wip

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

mod_google_tts/LICENSE

@@ -0,0 +1,8 @@
+Copyright 2023, Drachtio Communications Services, LLC
+
+Permission is hereby granted, free of charge, to any person obtaining a copy of this software and associated documentation files (the "Software"), to deal in the Software without restriction, including without limitation the rights to use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies of the Software, and to permit persons to whom the Software is furnished to do so, subject to the following conditions:
+
+The above copyright notice and this permission notice shall be included in all copies or substantial portions of the Software.
+
+THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
+

mod_google_tts/Makefile.am

@@ -0,0 +1,9 @@
+include $(top_srcdir)/build/modmake.rulesam
+MODNAME=mod_google_tts
+
+mod_LTLIBRARIES = mod_google_tts.la
+mod_google_tts_la_SOURCES  = mod_google_tts.c google_glue.cpp
+mod_google_tts_la_CFLAGS   = $(AM_CFLAGS)
+mod_google_tts_la_CXXFLAGS = -I $(top_srcdir)/libs/googleapis/gens $(AM_CXXFLAGS) -std=c++17
+mod_google_tts_la_LIBADD   = $(switch_builddir)/libfreeswitch.la
+mod_google_tts_la_LDFLAGS  = -avoid-version -module -no-undefined -shared `pkg-config --libs grpc++ grpc` 

mod_google_tts/README.md

@@ -0,0 +1,3 @@
+# mod_google_tts
+
+A Freeswitch module that allows speak text to speech audio from Google stream.

mod_google_tts/google_glue.cpp

@@ -0,0 +1,405 @@
+#include "mod_google_tts.h"
+#include <switch.h>
+
+#include <boost/circular_buffer.hpp>
+
+#include <cstdlib>
+#include <string>
+#include <chrono>
+#include <thread>
+
+#include "google/cloud/texttospeech/v1/cloud_tts.grpc.pb.h"
+#include <grpc++/grpc++.h>
+
+using google::cloud::texttospeech::v1::SynthesizeSpeechRequest;
+using google::cloud::texttospeech::v1::SynthesizeSpeechResponse;
+using google::cloud::texttospeech::v1::TextToSpeech;
+using google::cloud::texttospeech::v1::Voice;
+using google::cloud::texttospeech::v1::SsmlVoiceGender;
+using google::cloud::texttospeech::v1::SsmlVoiceGender_Name;
+using google::cloud::texttospeech::v1::SynthesisInput;
+using google::cloud::texttospeech::v1::AudioEncoding;
+
+#define BUFFER_SIZE 8129
+
+typedef boost::circular_buffer<uint16_t> CircularBuffer_t;
+
+static std::string fullDirPath;
+
+static std::shared_ptr<grpc::Channel> create_grpc_channel(switch_channel_t *channel) {
+  const char* google_uri = "texttospeech.googleapis.com";
+  const char* var;
+  if (var = switch_channel_get_variable(channel, "GOOGLE_TTS_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);
+  }
+}
+
+static void start_synthesis(const char* text, google_t* g) {
+    try {
+      SynthesizeSpeechRequest request;
+      SynthesizeSpeechResponse response;
+      grpc::ClientContext context;
+      auto input = request.mutable_input();
+      auto voice = request.mutable_voice();
+      auto custom_voice = voice->mutable_custom_voice();
+      auto audio_config = request.mutable_audio_config();
+      /* lock and unlock session */
+      switch_core_session_t *psession = switch_core_session_locate(g->session_id);
+      switch_channel_t *swChannel = switch_core_session_get_channel(psession);
+      switch_core_session_rwunlock(psession);
+      auto channel = create_grpc_channel(swChannel);
+      auto stub = TextToSpeech::NewStub(channel);
+
+      if (strstr(text, "<speak") == text) {
+        input->set_ssml(text);
+      }
+      else {
+        input->set_text(text);
+      }
+      if (g->gender) {
+        if (strcmp(g->gender, "MALE") == 0) {
+          voice->set_ssml_gender(google::cloud::texttospeech::v1::SsmlVoiceGender::MALE);
+        } else if (strcmp(g->gender, "FEMALE") == 0) {
+          voice->set_ssml_gender(google::cloud::texttospeech::v1::SsmlVoiceGender::FEMALE);
+        } else if (strcmp(g->gender, "NEUTRAL") == 0) {
+          voice->set_ssml_gender(google::cloud::texttospeech::v1::SsmlVoiceGender::NEUTRAL);
+        } else {
+          voice->set_ssml_gender(google::cloud::texttospeech::v1::SsmlVoiceGender::SSML_VOICE_GENDER_UNSPECIFIED);
+        }
+      }
+
+      if (g->model) {
+        custom_voice->set_model(g->model);
+        if (strcmp(g->reported_usage, "OFFLINE") == 0) {
+          custom_voice->set_reported_usage(google::cloud::texttospeech::v1::CustomVoiceParams_ReportedUsage::CustomVoiceParams_ReportedUsage_OFFLINE);
+        } else if (strcmp(g->reported_usage, "REALTIME") == 0) {
+          custom_voice->set_reported_usage(google::cloud::texttospeech::v1::CustomVoiceParams_ReportedUsage::CustomVoiceParams_ReportedUsage_REALTIME);
+        } else {
+          custom_voice->set_reported_usage(google::cloud::texttospeech::v1::CustomVoiceParams_ReportedUsage::CustomVoiceParams_ReportedUsage_REPORTED_USAGE_UNSPECIFIED);
+        }
+      } else {
+        voice->set_name(g->voice_name);
+      }
+      voice->set_language_code(g->language);
+      audio_config->set_audio_encoding(AudioEncoding::LINEAR16);
+      audio_config->set_sample_rate_hertz(8000);
+      grpc::Status status = stub->SynthesizeSpeech(&context, request, &response);
+      if (!status.ok()) {
+        switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, 
+          "start_synthesis: error synthesizing speech: %s: details: %s\n", 
+          status.error_message().c_str(), status.error_details().c_str()); 
+        return;
+      }
+      g->response_code = 200;
+      auto audioData = response.audio_content();
+      if (g->flushed) return;
+      bool fireEvent = false;
+      CircularBuffer_t *cBuffer = (CircularBuffer_t *) g->circularBuffer;
+      if (g->file) {
+        // google return linear16 wav data that contains 44 bytes wav header, let remove them
+        fwrite(audioData.data() + 44, 1, audioData.size() - 44, g->file);
+      }
+
+      /**
+       * this sort of reinterpretation can be dangerous as a general rule, but in this case we know that the data
+       * is 16-bit PCM, so it's safe to do this and its much faster than copying the data byte by byte
+       */
+      // google return linear16 wav data that contains 44 bytes wav header, let remove them
+      const uint16_t* begin = reinterpret_cast<const uint16_t*>(audioData.data() + 44);
+      const uint16_t* end = reinterpret_cast<const uint16_t*>(audioData.data() + audioData.size());
+
+      /* lock as briefly as possible */
+      switch_mutex_lock(g->mutex);
+      if (cBuffer->capacity() - cBuffer->size() < audioData.size()) {
+        cBuffer->set_capacity(cBuffer->size() + std::max( audioData.size(), (size_t)BUFFER_SIZE));
+      }
+      cBuffer->insert(cBuffer->end(), begin, end);
+      switch_mutex_unlock(g->mutex);
+
+      if (0 == g->reads++) {
+        fireEvent = true;
+      }
+
+      if (fireEvent && g->session_id) {
+        auto endTime = std::chrono::high_resolution_clock::now();
+        auto startTime = *static_cast<std::chrono::time_point<std::chrono::high_resolution_clock>*>(g->startTime);
+        auto duration = std::chrono::duration_cast<std::chrono::milliseconds>(endTime - startTime);
+        auto time_to_first_byte_ms = std::to_string(duration.count());
+        switch_core_session_t* session = switch_core_session_locate(g->session_id);
+        if (session) {
+          switch_channel_t *channel = switch_core_session_get_channel(session);
+          switch_core_session_rwunlock(session);
+          if (channel) {
+            switch_event_t *event;
+            if (switch_event_create(&event, SWITCH_EVENT_PLAYBACK_START) == SWITCH_STATUS_SUCCESS) {
+              switch_channel_event_set_data(channel, event);
+              switch_event_add_header_string(event, SWITCH_STACK_BOTTOM, "Playback-File-Type", "tts_stream");
+              switch_event_add_header_string(event, SWITCH_STACK_BOTTOM, "variable_tts_time_to_first_byte_ms", time_to_first_byte_ms.c_str());
+              if (g->cache_filename) {
+                switch_event_add_header_string(event, SWITCH_STACK_BOTTOM, "variable_tts_cache_filename", g->cache_filename);
+              }
+              switch_event_fire(&event);
+            } else {
+              switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "mod_google_tts start_synthesis: failed to create event\n");
+            }
+          }else {
+            switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "mod_google_tts start_synthesis: channel not found\n");
+          }
+        }
+      }
+    } catch (const std::exception& e) {
+        g->response_code = 500;
+        switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "mod_google_tts: Exception in start_synthesis %s\n",  e.what());
+    }
+    g->draining = 1;
+}
+
+extern "C" {
+  switch_status_t google_speech_load() {
+    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "google_speech_loading..\n");
+
+    /* create temp folder for cache files */
+    const char* baseDir = std::getenv("JAMBONZ_TMP_CACHE_FOLDER");
+    if (!baseDir) {
+      baseDir = "/tmp/";
+    }
+    if (strcmp(baseDir, "/") == 0) {
+      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "failed to create folder %s\n", baseDir);
+      return SWITCH_STATUS_FALSE;
+    }
+
+    fullDirPath = std::string(baseDir) + "jambonz-tts-cache-files";
+
+    // Create the directory with read, write, and execute permissions for everyone
+    mode_t oldMask = umask(0);
+    int result = mkdir(fullDirPath.c_str(), S_IRWXU | S_IRWXG | S_IRWXO);
+    umask(oldMask);
+    if (result != 0) {
+      if (errno != EEXIST) {
+        switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "failed to create folder %s\n", fullDirPath.c_str());
+        fullDirPath = "";
+      }
+      else switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "folder %s already exists\n", fullDirPath.c_str());
+    }
+    else {
+      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "created folder %s\n", fullDirPath.c_str());
+    }
+
+    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "google_speech_loaded..\n");
+
+    return SWITCH_STATUS_SUCCESS;
+  }
+
+    switch_status_t google_speech_open(google_t* google) {
+    return SWITCH_STATUS_SUCCESS;
+  }
+
+  switch_status_t google_speech_feed_tts(google_t* g, char* text, switch_speech_flag_t *flags) {
+    const int MAX_CHARS = 20;
+    char tempText[MAX_CHARS + 4]; // +4 for the ellipsis and null terminator
+
+    if (strlen(text) > MAX_CHARS) {
+        strncpy(tempText, text, MAX_CHARS);
+        strcpy(tempText + MAX_CHARS, "...");
+    } else {
+        strcpy(tempText, text);
+    }
+
+    /* open cache file */
+    if (g->cache_audio && fullDirPath.length() > 0) {
+      switch_uuid_t uuid;
+      char uuid_str[SWITCH_UUID_FORMATTED_LENGTH + 1];
+      char outfile[512] = "";
+      int fd;
+
+      switch_uuid_get(&uuid);
+      switch_uuid_format(uuid_str, &uuid);
+
+      switch_snprintf(outfile, sizeof(outfile), "%s%s%s.r8", fullDirPath.c_str(), SWITCH_PATH_SEPARATOR, uuid_str);
+      g->cache_filename = strdup(outfile);
+      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "writing audio cache file to %s\n", g->cache_filename);
+
+      mode_t oldMask = umask(0);
+      fd = open(outfile, O_WRONLY | O_CREAT | O_TRUNC, S_IRUSR | S_IWUSR | S_IRGRP | S_IWGRP | S_IROTH | S_IWOTH);
+      umask(oldMask);
+      if (fd == -1 ) {
+        switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error opening cache file %s: %s\n", outfile, strerror(errno));
+      }
+      else {
+        g->file = fdopen(fd, "wb");
+        if (!g->file) {
+          close(fd);
+          switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error opening cache file %s: %s\n", outfile, strerror(errno));
+        }
+      }
+    }
+
+    if (!g->language) {
+      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "google_speech_feed_tts: no language provided\n");
+      return SWITCH_STATUS_FALSE;
+    }
+
+    if (!g->voice_name && !g->model) {
+      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "google_speech_feed_tts: no voice_name or model provided\n");
+      return SWITCH_STATUS_FALSE;
+    }
+
+    if (g->rate != 8000 /*Hz*/) {
+      int err;
+      g->resampler = speex_resampler_init(1, 8000, g->rate, SWITCH_RESAMPLE_QUALITY, &err);
+      if (0 != err) {
+        switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error initializing resampler: %s.\n", speex_resampler_strerror(err));
+        return SWITCH_STATUS_FALSE;
+      }
+    }
+
+    std::chrono::time_point<std::chrono::high_resolution_clock>* ptr = new std::chrono::time_point<std::chrono::high_resolution_clock>(std::chrono::high_resolution_clock::now());
+    g->startTime = ptr;
+
+    g->circularBuffer = (void *) new CircularBuffer_t(BUFFER_SIZE);
+
+    std::thread(start_synthesis, text, g).detach();
+    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "google_speech_feed_tts sent synthesize request\n");
+    return SWITCH_STATUS_SUCCESS;
+  }
+
+  switch_status_t google_speech_read_tts(google_t* g, void *data, size_t *datalen, switch_speech_flag_t *flags) {
+    CircularBuffer_t *cBuffer = (CircularBuffer_t *) g->circularBuffer;
+    std::vector<uint16_t> pcm_data;
+
+    if (g->response_code > 0 && g->response_code != 200) {
+      switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "google_speech_read_tts, returning failure\n") ;
+      return SWITCH_STATUS_FALSE;
+    }
+    if (g->flushed) {
+      return SWITCH_STATUS_BREAK;
+    }
+    switch_mutex_lock(g->mutex);
+    size_t bufSize = cBuffer->size();
+    if (cBuffer->empty()) {
+      switch_mutex_unlock(g->mutex);
+      if (g->draining) {
+        return SWITCH_STATUS_BREAK;
+      }
+      /* no audio available yet so send silence */
+      memset(data, 255, *datalen);
+      return SWITCH_STATUS_SUCCESS;
+    }
+    // google returned 8000hz 16 bit data, we have to take enough data based on call sample rate.
+    size_t size = std::min((*datalen/(2 * g->rate / 8000)), bufSize);
+    pcm_data.insert(pcm_data.end(), cBuffer->begin(), cBuffer->begin() + size);
+    cBuffer->erase(cBuffer->begin(), cBuffer->begin() + size);
+    switch_mutex_unlock(g->mutex);
+
+    size_t data_size = pcm_data.size();
+
+    if (g->resampler) {
+      std::vector<int16_t> in(pcm_data.begin(), pcm_data.end());
+
+      std::vector<int16_t> out((*datalen));
+      spx_uint32_t in_len = data_size;
+      spx_uint32_t out_len = out.size();
+
+      speex_resampler_process_interleaved_int(g->resampler, in.data(), &in_len, out.data(), &out_len);
+
+      if (out_len > out.size()) {
+        switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Resampler output exceeded maximum buffer size!\n");
+        return SWITCH_STATUS_FALSE;
+      }
+
+      memcpy(data, out.data(), out_len * sizeof(int16_t));
+      *datalen = out_len * sizeof(int16_t);
+    } else {
+      memcpy(data, pcm_data.data(), data_size * sizeof(int16_t));
+      *datalen = data_size * sizeof(int16_t);
+    }
+
+    return SWITCH_STATUS_SUCCESS;
+  }
+
+  switch_status_t google_speech_flush_tts(google_t* g) {
+    bool download_complete = g->response_code == 200;
+    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "google_speech_flush_tts, download complete? %s\n", download_complete ? "yes" : "no") ;
+
+    CircularBuffer_t *cBuffer = (CircularBuffer_t *) g->circularBuffer;
+    delete cBuffer;
+    g->circularBuffer = nullptr ;
+    delete static_cast<std::chrono::time_point<std::chrono::high_resolution_clock>*>(g->startTime);
+    g->startTime = nullptr;
+
+    g->flushed = 1;
+    if (!download_complete) {
+      if (g->file) {
+        //switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "closing audio cache file %s because download was interrupted\n", g->cache_filename);
+        if (fclose(g->file) != 0) {
+          switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "error closing audio cache file\n");
+        }
+        g->file = nullptr ;
+      }
+
+      if (g->cache_filename) {
+        //switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "removing audio cache file %s because download was interrupted\n", g->cache_filename);
+        if (unlink(g->cache_filename) != 0) {
+          switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "cleanupConn: error removing audio cache file %s: %d:%s\n", 
+            g->cache_filename, errno, strerror(errno));
+        }
+        free(g->cache_filename);
+        g->cache_filename = nullptr ;
+      }
+    }
+    if (g->session_id) {
+      switch_core_session_t* session = switch_core_session_locate(g->session_id);
+      if (session) {
+        switch_channel_t *channel = switch_core_session_get_channel(session);
+
+        /* unlock as quickly as possible */
+        switch_core_session_rwunlock(session);
+        if (channel) {
+          switch_event_t *event;
+          if (switch_event_create(&event, SWITCH_EVENT_PLAYBACK_STOP) == SWITCH_STATUS_SUCCESS) {
+            switch_channel_event_set_data(channel, event);
+            switch_event_add_header_string(event, SWITCH_STACK_BOTTOM, "Playback-File-Type", "tts_stream");
+            switch_event_add_header_string(event, SWITCH_STACK_BOTTOM, "variable_tts_google_response_code", std::to_string(g->response_code).c_str());
+            if (g->cache_filename && download_complete) {
+              switch_event_add_header_string(event, SWITCH_STACK_BOTTOM, "variable_tts_cache_filename", g->cache_filename);
+            }
+            if (!download_complete && g->err_msg) {
+              switch_event_add_header_string(event, SWITCH_STACK_BOTTOM, "variable_tts_error", g->err_msg);
+            }
+            switch_event_fire(&event);
+          }
+          else {
+            switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "write_cb: failed to create event\n");
+          }
+        }
+        else {
+          switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "write_cb: channel not found\n");
+        }
+      }
+    }
+
+    return SWITCH_STATUS_SUCCESS;
+  }
+
+  switch_status_t google_speech_close(google_t* g) {
+    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "google_speech_close\n") ;
+    if (g->resampler) {
+      speex_resampler_destroy(g->resampler);
+    }
+
+    g->resampler = NULL;
+    return SWITCH_STATUS_SUCCESS;
+  }
+
+  switch_status_t google_speech_unload() {
+    return SWITCH_STATUS_SUCCESS;
+  }
+
+} 

mod_google_tts/google_glue.h

@@ -0,0 +1,12 @@
+#ifndef __GOOGLE_TTS_GLUE_H__
+#define __GOOGLE_TTS_GLUE_H__
+
+switch_status_t google_speech_load();
+switch_status_t google_speech_open(google_t* google);
+switch_status_t google_speech_feed_tts(google_t* google, char* text, switch_speech_flag_t *flags);
+switch_status_t google_speech_read_tts(google_t* google, void *data, size_t *datalen, switch_speech_flag_t *flags);
+switch_status_t google_speech_flush_tts(google_t* google);
+switch_status_t google_speech_close(google_t* google);
+switch_status_t google_speech_unload();
+
+#endif

mod_google_tts/mod_google_tts.c

@@ -0,0 +1,154 @@
+#include "mod_google_tts.h"
+#include "google_glue.h"
+
+SWITCH_MODULE_LOAD_FUNCTION(mod_google_tts_load);
+SWITCH_MODULE_SHUTDOWN_FUNCTION(mod_google_tts_shutdown);
+SWITCH_MODULE_DEFINITION(mod_google_tts, mod_google_tts_load, mod_google_tts_shutdown, NULL);
+
+static void clearGoogle(google_t* g, int freeAll) {
+  switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "clearGoogle\n");
+
+  if (g->cache_filename) free(g->cache_filename);
+  if (g->model) free(g->model);
+  if (g->reported_usage) free(g->reported_usage);
+  if (g->gender) free(g->gender);
+  if (g->language) free(g->language);
+  if (g->err_msg) free(g->err_msg);
+  
+  g->cache_filename = NULL;
+  g->model = NULL;
+  g->reported_usage = NULL;
+  g->gender = NULL;
+  g->language = NULL;
+  g->err_msg = NULL;
+
+
+  if (freeAll) {
+    if (g->voice_name) free(g->voice_name);
+    if (g->session_id) free(g->session_id);
+    g->voice_name = NULL;
+    g->session_id = NULL;
+  }
+
+}
+
+static google_t * createOrRetrievePrivateData(switch_speech_handle_t *sh) {
+  google_t *g = (google_t *) sh->private_info;  
+  if (!g) {
+    g = switch_core_alloc(sh->memory_pool, sizeof(*g));
+  	sh->private_info = g;
+    memset(g, 0, sizeof(*g));
+    switch_mutex_init(&g->mutex, SWITCH_MUTEX_NESTED, sh->memory_pool);
+    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "allocated google_t\n");
+  }
+  return g;
+}
+
+switch_status_t g_speech_open(switch_speech_handle_t *sh, const char *voice_name, int rate, int channels, switch_speech_flag_t *flags)
+{
+  google_t *g = createOrRetrievePrivateData(sh);
+  g->voice_name = strdup(voice_name);
+  g->rate = rate;
+  switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "g_speech_open voice: %s, rate %d, channels %d\n", voice_name, rate, channels);
+  return google_speech_open(g);
+}
+
+static switch_status_t g_speech_close(switch_speech_handle_t *sh, switch_speech_flag_t *flags)
+{
+  switch_status_t rc;
+  google_t *g = createOrRetrievePrivateData(sh);
+  switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "g_speech_close\n");
+
+  switch_mutex_destroy(g->mutex);
+
+  rc = google_speech_close(g);
+  clearGoogle(g, 1);
+  return rc;
+}
+
+/**
+ * Freeswitch will call this function to feed us text to speak
+ */
+static switch_status_t g_speech_feed_tts(switch_speech_handle_t *sh, char *text, switch_speech_flag_t *flags)
+{
+  google_t *g = createOrRetrievePrivateData(sh);
+  g->draining = 0;
+  g->reads = 0;
+  g->flushed = 0;
+
+  return google_speech_feed_tts(g, text, flags);
+}
+
+/**
+ * Freeswitch calls periodically to get some rendered audio in L16 format. We can provide up to 8k of audio at a time.
+ */
+static switch_status_t g_speech_read_tts(switch_speech_handle_t *sh, void *data, size_t *datalen, switch_speech_flag_t *flags)
+{
+  google_t *g = createOrRetrievePrivateData(sh);
+  return google_speech_read_tts(g, data, datalen, flags);
+}
+
+/**
+ * This is called at the end, not sure exactly what we need to do here..
+ */
+static void g_speech_flush_tts(switch_speech_handle_t *sh)
+{
+  google_t *g = createOrRetrievePrivateData(sh);
+  google_speech_flush_tts(g);
+
+  clearGoogle(g, 0);
+}
+
+static void g_text_param_tts(switch_speech_handle_t *sh, char *param, const char *val)
+{
+  google_t *g = createOrRetrievePrivateData(sh);
+  switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "g_text_param_tts: %s=%s\n", param, val);
+  if (0 == strcmp(param, "voice")) {
+    if (g->voice_name) free(g->voice_name);
+    g->voice_name = strdup(val);
+  } else if (0 == strcmp(param, "model")) {
+    if (g->model) free(g->model);
+    g->model = strdup(val);
+  } else if (0 == strcmp(param, "reported_usage")) {
+    if (g->reported_usage) free(g->reported_usage);
+    g->reported_usage = strdup(val);
+  } else if (0 == strcmp(param, "language")) {
+    if (g->language) free(g->language);
+    g->language = strdup(val);
+  } else if (0 == strcmp(param, "session-uuid")) {
+    if (g->session_id) free(g->session_id);
+    g->session_id = strdup(val);
+  } else if (0 == strcmp(param, "write_cache_file") && switch_true(val)) {
+    g->cache_audio = 1;
+  }
+}
+
+static void g_numeric_param_tts(switch_speech_handle_t *sh, char *param, int val)
+{
+}
+static void g_float_param_tts(switch_speech_handle_t *sh, char *param, double val)
+{
+}
+
+SWITCH_MODULE_LOAD_FUNCTION(mod_google_tts_load)
+{
+  switch_speech_interface_t *speech_interface;
+
+  *module_interface = switch_loadable_module_create_module_interface(pool, modname);
+  speech_interface = switch_loadable_module_create_interface(*module_interface, SWITCH_SPEECH_INTERFACE);
+  speech_interface->interface_name = "google";
+  speech_interface->speech_open = g_speech_open;
+  speech_interface->speech_close = g_speech_close;
+  speech_interface->speech_feed_tts = g_speech_feed_tts;
+  speech_interface->speech_read_tts = g_speech_read_tts;
+	speech_interface->speech_flush_tts = g_speech_flush_tts;
+	speech_interface->speech_text_param_tts = g_text_param_tts;
+	speech_interface->speech_numeric_param_tts = g_numeric_param_tts;
+	speech_interface->speech_float_param_tts = g_float_param_tts;
+  return google_speech_load();
+}
+
+SWITCH_MODULE_SHUTDOWN_FUNCTION(mod_google_tts_shutdown)
+{
+  return google_speech_unload();
+}

mod_google_tts/mod_google_tts.h

@@ -0,0 +1,34 @@
+#ifndef __MOD_GOOGLE_TTS_H__
+#define __MOD_GOOGLE_TTS_H__
+
+#include <switch.h>
+#include <speex/speex_resampler.h>
+
+typedef struct google_data {
+  char *voice_name;
+  char *model;
+  char *reported_usage;
+  char *language;
+  char *gender;
+
+  /* result data */
+  long response_code;
+  char *session_id;
+  char *cache_filename;
+  char *err_msg;
+
+  int rate;
+  int draining;
+  int reads;
+  int cache_audio;
+  int flushed;
+
+  void *startTime;
+
+  FILE *file;
+  SpeexResamplerState *resampler;
+  void *circularBuffer;
+  switch_mutex_t *mutex;
+} google_t;
+
+#endif