#include <switch.h>
#include <switch_json.h>
#include <string.h>
#include <string>
#include <mutex>
#include <thread>
#include <list>
#include <algorithm>
#include <functional>
#include <cassert>
#include <cstdlib>
#include <fstream>
#include <sstream>
#include <regex>

#include "mod_jambonz_transcribe.h"
#include "simple_buffer.h"
#include "parser.hpp"
#include "audio_pipe.hpp"

#define RTP_PACKETIZATION_PERIOD 20
#define FRAME_SIZE_8000  320 /*which means each 20ms frame as 320 bytes at 8 khz (1 channel only)*/

namespace {
  static bool hasDefaultCredentials = false;
  static const char* defaultApiKey = nullptr;
  static const char *requestedBufferSecs = std::getenv("MOD_AUDIO_FORK_BUFFER_SECS");
  static int nAudioBufferSecs = std::max(1, std::min(requestedBufferSecs ? ::atoi(requestedBufferSecs) : 2, 5));
  static const char *requestedNumServiceThreads = std::getenv("MOD_AUDIO_FORK_SERVICE_THREADS");
  static unsigned int nServiceThreads = std::max(1, std::min(requestedNumServiceThreads ? ::atoi(requestedNumServiceThreads) : 1, 5));
  static unsigned int idxCallCount = 0;
  static uint32_t playCount = 0;

  static int parse_ws_uri(switch_channel_t *channel, const char* szServerUri, char* host, char *path, unsigned int* pPort, int* pSslFlags) {
    int i = 0, offset;
    char server[MAX_WS_URL_LEN + MAX_PATH_LEN];
    char *saveptr;
    int flags = LCCSCF_USE_SSL;
    
    if (switch_true(switch_channel_get_variable(channel, "MOD_AUDIO_FORK_ALLOW_SELFSIGNED"))) {
      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "parse_ws_uri - allowing self-signed certs\n");
      flags |= LCCSCF_ALLOW_SELFSIGNED;
    }
    if (switch_true(switch_channel_get_variable(channel, "MOD_AUDIO_FORK_SKIP_SERVER_CERT_HOSTNAME_CHECK"))) {
      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "parse_ws_uri - skipping hostname check\n");
      flags |= LCCSCF_SKIP_SERVER_CERT_HOSTNAME_CHECK;
    }
    if (switch_true(switch_channel_get_variable(channel, "MOD_AUDIO_FORK_ALLOW_EXPIRED"))) {
      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "parse_ws_uri - allowing expired certs\n");
      flags |= LCCSCF_ALLOW_EXPIRED;
    }

    // get the scheme
    strncpy(server, szServerUri, MAX_WS_URL_LEN + MAX_PATH_LEN);
    if (0 == strncmp(server, "https://", 8) || 0 == strncmp(server, "HTTPS://", 8)) {
      *pSslFlags = flags;
      offset = 8;
      *pPort = 443;
    }
    else if (0 == strncmp(server, "wss://", 6) || 0 == strncmp(server, "WSS://", 6)) {
      *pSslFlags = flags;
      offset = 6;
      *pPort = 443;
    }
    else if (0 == strncmp(server, "http://", 7) || 0 == strncmp(server, "HTTP://", 7)) {
      offset = 7;
      *pSslFlags = 0;
      *pPort = 80;
    }
    else if (0 == strncmp(server, "ws://", 5) || 0 == strncmp(server, "WS://", 5)) {
      offset = 5;
      *pSslFlags = 0;
      *pPort = 80;
    }
    else {
      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "parse_ws_uri - error parsing uri %s: invalid scheme\n", szServerUri);;
      return 0;
    }

    std::string strHost(server + offset);
    std::regex re("^(.+?):?(\\d+)?(/.*)?$");
    std::smatch matches;
    if(std::regex_search(strHost, matches, re)) {
      /*
      for (int i = 0; i < matches.length(); i++) {
        switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "parse_ws_uri - %d: %s\n", i, matches[i].str().c_str());
      }
      */
      strncpy(host, matches[1].str().c_str(), MAX_WS_URL_LEN);
      if (matches[2].str().length() > 0) {
        *pPort = atoi(matches[2].str().c_str());
      }
      if (matches[3].str().length() > 0) {
        strncpy(path, matches[3].str().c_str(), MAX_PATH_LEN);
      }
      else {
        strcpy(path, "/");
      }
    } else {
      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "parse_ws_uri - invalid format %s\n", strHost.c_str());
      return 0;
    }
    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "parse_ws_uri - host %s, path %s\n", host, path);

    return 1;
  }

  static void reaper(private_t *tech_pvt) {
    std::shared_ptr<jambonz::AudioPipe> pAp;
    pAp.reset((jambonz::AudioPipe *)tech_pvt->pAudioPipe);
    tech_pvt->pAudioPipe = nullptr;

    std::thread t([pAp, tech_pvt]{
      pAp->finish();
      pAp->waitForClose();
      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "%s (%u) got remote close\n", tech_pvt->sessionId, tech_pvt->id);
    });
    t.detach();
  }

  static void destroy_tech_pvt(private_t *tech_pvt) {
    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "%s (%u) destroy_tech_pvt\n", tech_pvt->sessionId, tech_pvt->id);
    if (tech_pvt) {
      if (tech_pvt->mutex) {
        switch_mutex_destroy(tech_pvt->mutex);
        tech_pvt->mutex = nullptr;
      }
      if (tech_pvt->resampler) {
          speex_resampler_destroy(tech_pvt->resampler);
          tech_pvt->resampler = NULL;
      }
    }
  }

  static void sendStartMessage(switch_channel_t *channel, private_t* tech_pvt) {
    auto *pAudioPipe = static_cast<jambonz::AudioPipe*>(tech_pvt->pAudioPipe);
    const char* var;
    bool hasOptions = false;

    if (nullptr == pAudioPipe) return;

    cJSON* json = cJSON_CreateObject();
    cJSON_AddStringToObject(json, "type", "start");
    cJSON_AddStringToObject(json, "language", tech_pvt->language);
    cJSON_AddStringToObject(json, "format", "raw");
    cJSON_AddStringToObject(json, "encoding", "LINEAR16");
    cJSON_AddBoolToObject(json, "interimResults", tech_pvt->interim);
    cJSON_AddNumberToObject(json, "sampleRateHz", tech_pvt->sampling);
    if (var = switch_channel_get_variable(channel, "JAMBONZ_STT_OPTIONS")) {
      cJSON* jOptions = cJSON_Parse(var);
      if (jOptions)  {
        hasOptions = true;
        cJSON_AddItemToObject(json, "options", jOptions); 
      }
    }
    if (!hasOptions) {
      cJSON* jOptions = cJSON_CreateObject();   
      cJSON_AddItemToObject(json, "options", jOptions); 
    }
    char* jsonString = cJSON_PrintUnformatted(json);

    /* send */
    pAudioPipe->bufferForSending(jsonString);

    /* cleanup */
    free(jsonString);
    cJSON_Delete(json);
  }

  static void eventCallback(const char* sessionId, const char* bugname, jambonz::AudioPipe::NotifyEvent_t event, const char* message, bool finished) {
    switch_core_session_t* session = switch_core_session_locate(sessionId);
    if (session) {
      switch_channel_t *channel = switch_core_session_get_channel(session);
      switch_media_bug_t *bug = (switch_media_bug_t*) switch_channel_get_private(channel, bugname);
      if (bug) {
        private_t* tech_pvt = (private_t*) switch_core_media_bug_get_user_data(bug);
        if (tech_pvt) {
          switch (event) {
            case jambonz::AudioPipe::CONNECT_SUCCESS:
              switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_INFO, "connection successful\n");
              tech_pvt->responseHandler(session, TRANSCRIBE_EVENT_CONNECT_SUCCESS, NULL, tech_pvt->bugname, finished);
              sendStartMessage(channel, tech_pvt);
            break;
            case jambonz::AudioPipe::CONNECT_FAIL:
            {
              // first thing: we can no longer access the AudioPipe
              std::stringstream json;
              json << "{\"reason\":\"" << message << "\"}";
              tech_pvt->pAudioPipe = nullptr;
              tech_pvt->responseHandler(session, TRANSCRIBE_EVENT_CONNECT_FAIL, (char *) json.str().c_str(), tech_pvt->bugname, finished);
              switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_NOTICE, "connection failed: %s\n", message);
            }
            break;
            case jambonz::AudioPipe::CONNECTION_DROPPED:
              // first thing: we can no longer access the AudioPipe
              tech_pvt->pAudioPipe = nullptr;
              tech_pvt->responseHandler(session, TRANSCRIBE_EVENT_DISCONNECT, NULL, tech_pvt->bugname, finished);
              switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "connection dropped from far end\n");
            break;
            case jambonz::AudioPipe::CONNECTION_CLOSED_GRACEFULLY:
              // first thing: we can no longer access the AudioPipe
              tech_pvt->pAudioPipe = nullptr;
              switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "connection closed gracefully\n");
            break;
            case jambonz::AudioPipe::MESSAGE:
            {
              cJSON* jMessage = cJSON_Parse(message);
              if (!jMessage) {
                switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_ERROR, "discarding incoming message (not json): %s\n", message);
                break;
              }
              const char* type = cJSON_GetStringValue(cJSON_GetObjectItem(jMessage, "type"));
              if (type && 0 == strcmp(type, "error")) {
                tech_pvt->responseHandler(session, TRANSCRIBE_EVENT_ERROR, message, tech_pvt->bugname, finished);
              }
              else if (type && 0 == strcmp(type, "transcription")) {
                tech_pvt->responseHandler(session, TRANSCRIBE_EVENT_RESULTS, message, tech_pvt->bugname, finished);
              }
              else {
                switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_ERROR, "discarding incoming message: %s\n", message);
              }
              cJSON_Delete(jMessage);
            }
            break;

            default:
              switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_NOTICE, "got unexpected msg from jambonz %d:%s\n", event, message);
              break;
          }
        }
      }
      switch_core_session_rwunlock(session);
    }
  }
  switch_status_t fork_data_init(private_t *tech_pvt, switch_core_session_t *session, 
    int sampling, int desiredSampling, int channels, char *lang, int interim, 
    char* bugname, responseHandler_t responseHandler) {

    int err;
    switch_codec_implementation_t read_impl;
    switch_channel_t *channel = switch_core_session_get_channel(session);
    char host[MAX_WS_URL_LEN], path[MAX_PATH_LEN];
    unsigned int port;
    int sslFlags;

    const char* url = switch_channel_get_variable(channel, "JAMBONZ_STT_URL");
    if (!parse_ws_uri(channel, url, &host[0], &path[0], &port, &sslFlags)) {
      switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_ERROR, "invalid websocket uri: %s\n", url);
      return SWITCH_STATUS_FALSE;
    }

    switch_core_session_get_read_impl(session, &read_impl);
  
    memset(tech_pvt, 0, sizeof(private_t));
  
    switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "host: %s, port: %d, path: %s\n", host, port, path);

    strncpy(tech_pvt->sessionId, switch_core_session_get_uuid(session), MAX_SESSION_ID);
    strncpy(tech_pvt->host, host, MAX_WS_URL_LEN);
    strncpy(tech_pvt->path, path, MAX_PATH_LEN); 
    tech_pvt->port = port;
    tech_pvt->sslFlags = sslFlags;
    strncpy(tech_pvt->language, lang, MAX_LANG_LEN); 
    tech_pvt->interim = interim;
    tech_pvt->sampling = desiredSampling;
    tech_pvt->responseHandler = responseHandler;
    tech_pvt->channels = channels;
    tech_pvt->id = ++idxCallCount;
    tech_pvt->buffer_overrun_notified = 0;
    strncpy(tech_pvt->bugname, bugname, MAX_BUG_LEN);
    
    size_t buflen = LWS_PRE + (FRAME_SIZE_8000 * desiredSampling / 8000 * channels * 1000 / RTP_PACKETIZATION_PERIOD * nAudioBufferSecs);

    const char* apiKey = switch_channel_get_variable(channel, "JAMBONZ_STT_API_KEY");
    if (!apiKey && defaultApiKey) apiKey = defaultApiKey;
    else if (!apiKey) {
      switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_ERROR, "no jambonz api key provided\n");
      return SWITCH_STATUS_FALSE;
    }

    jambonz::AudioPipe* ap = new jambonz::AudioPipe(tech_pvt->sessionId, bugname, tech_pvt->host, tech_pvt->port, tech_pvt->path, 
      tech_pvt->sslFlags, buflen, read_impl.decoded_bytes_per_packet, apiKey, eventCallback);
    if (!ap) {
      switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_ERROR, "Error allocating AudioPipe\n");
      return SWITCH_STATUS_FALSE;
    }

    tech_pvt->pAudioPipe = static_cast<void *>(ap);

    switch_mutex_init(&tech_pvt->mutex, SWITCH_MUTEX_NESTED, switch_core_session_get_pool(session));

    if (desiredSampling != sampling) {
      switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "(%u) resampling from %u to %u\n", tech_pvt->id, sampling, desiredSampling);
      tech_pvt->resampler = speex_resampler_init(channels, sampling, desiredSampling, SWITCH_RESAMPLE_QUALITY, &err);
      if (0 != err) {
        switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_ERROR, "Error initializing resampler: %s.\n", speex_resampler_strerror(err));
        return SWITCH_STATUS_FALSE;
      }
    }
    else {
      switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "(%u) no resampling needed for this call\n", tech_pvt->id);
    }

    switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "(%u) fork_data_init\n", tech_pvt->id);

    return SWITCH_STATUS_SUCCESS;
  }

  void lws_logger(int level, const char *line) {
    switch_log_level_t llevel = SWITCH_LOG_DEBUG;

    switch (level) {
      case LLL_ERR: llevel = SWITCH_LOG_ERROR; break;
      case LLL_WARN: llevel = SWITCH_LOG_WARNING; break;
      case LLL_NOTICE: llevel = SWITCH_LOG_NOTICE; break;
      case LLL_INFO: llevel = SWITCH_LOG_INFO; break;
      break;
    }
	  switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "%s\n", line);
  }
}


extern "C" {
  switch_status_t jb_transcribe_init() {
    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "mod_jambonz_transcribe: audio buffer (in secs):    %d secs\n", nAudioBufferSecs);
 
    int logs = LLL_ERR | LLL_WARN | LLL_NOTICE ;
    // | LLL_INFO | LLL_PARSER | LLL_HEADER | LLL_EXT | LLL_CLIENT  | LLL_LATENCY | LLL_DEBUG ;
    
    jambonz::AudioPipe::initialize(logs, lws_logger);
		switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "AudioPipe::initialize completed\n");

		const char* apiKey = std::getenv("JAMBONZ_STT_API_KEY");
		if (NULL == apiKey) {
			switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, 
				"\"JAMBONZ_STT_API_KEY\" env var not set; authentication will expect channel variables of same names to be set\n");
		}
		else {
			hasDefaultCredentials = true;
      defaultApiKey = apiKey;
		}
		return SWITCH_STATUS_SUCCESS;
  }

  switch_status_t jb_transcribe_cleanup() {
    bool cleanup = false;
    cleanup = jambonz::AudioPipe::deinitialize();
    if (cleanup == true) {
        return SWITCH_STATUS_SUCCESS;
    }
    return SWITCH_STATUS_FALSE;
  }
	
  switch_status_t jb_transcribe_session_init(switch_core_session_t *session, 
    responseHandler_t responseHandler, uint32_t samples_per_second, int desiredSampling, uint32_t channels, 
    char* lang, int interim, char* bugname, void **ppUserData)
  {    	
    int err;

    // allocate per-session data structure
    private_t* tech_pvt = (private_t *) switch_core_session_alloc(session, sizeof(private_t));
    if (!tech_pvt) {
      switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_ERROR, "error allocating memory!\n");
      return SWITCH_STATUS_FALSE;
    }

    if (SWITCH_STATUS_SUCCESS != fork_data_init(tech_pvt, session, samples_per_second, desiredSampling, channels, lang, interim, bugname, responseHandler)) {
      destroy_tech_pvt(tech_pvt);
      return SWITCH_STATUS_FALSE;
    }

    *ppUserData = tech_pvt;

    jambonz::AudioPipe *pAudioPipe = static_cast<jambonz::AudioPipe *>(tech_pvt->pAudioPipe);
    switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "connecting now\n");
    pAudioPipe->connect();
    switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "connection in progress\n");
    return SWITCH_STATUS_SUCCESS;
  }

	switch_status_t jb_transcribe_session_stop(switch_core_session_t *session,int channelIsClosing, char* bugname) {
    switch_channel_t *channel = switch_core_session_get_channel(session);
    switch_media_bug_t *bug = (switch_media_bug_t*) switch_channel_get_private(channel, bugname);
    if (!bug) {
      switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "jb_transcribe_session_stop: no bug %s - websocket conection already closed\n", bugname);
      return SWITCH_STATUS_FALSE;
    }
    private_t* tech_pvt = (private_t*) switch_core_media_bug_get_user_data(bug);
    if (!tech_pvt) return SWITCH_STATUS_FALSE;

    // close connection and get final responses
    switch_mutex_lock(tech_pvt->mutex);
    {
      // get the bug again, now that we are under lock
      switch_media_bug_t *bug = (switch_media_bug_t*) switch_channel_get_private(channel, bugname);
      if (bug) {
        switch_channel_set_private(channel, bugname, NULL);
        uint32_t id = tech_pvt->id;
        switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "(%u) jb_transcribe_session_stop\n", id);
        if (!channelIsClosing) {
          switch_core_media_bug_remove(session, &bug);
        }
        jambonz::AudioPipe *pAudioPipe = static_cast<jambonz::AudioPipe *>(tech_pvt->pAudioPipe);
        if (pAudioPipe) reaper(tech_pvt);
        switch_mutex_unlock(tech_pvt->mutex);
        destroy_tech_pvt(tech_pvt);
        switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "(%u) jb_transcribe_session_stop, bug removed\n", id);
        return SWITCH_STATUS_SUCCESS;
      }
      else {
        switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "jb_transcribe_session_stop: race condition, previous close completed\n");
      }
      switch_mutex_unlock(tech_pvt->mutex);
    }
    return SWITCH_STATUS_FALSE;
  }
	
	switch_bool_t jb_transcribe_frame(switch_core_session_t *session, switch_media_bug_t *bug) {
    private_t* tech_pvt = (private_t*) switch_core_media_bug_get_user_data(bug);
    size_t inuse = 0;
    bool dirty = false;
    char *p = (char *) "{\"msg\": \"buffer overrun\"}";

    if (!tech_pvt) return SWITCH_TRUE;
    
    if (switch_mutex_trylock(tech_pvt->mutex) == SWITCH_STATUS_SUCCESS) {
      if (!tech_pvt->pAudioPipe) {
        switch_mutex_unlock(tech_pvt->mutex);
        return SWITCH_TRUE;
      }
      jambonz::AudioPipe *pAudioPipe = static_cast<jambonz::AudioPipe *>(tech_pvt->pAudioPipe);
      if (pAudioPipe->getLwsState() != jambonz::AudioPipe::LWS_CLIENT_CONNECTED) {
        switch_mutex_unlock(tech_pvt->mutex);
        return SWITCH_TRUE;
      }

      pAudioPipe->lockAudioBuffer();
      size_t available = pAudioPipe->binarySpaceAvailable();
      if (NULL == tech_pvt->resampler) {
        switch_frame_t frame = { 0 };
        frame.data = pAudioPipe->binaryWritePtr();
        frame.buflen = available;
        while (true) {

          // check if buffer would be overwritten; dump packets if so
          if (available < pAudioPipe->binaryMinSpace()) {
            if (!tech_pvt->buffer_overrun_notified) {
              tech_pvt->buffer_overrun_notified = 1;
              tech_pvt->responseHandler(session, TRANSCRIBE_EVENT_BUFFER_OVERRUN, NULL, tech_pvt->bugname, 0);
            }
            switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_ERROR, "(%u) dropping packets!\n", 
              tech_pvt->id);
            pAudioPipe->binaryWritePtrResetToZero();

            frame.data = pAudioPipe->binaryWritePtr();
            frame.buflen = available = pAudioPipe->binarySpaceAvailable();
          }

          switch_status_t rv = switch_core_media_bug_read(bug, &frame, SWITCH_TRUE);
          if (rv != SWITCH_STATUS_SUCCESS) break;
          if (frame.datalen) {
            pAudioPipe->binaryWritePtrAdd(frame.datalen);
            frame.buflen = available = pAudioPipe->binarySpaceAvailable();
            frame.data = pAudioPipe->binaryWritePtr();
            dirty = true;
          }
        }
      }
      else {
        uint8_t data[SWITCH_RECOMMENDED_BUFFER_SIZE];
        switch_frame_t frame = { 0 };
        frame.data = data;
        frame.buflen = SWITCH_RECOMMENDED_BUFFER_SIZE;
        while (switch_core_media_bug_read(bug, &frame, SWITCH_TRUE) == SWITCH_STATUS_SUCCESS) {
          if (frame.datalen) {
            spx_uint32_t out_len = available >> 1;  // space for samples which are 2 bytes
            spx_uint32_t in_len = frame.samples;

            speex_resampler_process_interleaved_int(tech_pvt->resampler, 
              (const spx_int16_t *) frame.data, 
              (spx_uint32_t *) &in_len, 
              (spx_int16_t *) ((char *) pAudioPipe->binaryWritePtr()),
              &out_len);

            if (out_len > 0) {
              // bytes written = num samples * 2 * num channels
              size_t bytes_written = out_len << tech_pvt->channels;
              pAudioPipe->binaryWritePtrAdd(bytes_written);
              available = pAudioPipe->binarySpaceAvailable();
              dirty = true;
            }
            if (available < pAudioPipe->binaryMinSpace()) {
              if (!tech_pvt->buffer_overrun_notified) {
                tech_pvt->buffer_overrun_notified = 1;
                switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_ERROR, "(%u) dropping packets!\n", 
                  tech_pvt->id);
                tech_pvt->responseHandler(session, TRANSCRIBE_EVENT_BUFFER_OVERRUN, NULL, tech_pvt->bugname, 0);
              }
              break;
            }
          }
        }
      }

      pAudioPipe->unlockAudioBuffer();
      switch_mutex_unlock(tech_pvt->mutex);
    }
    return SWITCH_TRUE;
  }
}