto ap = std::make_shared<AudioProducerHttp>(_mutex, _buffer, _sampleRate);
+    ap->queueHttpPostAudio(url, body, headers, gain, loop);
+  {
+    std::lock_guard<std::mutex> lock(_mutex);
+    _apQueue.push(ap);
+    startIt = _apQueue.size() == 1;
+  }
+
+  if (startIt) {
+    try {
+      ap->start(std::bind(&Track::onPlayDone, this, std::placeholders::_1, std::placeholders::_2));
+    } catch (std::exception& e) {
+      onPlayDone(true, e.what());
+    }
+  }
+}
+
+ bool Track::hasAudio() {
+  if (_stopping) return false;
+  std::lock_guard<std::mutex> lock(_mutex); 
+  return hasAudio_NoLock();
+}
+
+void Track::removeAllAudio() {
+  _stopping = true;
+  std::queue<std::shared_ptr<AudioProducer>> apQueueCopy;
+  {
+    std::lock_guard<std::mutex> lock(_mutex); 
+    apQueueCopy = _apQueue;
+    _apQueue = std::queue<std::shared_ptr<AudioProducer>>();
+  }
+  
+  while (!apQueueCopy.empty()) {
+    auto ap = apQueueCopy.front();
+    apQueueCopy.pop();
+    ap->stop();
+  }
+}

mod_dub/track.h

@@ -0,0 +1,52 @@
+#ifndef  __TRACK_H__
+#define __TRACK_H__
+
+#include <mutex>
+#include <queue>
+#include "common.h"
+#include "ap.h"
+
+class Track {
+public:
+  Track(const std::string& trackName, int sampleRate);
+  ~Track();
+
+  /* audio production methods */
+  void queueHttpGetAudio(const std::string& url, int gain = 0, bool loop = false);
+  void queueHttpPostAudio(const std::string& url, int gain = 0, bool loop = false);
+  void queueHttpPostAudio(const std::string& url, const std::string& body, std::vector<std::string>& headers, int gain = 0, bool loop = false);
+  void queueFileAudio(const std::string& path, int gain = 0, bool loop = false);
+  void removeAllAudio();
+
+  void onPlayDone(bool hasError, const std::string& errMsg);
+
+  std::string& getTrackName() { return _trackName; }
+
+  /* audio playout methods */
+  bool hasAudio();
+  inline bool hasAudio_NoLock() const {
+    return !_stopping && !_buffer.empty();
+  }
+
+
+  int retrieveAndClearAudio(int16_t* buf, int desiredSamples) {
+    std::lock_guard<std::mutex> lock(_mutex);
+    int samplesToCopy = std::min(static_cast<int>(_buffer.size()), desiredSamples);
+    std::copy_n(_buffer.begin(), samplesToCopy, buf);
+    _buffer.erase(_buffer.begin(), _buffer.begin() + samplesToCopy);
+    return samplesToCopy;
+  }
+
+private:
+  std::string _trackName;
+  int _sampleRate;
+  std::mutex _mutex;
+  CircularBuffer_t _buffer;
+  std::queue<std::shared_ptr<AudioProducer>> _apQueue;
+  bool _stopping;
+};
+
+
+
+
+#endif

mod_dub/tts_vendor_parser.cpp

@@ -0,0 +1,118 @@
+#include "tts_vendor_parser.h"
+
+#include <sstream>
+#include <switch_json.h>
+#include <map>
+
+switch_status_t elevenlabs_parse_text(const std::map<std::string, std::string>& params, const std::string& text, 
+  std::string& url, std::string& body, std::vector<std::string>& headers) {
+
+  std::string api_key;
+  std::string voice_name;
+  std::string model_id;
+  std::string similarity_boost;
+  std::string stability;
+  std::string style;
+  std::string use_speaker_boost;
+  std::string optimize_streaming_latency;
+
+  for (const auto& pair : params) {
+    if (pair.first == "api_key") {
+      api_key = pair.second;
+    } else if (pair.first == "voice") {
+      voice_name = pair.second;
+    } else if (pair.first == "model_id") {
+      model_id = pair.second;
+    } else if (pair.first == "similarity_boost") {
+      similarity_boost = pair.second;
+    } else if (pair.first == "stability") {
+      stability = pair.second;
+    } else if (pair.first == "style") {
+      style = pair.second;
+    } else if (pair.first == "use_speaker_boost") {
+      use_speaker_boost = pair.second;
+    } else if (pair.first == "modeloptimize_streaming_latency_id") {
+      optimize_streaming_latency = pair.second;
+    }
+  }
+
+  if (api_key.empty()) {
+    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "elevenlabs_parse_text: no api_key provided\n");
+    return SWITCH_STATUS_FALSE;
+  }
+  if (model_id.empty()) {
+    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "elevenlabs_parse_text: no model_id provided\n");
+    return SWITCH_STATUS_FALSE;
+  }
+  if (optimize_streaming_latency.empty()) {
+    optimize_streaming_latency = "2";
+  }
+
+  // URL
+  std::ostringstream url_stream;
+  url_stream << "https://api.elevenlabs.io/v1/text-to-speech/" << voice_name << "/stream?";
+  url_stream << "optimize_streaming_latency=" << optimize_streaming_latency << "&output_format=mp3_44100_128";
+  url = url_stream.str();
+
+  /* create the JSON body */
+  cJSON * jResult = cJSON_CreateObject();
+  cJSON_AddStringToObject(jResult, "model_id", model_id.c_str());
+  cJSON_AddStringToObject(jResult, "text", text.c_str());
+  if (!similarity_boost.empty() || !style.empty() || !use_speaker_boost.empty() || !stability.empty()) {
+    cJSON * jVoiceSettings = cJSON_CreateObject();
+    cJSON_AddItemToObject(jResult, "voice_settings", jVoiceSettings);
+    if (!similarity_boost.empty()) {
+      cJSON_AddStringToObject(jVoiceSettings, "similarity_boost", similarity_boost.c_str());
+    }
+    if (!style.empty()) {
+      cJSON_AddStringToObject(jVoiceSettings, "style", style.c_str());
+    }
+    if (!use_speaker_boost.empty()) {
+      cJSON_AddStringToObject(jVoiceSettings, "use_speaker_boost", use_speaker_boost.c_str());
+    }
+    if (!stability.empty()) {
+      cJSON_AddStringToObject(jVoiceSettings, "stability", stability.c_str());
+    }
+  }
+  char* _body = cJSON_PrintUnformatted(jResult);
+  body = _body;
+
+  cJSON_Delete(jResult);
+  free(_body);
+
+  // Create headers
+  headers.push_back("xi-api-key: " + api_key);
+  headers.push_back("Content-Type: application/json");
+
+  return SWITCH_STATUS_SUCCESS;
+}
+
+switch_status_t tts_vendor_parse_text(const std::string& say, std::string& url, std::string& body, std::vector<std::string>& headers) {
+  size_t start = say.find("{") + 1;
+  size_t end = say.find("}");
+
+  std::string text = say.substr(end + 1);
+
+  std::string params_string = say.substr(start, end - start);
+  std::istringstream ss(params_string);
+  std::map<std::string, std::string> params;
+
+  while (ss.good()) {
+    std::string substr;
+    getline(ss, substr, ',');
+    substr.erase(0, substr.find_first_not_of(' '));
+
+    size_t equal_pos = substr.find("=");
+    std::string key = substr.substr(0, equal_pos);
+    std::string value = substr.substr(equal_pos + 1, substr.size());
+
+    params[key] = value;
+  }
+
+  if (params["vendor"] == "elevenlabs") {
+    return elevenlabs_parse_text(params, text, url, body, headers);
+  } else {
+    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "tts_vendor_parse_text: There is no available parser for text\n");
+    return SWITCH_STATUS_FALSE;
+  }
+}

mod_dub/tts_vendor_parser.h

@@ -0,0 +1,12 @@
+#ifndef __MOD_DUB_TTS_VENDOR_PARSER_H__
+#define __MOD_DUB_TTS_VENDOR_PARSER_H__
+
+#include <string>
+#include <vector>
+#include <switch.h>
+#include "common.h"
+
+
+switch_status_t tts_vendor_parse_text(const std::string& say, std::string& url, std::string& body, std::vector<std::string>& headers);
+
+#endif

mod_dub/vector_math.cpp

@@ -0,0 +1,211 @@
+#include "vector_math.h"
+#include <assert.h>
+#include <string.h>
+
+#define GRANULAR_VOLUME_MAX (50)
+#define SMAX 32767
+#define SMIN (-32768)
+#define normalize_to_16bit_basic(n) if (n > SMAX) n = SMAX; else if (n < SMIN) n = SMIN;
+#define normalize_volume_granular(x) if (x > GRANULAR_VOLUME_MAX) x = GRANULAR_VOLUME_MAX; if (x < -GRANULAR_VOLUME_MAX) x = -GRANULAR_VOLUME_MAX;
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+#if defined(USE_AVX2)
+#include <immintrin.h>
+#pragma message("Using AVX2 SIMD.")
+void vector_add(int16_t* a, int16_t* b, size_t len) {
+    size_t i = 0;
+    for (; i + 15 < len; i += 16) {
+        __m256i va = _mm256_loadu_si256((const __m256i*)(a + i));
+        __m256i vb = _mm256_loadu_si256((const __m256i*)(b + i));
+        __m256i vc = _mm256_add_epi16(va, vb);
+        _mm256_storeu_si256((__m256i*)(a + i), vc);
+    }
+    for (; i < len; ++i) {
+        a[i] += b[i];
+    }
+}
+void vector_normalize(int16_t* a, size_t len) {
+  __m256i max_val = _mm256_set1_epi16(SMAX);
+  __m256i min_val = _mm256_set1_epi16(SMIN);
+
+  size_t i = 0;
+  for (; i + 15 < len; i += 16) {
+      __m256i values = _mm256_loadu_si256((__m256i*)(a + i));
+      __m256i gt_max = _mm256_cmpgt_epi16(values, max_val);
+      __m256i lt_min = _mm256_cmpgt_epi16(min_val, values);
+      values = _mm256_blendv_epi8(values, max_val, gt_max);
+      values = _mm256_blendv_epi8(values, min_val, lt_min);
+      _mm256_storeu_si256((__m256i*)(a + i), values);
+  }
+
+  // Process remaining elements
+  for (; i < len; ++i) {
+      if (a[i] > SMAX) a[i] = SMAX;
+      else if (a[i] < SMIN) a[i] = SMIN;
+  }
+}
+typedef union {
+    int16_t* data;
+    __m256i* fp_avx2;
+} vector_data_t;
+
+void vector_change_sln_volume_granular(int16_t* data, uint32_t samples, int32_t vol) {
+  float newrate = 0;
+  static const float pos[GRANULAR_VOLUME_MAX] = {
+		1.122018,   1.258925,   1.412538,   1.584893,   1.778279,   1.995262,   2.238721,   2.511887,   2.818383,   3.162278,
+		3.548134,   3.981072,   4.466835,   5.011872,   5.623413,   6.309574,   7.079458,   7.943282,   8.912509,  10.000000,
+		11.220183,  12.589254,  14.125375,  15.848933,  17.782795,  19.952621,  22.387213,  25.118862,  28.183832,  31.622776,
+		35.481335,  39.810719,  44.668358,  50.118729,  56.234131,  63.095726,  70.794586,  79.432816,  89.125107, 100.000000,
+		112.201836, 125.892517, 141.253784, 158.489334, 177.827942, 199.526215, 223.872070, 251.188705, 281.838318, 316.227753
+  };
+  static const float neg[GRANULAR_VOLUME_MAX] = {
+		0.891251, 0.794328, 0.707946, 0.630957, 0.562341, 0.501187, 0.446684, 0.398107, 0.354813, 0.316228,
+		0.281838, 0.251189, 0.223872, 0.199526, 0.177828, 0.158489, 0.141254, 0.125893, 0.112202, 0.100000,
+		0.089125, 0.079433, 0.070795, 0.063096, 0.056234, 0.050119, 0.044668, 0.039811, 0.035481, 0.031623,
+		0.028184, 0.025119, 0.022387, 0.019953, 0.017783, 0.015849, 0.014125, 0.012589, 0.011220, 0.010000,
+		0.008913, 0.007943, 0.007079, 0.006310, 0.005623, 0.005012, 0.004467, 0.003981, 0.003548, 0.000000  // NOTE mapped -50 dB ratio to total silence instead of 0.003162
+  };
+  const float* chart;
+  uint32_t i = abs(vol) - 1;
+
+  if (vol == 0) return;
+  normalize_volume_granular(vol);
+
+  chart = vol > 0 ? pos : neg;
+
+  newrate = chart[i];
+
+  if (newrate) {
+    __m256 scale_factor_reg = _mm256_set1_ps(newrate);
+    uint32_t processed_samples = samples - (samples % 8); // Ensure we process only multiples of 8
+    for (uint32_t i = 0; i < processed_samples; i += 8) {
+      __m128i data_ = _mm_loadu_si128((__m128i*)(data + i));
+      __m256i data_32 = _mm256_cvtepi16_epi32(data_);
+
+      __m256 data_float = _mm256_cvtepi32_ps(data_32);
+      __m256 result = _mm256_mul_ps(data_float, scale_factor_reg);
+
+      __m256i result_32 = _mm256_cvtps_epi32(result);
+
+      // Handle saturation
+      __m256i min_val = _mm256_set1_epi32(SMIN);
+      __m256i max_val = _mm256_set1_epi32(SMAX);
+      result_32 = _mm256_min_epi32(result_32, max_val);
+      result_32 = _mm256_max_epi32(result_32, min_val);
+
+      __m128i result_16 = _mm_packs_epi32(_mm256_castsi256_si128(result_32), _mm256_extractf128_si256(result_32, 1));
+
+      _mm_storeu_si128((__m128i*)(data + i), result_16);
+    }
+
+    // Process any remaining samples
+    for (uint32_t i = processed_samples; i < samples; i++) {
+      int32_t tmp = (int32_t)(data[i] * newrate);
+      tmp = tmp > SMAX ? SMAX : (tmp < SMIN ? SMIN : tmp);
+      data[i] = (int16_t)tmp;
+    }
+  }
+}
+#elif defined(USE_SSE2)
+#include <emmintrin.h>
+#pragma message("Using SSE2 SIMD.")
+void vector_add(int16_t* a, int16_t* b, size_t len) {
+  size_t i = 0;
+  for (; i + 7 < len; i += 8) {
+    __m128i va = _mm_loadu_si128((const __m128i*)(a + i));
+    __m128i vb = _mm_loadu_si128((const __m128i*)(b + i));
+    __m128i vc = _mm_add_epi16(va, vb);
+    _mm_storeu_si128((__m128i*)(a + i), vc);
+  }
+  for (; i < len; ++i) {
+    a[i] += b[i];
+  }
+}
+void vector_normalize(int16_t* a, size_t len) {
+  __m128i max_val = _mm_set1_epi16(SMAX);
+  __m128i min_val = _mm_set1_epi16(SMIN);
+
+  size_t i = 0;
+  for (; i + 7 < len; i += 8) {
+    __m128i values = _mm_loadu_si128((__m128i*)(a + i));
+    __m128i gt_max = _mm_cmpgt_epi16(values, max_val);
+    __m128i lt_min = _mm_cmpgt_epi16(min_val, values);
+    __m128i max_masked = _mm_and_si128(gt_max, max_val);
+    __m128i min_masked = _mm_and_si128(lt_min, min_val);
+    __m128i other_masked = _mm_andnot_si128(_mm_or_si128(gt_max, lt_min), values);
+    values = _mm_or_si128(_mm_or_si128(max_masked, min_masked), other_masked);
+    _mm_storeu_si128((__m128i*)(a + i), values);
+  }
+
+  // Process remaining elements
+  for (; i < len; ++i) {
+    if (a[i] > SMAX) a[i] = SMAX;
+    else if (a[i] < SMIN) a[i] = SMIN;
+  }
+}
+
+typedef union {
+    int16_t* data;
+    __m128i* fp_sse2;
+} vector_data_t;
+
+#else
+void vector_add(int16_t* a, int16_t* b, size_t len) {
+    for (size_t i = 0; i < len; i++) {
+        a[i] += b[i];
+    }
+}
+void vector_normalize(int16_t* a, size_t len) {
+  for (size_t i = 0; i < len; i++) {
+    normalize_to_16bit_basic(a[i]);
+  }
+}
+void vector_change_sln_volume_granular(int16_t* data, uint32_t samples, int32_t vol) {
+  float newrate = 0;
+  static const float pos[GRANULAR_VOLUME_MAX] = {
+		1.122018,   1.258925,   1.412538,   1.584893,   1.778279,   1.995262,   2.238721,   2.511887,   2.818383,   3.162278,
+		3.548134,   3.981072,   4.466835,   5.011872,   5.623413,   6.309574,   7.079458,   7.943282,   8.912509,  10.000000,
+		11.220183,  12.589254,  14.125375,  15.848933,  17.782795,  19.952621,  22.387213,  25.118862,  28.183832,  31.622776,
+		35.481335,  39.810719,  44.668358,  50.118729,  56.234131,  63.095726,  70.794586,  79.432816,  89.125107, 100.000000,
+		112.201836, 125.892517, 141.253784, 158.489334, 177.827942, 199.526215, 223.872070, 251.188705, 281.838318, 316.227753
+  };
+  static const float neg[GRANULAR_VOLUME_MAX] = {
+		0.891251, 0.794328, 0.707946, 0.630957, 0.562341, 0.501187, 0.446684, 0.398107, 0.354813, 0.316228,
+		0.281838, 0.251189, 0.223872, 0.199526, 0.177828, 0.158489, 0.141254, 0.125893, 0.112202, 0.100000,
+		0.089125, 0.079433, 0.070795, 0.063096, 0.056234, 0.050119, 0.044668, 0.039811, 0.035481, 0.031623,
+		0.028184, 0.025119, 0.022387, 0.019953, 0.017783, 0.015849, 0.014125, 0.012589, 0.011220, 0.010000,
+		0.008913, 0.007943, 0.007079, 0.006310, 0.005623, 0.005012, 0.004467, 0.003981, 0.003548, 0.000000  // NOTE mapped -50 dB ratio to total silence instead of 0.003162
+  };
+  const float* chart;
+  uint32_t i;
+
+  if (vol == 0) return;
+
+  normalize_volume_granular(vol);
+
+  chart = vol > 0 ? pos : neg;
+
+  i = abs(vol) - 1;
+  assert(i < GRANULAR_VOLUME_MAX);
+  newrate = chart[i];
+  if (newrate) {
+    int32_t tmp;
+    uint32_t x;
+    int16_t *fp = data;
+
+    for (x = 0; x < samples; x++) {
+      tmp = (int32_t) (fp[x] * newrate);
+      normalize_to_16bit_basic(tmp);
+      fp[x] = (int16_t) tmp;
+    }
+  }
+}
+
+#endif
+
+#ifdef __cplusplus
+}
+#endif

mod_dub/vector_math.h

@@ -0,0 +1,20 @@
+#ifndef VECTOR_MATH_H
+#define VECTOR_MATH_H
+
+#include <stddef.h>
+#include <stdint.h>
+
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+void vector_add(int16_t* a, int16_t* b, size_t len);
+void vector_normalize(int16_t* a, size_t len);
+void vector_change_sln_volume_granular(int16_t* data, uint32_t samples, int32_t vol);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif

mod_google_transcribe/Makefile.am

@@ -2,7 +2,7 @@ include $(top_srcdir)/build/modmake.rulesam
 MODNAME=mod_google_transcribe
 
 mod_LTLIBRARIES = mod_google_transcribe.la
-mod_google_transcribe_la_SOURCES  = mod_google_transcribe.c google_glue.cpp
+mod_google_transcribe_la_SOURCES  = mod_google_transcribe.c google_glue.cpp google_glue_v1.cpp google_glue_v2.cpp
 mod_google_transcribe_la_CFLAGS   = $(AM_CFLAGS)
 mod_google_transcribe_la_CXXFLAGS = -I $(top_srcdir)/libs/googleapis/gens $(AM_CXXFLAGS) -std=c++17
 

mod_google_transcribe/generic_google_glue.h

@@ -0,0 +1,248 @@
+#ifndef __GENERIC_GOOGLE_GLUE_H__
+#define __GENERIC_GOOGLE_GLUE_H__
+
+#include <switch_json.h>
+
+template<typename Streamer>
+switch_bool_t google_speech_frame(switch_media_bug_t *bug, void* user_data) {
+	switch_core_session_t *session = switch_core_media_bug_get_session(bug);
+	struct cap_cb *cb = (struct cap_cb *) user_data;
+    if (cb->streamer && (!cb->wants_single_utterance || !cb->got_end_of_utterance)) {
+        Streamer* streamer = (Streamer *) cb->streamer;
+        uint8_t data[SWITCH_RECOMMENDED_BUFFER_SIZE];
+        switch_frame_t frame = {};
+        frame.data = data;
+        frame.buflen = SWITCH_RECOMMENDED_BUFFER_SIZE;
+
+        if (switch_mutex_trylock(cb->mutex) == SWITCH_STATUS_SUCCESS) {
+            while (switch_core_media_bug_read(bug, &frame, SWITCH_TRUE) == SWITCH_STATUS_SUCCESS && !switch_test_flag((&frame), SFF_CNG)) {
+                if (frame.datalen) {
+                    if (cb->vad && !streamer->isConnected()) {
+                        switch_vad_state_t state = switch_vad_process(cb->vad, (int16_t*) frame.data, frame.samples);
+                        if (state == SWITCH_VAD_STATE_START_TALKING) {
+                            switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_INFO, "detected speech, connect to google speech now\n");
+                            streamer->connect();
+                            cb->responseHandler(session, "vad_detected", cb->bugname);
+                        }
+                    }
+
+                    if (cb->resampler) {
+                        spx_int16_t out[SWITCH_RECOMMENDED_BUFFER_SIZE];
+                        spx_uint32_t out_len = SWITCH_RECOMMENDED_BUFFER_SIZE;
+                        spx_uint32_t in_len = frame.samples;
+                        size_t written;
+
+                        speex_resampler_process_interleaved_int(cb->resampler,
+                            (const spx_int16_t *) frame.data,
+                            (spx_uint32_t *) &in_len,
+                            &out[0],
+                            &out_len);
+                        streamer->write( &out[0], sizeof(spx_int16_t) * out_len);
+                    }
+                    else {
+                        streamer->write( frame.data, sizeof(spx_int16_t) * frame.samples);
+                    }
+                }
+            }
+            switch_mutex_unlock(cb->mutex);
+        }
+	}
+	return SWITCH_TRUE;
+}
+
+template<typename Streamer>
+switch_status_t google_speech_session_init(switch_core_session_t *session, responseHandler_t responseHandler,
+		switch_thread_start_t func, uint32_t to_rate, uint32_t samples_per_second, uint32_t channels, char* lang,
+		int interim, char *bugname, int single_utterance, int separate_recognition, int max_alternatives,
+		int profanity_filter, int word_time_offset, int punctuation, const char* model, int enhanced,
+		const char* hints, char* play_file, void **ppUserData) {
+
+	switch_channel_t *channel = switch_core_session_get_channel(session);
+	auto read_codec = switch_core_session_get_read_codec(session);
+	uint32_t sampleRate = read_codec->implementation->actual_samples_per_second;
+	struct cap_cb *cb;
+	int err;
+
+	cb =(struct cap_cb *) switch_core_session_alloc(session, sizeof(*cb));
+	strncpy(cb->sessionId, switch_core_session_get_uuid(session), MAX_SESSION_ID);
+	strncpy(cb->bugname, bugname, MAX_BUG_LEN);
+	cb->got_end_of_utterance = 0;
+	cb->wants_single_utterance = single_utterance;
+	if (play_file != NULL){
+		cb->play_file = 1;
+	}
+	
+	switch_mutex_init(&cb->mutex, SWITCH_MUTEX_NESTED, switch_core_session_get_pool(session));
+	if (sampleRate != to_rate) {
+		cb->resampler = speex_resampler_init(channels, sampleRate, to_rate, SWITCH_RESAMPLE_QUALITY, &err);
+	if (0 != err) {
+		switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_ERROR, "%s: Error initializing resampler: %s.\n",
+								switch_channel_get_name(channel), speex_resampler_strerror(err));
+		return SWITCH_STATUS_FALSE;
+	}
+	} else {
+		switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "%s: no resampling needed for this call\n", switch_channel_get_name(channel));
+	}
+	cb->responseHandler = responseHandler;
+
+	// allocate vad if we are delaying connecting to the recognizer until we detect speech
+	if (switch_channel_var_true(channel, "START_RECOGNIZING_ON_VAD")) {
+		cb->vad = switch_vad_init(sampleRate, channels);
+		if (cb->vad) {
+			const char* var;
+			int mode = 2;
+			int silence_ms = 150;
+			int voice_ms = 250;
+			int debug = 0;
+
+			if (var = switch_channel_get_variable(channel, "RECOGNIZER_VAD_MODE")) {
+				mode = atoi(var);
+			}
+			if (var = switch_channel_get_variable(channel, "RECOGNIZER_VAD_SILENCE_MS")) {
+				silence_ms = atoi(var);
+			}
+			if (var = switch_channel_get_variable(channel, "RECOGNIZER_VAD_VOICE_MS")) {
+				voice_ms = atoi(var);
+			}
+			if (var = switch_channel_get_variable(channel, "RECOGNIZER_VAD_VOICE_MS")) {
+				voice_ms = atoi(var);
+			}
+			switch_vad_set_mode(cb->vad, mode);
+			switch_vad_set_param(cb->vad, "silence_ms", silence_ms);
+			switch_vad_set_param(cb->vad, "voice_ms", voice_ms);
+			switch_vad_set_param(cb->vad, "debug", debug);
+		}
+	}
+
+	Streamer *streamer = NULL;
+	try {
+	    streamer = new Streamer(session, channels, lang, interim, to_rate, sampleRate, single_utterance, separate_recognition, max_alternatives,
+		    profanity_filter, word_time_offset, punctuation, model, enhanced, hints);
+	    cb->streamer = streamer;
+	} catch (std::exception& e) {
+	    switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_ERROR, "%s: Error initializing gstreamer: %s.\n", 
+		switch_channel_get_name(channel), e.what());
+	    return SWITCH_STATUS_FALSE;
+	}
+
+	if (!cb->vad) streamer->connect();
+
+	// create the read thread
+	switch_threadattr_t *thd_attr = NULL;
+	switch_memory_pool_t *pool = switch_core_session_get_pool(session);
+
+	switch_threadattr_create(&thd_attr, pool);
+	switch_threadattr_stacksize_set(thd_attr, SWITCH_THREAD_STACKSIZE);
+	switch_thread_create(&cb->thread, thd_attr, func, cb, pool);
+
+	*ppUserData = cb;
+	return SWITCH_STATUS_SUCCESS;
+}
+
+template<typename Streamer>
+switch_status_t google_speech_session_cleanup(switch_core_session_t *session, int channelIsClosing, switch_media_bug_t *bug) {
+	switch_channel_t *channel = switch_core_session_get_channel(session);
+
+	if (bug) {
+		struct cap_cb *cb = (struct cap_cb *) switch_core_media_bug_get_user_data(bug);
+		switch_mutex_lock(cb->mutex);
+
+		if (!switch_channel_get_private(channel, cb->bugname)) {
+			// race condition
+			switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_INFO, "%s Bug is not attached (race).\n", switch_channel_get_name(channel));
+			switch_mutex_unlock(cb->mutex);
+			return SWITCH_STATUS_FALSE;
+		}
+		switch_channel_set_private(channel, cb->bugname, NULL);
+
+		// stop playback if available
+		if (cb->play_file == 1){ 
+			if (switch_channel_test_flag(channel, CF_BROADCAST)) {
+				switch_channel_stop_broadcast(channel);
+			} else {
+				switch_channel_set_flag_value(channel, CF_BREAK, 1);
+			}
+		}
+
+		// close connection and get final responses
+		Streamer* streamer = (Streamer *) cb->streamer;
+
+		if (streamer) {
+			streamer->writesDone();
+
+			switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "google_speech_session_cleanup: GStreamer (%p) waiting for read thread to complete\n", (void*)streamer);
+			switch_status_t st;
+			switch_thread_join(&st, cb->thread);
+			switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "google_speech_session_cleanup:  GStreamer (%p) read thread completed\n", (void*)streamer);
+
+			delete streamer;
+			cb->streamer = NULL;
+		}
+
+		if (cb->resampler) {
+			speex_resampler_destroy(cb->resampler);
+		}
+		if (cb->vad) {
+			switch_vad_destroy(&cb->vad);
+			cb->vad = nullptr;
+		}
+		if (!channelIsClosing) {
+			switch_core_media_bug_remove(session, &bug);
+		}
+
+		switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "google_speech_session_cleanup: Closed stream\n");
+
+		switch_mutex_unlock(cb->mutex);
+
+		return SWITCH_STATUS_SUCCESS;
+	}
+
+	switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_INFO, "%s Bug is not attached.\n", switch_channel_get_name(channel));
+	return SWITCH_STATUS_FALSE;
+}
+
+template<typename PhraseSet>
+void google_speech_configure_grammar_hints(switch_core_session_t *session, switch_channel_t *channel, const char* hints, PhraseSet* phrase_set) {
+    float boost = -1;
+
+    // get boost setting for the phrase set in its entirety
+    if (switch_true(switch_channel_get_variable(channel, "GOOGLE_SPEECH_HINTS_BOOST"))) {
+        boost = (float) atof(switch_channel_get_variable(channel, "GOOGLE_SPEECH_HINTS_BOOST"));
+        switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "boost value: %f\n", boost);
+        phrase_set->set_boost(boost);
+    }
+
+    // hints are either a simple comma-separated list of phrases, or a json array of objects
+    // containing a phrase and a boost value
+    auto *jHint = cJSON_Parse((char *) hints);
+    if (jHint) {
+        int i = 0;
+        cJSON *jPhrase = NULL;
+        cJSON_ArrayForEach(jPhrase, jHint) {
+            cJSON *jItem = cJSON_GetObjectItem(jPhrase, "phrase");
+            if (jItem) {
+                auto* phrase = phrase_set->add_phrases();
+                phrase->set_value(cJSON_GetStringValue(jItem));
+                switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "phrase: %s\n", phrase->value().c_str());
+                if (cJSON_GetObjectItem(jPhrase, "boost")) {
+                    phrase->set_boost((float) cJSON_GetObjectItem(jPhrase, "boost")->valuedouble);
+                    switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "boost value: %f\n", phrase->boost());
+                }
+                i++;
+            }
+        }
+        cJSON_Delete(jHint);
+        switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "added %d hints\n", i);
+    }
+    else {
+        char *phrases[500] = { 0 };
+        int argc = switch_separate_string((char *) hints, ',', phrases, 500);
+        for (int i = 0; i < argc; i++) {
+            auto* phrase = phrase_set->add_phrases();
+            phrase->set_value(phrases[i]);
+        }
+        switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "added %d hints\n", argc);
+    }
+}
+
+#endif

mod_google_transcribe/google_glue.cpp

@@ -1,727 +1,27 @@
-#include <cstdlib>
-#include <algorithm>
-#include <future>
-
 #include <switch.h>
-#include <switch_json.h>
 #include <grpc++/grpc++.h>
-
-#include "google/cloud/speech/v1p1beta1/cloud_speech.grpc.pb.h"
-
-#include <switch_json.h>
+#include <unistd.h>
 
 #include "mod_google_transcribe.h"
-#include "simple_buffer.h"
-
-using google::cloud::speech::v1p1beta1::RecognitionConfig;
-using google::cloud::speech::v1p1beta1::Speech;
-using google::cloud::speech::v1p1beta1::SpeechContext;
-using google::cloud::speech::v1p1beta1::StreamingRecognizeRequest;
-using google::cloud::speech::v1p1beta1::StreamingRecognizeResponse;
-using google::cloud::speech::v1p1beta1::SpeakerDiarizationConfig;
-using google::cloud::speech::v1p1beta1::SpeechAdaptation;
-using google::cloud::speech::v1p1beta1::PhraseSet;
-using google::cloud::speech::v1p1beta1::PhraseSet_Phrase;
-using google::cloud::speech::v1p1beta1::RecognitionMetadata;
-using google::cloud::speech::v1p1beta1::RecognitionMetadata_InteractionType_DISCUSSION;
-using google::cloud::speech::v1p1beta1::RecognitionMetadata_InteractionType_PRESENTATION;
-using google::cloud::speech::v1p1beta1::RecognitionMetadata_InteractionType_PHONE_CALL;
-using google::cloud::speech::v1p1beta1::RecognitionMetadata_InteractionType_VOICEMAIL;
-using google::cloud::speech::v1p1beta1::RecognitionMetadata_InteractionType_PROFESSIONALLY_PRODUCED;
-using google::cloud::speech::v1p1beta1::RecognitionMetadata_InteractionType_VOICE_SEARCH;
-using google::cloud::speech::v1p1beta1::RecognitionMetadata_InteractionType_VOICE_COMMAND;
-using google::cloud::speech::v1p1beta1::RecognitionMetadata_InteractionType_DICTATION;
-using google::cloud::speech::v1p1beta1::RecognitionMetadata_MicrophoneDistance_NEARFIELD;
-using google::cloud::speech::v1p1beta1::RecognitionMetadata_MicrophoneDistance_MIDFIELD;
-using google::cloud::speech::v1p1beta1::RecognitionMetadata_MicrophoneDistance_FARFIELD;
-using google::cloud::speech::v1p1beta1::RecognitionMetadata_OriginalMediaType_AUDIO;
-using google::cloud::speech::v1p1beta1::RecognitionMetadata_OriginalMediaType_VIDEO;
-using google::cloud::speech::v1p1beta1::RecognitionMetadata_RecordingDeviceType_SMARTPHONE;
-using google::cloud::speech::v1p1beta1::RecognitionMetadata_RecordingDeviceType_PC;
-using google::cloud::speech::v1p1beta1::RecognitionMetadata_RecordingDeviceType_PHONE_LINE;
-using google::cloud::speech::v1p1beta1::RecognitionMetadata_RecordingDeviceType_VEHICLE;
-using google::cloud::speech::v1p1beta1::RecognitionMetadata_RecordingDeviceType_OTHER_OUTDOOR_DEVICE;
-using google::cloud::speech::v1p1beta1::RecognitionMetadata_RecordingDeviceType_OTHER_INDOOR_DEVICE;
-using google::cloud::speech::v1p1beta1::StreamingRecognizeResponse_SpeechEventType_END_OF_SINGLE_UTTERANCE;
-using google::rpc::Status;
-
-#define CHUNKSIZE (320)
-
-namespace {
-  int case_insensitive_match(std::string s1, std::string s2) {
-   std::transform(s1.begin(), s1.end(), s1.begin(), ::tolower);
-   std::transform(s2.begin(), s2.end(), s2.begin(), ::tolower);
-   if(s1.compare(s2) == 0)
-      return 1; //The strings are same
-   return 0; //not matched
-  }
-}
-class GStreamer;
-
-class GStreamer {
-public:
-	GStreamer(
-    switch_core_session_t *session, 
-    uint32_t channels, 
-    char* lang, 
-    int interim, 
-    uint32_t config_sample_rate,
-		uint32_t samples_per_second,
-    int single_utterance, 
-    int separate_recognition,
-		int max_alternatives, 
-    int profanity_filter, 
-    int word_time_offset, 
-    int punctuation, 
-    const char* model, 
-    int enhanced, 
-		const char* hints) : m_session(session), m_writesDone(false), m_connected(false), 
-      m_audioBuffer(CHUNKSIZE, 15) {
-  
-    const char* var;
-    const char* google_uri;
-    switch_channel_t *channel = switch_core_session_get_channel(session);
-
-    if (!(google_uri = switch_channel_get_variable(channel, "GOOGLE_SPEECH_TO_TEXT_URI"))) {
-      google_uri = "speech.googleapis.com";
-    }
-		if (var = switch_channel_get_variable(channel, "GOOGLE_APPLICATION_CREDENTIALS")) {
-			auto channelCreds = grpc::SslCredentials(grpc::SslCredentialsOptions());
-			auto callCreds = grpc::ServiceAccountJWTAccessCredentials(var);
-			auto creds = grpc::CompositeChannelCredentials(channelCreds, callCreds);
-			m_channel = grpc::CreateChannel(google_uri, creds);
-		}
-		else {
-			auto creds = grpc::GoogleDefaultCredentials();
-			m_channel = grpc::CreateChannel(google_uri, creds);
-		}
-
-  	m_stub = Speech::NewStub(m_channel);
-  		
-		auto* streaming_config = m_request.mutable_streaming_config();
-		RecognitionConfig* config = streaming_config->mutable_config();
-
-    streaming_config->set_interim_results(interim);
-    if (single_utterance == 1) {
-      switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "enable_single_utterance\n");
-      streaming_config->set_single_utterance(true);
-    }
-    else {
-      switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "enable_single_utterance is FALSE\n");
-      streaming_config->set_single_utterance(false);
-    }
-
-		config->set_language_code(lang);
-    switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "transcribe language %s \n", lang);
-    
-  	config->set_sample_rate_hertz(config_sample_rate);
-
-		config->set_encoding(RecognitionConfig::LINEAR16);
-
-    // the rest of config comes from channel vars
-
-    // number of channels in the audio stream (default: 1)
-    if (channels > 1) {
-      config->set_audio_channel_count(channels);
-      switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "audio_channel_count %d\n", channels);
-
-      // transcribe each separately?
-      if (separate_recognition == 1) {
-        config->set_enable_separate_recognition_per_channel(true);
-        switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "enable_separate_recognition_per_channel on\n");
-      }
-    }
-
-    // max alternatives
-    if (max_alternatives > 1) {
-      config->set_max_alternatives(max_alternatives);
-      switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "max_alternatives %d\n", max_alternatives);
-    }
-
-    // profanity filter
-    if (profanity_filter == 1) {
-      config->set_profanity_filter(true);
-      switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "profanity_filter\n");
-    }
-
-    // enable word offsets
-    if (word_time_offset == 1) {
-      config->set_enable_word_time_offsets(true);
-      switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "enable_word_time_offsets\n");
-    }
-
-    // enable automatic punctuation
-    if (punctuation == 1) {
-      config->set_enable_automatic_punctuation(true);
-      switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "enable_automatic_punctuation\n");
-    }
-    else {
-      config->set_enable_automatic_punctuation(false);
-      switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "disable_automatic_punctuation\n");
-    }
-
-    // speech model
-    if (model != NULL) {
-      config->set_model(model);
-      switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "speech model %s\n", model);
-    }
-
-    // use enhanced model
-    if (enhanced == 1) {
-      config->set_use_enhanced(true);
-      switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "use_enhanced\n");
-    }
-
-    // hints  
-    if (hints != NULL) {
-      auto* adaptation = config->mutable_adaptation();
-      auto* phrase_set = adaptation->add_phrase_sets();
-      auto *context = config->add_speech_contexts();
-      float boost = -1;
-
-      // get boost setting for the phrase set in its entirety
-      if (switch_true(switch_channel_get_variable(channel, "GOOGLE_SPEECH_HINTS_BOOST"))) {
-     	  boost = (float) atof(switch_channel_get_variable(channel, "GOOGLE_SPEECH_HINTS_BOOST"));
-        switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "boost value: %f\n", boost);
-        phrase_set->set_boost(boost);
-      }
-
-      // hints are either a simple comma-separated list of phrases, or a json array of objects
-      // containing a phrase and a boost value
-      auto *jHint = cJSON_Parse((char *) hints);
-      if (jHint) {
-        int i = 0;
-        cJSON *jPhrase = NULL;
-        cJSON_ArrayForEach(jPhrase, jHint) {
-          auto* phrase = phrase_set->add_phrases();
-          cJSON *jItem = cJSON_GetObjectItem(jPhrase, "phrase");
-          if (jItem) {
-            phrase->set_value(cJSON_GetStringValue(jItem));
-            switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "phrase: %s\n", phrase->value().c_str());
-            if (cJSON_GetObjectItem(jPhrase, "boost")) {
-              phrase->set_boost((float) cJSON_GetObjectItem(jPhrase, "boost")->valuedouble);
-              switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "boost value: %f\n", phrase->boost());
-            }
-            i++;
-          }
-        }
-        cJSON_Delete(jHint);
-        switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "added %d hints\n", i);
-      }
-      else {
-        char *phrases[500] = { 0 };
-        int argc = switch_separate_string((char *) hints, ',', phrases, 500);
-        for (int i = 0; i < argc; i++) {
-          auto* phrase = phrase_set->add_phrases();
-          phrase->set_value(phrases[i]);
-        }
-        switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "added %d hints\n", argc);
-      }
-    }
-
-    // alternative language
-    if (var = switch_channel_get_variable(channel, "GOOGLE_SPEECH_ALTERNATIVE_LANGUAGE_CODES")) {
-      char *alt_langs[3] = { 0 };
-      int argc = switch_separate_string((char *) var, ',', alt_langs, 3);
-      for (int i = 0; i < argc; i++) {
-        config->add_alternative_language_codes(alt_langs[i]);
-        switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "added alternative lang %s\n", alt_langs[i]);
-      }
-    }
-
-    // speaker diarization
-    if (var = switch_channel_get_variable(channel, "GOOGLE_SPEECH_SPEAKER_DIARIZATION")) {
-      auto* diarization_config = config->mutable_diarization_config();
-      diarization_config->set_enable_speaker_diarization(true);
-      switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "enabling speaker diarization\n", var);
-      if (var = switch_channel_get_variable(channel, "GOOGLE_SPEECH_SPEAKER_DIARIZATION_MIN_SPEAKER_COUNT")) {
-        int count = std::max(atoi(var), 1);
-        switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "setting min speaker count to %d\n", count);
-        diarization_config->set_min_speaker_count(count);
-      }
-      if (var = switch_channel_get_variable(channel, "GOOGLE_SPEECH_SPEAKER_DIARIZATION_MAX_SPEAKER_COUNT")) {
-        int count = std::max(atoi(var), 2);
-        switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "setting max speaker count to %d\n", count);
-        diarization_config->set_max_speaker_count(count);
-      }
-    }
-
-    // recognition metadata
-    auto* metadata = config->mutable_metadata();
-    if (var = switch_channel_get_variable(channel, "GOOGLE_SPEECH_METADATA_INTERACTION_TYPE")) {
-      if (case_insensitive_match("discussion", var)) metadata->set_interaction_type(RecognitionMetadata_InteractionType_DISCUSSION);
-      if (case_insensitive_match("presentation", var)) metadata->set_interaction_type(RecognitionMetadata_InteractionType_PRESENTATION);
-      if (case_insensitive_match("phone_call", var)) metadata->set_interaction_type(RecognitionMetadata_InteractionType_PHONE_CALL);
-      if (case_insensitive_match("voicemail", var)) metadata->set_interaction_type(RecognitionMetadata_InteractionType_VOICEMAIL);
-      if (case_insensitive_match("professionally_produced", var)) metadata->set_interaction_type(RecognitionMetadata_InteractionType_PROFESSIONALLY_PRODUCED);
-      if (case_insensitive_match("voice_search", var)) metadata->set_interaction_type(RecognitionMetadata_InteractionType_VOICE_SEARCH);
-      if (case_insensitive_match("voice_command", var)) metadata->set_interaction_type(RecognitionMetadata_InteractionType_VOICE_COMMAND);
-      if (case_insensitive_match("dictation", var)) metadata->set_interaction_type(RecognitionMetadata_InteractionType_DICTATION);
-    }
-    if (var = switch_channel_get_variable(channel, "GOOGLE_SPEECH_METADATA_INDUSTRY_NAICS_CODE")) {
-      metadata->set_industry_naics_code_of_audio(atoi(var));
-    }
-    if (var = switch_channel_get_variable(channel, "GOOGLE_SPEECH_METADATA_MICROPHONE_DISTANCE")) {
-      if (case_insensitive_match("nearfield", var)) metadata->set_microphone_distance(RecognitionMetadata_MicrophoneDistance_NEARFIELD);
-      if (case_insensitive_match("midfield", var)) metadata->set_microphone_distance(RecognitionMetadata_MicrophoneDistance_MIDFIELD);
-      if (case_insensitive_match("farfield", var)) metadata->set_microphone_distance(RecognitionMetadata_MicrophoneDistance_FARFIELD);
-    }
-    if (var = switch_channel_get_variable(channel, "GOOGLE_SPEECH_METADATA_ORIGINAL_MEDIA_TYPE")) {
-      if (case_insensitive_match("audio", var)) metadata->set_original_media_type(RecognitionMetadata_OriginalMediaType_AUDIO);
-      if (case_insensitive_match("video", var)) metadata->set_original_media_type(RecognitionMetadata_OriginalMediaType_VIDEO);
-    }
-    if (var = switch_channel_get_variable(channel, "GOOGLE_SPEECH_METADATA_RECORDING_DEVICE_TYPE")) {
-      if (case_insensitive_match("smartphone", var)) metadata->set_recording_device_type(RecognitionMetadata_RecordingDeviceType_SMARTPHONE);
-      if (case_insensitive_match("pc", var)) metadata->set_recording_device_type(RecognitionMetadata_RecordingDeviceType_PC);
-      if (case_insensitive_match("phone_line", var)) metadata->set_recording_device_type(RecognitionMetadata_RecordingDeviceType_PHONE_LINE);
-      if (case_insensitive_match("vehicle", var)) metadata->set_recording_device_type(RecognitionMetadata_RecordingDeviceType_VEHICLE);
-      if (case_insensitive_match("other_outdoor_device", var)) metadata->set_recording_device_type(RecognitionMetadata_RecordingDeviceType_OTHER_OUTDOOR_DEVICE);
-      if (case_insensitive_match("other_indoor_device", var)) metadata->set_recording_device_type(RecognitionMetadata_RecordingDeviceType_OTHER_INDOOR_DEVICE);
-    }
-	}
-
-	~GStreamer() {
-		//switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_INFO, "GStreamer::~GStreamer - deleting channel and stub: %p\n", (void*)this);
-	}
-
-  void connect() {
-    assert(!m_connected);
-    // Begin a stream.
-  	m_streamer = m_stub->StreamingRecognize(&m_context);
-    m_connected = true;
-
-    // read thread is waiting on this
-    m_promise.set_value();
-
-  	// Write the first request, containing the config only.
-  	m_streamer->Write(m_request);
-
-    // send any buffered audio
-    int nFrames = m_audioBuffer.getNumItems();
-    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "GStreamer %p got stream ready, %d buffered frames\n", this, nFrames);	
-    if (nFrames) {
-      char *p;
-      do {
-        p = m_audioBuffer.getNextChunk();
-        if (p) {
-          write(p, CHUNKSIZE);
-        }
-      } while (p);
-    }
-  }
-
-	bool write(void* data, uint32_t datalen) {
-    if (!m_connected) {
-      if (datalen % CHUNKSIZE == 0) {
-        m_audioBuffer.add(data, datalen);
-      }
-      return true;
-    }
-    m_request.set_audio_content(data, datalen);
-    bool ok = m_streamer->Write(m_request);
-    return ok;
-  }
-
-	uint32_t nextMessageSize(void) {
-		uint32_t size = 0;
-		m_streamer->NextMessageSize(&size);
-		return size;
-	}
-
-	bool read(StreamingRecognizeResponse* response) {
-		return m_streamer->Read(response);
-	}
-
-	grpc::Status finish() {
-		return m_streamer->Finish();
-	}
-
-	void writesDone() {
-    // grpc crashes if we call this twice on a stream
-    if (!m_connected) {
-      cancelConnect();
-    }
-    else if (!m_writesDone) {
-      m_streamer->WritesDone();
-      m_writesDone = true;
-    }
-	}
-
-  bool waitForConnect() {
-    std::shared_future<void> sf(m_promise.get_future());
-    sf.wait();
-    return m_connected;
-  }
-
-  void cancelConnect() {
-    assert(!m_connected);
-    m_promise.set_value();
-  }
-
-  bool isConnected() {
-    return m_connected;
-  }
-
-private:
-	switch_core_session_t* m_session;
-  grpc::ClientContext m_context;
-	std::shared_ptr<grpc::Channel> m_channel;
-	std::unique_ptr<Speech::Stub> 	m_stub;
-	std::unique_ptr< grpc::ClientReaderWriterInterface<StreamingRecognizeRequest, StreamingRecognizeResponse> > m_streamer;
-	StreamingRecognizeRequest m_request;
-  bool m_writesDone;
-  bool m_connected;
-  std::promise<void> m_promise;
-  SimpleBuffer m_audioBuffer;
-};
-
-static void *SWITCH_THREAD_FUNC grpc_read_thread(switch_thread_t *thread, void *obj) {
-  static int count;
-	struct cap_cb *cb = (struct cap_cb *) obj;
-	GStreamer* streamer = (GStreamer *) cb->streamer;
-
-  bool connected = streamer->waitForConnect();
-  if (!connected) {
-    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "google transcribe grpc read thread exiting since we didnt connect\n") ;
-    return nullptr;
-  }
-
-  // Read responses.
-  StreamingRecognizeResponse response;
-  while (streamer->read(&response)) {  // Returns false when no more to read.
-    switch_core_session_t* session = switch_core_session_locate(cb->sessionId);
-    if (!session) {
-      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "grpc_read_thread: session %s is gone!\n", cb->sessionId) ;
-      return nullptr;
-    }
-    count++;
-    auto speech_event_type = response.speech_event_type();
-    if (response.has_error()) {
-      Status status = response.error();
-      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "grpc_read_thread: error %s (%d)\n", status.message().c_str(), status.code()) ;
-      cJSON* json = cJSON_CreateObject();
-      cJSON_AddStringToObject(json, "type", "error");
-      cJSON_AddStringToObject(json, "error", status.message().c_str());
-      char* jsonString = cJSON_PrintUnformatted(json);
-      cb->responseHandler(session, jsonString, cb->bugname);
-      free(jsonString);
-      cJSON_Delete(json);
-    }
-    
-    if (cb->play_file == 1){
-      cb->responseHandler(session, "play_interrupt", cb->bugname);
-    }
-    
-    for (int r = 0; r < response.results_size(); ++r) {
-      auto result = response.results(r);
-      cJSON * jResult = cJSON_CreateObject();
-      cJSON * jAlternatives = cJSON_CreateArray();
-      cJSON * jStability = cJSON_CreateNumber(result.stability());
-      cJSON * jIsFinal = cJSON_CreateBool(result.is_final());
-      cJSON * jLanguageCode = cJSON_CreateString(result.language_code().c_str());
-      cJSON * jChannelTag = cJSON_CreateNumber(result.channel_tag());
-
-      auto duration = result.result_end_time();
-      int32_t seconds = duration.seconds();
-      int64_t nanos = duration.nanos();
-      int span = (int) trunc(seconds * 1000. + ((float) nanos / 1000000.));
-      cJSON * jResultEndTime = cJSON_CreateNumber(span);
-
-      cJSON_AddItemToObject(jResult, "stability", jStability);
-      cJSON_AddItemToObject(jResult, "is_final", jIsFinal);
-      cJSON_AddItemToObject(jResult, "alternatives", jAlternatives);
-      cJSON_AddItemToObject(jResult, "language_code", jLanguageCode);
-      cJSON_AddItemToObject(jResult, "channel_tag", jChannelTag);
-      cJSON_AddItemToObject(jResult, "result_end_time", jResultEndTime);
-
-      for (int a = 0; a < result.alternatives_size(); ++a) {
-        auto alternative = result.alternatives(a);
-        cJSON* jAlt = cJSON_CreateObject();
-        cJSON* jConfidence = cJSON_CreateNumber(alternative.confidence());
-        cJSON* jTranscript = cJSON_CreateString(alternative.transcript().c_str());
-        cJSON_AddItemToObject(jAlt, "confidence", jConfidence);
-        cJSON_AddItemToObject(jAlt, "transcript", jTranscript);
-
-        if (alternative.words_size() > 0) {
-          cJSON * jWords = cJSON_CreateArray();
-          switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "grpc_read_thread: %d words\n", alternative.words_size()) ;
-          for (int b = 0; b < alternative.words_size(); b++) {
-            auto words = alternative.words(b);
-            cJSON* jWord = cJSON_CreateObject();
-            cJSON_AddItemToObject(jWord, "word", cJSON_CreateString(words.word().c_str()));
-            if (words.has_start_time()) {
-              cJSON_AddItemToObject(jWord, "start_time", cJSON_CreateNumber(words.start_time().seconds()));
-            }
-            if (words.has_end_time()) {
-              cJSON_AddItemToObject(jWord, "end_time", cJSON_CreateNumber(words.end_time().seconds()));
-            }
-            int speaker_tag = words.speaker_tag();
-            if (speaker_tag > 0) {
-              cJSON_AddItemToObject(jWord, "speaker_tag", cJSON_CreateNumber(speaker_tag));
-            }
-            float confidence = words.confidence();
-            if (confidence > 0.0) {
-              cJSON_AddItemToObject(jWord, "confidence", cJSON_CreateNumber(confidence));
-            }
-
-            cJSON_AddItemToArray(jWords, jWord);
-          }
-          cJSON_AddItemToObject(jAlt, "words", jWords);
-        }
-        cJSON_AddItemToArray(jAlternatives, jAlt);
-      }
-
-      char* json = cJSON_PrintUnformatted(jResult);
-      cb->responseHandler(session, (const char *) json, cb->bugname);
-      free(json);
-
-      cJSON_Delete(jResult);
-    }
-
-    if (speech_event_type == StreamingRecognizeResponse_SpeechEventType_END_OF_SINGLE_UTTERANCE) {
-      // we only get this when we have requested it, and recognition stops after we get this
-      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "grpc_read_thread: got end_of_utterance\n") ;
-      cb->got_end_of_utterance = 1;
-      cb->responseHandler(session, "end_of_utterance", cb->bugname);
-      if (cb->wants_single_utterance) {
-        switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "grpc_read_thread: sending writesDone because we want only a single utterance\n") ;
-        streamer->writesDone();
-      }
-    }
-    switch_core_session_rwunlock(session);
-    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "grpc_read_thread: got %d responses\n", response.results_size());
-  }
-
-  {
-    switch_core_session_t* session = switch_core_session_locate(cb->sessionId);
-    if (session) {
-      grpc::Status status = streamer->finish();
-      if (11 == status.error_code()) {
-        if (std::string::npos != status.error_message().find("Exceeded maximum allowed stream duration")) {
-          cb->responseHandler(session, "max_duration_exceeded", cb->bugname);
-        }
-        else {
-          cb->responseHandler(session, "no_audio", cb->bugname);
-        }
-      }
-      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "grpc_read_thread: finish() status %s (%d)\n", status.error_message().c_str(), status.error_code()) ;
-      switch_core_session_rwunlock(session);
-    }
-  }
-  return nullptr;
-}
+#include "google_glue.h"
+#include "generic_google_glue.h"
 
 extern "C" {
+	switch_status_t google_speech_init() {
+		const char* gcsServiceKeyFile = std::getenv("GOOGLE_APPLICATION_CREDENTIALS");
+		if (gcsServiceKeyFile) {
+		try {
+			auto creds = grpc::GoogleDefaultCredentials();
+		} catch (const std::exception& e) {
+			switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, 
+			"Error initializing google api with provided credentials in %s: %s\n", gcsServiceKeyFile, e.what());
+			return SWITCH_STATUS_FALSE;
+		}
+		}
+		return SWITCH_STATUS_SUCCESS;
+	}
 
-    switch_status_t google_speech_init() {
-      const char* gcsServiceKeyFile = std::getenv("GOOGLE_APPLICATION_CREDENTIALS");
-      if (gcsServiceKeyFile) {
-        try {
-          auto creds = grpc::GoogleDefaultCredentials();
-        } catch (const std::exception& e) {
-          switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, 
-            "Error initializing google api with provided credentials in %s: %s\n", gcsServiceKeyFile, e.what());
-          return SWITCH_STATUS_FALSE;
-        }
-      }
-      return SWITCH_STATUS_SUCCESS;
-    }
-
-    switch_status_t google_speech_cleanup() {
-      return SWITCH_STATUS_SUCCESS;
-    }
-    switch_status_t google_speech_session_init(switch_core_session_t *session, responseHandler_t responseHandler, 
-          uint32_t to_rate, uint32_t samples_per_second, uint32_t channels, char* lang, int interim, char *bugname,
-          int single_utterance, int separate_recognition, int max_alternatives, int profanity_filter, int word_time_offset,
-          int punctuation, const char* model, int enhanced, const char* hints, char* play_file, void **ppUserData) {
-
-      switch_channel_t *channel = switch_core_session_get_channel(session);
-      auto read_codec = switch_core_session_get_read_codec(session);
-      uint32_t sampleRate = read_codec->implementation->actual_samples_per_second;
-      struct cap_cb *cb;
-      int err;
-
-      cb =(struct cap_cb *) switch_core_session_alloc(session, sizeof(*cb));
-      strncpy(cb->sessionId, switch_core_session_get_uuid(session), MAX_SESSION_ID);
-      strncpy(cb->bugname, bugname, MAX_BUG_LEN);
-      cb->got_end_of_utterance = 0;
-      cb->wants_single_utterance = single_utterance;
-      if (play_file != NULL){
-        cb->play_file = 1;
-      }
-      
-      switch_mutex_init(&cb->mutex, SWITCH_MUTEX_NESTED, switch_core_session_get_pool(session));
-      if (sampleRate != to_rate) {
-          cb->resampler = speex_resampler_init(channels, sampleRate, to_rate, SWITCH_RESAMPLE_QUALITY, &err);
-        if (0 != err) {
-           switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_ERROR, "%s: Error initializing resampler: %s.\n",
-                                 switch_channel_get_name(channel), speex_resampler_strerror(err));
-          return SWITCH_STATUS_FALSE;
-        }
-      } else {
-        switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "%s: no resampling needed for this call\n", switch_channel_get_name(channel));
-      }
-      cb->responseHandler = responseHandler;
-
-      // allocate vad if we are delaying connecting to the recognizer until we detect speech
-      if (switch_channel_var_true(channel, "START_RECOGNIZING_ON_VAD")) {
-        cb->vad = switch_vad_init(sampleRate, channels);
-        if (cb->vad) {
-          const char* var;
-          int mode = 2;
-          int silence_ms = 150;
-          int voice_ms = 250;
-          int debug = 0;
-
-          if (var = switch_channel_get_variable(channel, "RECOGNIZER_VAD_MODE")) {
-            mode = atoi(var);
-          }
-          if (var = switch_channel_get_variable(channel, "RECOGNIZER_VAD_SILENCE_MS")) {
-            silence_ms = atoi(var);
-          }
-          if (var = switch_channel_get_variable(channel, "RECOGNIZER_VAD_VOICE_MS")) {
-            voice_ms = atoi(var);
-          }
-          if (var = switch_channel_get_variable(channel, "RECOGNIZER_VAD_VOICE_MS")) {
-            voice_ms = atoi(var);
-          }
-          switch_vad_set_mode(cb->vad, mode);
-          switch_vad_set_param(cb->vad, "silence_ms", silence_ms);
-          switch_vad_set_param(cb->vad, "voice_ms", voice_ms);
-          switch_vad_set_param(cb->vad, "debug", debug);
-        }
-      }
-
-      GStreamer *streamer = NULL;
-      try {
-        streamer = new GStreamer(session, channels, lang, interim, to_rate, sampleRate, single_utterance, separate_recognition, max_alternatives,
-         profanity_filter, word_time_offset, punctuation, model, enhanced, hints);
-        cb->streamer = streamer;
-      } catch (std::exception& e) {
-        switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_ERROR, "%s: Error initializing gstreamer: %s.\n", 
-          switch_channel_get_name(channel), e.what());
-        return SWITCH_STATUS_FALSE;
-      }
-
-      if (!cb->vad) streamer->connect();
-
-      // create the read thread
-      switch_threadattr_t *thd_attr = NULL;
-      switch_memory_pool_t *pool = switch_core_session_get_pool(session);
-
-      switch_threadattr_create(&thd_attr, pool);
-      switch_threadattr_stacksize_set(thd_attr, SWITCH_THREAD_STACKSIZE);
-      switch_thread_create(&cb->thread, thd_attr, grpc_read_thread, cb, pool);
-
-      *ppUserData = cb;
-      return SWITCH_STATUS_SUCCESS;
-    }
-
-    switch_status_t google_speech_session_cleanup(switch_core_session_t *session, int channelIsClosing, switch_media_bug_t *bug) {
-      switch_channel_t *channel = switch_core_session_get_channel(session);
-
-      if (bug) {
-        struct cap_cb *cb = (struct cap_cb *) switch_core_media_bug_get_user_data(bug);
-        switch_mutex_lock(cb->mutex);
-
-        if (!switch_channel_get_private(channel, cb->bugname)) {
-          // race condition
-          switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_INFO, "%s Bug is not attached (race).\n", switch_channel_get_name(channel));
-          switch_mutex_unlock(cb->mutex);
-          return SWITCH_STATUS_FALSE;
-        }
-        switch_channel_set_private(channel, cb->bugname, NULL);
-
-      // stop playback if available
-       if (cb->play_file == 1){ 
-          if (switch_channel_test_flag(channel, CF_BROADCAST)) {
-		        switch_channel_stop_broadcast(channel);
-	        } else {
-		        switch_channel_set_flag_value(channel, CF_BREAK, 1);
-        	}
-        }
-
-        // close connection and get final responses
-        GStreamer* streamer = (GStreamer *) cb->streamer;
-
-        if (streamer) {
-          streamer->writesDone();
-
-          switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "google_speech_session_cleanup: GStreamer (%p) waiting for read thread to complete\n", (void*)streamer);
-          switch_status_t st;
-          switch_thread_join(&st, cb->thread);
-          switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "google_speech_session_cleanup:  GStreamer (%p) read thread completed\n", (void*)streamer);
-
-          delete streamer;
-          cb->streamer = NULL;
-        }
-
-        if (cb->resampler) {
-          speex_resampler_destroy(cb->resampler);
-        }
-        if (cb->vad) {
-          switch_vad_destroy(&cb->vad);
-          cb->vad = nullptr;
-        }
-        if (!channelIsClosing) {
-          switch_core_media_bug_remove(session, &bug);
-        }
-
-			  switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "google_speech_session_cleanup: Closed stream\n");
-
-			  switch_mutex_unlock(cb->mutex);
-
-
-			  return SWITCH_STATUS_SUCCESS;
-      }
-
-      switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_INFO, "%s Bug is not attached.\n", switch_channel_get_name(channel));
-      return SWITCH_STATUS_FALSE;
-    }
-
-    switch_bool_t google_speech_frame(switch_media_bug_t *bug, void* user_data) {
-    	switch_core_session_t *session = switch_core_media_bug_get_session(bug);
-    	struct cap_cb *cb = (struct cap_cb *) user_data;
-		  if (cb->streamer && (!cb->wants_single_utterance || !cb->got_end_of_utterance)) {
-        GStreamer* streamer = (GStreamer *) cb->streamer;
-        uint8_t data[SWITCH_RECOMMENDED_BUFFER_SIZE];
-        switch_frame_t frame = {};
-        frame.data = data;
-        frame.buflen = SWITCH_RECOMMENDED_BUFFER_SIZE;
-
-        if (switch_mutex_trylock(cb->mutex) == SWITCH_STATUS_SUCCESS) {
-          while (switch_core_media_bug_read(bug, &frame, SWITCH_TRUE) == SWITCH_STATUS_SUCCESS && !switch_test_flag((&frame), SFF_CNG)) {
-            if (frame.datalen) {
-              if (cb->vad && !streamer->isConnected()) {
-                switch_vad_state_t state = switch_vad_process(cb->vad, (int16_t*) frame.data, frame.samples);
-                if (state == SWITCH_VAD_STATE_START_TALKING) {
-                  switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_INFO, "detected speech, connect to google speech now\n");
-                  streamer->connect();
-                  cb->responseHandler(session, "vad_detected", cb->bugname);
-                }
-              }
-
-              if (cb->resampler) {
-                spx_int16_t out[SWITCH_RECOMMENDED_BUFFER_SIZE];
-                spx_uint32_t out_len = SWITCH_RECOMMENDED_BUFFER_SIZE;
-                spx_uint32_t in_len = frame.samples;
-                size_t written;
-
-                speex_resampler_process_interleaved_int(cb->resampler,
-                  (const spx_int16_t *) frame.data,
-                  (spx_uint32_t *) &in_len,
-                  &out[0],
-                  &out_len);
-                streamer->write( &out[0], sizeof(spx_int16_t) * out_len);
-              }
-              else {
-                streamer->write( frame.data, sizeof(spx_int16_t) * frame.samples);
-              }
-            }
-          }
-          switch_mutex_unlock(cb->mutex);
-        }
-      }
-      return SWITCH_TRUE;
-    }
+	switch_status_t google_speech_cleanup() {
+		return SWITCH_STATUS_SUCCESS;
+	}
 }

mod_google_transcribe/google_glue.h

@@ -1,13 +1,28 @@
 #ifndef __GOOGLE_GLUE_H__
 #define __GOOGLE_GLUE_H__
 
+#ifdef __cplusplus
+extern "C" {
+#endif
+
 switch_status_t google_speech_init();
 switch_status_t google_speech_cleanup();
-switch_status_t google_speech_session_init(switch_core_session_t *session, responseHandler_t responseHandler, 
+switch_status_t google_speech_session_init_v1(switch_core_session_t *session, responseHandler_t responseHandler, 
 		uint32_t to_rate, uint32_t samples_per_second, uint32_t channels, char* lang, int interim, char *bugname, int single_utterence,
-		int separate_recognition, int max_alternatives, int profinity_filter, int word_time_offset, int punctuation, const char* model, int enhanced, 
+		int separate_recognition, int max_alternatives, int profanity_filter, int word_time_offset, int punctuation, const char* model, int enhanced, 
 		const char* hints, char* play_file, void **ppUserData);
-switch_status_t google_speech_session_cleanup(switch_core_session_t *session, int channelIsClosing, switch_media_bug_t *bug);
-switch_bool_t google_speech_frame(switch_media_bug_t *bug, void* user_data);
+switch_status_t google_speech_session_cleanup_v1(switch_core_session_t *session, int channelIsClosing, switch_media_bug_t *bug);
+switch_status_t google_speech_session_init_v2(switch_core_session_t *session, responseHandler_t responseHandler, 
+		uint32_t to_rate, uint32_t samples_per_second, uint32_t channels, char* lang, int interim, char *bugname, int single_utterence,
+		int separate_recognition, int max_alternatives, int profanity_filter, int word_time_offset, int punctuation, const char* model, int enhanced, 
+		const char* hints, char* play_file, void **ppUserData);
+switch_status_t google_speech_session_cleanup_v2(switch_core_session_t *session, int channelIsClosing, switch_media_bug_t *bug);
+
+switch_bool_t google_speech_frame_v1(switch_media_bug_t *bug, void* user_data);
+switch_bool_t google_speech_frame_v2(switch_media_bug_t *bug, void* user_data);
+
+#ifdef __cplusplus
+}
+#endif
 
 #endif

mod_google_transcribe/google_glue_v1.cpp

@@ -0,0 +1,381 @@
+#include <switch.h>
+#include <switch_json.h>
+#include <grpc++/grpc++.h>
+
+#include "mod_google_transcribe.h"
+
+#include "gstreamer.h"
+#include "generic_google_glue.h"
+
+#include "google/cloud/speech/v1p1beta1/cloud_speech.grpc.pb.h"
+
+using google::cloud::speech::v1p1beta1::RecognitionConfig;
+using google::cloud::speech::v1p1beta1::Speech;
+using google::cloud::speech::v1p1beta1::SpeechContext;
+using google::cloud::speech::v1p1beta1::StreamingRecognizeRequest;
+using google::cloud::speech::v1p1beta1::StreamingRecognizeResponse;
+using google::cloud::speech::v1p1beta1::SpeakerDiarizationConfig;
+using google::cloud::speech::v1p1beta1::SpeechAdaptation;
+using google::cloud::speech::v1p1beta1::PhraseSet;
+using google::cloud::speech::v1p1beta1::PhraseSet_Phrase;
+using google::cloud::speech::v1p1beta1::RecognitionMetadata;
+using google::cloud::speech::v1p1beta1::RecognitionMetadata_InteractionType_DISCUSSION;
+using google::cloud::speech::v1p1beta1::RecognitionMetadata_InteractionType_PRESENTATION;
+using google::cloud::speech::v1p1beta1::RecognitionMetadata_InteractionType_PHONE_CALL;
+using google::cloud::speech::v1p1beta1::RecognitionMetadata_InteractionType_VOICEMAIL;
+using google::cloud::speech::v1p1beta1::RecognitionMetadata_InteractionType_PROFESSIONALLY_PRODUCED;
+using google::cloud::speech::v1p1beta1::RecognitionMetadata_InteractionType_VOICE_SEARCH;
+using google::cloud::speech::v1p1beta1::RecognitionMetadata_InteractionType_VOICE_COMMAND;
+using google::cloud::speech::v1p1beta1::RecognitionMetadata_InteractionType_DICTATION;
+using google::cloud::speech::v1p1beta1::RecognitionMetadata_MicrophoneDistance_NEARFIELD;
+using google::cloud::speech::v1p1beta1::RecognitionMetadata_MicrophoneDistance_MIDFIELD;
+using google::cloud::speech::v1p1beta1::RecognitionMetadata_MicrophoneDistance_FARFIELD;
+using google::cloud::speech::v1p1beta1::RecognitionMetadata_OriginalMediaType_AUDIO;
+using google::cloud::speech::v1p1beta1::RecognitionMetadata_OriginalMediaType_VIDEO;
+using google::cloud::speech::v1p1beta1::RecognitionMetadata_RecordingDeviceType_SMARTPHONE;
+using google::cloud::speech::v1p1beta1::RecognitionMetadata_RecordingDeviceType_PC;
+using google::cloud::speech::v1p1beta1::RecognitionMetadata_RecordingDeviceType_PHONE_LINE;
+using google::cloud::speech::v1p1beta1::RecognitionMetadata_RecordingDeviceType_VEHICLE;
+using google::cloud::speech::v1p1beta1::RecognitionMetadata_RecordingDeviceType_OTHER_OUTDOOR_DEVICE;
+using google::cloud::speech::v1p1beta1::RecognitionMetadata_RecordingDeviceType_OTHER_INDOOR_DEVICE;
+using google::cloud::speech::v1p1beta1::StreamingRecognizeResponse_SpeechEventType_END_OF_SINGLE_UTTERANCE;
+using google::rpc::Status;
+
+typedef GStreamer<StreamingRecognizeRequest, StreamingRecognizeResponse, Speech::Stub> GStreamer_V1;
+
+template<>
+GStreamer<StreamingRecognizeRequest, StreamingRecognizeResponse, Speech::Stub>::GStreamer(
+    switch_core_session_t *session, 
+    uint32_t channels, 
+    char* lang, 
+    int interim, 
+    uint32_t config_sample_rate,
+		uint32_t samples_per_second,
+    int single_utterance, 
+    int separate_recognition,
+		int max_alternatives, 
+    int profanity_filter, 
+    int word_time_offset, 
+    int punctuation, 
+    const char* model, 
+    int enhanced, 
+		const char* hints) : m_session(session), m_writesDone(false), m_connected(false), 
+      m_audioBuffer(CHUNKSIZE, 15) {
+  
+    switch_channel_t *channel = switch_core_session_get_channel(session);
+    m_channel = create_grpc_channel(channel);
+  	m_stub = Speech::NewStub(m_channel);
+  		
+		auto* streaming_config = m_request.mutable_streaming_config();
+		RecognitionConfig* config = streaming_config->mutable_config();
+
+    streaming_config->set_interim_results(interim);
+    if (single_utterance == 1) {
+      switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "enable_single_utterance\n");
+      streaming_config->set_single_utterance(true);
+    }
+    else {
+      switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "enable_single_utterance is FALSE\n");
+      streaming_config->set_single_utterance(false);
+    }
+
+		config->set_language_code(lang);
+    switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "transcribe language %s \n", lang);
+    
+  	config->set_sample_rate_hertz(config_sample_rate);
+
+		config->set_encoding(RecognitionConfig::LINEAR16);
+
+    // the rest of config comes from channel vars
+
+    // number of channels in the audio stream (default: 1)
+    if (channels > 1) {
+      config->set_audio_channel_count(channels);
+      switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "audio_channel_count %d\n", channels);
+
+      // transcribe each separately?
+      if (separate_recognition == 1) {
+        config->set_enable_separate_recognition_per_channel(true);
+        switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "enable_separate_recognition_per_channel on\n");
+      }
+    }
+
+    // max alternatives
+    if (max_alternatives > 1) {
+      config->set_max_alternatives(max_alternatives);
+      switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "max_alternatives %d\n", max_alternatives);
+    }
+
+    // profanity filter
+    if (profanity_filter == 1) {
+      config->set_profanity_filter(true);
+      switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "profanity_filter\n");
+    }
+
+    // enable word offsets
+    if (word_time_offset == 1) {
+      config->set_enable_word_time_offsets(true);
+      switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "enable_word_time_offsets\n");
+    }
+
+    // enable automatic punctuation
+    if (punctuation == 1) {
+      config->set_enable_automatic_punctuation(true);
+      switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "enable_automatic_punctuation\n");
+    }
+    else {
+      config->set_enable_automatic_punctuation(false);
+      switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "disable_automatic_punctuation\n");
+    }
+
+    // speech model
+    if (model != NULL) {
+      config->set_model(model);
+      switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "speech model %s\n", model);
+    }
+
+    // use enhanced model
+    if (enhanced == 1) {
+      config->set_use_enhanced(true);
+      switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "use_enhanced\n");
+    }
+
+    // hints  
+    if (hints != NULL) {
+      auto* adaptation = config->mutable_adaptation();
+      auto* phrase_set = adaptation->add_phrase_sets();
+      google_speech_configure_grammar_hints(m_session, channel, hints, phrase_set);
+    }
+
+    const char* var;
+    
+    // alternative language
+    if (var = switch_channel_get_variable(channel, "GOOGLE_SPEECH_ALTERNATIVE_LANGUAGE_CODES")) {
+      char *alt_langs[3] = { 0 };
+      int argc = switch_separate_string((char *) var, ',', alt_langs, 3);
+      for (int i = 0; i < argc; i++) {
+        config->add_alternative_language_codes(alt_langs[i]);
+        switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "added alternative lang %s\n", alt_langs[i]);
+      }
+    }
+
+    // speaker diarization
+    if (var = switch_channel_get_variable(channel, "GOOGLE_SPEECH_SPEAKER_DIARIZATION")) {
+      auto* diarization_config = config->mutable_diarization_config();
+      diarization_config->set_enable_speaker_diarization(true);
+      switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "enabling speaker diarization\n", var);
+      if (var = switch_channel_get_variable(channel, "GOOGLE_SPEECH_SPEAKER_DIARIZATION_MIN_SPEAKER_COUNT")) {
+        int count = std::max(atoi(var), 1);
+        switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "setting min speaker count to %d\n", count);
+        diarization_config->set_min_speaker_count(count);
+      }
+      if (var = switch_channel_get_variable(channel, "GOOGLE_SPEECH_SPEAKER_DIARIZATION_MAX_SPEAKER_COUNT")) {
+        int count = std::max(atoi(var), 2);
+        switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "setting max speaker count to %d\n", count);
+        diarization_config->set_max_speaker_count(count);
+      }
+    }
+
+    // recognition metadata
+    auto* metadata = config->mutable_metadata();
+    if (var = switch_channel_get_variable(channel, "GOOGLE_SPEECH_METADATA_INTERACTION_TYPE")) {
+      if (case_insensitive_match("discussion", var)) metadata->set_interaction_type(RecognitionMetadata_InteractionType_DISCUSSION);
+      if (case_insensitive_match("presentation", var)) metadata->set_interaction_type(RecognitionMetadata_InteractionType_PRESENTATION);
+      if (case_insensitive_match("phone_call", var)) metadata->set_interaction_type(RecognitionMetadata_InteractionType_PHONE_CALL);
+      if (case_insensitive_match("voicemail", var)) metadata->set_interaction_type(RecognitionMetadata_InteractionType_VOICEMAIL);
+      if (case_insensitive_match("professionally_produced", var)) metadata->set_interaction_type(RecognitionMetadata_InteractionType_PROFESSIONALLY_PRODUCED);
+      if (case_insensitive_match("voice_search", var)) metadata->set_interaction_type(RecognitionMetadata_InteractionType_VOICE_SEARCH);
+      if (case_insensitive_match("voice_command", var)) metadata->set_interaction_type(RecognitionMetadata_InteractionType_VOICE_COMMAND);
+      if (case_insensitive_match("dictation", var)) metadata->set_interaction_type(RecognitionMetadata_InteractionType_DICTATION);
+    }
+    if (var = switch_channel_get_variable(channel, "GOOGLE_SPEECH_METADATA_INDUSTRY_NAICS_CODE")) {
+      metadata->set_industry_naics_code_of_audio(atoi(var));
+    }
+    if (var = switch_channel_get_variable(channel, "GOOGLE_SPEECH_METADATA_MICROPHONE_DISTANCE")) {
+      if (case_insensitive_match("nearfield", var)) metadata->set_microphone_distance(RecognitionMetadata_MicrophoneDistance_NEARFIELD);
+      if (case_insensitive_match("midfield", var)) metadata->set_microphone_distance(RecognitionMetadata_MicrophoneDistance_MIDFIELD);
+      if (case_insensitive_match("farfield", var)) metadata->set_microphone_distance(RecognitionMetadata_MicrophoneDistance_FARFIELD);
+    }
+    if (var = switch_channel_get_variable(channel, "GOOGLE_SPEECH_METADATA_ORIGINAL_MEDIA_TYPE")) {
+      if (case_insensitive_match("audio", var)) metadata->set_original_media_type(RecognitionMetadata_OriginalMediaType_AUDIO);
+      if (case_insensitive_match("video", var)) metadata->set_original_media_type(RecognitionMetadata_OriginalMediaType_VIDEO);
+    }
+    if (var = switch_channel_get_variable(channel, "GOOGLE_SPEECH_METADATA_RECORDING_DEVICE_TYPE")) {
+      if (case_insensitive_match("smartphone", var)) metadata->set_recording_device_type(RecognitionMetadata_RecordingDeviceType_SMARTPHONE);
+      if (case_insensitive_match("pc", var)) metadata->set_recording_device_type(RecognitionMetadata_RecordingDeviceType_PC);
+      if (case_insensitive_match("phone_line", var)) metadata->set_recording_device_type(RecognitionMetadata_RecordingDeviceType_PHONE_LINE);
+      if (case_insensitive_match("vehicle", var)) metadata->set_recording_device_type(RecognitionMetadata_RecordingDeviceType_VEHICLE);
+      if (case_insensitive_match("other_outdoor_device", var)) metadata->set_recording_device_type(RecognitionMetadata_RecordingDeviceType_OTHER_OUTDOOR_DEVICE);
+      if (case_insensitive_match("other_indoor_device", var)) metadata->set_recording_device_type(RecognitionMetadata_RecordingDeviceType_OTHER_INDOOR_DEVICE);
+    }
+}
+
+static void *SWITCH_THREAD_FUNC grpc_read_thread(switch_thread_t *thread, void *obj) {
+  static int count;
+	struct cap_cb *cb = (struct cap_cb *) obj;
+	GStreamer_V1* streamer = (GStreamer_V1 *) cb->streamer;
+
+  bool connected = streamer->waitForConnect();
+  if (!connected) {
+    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "google transcribe grpc read thread exiting since we didn't connect\n") ;
+    return nullptr;
+  }
+
+  // Read responses.
+  StreamingRecognizeResponse response;
+  while (streamer->read(&response)) {  // Returns false when no more to read.
+    switch_core_session_t* session = switch_core_session_locate(cb->sessionId);
+    if (!session) {
+      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "grpc_read_thread: session %s is gone!\n", cb->sessionId) ;
+      return nullptr;
+    }
+    count++;
+    auto speech_event_type = response.speech_event_type();
+    if (response.has_error()) {
+      Status status = response.error();
+      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "grpc_read_thread: error %s (%d)\n", status.message().c_str(), status.code()) ;
+      cJSON* json = cJSON_CreateObject();
+      cJSON_AddStringToObject(json, "type", "error");
+      cJSON_AddStringToObject(json, "error", status.message().c_str());
+      char* jsonString = cJSON_PrintUnformatted(json);
+      cb->responseHandler(session, jsonString, cb->bugname);
+      free(jsonString);
+      cJSON_Delete(json);
+    }
+    
+    if (cb->play_file == 1) {
+      cb->responseHandler(session, "play_interrupt", cb->bugname);
+    }
+    
+    for (int r = 0; r < response.results_size(); ++r) {
+      auto result = response.results(r);
+      cJSON * jResult = cJSON_CreateObject();
+      cJSON * jAlternatives = cJSON_CreateArray();
+      cJSON * jStability = cJSON_CreateNumber(result.stability());
+      cJSON * jIsFinal = cJSON_CreateBool(result.is_final());
+      cJSON * jLanguageCode = cJSON_CreateString(result.language_code().c_str());
+      cJSON * jChannelTag = cJSON_CreateNumber(result.channel_tag());
+
+      auto duration = result.result_end_time();
+      int32_t seconds = duration.seconds();
+      int64_t nanos = duration.nanos();
+      int span = (int) trunc(seconds * 1000. + ((float) nanos / 1000000.));
+      cJSON * jResultEndTime = cJSON_CreateNumber(span);
+
+      cJSON_AddItemToObject(jResult, "stability", jStability);
+      cJSON_AddItemToObject(jResult, "is_final", jIsFinal);
+      cJSON_AddItemToObject(jResult, "alternatives", jAlternatives);
+      cJSON_AddItemToObject(jResult, "language_code", jLanguageCode);
+      cJSON_AddItemToObject(jResult, "channel_tag", jChannelTag);
+      cJSON_AddItemToObject(jResult, "result_end_time", jResultEndTime);
+
+      for (int a = 0; a < result.alternatives_size(); ++a) {
+        auto alternative = result.alternatives(a);
+        cJSON* jAlt = cJSON_CreateObject();
+        cJSON* jConfidence = cJSON_CreateNumber(alternative.confidence());
+        cJSON* jTranscript = cJSON_CreateString(alternative.transcript().c_str());
+        cJSON_AddItemToObject(jAlt, "confidence", jConfidence);
+        cJSON_AddItemToObject(jAlt, "transcript", jTranscript);
+
+        if (alternative.words_size() > 0) {
+          cJSON * jWords = cJSON_CreateArray();
+          switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "grpc_read_thread: %d words\n", alternative.words_size()) ;
+          for (int b = 0; b < alternative.words_size(); b++) {
+            auto words = alternative.words(b);
+            cJSON* jWord = cJSON_CreateObject();
+            cJSON_AddItemToObject(jWord, "word", cJSON_CreateString(words.word().c_str()));
+            if (words.has_start_time()) {
+              cJSON_AddItemToObject(jWord, "start_time", cJSON_CreateNumber(words.start_time().seconds()));
+            }
+            if (words.has_end_time()) {
+              cJSON_AddItemToObject(jWord, "end_time", cJSON_CreateNumber(words.end_time().seconds()));
+            }
+            int speaker_tag = words.speaker_tag();
+            if (speaker_tag > 0) {
+              cJSON_AddItemToObject(jWord, "speaker_tag", cJSON_CreateNumber(speaker_tag));
+            }
+            float confidence = words.confidence();
+            if (confidence > 0.0) {
+              cJSON_AddItemToObject(jWord, "confidence", cJSON_CreateNumber(confidence));
+            }
+
+            cJSON_AddItemToArray(jWords, jWord);
+          }
+          cJSON_AddItemToObject(jAlt, "words", jWords);
+        }
+        cJSON_AddItemToArray(jAlternatives, jAlt);
+      }
+
+      char* json = cJSON_PrintUnformatted(jResult);
+      cb->responseHandler(session, (const char *) json, cb->bugname);
+      free(json);
+
+      cJSON_Delete(jResult);
+    }
+
+    if (speech_event_type == StreamingRecognizeResponse_SpeechEventType_END_OF_SINGLE_UTTERANCE) {
+      // we only get this when we have requested it, and recognition stops after we get this
+      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "grpc_read_thread: got end_of_utterance\n") ;
+      cb->got_end_of_utterance = 1;
+      cb->responseHandler(session, "end_of_utterance", cb->bugname);
+      if (cb->wants_single_utterance) {
+        switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "grpc_read_thread: sending writesDone because we want only a single utterance\n") ;
+        streamer->writesDone();
+      }
+    }
+    switch_core_session_rwunlock(session);
+    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "grpc_read_thread: got %d responses\n", response.results_size());
+  }
+
+  {
+    switch_core_session_t* session = switch_core_session_locate(cb->sessionId);
+    if (session) {
+      grpc::Status status = streamer->finish();
+      if (11 == status.error_code()) {
+        if (std::string::npos != status.error_message().find("Exceeded maximum allowed stream duration")) {
+          cb->responseHandler(session, "max_duration_exceeded", cb->bugname);
+        }
+        else {
+          cb->responseHandler(session, "no_audio", cb->bugname);
+        }
+      }
+      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "grpc_read_thread: finish() status %s (%d)\n", status.error_message().c_str(), status.error_code()) ;
+      switch_core_session_rwunlock(session);
+    }
+  }
+  return nullptr;
+}
+
+template<>
+bool GStreamer<StreamingRecognizeRequest, StreamingRecognizeResponse, Speech::Stub>::write(void* data, uint32_t datalen) {
+  if (!m_connected) {
+    if (datalen % CHUNKSIZE == 0) {
+      m_audioBuffer.add(data, datalen);
+    }
+    return true;
+  }
+  m_request.set_audio_content(data, datalen);
+  bool ok = m_streamer->Write(m_request);
+  return ok;
+}
+
+extern "C" {
+
+    switch_status_t google_speech_session_cleanup_v1(switch_core_session_t *session, int channelIsClosing, switch_media_bug_t *bug) {
+      return google_speech_session_cleanup<GStreamer_V1>(session, channelIsClosing, bug);
+    }
+
+    switch_bool_t google_speech_frame_v1(switch_media_bug_t *bug, void* user_data) {
+      return google_speech_frame<GStreamer_V1>(bug, user_data);
+    }
+
+    switch_status_t google_speech_session_init_v1(switch_core_session_t *session, responseHandler_t responseHandler,
+		  uint32_t to_rate, uint32_t samples_per_second, uint32_t channels, char* lang, int interim, char *bugname, int single_utterance,
+		  int separate_recognition, int max_alternatives, int profanity_filter, int word_time_offset, int punctuation, const char* model, int enhanced, 
+		  const char* hints, char* play_file, void **ppUserData) {
+      return google_speech_session_init<GStreamer_V1>(session, responseHandler, grpc_read_thread, to_rate, samples_per_second, channels,
+        lang, interim, bugname, single_utterance, separate_recognition, max_alternatives, profanity_filter,
+        word_time_offset, punctuation, model, enhanced, hints, play_file, ppUserData);
+    }
+
+}

mod_google_transcribe/google_glue_v2.cpp

@@ -0,0 +1,339 @@
+#include <switch.h>
+#include <switch_json.h>
+#include <grpc++/grpc++.h>
+
+#include "mod_google_transcribe.h"
+#include "gstreamer.h"
+#include "generic_google_glue.h"
+
+#include "google/cloud/speech/v2/cloud_speech.grpc.pb.h"
+
+using google::cloud::speech::v2::RecognitionConfig;
+using google::cloud::speech::v2::Speech;
+using google::cloud::speech::v2::StreamingRecognizeRequest;
+using google::cloud::speech::v2::StreamingRecognizeResponse;
+using google::cloud::speech::v2::SpeakerDiarizationConfig;
+using google::cloud::speech::v2::SpeechAdaptation;
+using google::cloud::speech::v2::SpeechRecognitionAlternative;
+using google::cloud::speech::v2::PhraseSet;
+using google::cloud::speech::v2::PhraseSet_Phrase;
+using google::cloud::speech::v2::StreamingRecognizeResponse_SpeechEventType_END_OF_SINGLE_UTTERANCE;
+using google::cloud::speech::v2::ExplicitDecodingConfig_AudioEncoding_LINEAR16;
+using google::cloud::speech::v2::RecognitionFeatures_MultiChannelMode_SEPARATE_RECOGNITION_PER_CHANNEL;
+using google::cloud::speech::v2::SpeechAdaptation_AdaptationPhraseSet;
+using google::rpc::Status;
+
+typedef GStreamer<StreamingRecognizeRequest, StreamingRecognizeResponse, Speech::Stub> GStreamer_V2;
+
+template<>
+GStreamer<StreamingRecognizeRequest, StreamingRecognizeResponse, Speech::Stub>::GStreamer(
+    switch_core_session_t *session, 
+    uint32_t channels, 
+    char* lang, 
+    int interim, 
+    uint32_t config_sample_rate,
+	uint32_t samples_per_second,
+    int single_utterance, 
+    int separate_recognition,
+	int max_alternatives, 
+    int profanity_filter, 
+    int word_time_offset, 
+    int punctuation, 
+    const char* model, 
+    int enhanced, 
+	const char* hints) : m_session(session), m_writesDone(false), m_connected(false),
+    m_audioBuffer(CHUNKSIZE, 15) {
+  
+    switch_channel_t *channel = switch_core_session_get_channel(session);
+    m_channel = create_grpc_channel(channel);
+  	m_stub = Speech::NewStub(m_channel);
+
+	auto streaming_config = m_request.mutable_streaming_config();
+    const char* var;
+
+    // The parent of the recognizer must still be provided even if the wildcard
+    // recognizer is used rather than a pre-prepared recognizer.
+    std::string recognizer;
+    if (var = switch_channel_get_variable(channel, "GOOGLE_SPEECH_RECOGNIZER_PARENT")) {
+        recognizer = var;
+        recognizer += "/recognizers/";
+    } else {
+        throw std::runtime_error("The v2 Speech-To-Text library requires GOOGLE_SPEECH_RECOGNIZER_PARENT to be set");
+    }
+
+    // Use the recognizer specified in the variable or just use the wildcard if this is not set.
+    if (var = switch_channel_get_variable(channel, "GOOGLE_SPEECH_RECOGNIZER_ID")) {
+        recognizer += var;
+    } else {
+        recognizer += "_";
+
+        RecognitionConfig* config = streaming_config->mutable_config();
+        config->add_language_codes(lang);
+        switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "transcribe language %s\n", lang);
+        
+        // alternative language
+        if (var = switch_channel_get_variable(channel, "GOOGLE_SPEECH_ALTERNATIVE_LANGUAGE_CODES")) {
+            char *alt_langs[3] = { 0 };
+            int argc = switch_separate_string((char *) var, ',', alt_langs, 3);
+            for (int i = 0; i < argc; i++) {
+                config->add_language_codes(alt_langs[i]);
+                switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "added alternative lang %s\n", alt_langs[i]);
+            }
+        }
+
+        config->mutable_explicit_decoding_config()->set_sample_rate_hertz(config_sample_rate);
+        config->mutable_explicit_decoding_config()->set_encoding(ExplicitDecodingConfig_AudioEncoding_LINEAR16);
+
+        // number of channels in the audio stream (default: 1)
+        // N.B. It is essential to set this configuration value in v2 even if it doesn't deviate from the default.
+        config->mutable_explicit_decoding_config()->set_audio_channel_count(channels);
+        switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "audio_channel_count %d\n", channels);
+        if (channels > 1) {
+            // transcribe each separately?
+            if (separate_recognition == 1) {
+                config->mutable_features()->set_multi_channel_mode(RecognitionFeatures_MultiChannelMode_SEPARATE_RECOGNITION_PER_CHANNEL);
+                switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "enable_separate_recognition_per_channel on\n");
+            }
+        }
+
+        // max alternatives
+        if (max_alternatives > 1) {
+            config->mutable_features()->set_max_alternatives(max_alternatives);
+            switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "max_alternatives %d\n", max_alternatives);
+        }
+
+        // profanity filter
+        if (profanity_filter == 1) {
+            config->mutable_features()->set_profanity_filter(true);
+            switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "profanity_filter\n");
+        }
+
+        // enable word offsets
+        if (word_time_offset == 1) {
+            config->mutable_features()->set_enable_word_time_offsets(true);
+            switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "enable_word_time_offsets\n");
+        }
+
+        // enable automatic punctuation
+        if (punctuation == 1) {
+            config->mutable_features()->set_enable_automatic_punctuation(true);
+            switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "enable_automatic_punctuation\n");
+        }
+        else {
+            config->mutable_features()->set_enable_automatic_punctuation(false);
+            switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "disable_automatic_punctuation\n");
+        }
+
+        // speech model
+        if (model != NULL) {
+            config->set_model(model);
+            switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "speech model %s\n", model);
+        }
+
+        // hints  
+        if (hints != NULL) {
+            auto* adaptation = config->mutable_adaptation();
+            auto* phrase_set = adaptation->add_phrase_sets()->mutable_inline_phrase_set();
+            google_speech_configure_grammar_hints(m_session, channel, hints, phrase_set);
+        }
+
+        // the rest of config comes from channel vars
+
+        // speaker diarization
+        // N.B. At the moment there does not seem to be any combination of model, language and location which supports diarization for STT v2.
+        // See https://stackoverflow.com/questions/76779418/speaker-diarization-is-disabled-even-for-supported-languages-in-google-speech-to
+        if (var = switch_channel_get_variable(channel, "GOOGLE_SPEECH_SPEAKER_DIARIZATION")) {
+            auto* diarization_config = config->mutable_features()->mutable_diarization_config();
+            // There is no enable function in v2
+            // diarization_config->set_enable_speaker_diarization(true);
+            // switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "enabling speaker diarization\n", var);
+            if (var = switch_channel_get_variable(channel, "GOOGLE_SPEECH_SPEAKER_DIARIZATION_MIN_SPEAKER_COUNT")) {
+                int count = std::max(atoi(var), 1);
+                switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "setting min speaker count to %d\n", count);
+                diarization_config->set_min_speaker_count(count);
+            }
+            if (var = switch_channel_get_variable(channel, "GOOGLE_SPEECH_SPEAKER_DIARIZATION_MAX_SPEAKER_COUNT")) {
+                int count = std::max(atoi(var), 2);
+                switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "setting max speaker count to %d\n", count);
+                diarization_config->set_max_speaker_count(count);
+            }
+        }
+    }
+
+    m_request.set_recognizer(recognizer);
+    switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_DEBUG, "using recognizer: %s\n", recognizer.c_str());
+
+    // This must be set whether a recognizer id is provided orr not, because it cannot be configured as part of a recognizer.
+    if (interim > 0) {
+        streaming_config->mutable_streaming_features()->set_interim_results(interim > 0);
+    }
+}
+
+static void *SWITCH_THREAD_FUNC grpc_read_thread(switch_thread_t *thread, void *obj) {
+  static int count;
+	struct cap_cb *cb = (struct cap_cb *) obj;
+	GStreamer_V2* streamer = (GStreamer_V2 *) cb->streamer;
+
+  bool connected = streamer->waitForConnect();
+  if (!connected) {
+    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "google transcribe grpc read thread exiting since we didn't connect\n") ;
+    return nullptr;
+  }
+
+  // Read responses.
+  StreamingRecognizeResponse response;
+  while (streamer->read(&response)) {  // Returns false when no more to read.
+    switch_core_session_t* session = switch_core_session_locate(cb->sessionId);
+    if (!session) {
+      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "grpc_read_thread: session %s is gone!\n", cb->sessionId) ;
+      return nullptr;
+    }
+    count++;
+    
+    if (cb->play_file == 1){
+      cb->responseHandler(session, "play_interrupt", cb->bugname);
+    }
+    
+    for (int r = 0; r < response.results_size(); ++r) {
+      auto result = response.results(r);
+      cJSON * jResult = cJSON_CreateObject();
+      cJSON * jAlternatives = cJSON_CreateArray();
+      cJSON * jStability = cJSON_CreateNumber(result.stability());
+      cJSON * jIsFinal = cJSON_CreateBool(result.is_final());
+      cJSON * jLanguageCode = cJSON_CreateString(result.language_code().c_str());
+      cJSON * jChannelTag = cJSON_CreateNumber(result.channel_tag());
+
+      auto duration = result.result_end_offset();
+      int32_t seconds = duration.seconds();
+      int64_t nanos = duration.nanos();
+      int span = (int) trunc(seconds * 1000. + ((float) nanos / 1000000.));
+      cJSON * jResultEndTime = cJSON_CreateNumber(span);
+
+      cJSON_AddItemToObject(jResult, "stability", jStability);
+      cJSON_AddItemToObject(jResult, "is_final", jIsFinal);
+      cJSON_AddItemToObject(jResult, "alternatives", jAlternatives);
+      cJSON_AddItemToObject(jResult, "language_code", jLanguageCode);
+      cJSON_AddItemToObject(jResult, "channel_tag", jChannelTag);
+      cJSON_AddItemToObject(jResult, "result_end_time", jResultEndTime);
+
+      if (result.alternatives_size() == 0) {
+        SpeechRecognitionAlternative alternative;
+        alternative.set_confidence(0.0);
+        alternative.set_transcript("");
+        *result.add_alternatives() = alternative;
+      }
+      for (int a = 0; a < result.alternatives_size(); ++a) {
+        auto alternative = result.alternatives(a);
+        cJSON* jAlt = cJSON_CreateObject();
+        cJSON* jConfidence = cJSON_CreateNumber(alternative.confidence());
+        cJSON* jTranscript = cJSON_CreateString(alternative.transcript().c_str());
+        cJSON_AddItemToObject(jAlt, "confidence", jConfidence);
+        cJSON_AddItemToObject(jAlt, "transcript", jTranscript);
+
+        if (alternative.words_size() > 0) {
+          cJSON * jWords = cJSON_CreateArray();
+          switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "grpc_read_thread: %d words\n", alternative.words_size()) ;
+          for (int b = 0; b < alternative.words_size(); b++) {
+            auto words = alternative.words(b);
+            cJSON* jWord = cJSON_CreateObject();
+            cJSON_AddItemToObject(jWord, "word", cJSON_CreateString(words.word().c_str()));
+            if (words.has_start_offset()) {
+              cJSON_AddItemToObject(jWord, "start_offset", cJSON_CreateNumber(words.start_offset().seconds()));
+            }
+            if (words.has_end_offset()) {
+              cJSON_AddItemToObject(jWord, "end_offset", cJSON_CreateNumber(words.end_offset().seconds()));
+            }
+            auto speaker_label = words.speaker_label();
+            if (speaker_label.size() > 0) {
+              cJSON_AddItemToObject(jWord, "speaker_label", cJSON_CreateString(speaker_label.c_str()));
+            }
+            float confidence = words.confidence();
+            if (confidence > 0.0) {
+              cJSON_AddItemToObject(jWord, "confidence", cJSON_CreateNumber(confidence));
+            }
+
+            cJSON_AddItemToArray(jWords, jWord);
+          }
+          cJSON_AddItemToObject(jAlt, "words", jWords);
+        }
+        cJSON_AddItemToArray(jAlternatives, jAlt);
+      }
+
+      char* json = cJSON_PrintUnformatted(jResult);
+      cb->responseHandler(session, (const char *) json, cb->bugname);
+      free(json);
+
+      cJSON_Delete(jResult);
+    }
+
+    auto speech_event_type = response.speech_event_type();
+    if (speech_event_type == StreamingRecognizeResponse_SpeechEventType_END_OF_SINGLE_UTTERANCE) {
+      // we only get this when we have requested it, and recognition stops after we get this
+      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "grpc_read_thread: got end_of_utterance\n") ;
+      cb->got_end_of_utterance = 1;
+      cb->responseHandler(session, "end_of_utterance", cb->bugname);
+      if (cb->wants_single_utterance) {
+        switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "grpc_read_thread: sending writesDone because we want only a single utterance\n") ;
+        streamer->writesDone();
+      }
+    }
+    switch_core_session_rwunlock(session);
+    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "grpc_read_thread: got %d responses\n", response.results_size());
+  }
+
+  {
+    switch_core_session_t* session = switch_core_session_locate(cb->sessionId);
+    if (session) {
+      grpc::Status status = streamer->finish();
+      // TODO: This works on the same principle as that used in the v1 equivalent, in that we search for the textual
+      // error message to determine whether the cause of the problem is the expiration of the session.
+      // It would be better if we could find a more reliable way of detecting this.
+      if (10 == status.error_code()) {
+        if (std::string::npos != status.error_message().find("Max duration of 5 minutes reached")) {
+          cb->responseHandler(session, "max_duration_exceeded", cb->bugname);
+        }
+        else {
+          cb->responseHandler(session, "no_audio", cb->bugname);
+        }
+      }
+      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "grpc_read_thread: finish() status %s (%d)\n", status.error_message().c_str(), status.error_code()) ;
+      switch_core_session_rwunlock(session);
+    }
+  }
+  return nullptr;
+}
+
+template <>
+bool GStreamer<StreamingRecognizeRequest, StreamingRecognizeResponse, Speech::Stub>::write(void* data, uint32_t datalen) {
+	if (!m_connected) {
+		if (datalen % CHUNKSIZE == 0) {
+			m_audioBuffer.add(data, datalen);
+		}
+		return true;
+	}
+    m_request.clear_streaming_config();
+	m_request.set_audio(data, datalen);
+	bool ok = m_streamer->Write(m_request);
+	return ok;
+}
+
+extern "C" {
+
+    switch_status_t google_speech_session_cleanup_v2(switch_core_session_t *session, int channelIsClosing, switch_media_bug_t *bug) {
+      return google_speech_session_cleanup<GStreamer_V2>(session, channelIsClosing, bug);
+    }
+
+    switch_bool_t google_speech_frame_v2(switch_media_bug_t *bug, void* user_data) {
+      return google_speech_frame<GStreamer_V2>(bug, user_data);
+    }
+
+    switch_status_t google_speech_session_init_v2(switch_core_session_t *session, responseHandler_t responseHandler,
+		  uint32_t to_rate, uint32_t samples_per_second, uint32_t channels, char* lang, int interim, char *bugname, int single_utterance,
+		  int separate_recognition, int max_alternatives, int profanity_filter, int word_time_offset, int punctuation, const char* model, int enhanced, 
+		  const char* hints, char* play_file, void **ppUserData) {
+      return google_speech_session_init<GStreamer_V2>(session, responseHandler, grpc_read_thread, to_rate, samples_per_second, channels,
+        lang, interim, bugname, single_utterance, separate_recognition, max_alternatives, profanity_filter,
+        word_time_offset, punctuation, model, enhanced, hints, play_file, ppUserData);
+    }
+
+}

mod_google_transcribe/gstreamer.h

@@ -0,0 +1,147 @@
+#include <cstdlib>
+#include <algorithm>
+#include <future>
+
+#include <switch.h>
+#include <switch_json.h>
+#include <grpc++/grpc++.h>
+#include <grpcpp/impl/codegen/sync_stream.h>
+
+#include "mod_google_transcribe.h"
+#include "simple_buffer.h"
+
+#define CHUNKSIZE (320)
+
+namespace {
+  int case_insensitive_match(std::string s1, std::string s2) {
+   std::transform(s1.begin(), s1.end(), s1.begin(), ::tolower);
+   std::transform(s2.begin(), s2.end(), s2.begin(), ::tolower);
+   if(s1.compare(s2) == 0)
+      return 1; //The strings are same
+   return 0; //not matched
+  }
+}
+
+template <typename Request, typename Response, typename Stub>
+class GStreamer {
+public:
+	GStreamer(
+    switch_core_session_t *session, 
+    uint32_t channels, 
+    char* lang, 
+    int interim, 
+    uint32_t config_sample_rate,
+		uint32_t samples_per_second,
+    int single_utterance, 
+    int separate_recognition,
+		int max_alternatives, 
+    int profanity_filter, 
+    int word_time_offset, 
+    int punctuation, 
+    const char* model, 
+    int enhanced, 
+	const char* hints);
+
+	~GStreamer() {
+		//switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(m_session), SWITCH_LOG_INFO, "GStreamer::~GStreamer - deleting channel and stub: %p\n", (void*)this);
+	}
+
+	bool write(void* data, uint32_t datalen);
+
+	void connect() {
+		assert(!m_connected);
+		// Begin a stream.
+		m_streamer = m_stub->StreamingRecognize(&m_context);
+		m_connected = true;
+
+		// read thread is waiting on this
+		m_promise.set_value();
+
+		// Write the first request, containing the config only.
+		m_streamer->Write(m_request);
+
+		// send any buffered audio
+		int nFrames = m_audioBuffer.getNumItems();
+		switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "GStreamer %p got stream ready, %d buffered frames\n", this, nFrames);	
+		if (nFrames) {
+			char *p;
+			do {
+				p = m_audioBuffer.getNextChunk();
+				if (p) {
+					write(p, CHUNKSIZE);
+				}
+			} while (p);
+		}
+	}
+
+	uint32_t nextMessageSize(void) {
+		uint32_t size = 0;
+		m_streamer->NextMessageSize(&size);
+		return size;
+	}
+
+	bool read(Response* response) {
+		return m_streamer->Read(response);
+	}
+
+	grpc::Status finish() {
+		return m_streamer->Finish();
+	}
+
+	void writesDone() {
+		// grpc crashes if we call this twice on a stream
+		if (!m_connected) {
+			cancelConnect();
+		}
+		else if (!m_writesDone) {
+			m_streamer->WritesDone();
+			m_writesDone = true;
+		}
+	}
+
+	bool waitForConnect() {
+		std::shared_future<void> sf(m_promise.get_future());
+		sf.wait();
+		return m_connected;
+	}
+
+	void cancelConnect() {
+		assert(!m_connected);
+		m_promise.set_value();
+	}
+
+	bool isConnected() {
+		return m_connected;
+	}
+
+private:
+	std::shared_ptr<grpc::Channel> create_grpc_channel(switch_channel_t *channel) {
+	    const char* google_uri;
+		if (!(google_uri = switch_channel_get_variable(channel, "GOOGLE_SPEECH_TO_TEXT_URI"))) {
+			google_uri = "speech.googleapis.com";
+		}
+
+	    const char* var;
+		if (var = switch_channel_get_variable(channel, "GOOGLE_APPLICATION_CREDENTIALS")) {
+			auto channelCreds = grpc::SslCredentials(grpc::SslCredentialsOptions());
+			auto callCreds = grpc::ServiceAccountJWTAccessCredentials(var);
+			auto creds = grpc::CompositeChannelCredentials(channelCreds, callCreds);
+			return grpc::CreateChannel(google_uri, creds);
+		}
+		else {
+			auto creds = grpc::GoogleDefaultCredentials();
+			return grpc::CreateChannel(google_uri, creds);
+		}
+	}
+
+	switch_core_session_t* m_session;
+	grpc::ClientContext m_context;
+	std::shared_ptr<grpc::Channel> m_channel;
+	std::unique_ptr<Stub> 	m_stub;
+	std::unique_ptr< grpc::ClientReaderWriterInterface<Request, Response> > m_streamer;
+	Request m_request;
+	bool m_writesDone;
+	bool m_connected;
+	std::promise<void> m_promise;
+	SimpleBuffer m_audioBuffer;
+};

mod_google_transcribe/mod_google_transcribe.c

@@ -10,6 +10,13 @@
 
 static const uint32_t DEFAULT_SAMPLE_RATE = 8000;
 
+/* Callback Type Definitions */
+typedef switch_status_t (*speech_cleanup_callback_t) (switch_core_session_t *, int, switch_media_bug_t *);
+typedef switch_bool_t (*speech_frame_callback_t) (switch_media_bug_t *, void *);
+typedef switch_status_t (*speech_init_callback_t) (switch_core_session_t *, responseHandler_t,
+		  uint32_t, uint32_t, uint32_t, char *, int, char *, int, int, int, int, int, int, const char *,
+		  int, const char *, char *, void **);
+
 /* Prototypes */
 SWITCH_MODULE_SHUTDOWN_FUNCTION(mod_transcribe_shutdown);
 SWITCH_MODULE_RUNTIME_FUNCTION(mod_transcribe_runtime);
@@ -17,7 +24,42 @@ SWITCH_MODULE_LOAD_FUNCTION(mod_transcribe_load);
 
 SWITCH_MODULE_DEFINITION(mod_google_transcribe, mod_transcribe_load, mod_transcribe_shutdown, NULL);
 
-static switch_status_t do_stop(switch_core_session_t *session, char* bugname);
+static switch_bool_t capture_callback_v1(switch_media_bug_t *bug, void *user_data, switch_abc_type_t type);
+static switch_bool_t capture_callback_v2(switch_media_bug_t *bug, void *user_data, switch_abc_type_t type);
+static switch_status_t do_stop(switch_core_session_t *session, char* bugname, speech_cleanup_callback_t cleanup_callback);
+
+static switch_media_bug_callback_t get_bug_callback_from_version(GoogleCloudServiceVersion version) {
+	switch (version) {
+		case GoogleCloudServiceVersion_v1:
+			return capture_callback_v1;
+		case GoogleCloudServiceVersion_v2:
+			return capture_callback_v2;
+		default:
+			return capture_callback_v1;
+	}
+}
+
+static speech_cleanup_callback_t get_cleanup_callback_from_version(GoogleCloudServiceVersion version) {
+	switch (version) {
+		case GoogleCloudServiceVersion_v1:
+			return google_speech_session_cleanup_v1;
+		case GoogleCloudServiceVersion_v2:
+			return google_speech_session_cleanup_v2;
+		default:
+			return google_speech_session_cleanup_v1;
+	}
+}
+
+static speech_init_callback_t get_init_callback_from_version(GoogleCloudServiceVersion version) {
+	switch (version) {
+		case GoogleCloudServiceVersion_v1:
+			return google_speech_session_init_v1;
+		case GoogleCloudServiceVersion_v2:
+			return google_speech_session_init_v2;
+		default:
+			return google_speech_session_init_v1;
+	}
+}
 
 
 static void responseHandler(switch_core_session_t* session, const char * json, const char* bugname) {
@@ -69,19 +111,19 @@ static void responseHandler(switch_core_session_t* session, const char * json, c
 		switch_event_add_header_string(event, SWITCH_STACK_BOTTOM, "transcription-vendor", "google");
 	}
 	else {
-    int error = 0;
-    cJSON* jMessage = cJSON_Parse(json);
-    if (jMessage) {
-      const char* type = cJSON_GetStringValue(cJSON_GetObjectItem(jMessage, "type"));
-      if (type && 0 == strcmp(type, "error")) {
-        error = 1;
-    		switch_event_create_subclass(&event, SWITCH_EVENT_CUSTOM, TRANSCRIBE_EVENT_ERROR);
-      }
-      cJSON_Delete(jMessage);
-    }
-    if (!error) {
-    		switch_event_create_subclass(&event, SWITCH_EVENT_CUSTOM, TRANSCRIBE_EVENT_RESULTS);
-    }
+		int error = 0;
+		cJSON* jMessage = cJSON_Parse(json);
+		if (jMessage) {
+			const char* type = cJSON_GetStringValue(cJSON_GetObjectItem(jMessage, "type"));
+			if (type && 0 == strcmp(type, "error")) {
+				error = 1;
+				switch_event_create_subclass(&event, SWITCH_EVENT_CUSTOM, TRANSCRIBE_EVENT_ERROR);
+			}
+			cJSON_Delete(jMessage);
+		}
+		if (!error) {
+			switch_event_create_subclass(&event, SWITCH_EVENT_CUSTOM, TRANSCRIBE_EVENT_RESULTS);
+		}
 		switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "%s json payload: %s.\n", bugname ? bugname : "google_transcribe", json);
 		switch_channel_event_set_data(channel, event);
 		switch_event_add_header_string(event, SWITCH_STACK_BOTTOM, "transcription-vendor", "google");
@@ -91,29 +133,29 @@ static void responseHandler(switch_core_session_t* session, const char * json, c
 	switch_event_fire(&event);
 }
 
-static switch_bool_t capture_callback(switch_media_bug_t *bug, void *user_data, switch_abc_type_t type)
+static switch_bool_t capture_callback(switch_media_bug_t *bug, void *user_data, switch_abc_type_t type,
+	speech_frame_callback_t frame_callback, speech_cleanup_callback_t cleanup_callback)
 {
 	switch_core_session_t *session = switch_core_media_bug_get_session(bug);
 	struct cap_cb* cb = (struct cap_cb*) switch_core_media_bug_get_user_data(bug);
 
 	switch (type) {
 	case SWITCH_ABC_TYPE_INIT:
-			switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "Got SWITCH_ABC_TYPE_INIT.\n");
-			responseHandler(session, "start_of_transcript", cb->bugname);
+		switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "Got SWITCH_ABC_TYPE_INIT.\n");
+		responseHandler(session, "start_of_transcript", cb->bugname);
 		break;
 
 	case SWITCH_ABC_TYPE_CLOSE:
 		{
 			switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "Got SWITCH_ABC_TYPE_CLOSE, calling google_speech_session_cleanup.\n");
 			responseHandler(session, "end_of_transcript", cb->bugname);
-			google_speech_session_cleanup(session, 1, bug);
+			cleanup_callback(session, 1, bug);
 			switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "Finished SWITCH_ABC_TYPE_CLOSE.\n");
 		}
 		break;
 	
 	case SWITCH_ABC_TYPE_READ:
-
-		return google_speech_frame(bug, user_data);
+		return frame_callback(bug, user_data);
 		break;
 
 	case SWITCH_ABC_TYPE_WRITE:
@@ -124,19 +166,27 @@ static switch_bool_t capture_callback(switch_media_bug_t *bug, void *user_data,
 	return SWITCH_TRUE;
 }
 
+static switch_bool_t capture_callback_v1(switch_media_bug_t *bug, void *user_data, switch_abc_type_t type) {
+	return capture_callback(bug, user_data, type, google_speech_frame_v1, google_speech_session_cleanup_v1);
+}
+
+static switch_bool_t capture_callback_v2(switch_media_bug_t *bug, void *user_data, switch_abc_type_t type) {
+	return capture_callback(bug, user_data, type, google_speech_frame_v2, google_speech_session_cleanup_v2);
+}
+
 static switch_status_t transcribe_input_callback(switch_core_session_t *session, void *input, switch_input_type_t input_type, void *data, unsigned int len){
 	if (input_type == SWITCH_INPUT_TYPE_EVENT) {
-			switch_event_t *event;
-			event = (switch_event_t *)input;
-			if (event->event_id == SWITCH_EVENT_DETECTED_SPEECH) {
-				return SWITCH_STATUS_BREAK;
-			}
+		switch_event_t *event;
+		event = (switch_event_t *)input;
+		if (event->event_id == SWITCH_EVENT_DETECTED_SPEECH) {
+			return SWITCH_STATUS_BREAK;
+		}
 	}
 	
 	return SWITCH_STATUS_SUCCESS;
 }
 
-static switch_status_t do_stop(switch_core_session_t *session, char *bugname)
+static switch_status_t do_stop(switch_core_session_t *session, char *bugname, speech_cleanup_callback_t cleanup_callback)
 {
 	switch_status_t status = SWITCH_STATUS_SUCCESS;
 	switch_channel_t *channel = switch_core_session_get_channel(session);
@@ -144,7 +194,7 @@ static switch_status_t do_stop(switch_core_session_t *session, char *bugname)
 
 	if (bug) {
 		switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "Received user command command, calling google_speech_session_cleanup (possibly to stop prev transcribe)\n");
-		status = google_speech_session_cleanup(session, 0, bug);
+		status = cleanup_callback(session, 0, bug);
 		switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "stopped transcription.\n");
 	}
 
@@ -153,7 +203,8 @@ static switch_status_t do_stop(switch_core_session_t *session, char *bugname)
 
 static switch_status_t start_capture2(switch_core_session_t *session, switch_media_bug_flag_t flags, 
   uint32_t sample_rate, char* lang, int interim, int single_utterance, int separate_recognition, int max_alternatives,
-  int profinity_filter, int word_time_offset, int punctuation, const char* model, int enhanced, const char* hints, char* play_file)
+  int profanity_filter, int word_time_offset, int punctuation, const char* model, int enhanced, const char* hints,
+  char* play_file, GoogleCloudServiceVersion version)
 {
 	switch_channel_t *channel = switch_core_session_get_channel(session);
 	switch_media_bug_t *bug;
@@ -165,7 +216,7 @@ static switch_status_t start_capture2(switch_core_session_t *session, switch_med
 
 	if (switch_channel_get_private(channel, MY_BUG_NAME)) {
 		switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "removing bug from previous transcribe\n");
-		do_stop(session, MY_BUG_NAME);
+		do_stop(session, MY_BUG_NAME, get_cleanup_callback_from_version(version));
 	}
 
 	switch_core_session_get_read_impl(session, &read_impl);
@@ -175,13 +226,15 @@ static switch_status_t start_capture2(switch_core_session_t *session, switch_med
 	}
 
 	samples_per_second = !strcasecmp(read_impl.iananame, "g722") ? read_impl.actual_samples_per_second : read_impl.samples_per_second;
+	status = get_init_callback_from_version(version)(session, responseHandler, sample_rate, samples_per_second, flags & SMBF_STEREO ? 2 : 1, lang, interim, MY_BUG_NAME, single_utterance,
+			separate_recognition, max_alternatives, profanity_filter, word_time_offset, punctuation, model, enhanced, hints, play_file, &pUserData);
 
-	if (SWITCH_STATUS_FALSE == google_speech_session_init(session, responseHandler, sample_rate, samples_per_second, flags & SMBF_STEREO ? 2 : 1, lang, interim, MY_BUG_NAME, single_utterance,
-	 separate_recognition, max_alternatives, profinity_filter, word_time_offset, punctuation, model, enhanced, hints, play_file, &pUserData)) {
+	if (SWITCH_STATUS_FALSE == status) {
 		switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error initializing google speech session.\n");
 		return SWITCH_STATUS_FALSE;
 	}
-	if ((status = switch_core_media_bug_add(session, "google_transcribe", NULL, capture_callback, pUserData, 0, flags, &bug)) != SWITCH_STATUS_SUCCESS) {
+
+	if ((status = switch_core_media_bug_add(session, "google_transcribe", NULL, get_bug_callback_from_version(version), pUserData, 0, flags, &bug)) != SWITCH_STATUS_SUCCESS) {
 		return status;
 	}
 
@@ -197,22 +250,23 @@ static switch_status_t start_capture2(switch_core_session_t *session, switch_med
 }
 
 static switch_status_t start_capture(switch_core_session_t *session, switch_media_bug_flag_t flags, 
-  char* lang, int interim, char* bugname)
+  char* lang, int interim, char* bugname, GoogleCloudServiceVersion version)
 {
 	switch_channel_t *channel = switch_core_session_get_channel(session);
 	switch_media_bug_t *bug;
 	switch_status_t status;
 	switch_codec_implementation_t read_impl = { 0 };
 	void *pUserData;
+	uint32_t sample_rate = DEFAULT_SAMPLE_RATE;
 	uint32_t samples_per_second;
 	int single_utterance = 0, separate_recognition = 0, max_alternatives = 0, profanity_filter = 0, word_time_offset = 0, punctuation = 0, enhanced = 0;
 	const char* hints = NULL;
-  const char* model = NULL;
+  	const char* model = NULL;
 	const char* var;
-
+	
 	if (switch_channel_get_private(channel, bugname)) {
 		switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "removing bug from previous transcribe\n");
-		do_stop(session, bugname);
+		do_stop(session, bugname, get_cleanup_callback_from_version(version));
 	}
 
 	if (switch_true(switch_channel_get_variable(channel, "GOOGLE_SPEECH_SINGLE_UTTERANCE"))) {
@@ -229,6 +283,11 @@ static switch_status_t start_capture(switch_core_session_t *session, switch_medi
      	max_alternatives = atoi(var);
     }
 
+	// sample rate
+	if ((var = switch_channel_get_variable(channel, "GOOGLE_SPEECH_SAMPLE_RATE"))) {
+     	sample_rate = atoi(var);
+    }
+
 	// profanity filter
 	if (switch_true(switch_channel_get_variable(channel, "GOOGLE_SPEECH_PROFANITY_FILTER"))) {
       profanity_filter = 1;
@@ -266,14 +325,16 @@ static switch_status_t start_capture(switch_core_session_t *session, switch_medi
 	}
 
 	samples_per_second = !strcasecmp(read_impl.iananame, "g722") ? read_impl.actual_samples_per_second : read_impl.samples_per_second;
+	status = SWITCH_STATUS_FALSE;
+	status = get_init_callback_from_version(version)(session, responseHandler, sample_rate, samples_per_second, flags & SMBF_STEREO ? 2 : 1, lang, interim, bugname, single_utterance,
+			separate_recognition, max_alternatives, profanity_filter, word_time_offset, punctuation, model, enhanced, hints, NULL, &pUserData);
 
-	if (SWITCH_STATUS_FALSE == google_speech_session_init(session, responseHandler, DEFAULT_SAMPLE_RATE, samples_per_second, flags & SMBF_STEREO ? 2 : 1, lang, interim, bugname, single_utterance,
-	 separate_recognition, max_alternatives, profanity_filter, word_time_offset, punctuation, model, enhanced, hints, NULL, &pUserData)) {
+	if (SWITCH_STATUS_FALSE == status) {
 		switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error initializing google speech session.\n");
 		return SWITCH_STATUS_FALSE;
 	}
 
-	if ((status = switch_core_media_bug_add(session, bugname, NULL, capture_callback, pUserData, 0, flags, &bug)) != SWITCH_STATUS_SUCCESS) {
+	if ((status = switch_core_media_bug_add(session, bugname, NULL, get_bug_callback_from_version(version), pUserData, 0, flags, &bug)) != SWITCH_STATUS_SUCCESS) {
 		return status;
 	}
 
@@ -283,7 +344,7 @@ static switch_status_t start_capture(switch_core_session_t *session, switch_medi
 }
 
 // #define TRANSCRIBE_API_SYNTAX "<uuid> [start|stop] [lang-code] [interim] [single-utterance](bool) [seperate-recognition](bool) [max-alternatives](int) [profinity-filter](bool) [word-time](bool) [punctuation](bool) [model](string) [enhanced](true) [hints](string without space) [play-file]"
-#define TRANSCRIBE2_API_SYNTAX "<uuid> [start|stop] [lang-code] [interim]  [single-utterance] [seperate-recognition] [max-alternatives] [profinity-filter] [word-time] [punctuation] [sample-rate] [model] [enhanced] [hints] [play-file]"
+#define TRANSCRIBE2_API_SYNTAX "<uuid> [start|stop] [lang-code] [interim]  [single-utterance] [separate-recognition] [max-alternatives] [profanity-filter] [word-time] [punctuation] [sample-rate] [model] [enhanced] [hints] [play-file]"
 SWITCH_STANDARD_API(transcribe2_function)
 {
 	char *mycmd = NULL, *argv[20] = { 0 };
@@ -295,6 +356,9 @@ SWITCH_STANDARD_API(transcribe2_function)
 	
 	switch_status_t status = SWITCH_STATUS_FALSE;
 	switch_media_bug_flag_t flags = SMBF_READ_STREAM /* | SMBF_WRITE_STREAM | SMBF_READ_PING */;
+	switch_channel_t *channel;
+	const char* var;
+	GoogleCloudServiceVersion version = GoogleCloudServiceVersion_v1;
 
 	if (!zstr(cmd) && (mycmd = strdup(cmd))) {
 		argc = switch_separate_string(mycmd, ' ', argv, (sizeof(argv) / sizeof(argv[0])));
@@ -311,16 +375,24 @@ SWITCH_STANDARD_API(transcribe2_function)
 		switch_core_session_t *lsession = NULL;
 
 		if ((lsession = switch_core_session_locate(argv[0]))) {
+			channel = switch_core_session_get_channel(lsession);
+			if ((var = switch_channel_get_variable(channel, "GOOGLE_SPEECH_CLOUD_SERVICES_VERSION"))) {
+				if (!strcasecmp(var, "v1"))
+					version = GoogleCloudServiceVersion_v1;
+				else if (!strcasecmp(var, "v2"))
+					version = GoogleCloudServiceVersion_v2;
+			}
+
 			if (!strcasecmp(argv[1], "stop")) {
     		switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_INFO, "stop transcribing\n");
-				status = do_stop(lsession, MY_BUG_NAME);
+				status = do_stop(lsession, MY_BUG_NAME, get_cleanup_callback_from_version(version));
 			} else if (!strcasecmp(argv[1], "start")) {
-        char* lang = argv[2];
-        int interim = argc > 3 && !strcmp(argv[3], "true");
+        		char* lang = argv[2];
+        		int interim = argc > 3 && !strcmp(argv[3], "true");
 				int single_utterance =  !strcmp(argv[4], "true"); // single-utterance
-				int separate_recognition = !strcmp(argv[5], "true"); // sepreate-recognition
+				int separate_recognition = !strcmp(argv[5], "true"); // separate-recognition
 				int max_alternatives = atoi(argv[6]); // max-alternatives
-				int profinity_filter = !strcmp(argv[7], "true"); // profinity-filter
+				int profanity_filter = !strcmp(argv[7], "true"); // profanity-filter
 				int word_time_offset = !strcmp(argv[8], "true"); // word-time
 				int punctuation      = !strcmp(argv[9], "true");  //punctuation
 				if (argc > 10) {
@@ -336,9 +408,9 @@ SWITCH_STANDARD_API(transcribe2_function)
 				if (argc > 14){
 					play_file = argv[14];
 				}
-    		switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_INFO, "start transcribing %s %s\n", lang, interim ? "interim": "complete");
+    			switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_INFO, "start transcribing %s %s\n", lang, interim ? "interim": "complete");
 				status = start_capture2(lsession, flags, sample_rate, lang, interim, single_utterance, separate_recognition,max_alternatives,
-				profinity_filter, word_time_offset, punctuation, model, enhanced, hints, play_file);
+				profanity_filter, word_time_offset, punctuation, model, enhanced, hints, play_file, version);
 			}
 			switch_core_session_rwunlock(lsession);
 		}
@@ -363,6 +435,9 @@ SWITCH_STANDARD_API(transcribe_function)
 	int argc = 0;
 	switch_status_t status = SWITCH_STATUS_FALSE;
 	switch_media_bug_flag_t flags = SMBF_READ_STREAM /* | SMBF_WRITE_STREAM | SMBF_READ_PING */;
+	switch_channel_t *channel;
+	const char* var;
+	GoogleCloudServiceVersion version = GoogleCloudServiceVersion_v1;
 
 	if (!zstr(cmd) && (mycmd = strdup(cmd))) {
 		argc = switch_separate_string(mycmd, ' ', argv, (sizeof(argv) / sizeof(argv[0])));
@@ -379,20 +454,28 @@ SWITCH_STANDARD_API(transcribe_function)
 		switch_core_session_t *lsession = NULL;
 
 		if ((lsession = switch_core_session_locate(argv[0]))) {
+			channel = switch_core_session_get_channel(lsession);
+			if ((var = switch_channel_get_variable(channel, "GOOGLE_SPEECH_CLOUD_SERVICES_VERSION"))) {
+				if (!strcasecmp(var, "v1"))
+					version = GoogleCloudServiceVersion_v1;
+				else if (!strcasecmp(var, "v2"))
+					version = GoogleCloudServiceVersion_v2;
+			}
+
 			if (!strcasecmp(argv[1], "stop")) {
 				char *bugname = argc > 2 ? argv[2] : MY_BUG_NAME;
-    		switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_INFO, "stop transcribing\n");
-				status = do_stop(lsession, bugname);
+	    		switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_INFO, "stop transcribing\n");
+				status = do_stop(lsession, bugname, get_cleanup_callback_from_version(version));
 			} else if (!strcasecmp(argv[1], "start")) {
-        char* lang = argv[2];
-        int interim = argc > 3 && !strcmp(argv[3], "interim");
+    		    char* lang = argv[2];
+        		int interim = argc > 3 && !strcmp(argv[3], "interim");
 				char *bugname = argc > 5 ? argv[5] : MY_BUG_NAME;
 				if (argc > 4 && !strcmp(argv[4], "stereo")) {
-          flags |= SMBF_WRITE_STREAM ;
-          flags |= SMBF_STEREO;
+          			flags |= SMBF_WRITE_STREAM ;
+          			flags |= SMBF_STEREO;
 				}
-    		switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_INFO, "%s start transcribing %s %s\n", bugname, lang, interim ? "interim": "complete");
-				status = start_capture(lsession, flags, lang, interim, bugname);
+    			switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_INFO, "%s start transcribing %s %s\n", bugname, lang, interim ? "interim": "complete");
+				status = start_capture(lsession, flags, lang, interim, bugname, version);
 			}
 			switch_core_session_rwunlock(lsession);
 		}
@@ -481,4 +564,3 @@ SWITCH_MODULE_SHUTDOWN_FUNCTION(mod_transcribe_shutdown)
 	switch_event_free_subclass(TRANSCRIBE_EVENT_PLAY_INTERRUPT);
 	return SWITCH_STATUS_SUCCESS;
 }
-

mod_google_transcribe/mod_google_transcribe.h

@@ -37,6 +37,10 @@ struct cap_cb {
 #else
 /* per-channel data */
 typedef void (*responseHandler_t)(switch_core_session_t* session, const char* json, const char* bugname);
+typedef enum GoogleCloudServiceVersion {
+    GoogleCloudServiceVersion_v1,
+    GoogleCloudServiceVersion_v2
+} GoogleCloudServiceVersion;
 
 struct cap_cb {
 	switch_mutex_t *mutex;