mirror of
https://github.com/jambonz/freeswitch-modules.git
synced 2026-01-25 02:08:27 +00:00
Compare commits
6 Commits
| Author | SHA1 | Date | |
|---|---|---|---|
| 30f2189986 | |||
| ffde303446 | |||
| 7b4520c070 | |||
| 9f7a06ce56 | |||
| f7f8f52283 | |||
| 3f06a24b5d |
@@ -557,7 +557,7 @@ namespace {
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tech_pvt->streamingPreBuffer = nullptr;
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}
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if (nullptr == tech_pvt->pVecMarksInInventory) {
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if (tech_pvt->pVecMarksInInventory) {
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delete static_cast<std::deque<std::string>*>(tech_pvt->pVecMarksInInventory);
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tech_pvt->pVecMarksInInventory = nullptr;
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delete static_cast<std::deque<std::string>*>(tech_pvt->pVecMarksInUse);
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@@ -979,7 +979,7 @@ extern "C" {
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std::deque<std::string>* pVecInventory = nullptr;
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std::deque<std::string>* pVecInUse = nullptr;
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std::deque<std::string>* pVecCleared = nullptr;
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if (nullptr != tech_pvt->pVecMarksInUse) {
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if (tech_pvt->pVecMarksInInventory && tech_pvt->pVecMarksInUse && tech_pvt->pVecMarksCleared) {
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pVecInventory = static_cast<std::deque<std::string>*>(tech_pvt->pVecMarksInInventory);
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pVecInUse = static_cast<std::deque<std::string>*>(tech_pvt->pVecMarksInUse);
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pVecCleared = static_cast<std::deque<std::string>*>(tech_pvt->pVecMarksCleared);
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@@ -1032,7 +1032,7 @@ extern "C" {
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if (samplesToCopy > 0) {
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vector_add(fp, data, rframe->samples);
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} else if (pVecInventory != nullptr && pVecInventory->size()) {
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} else if (pVecInventory && pVecInventory->size()) {
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// no bidirectional audio to dub but still have some mark in inventory, send them now
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auto name = pVecInventory->front();
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pVecInventory->pop_front();
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@@ -72,10 +72,7 @@ public:
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if (switch_true(switch_channel_get_variable(channel, "AZURE_USE_OUTPUT_FORMAT_DETAILED"))) {
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speechConfig->SetOutputFormat(OutputFormat::Detailed);
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}
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if (nullptr != endpointId) {
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switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(psession), SWITCH_LOG_DEBUG, "setting endpoint id: %s\n", endpointId);
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speechConfig->SetEndpointId(endpointId);
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}
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if (!sdkInitialized && sdkLog) {
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sdkInitialized = true;
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speechConfig->SetProperty(PropertyId::Speech_LogFilename, sdkLog);
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@@ -104,11 +101,31 @@ public:
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languages.push_back( alt_langs[i]);
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switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(psession), SWITCH_LOG_DEBUG, "added alternative lang %s\n", alt_langs[i]);
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}
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auto autoDetectSourceLanguageConfig = AutoDetectSourceLanguageConfig::FromLanguages(languages);
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std::vector<std::shared_ptr<SourceLanguageConfig>> sourceLanguageConfigs;
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for (const auto& language : languages) {
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std::shared_ptr<SourceLanguageConfig> sourceLanguageConfig;
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if (endpointId != nullptr) {
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sourceLanguageConfig = SourceLanguageConfig::FromLanguage(language, endpointId);
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} else {
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sourceLanguageConfig = SourceLanguageConfig::FromLanguage(language);
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}
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sourceLanguageConfigs.push_back(sourceLanguageConfig);
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}
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// Create AutoDetectSourceLanguageConfig from SourceLanguageConfigs
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auto autoDetectSourceLanguageConfig = AutoDetectSourceLanguageConfig::FromSourceLanguageConfigs(sourceLanguageConfigs);
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m_recognizer = SpeechRecognizer::FromConfig(speechConfig, autoDetectSourceLanguageConfig, audioConfig);
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}
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else {
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auto sourceLanguageConfig = SourceLanguageConfig::FromLanguage(lang);
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std::shared_ptr<SourceLanguageConfig> sourceLanguageConfig;
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if (endpointId != nullptr) {
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sourceLanguageConfig = SourceLanguageConfig::FromLanguage(lang, endpointId);
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} else {
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sourceLanguageConfig = SourceLanguageConfig::FromLanguage(lang);
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}
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m_recognizer = SpeechRecognizer::FromConfig(speechConfig, sourceLanguageConfig, audioConfig);
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}
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@@ -487,20 +504,27 @@ extern "C" {
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auto read_codec = switch_core_session_get_read_codec(session);
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uint32_t sampleRate = read_codec->implementation->actual_samples_per_second;
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if (bug) {
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struct cap_cb* existing_cb = (struct cap_cb*) switch_core_media_bug_get_user_data(bug);
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GStreamer* existing_streamer = (GStreamer*) existing_cb->streamer;
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existing_cb->is_keep_alive = 0;
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if (!existing_streamer->hasConfigurationChanged(channels, lang, interim, sampleRate, region, subscriptionKey)) {
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switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "Reuse active azure connection.\n");
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try {
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struct cap_cb* existing_cb = (struct cap_cb*) switch_core_media_bug_get_user_data(bug);
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GStreamer* existing_streamer = (GStreamer*) existing_cb->streamer;
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existing_cb->is_keep_alive = 0;
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if (!existing_streamer->hasConfigurationChanged(channels, lang, interim, sampleRate, region, subscriptionKey)) {
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switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "Reuse active azure connection.\n");
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return SWITCH_STATUS_SUCCESS;
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}
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switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "Azure configuration is changed, destroy old and create new azure connection\n");
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reaper(existing_cb);
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streamer = new GStreamer(sessionId, bugname, channels, lang, interim, sampleRate, region, subscriptionKey, responseHandler);
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if (!existing_cb->vad) streamer->connect();
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existing_cb->streamer = streamer;
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*ppUserData = existing_cb;
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return SWITCH_STATUS_SUCCESS;
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} catch (std::exception& e) {
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switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_ERROR, "%s: Error initializing gstreamer: %s.\n",
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switch_channel_get_name(channel), e.what());
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return SWITCH_STATUS_FALSE;
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}
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switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "Azure configuration is changed, destroy old and create new azure connection\n");
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reaper(existing_cb);
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streamer = new GStreamer(sessionId, bugname, channels, lang, interim, sampleRate, region, subscriptionKey, responseHandler);
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if (!existing_cb->vad) streamer->connect();
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existing_cb->streamer = streamer;
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*ppUserData = existing_cb;
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return SWITCH_STATUS_SUCCESS;
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}
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int err;
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switch_threadattr_t *thd_attr = NULL;
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@@ -1,11 +1,11 @@
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# mod_google_tts
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A Freeswitch module that allows Google Text-to-Speech API to be used as a tts provider.
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A Freeswitch module that allows Eleven Labs' Text-to-Speech API to be used as a tts provider.
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## API
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### Commands
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This freeswitch module does not add any new commands, per se. Rather, it integrates into the Freeswitch TTS interface such that it is invoked when an application uses the mod_dptools `speak` command with a tts engine of `google_tts` and a voice equal to the language code associated to one of the [supported Wavenet voices](https://cloud.google.com/text-to-speech/docs/voices)
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This freeswitch module does not add any new commands, per se. Rather, it integrates into the Freeswitch TTS interface such that it is invoked when an application uses the mod_dptools `speak` command with a tts engine of `elevenlabs` and a voice equal to the language code associated to one of the [supported Eleven Labs voices](https://elevenlabs.io/docs/api-reference/query-library)
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### Events
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None.
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@@ -13,11 +13,19 @@ None.
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## Usage
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When using [drachtio-fsrmf](https://www.npmjs.com/package/drachtio-fsmrf), you can access this functionality via the speak method on the 'endpoint' object.
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```js
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ep.speak({
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ttsEngine: 'google_tts',
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voice: 'en-GB-Wavenet-A',
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text: 'This aggression will not stand'
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});
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var text = "Hello World";
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await endpoint.speak({
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"ttsEngine": 'elevenlabs',
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"voice": "W9OIfHh5DtdYiZUcFiql",
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"text": `{use_speaker_boost=1,optimize_streaming_latency=4,style=0.5,stability=0.5,similarity_boost=0.75,api_key=XXYYZZ,model_id=eleven_turbo_v2}${text}`,
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});
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```
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## Examples
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[google_tts.js](../../examples/google_tts.js)
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## Options
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Documentation on these options can be found in Eleven Labs API docs: [Voice Settings](https://elevenlabs.io/docs/speech-synthesis/voice-settings)
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- use_speaker_boost
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- optimize_streaming_latency
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- style
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- stability
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- similarity_boost
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@@ -233,6 +233,10 @@ static void *SWITCH_THREAD_FUNC grpc_read_thread(switch_thread_t *thread, void *
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auto speech_event_type = response.speech_event_type();
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if (response.has_error()) {
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Status status = response.error();
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//error 11 is handled in finished session, avoid sending jambonz_transcribe::error event for this here.
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if (11 == status.code()) {
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continue;
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}
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switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "grpc_read_thread: error %s (%d)\n", status.message().c_str(), status.code()) ;
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cJSON* json = cJSON_CreateObject();
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cJSON_AddStringToObject(json, "type", "error");
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@@ -14,6 +14,10 @@ static void clearPlayht(playht_t* p, int freeAll) {
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if (p->seed) free(p->seed);
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if (p->temperature) free(p->temperature);
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if (p->voice_engine) free(p->voice_engine);
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if (p->synthesize_url) free(p->synthesize_url);
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if (p->language) free(p->language);
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if (p->top_p) free(p->top_p);
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if (p->repetition_penalty) free(p->repetition_penalty);
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if (p->emotion) free(p->emotion);
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if (p->voice_guidance) free(p->voice_guidance);
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if (p->style_guidance) free(p->style_guidance);
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@@ -36,6 +40,10 @@ static void clearPlayht(playht_t* p, int freeAll) {
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p->seed = NULL;
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p->temperature = NULL;
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p->voice_engine = NULL;
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p->synthesize_url = NULL;
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p->language = NULL;
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p->top_p = NULL;
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p->repetition_penalty = NULL;
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p->emotion = NULL;
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p->voice_guidance = NULL;
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p->style_guidance = NULL;
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@@ -154,6 +162,18 @@ static void p_text_param_tts(switch_speech_handle_t *sh, char *param, const char
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} else if (0 == strcmp(param, "voice_engine")) {
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if (p->voice_engine) free(p->voice_engine);
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p->voice_engine = strdup(val);
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} else if (0 == strcmp(param, "synthesize_url")) {
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if (p->synthesize_url) free(p->synthesize_url);
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p->synthesize_url = strdup(val);
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} else if (0 == strcmp(param, "language")) {
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if (p->language) free(p->language);
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p->language = strdup(val);
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} else if (0 == strcmp(param, "top_p")) {
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if (p->top_p) free(p->top_p);
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p->top_p = strdup(val);
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} else if (0 == strcmp(param, "repetition_penalty")) {
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if (p->repetition_penalty) free(p->repetition_penalty);
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p->repetition_penalty = strdup(val);
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} else if (0 == strcmp(param, "emotion")) {
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if (p->emotion) free(p->emotion);
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p->emotion = strdup(val);
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@@ -11,10 +11,14 @@ typedef struct playht_data {
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char *seed;
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char *temperature;
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char *voice_engine;
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char *synthesize_url;
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char *language;
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char *emotion;
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char *voice_guidance;
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char *style_guidance;
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char *text_guidance;
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char *top_p;
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char *repetition_penalty;
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/* result data */
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long response_code;
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@@ -780,7 +780,7 @@ extern "C" {
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}
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/* format url*/
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std::string url = "https://api.play.ht/api/v2/tts/stream";
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std::string url = p->synthesize_url;
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/* create the JSON body */
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cJSON * jResult = cJSON_CreateObject();
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@@ -800,9 +800,6 @@ extern "C" {
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cJSON_AddNumberToObject(jResult, "speed", val);
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}
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}
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if (p->seed) {
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cJSON_AddNumberToObject(jResult, "seed", atoi(p->seed));
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}
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if (p->temperature) {
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cJSON_AddNumberToObject(jResult, "temperature", std::strtof(p->temperature, nullptr));
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}
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@@ -818,6 +815,15 @@ extern "C" {
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if (p->text_guidance) {
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cJSON_AddNumberToObject(jResult, "text_guidance", atoi(p->text_guidance));
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}
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if (strcmp(p->voice_engine, "Play3.0") == 0 && p->language) {
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cJSON_AddStringToObject(jResult, "language", p->language);
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}
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if (p->repetition_penalty) {
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cJSON_AddNumberToObject(jResult, "repetition_penalty", std::strtof(p->repetition_penalty, nullptr));
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}
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if (p->top_p) {
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cJSON_AddNumberToObject(jResult, "top_p", std::strtof(p->top_p, nullptr));
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}
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char *json = cJSON_PrintUnformatted(jResult);
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cJSON_Delete(jResult);
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@@ -889,9 +895,12 @@ extern "C" {
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/* call this function to close a socket */
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curl_easy_setopt(easy, CURLOPT_CLOSESOCKETFUNCTION, close_socket);
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// Play3.0 voice engine doesn't need authorization
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if (strcmp(p->voice_engine, "Play3.0") != 0) {
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conn->hdr_list = curl_slist_append(conn->hdr_list, api_key_stream.str().c_str());
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conn->hdr_list = curl_slist_append(conn->hdr_list, user_id_stream.str().c_str());
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}
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conn->hdr_list = curl_slist_append(conn->hdr_list, api_key_stream.str().c_str());
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conn->hdr_list = curl_slist_append(conn->hdr_list, user_id_stream.str().c_str());
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conn->hdr_list = curl_slist_append(conn->hdr_list, "Accept: audio/mpeg");
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conn->hdr_list = curl_slist_append(conn->hdr_list, "Content-Type: application/json");
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curl_easy_setopt(easy, CURLOPT_HTTPHEADER, conn->hdr_list);
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@@ -959,8 +968,11 @@ extern "C" {
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switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "playht_speech_flush_tts, download complete? %s\n", download_complete ? "yes" : "no") ;
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ConnInfo_t *conn = (ConnInfo_t *) p->conn;
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CircularBuffer_t *cBuffer = (CircularBuffer_t *) p->circularBuffer;
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// In multi threads, only delete the circular buffer when write and read buffer action finished using it.
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switch_mutex_lock(p->mutex);
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delete cBuffer;
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p->circularBuffer = nullptr ;
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switch_mutex_unlock(p->mutex);
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if (conn) {
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conn->flushed = true;
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