ben.carrasco/freeswitch-modules
1
0
Fork 0
mirror of https://github.com/jambonz/freeswitch-modules.git synced 2025-12-19 08:57:44 +00:00
Code Issues Packages Projects Releases Wiki Activity

Fixes/mod azure tts (#36)

Browse Source 
* fixes for unlocking mutex and minimizing time under lock

* call .get() on future returned from speechSynthesizer->SpeakTextAsync

* mod_azure_tts: various fixes, including dangling session lock and reducing latency

* mod_deepgram_transcribe: add support for transcribing filler words
This commit is contained in:
Dave Horton
2024-04-06 12:26:56 -04:00
committed by GitHub
parent 94361f1d20
commit a96fb2b4b2
3 changed files with 81 additions and 61 deletions
Download Patch File Download Diff File Expand all files Collapse all files

135
mod_azure_tts/azure_glue.cpp
View File

@@ -7,6 +7,7 @@
#include <cstdlib>
#include <string>
#include <chrono>
#include <thread>
#define BUFFER_SIZE 8129
@@ -16,6 +17,30 @@ using namespace Microsoft::CognitiveServices::Speech;
static std::string fullDirPath;
static void start_synthesis(std::shared_ptr<SpeechSynthesizer> speechSynthesizer, const char* text) {
try {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "start_synthesis calling \n");
auto result = std::strncmp(text, "<speak", 6) == 0 ?
speechSynthesizer->SpeakSsmlAsync(text).get() :
speechSynthesizer->SpeakTextAsync(text).get();
if (result->Reason == ResultReason::SynthesizingAudioCompleted) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "start_synthesis completed id %s, audio data - bytes: %ld, milliseconds: %ld milliseconds\n",
result->ResultId.c_str(), result->GetAudioLength(), result->AudioDuration.count());
} else if (result->Reason == ResultReason::Canceled) {
auto cancellation = SpeechSynthesisCancellationDetails::FromResult(result);
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR,
"Error synthesizing text %s: (%d) %s.\n", text, static_cast<int>(cancellation->ErrorCode), cancellation->ErrorDetails.c_str());
} else {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error synthsize text %s (%d).\n", text, static_cast<int>(result->Reason));
}
} catch (const std::exception& e) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "mod_azure_tts: Exception in start_synthesis %s\n", e.what());
}
}
extern "C" {
switch_status_t azure_speech_load() {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "azure_speech_loading..\n");
@@ -109,8 +134,12 @@ extern "C" {
if (a->session_id) {
int err;
switch_codec_implementation_t read_impl;
/* lock and unlock session */
switch_core_session_t *psession = switch_core_session_locate(a->session_id);
switch_core_session_get_read_impl(psession, &read_impl);
switch_core_session_rwunlock(psession);
uint32_t samples_per_second = !strcasecmp(read_impl.iananame, "g722") ? read_impl.actual_samples_per_second : read_impl.samples_per_second;
a->samples_rate = samples_per_second;
if (samples_per_second != 8000 /*Hz*/) {
@@ -154,39 +183,29 @@ extern "C" {
};
speechSynthesizer->Synthesizing += [a](const SpeechSynthesisEventArgs& e) {
if (a->flushed) return;
bool fireEvent = false;
CircularBuffer_t *cBuffer = (CircularBuffer_t *) a->circularBuffer;
std::vector<uint16_t> pcm_data;
//switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Synthesizing: received data\n");
if (a->flushed) {
return;
auto audioData = e.Result->GetAudioData();
if (a->file) {
fwrite(audioData->data(), 1, audioData->size(), a->file);
}
{
switch_mutex_lock(a->mutex);
auto audioData = e.Result->GetAudioData();
for (size_t i = 0; i < audioData->size(); i += sizeof(int16_t)) {
int16_t value = static_cast<int16_t>((*audioData)[i]) | (static_cast<int16_t>((*audioData)[i + 1]) << 8);
pcm_data.push_back(value);
}
/* and write to the file */
size_t bytesResampled = pcm_data.size() * sizeof(uint16_t);
if (a->file) fwrite(pcm_data.data(), sizeof(uint16_t), pcm_data.size(), a->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
*/
const uint16_t* begin = reinterpret_cast<const uint16_t*>(audioData->data());
const uint16_t* end = reinterpret_cast<const uint16_t*>(audioData->data() + audioData->size());
// Resize the buffer if necessary
if (cBuffer->capacity() - cBuffer->size() < (bytesResampled / sizeof(uint16_t))) {
//TODO: if buffer exceeds some max size, return CURL_WRITEFUNC_ERROR to abort the transfer
cBuffer->set_capacity(cBuffer->size() + std::max((bytesResampled / sizeof(uint16_t)), (size_t)BUFFER_SIZE));
}
/* Push the data into the buffer */
cBuffer->insert(cBuffer->end(), pcm_data.data(), pcm_data.data() + pcm_data.size());
switch_mutex_unlock(a->mutex);
/* lock as briefly as possible */
switch_mutex_lock(a->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(a->mutex);
if (0 == a->reads++) {
fireEvent = true;
@@ -200,6 +219,7 @@ extern "C" {
switch_core_session_t* session = switch_core_session_locate(a->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) {
@@ -216,7 +236,6 @@ extern "C" {
}else {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "speechSynthesizer->Synthesizing: channel not found\n");
}
switch_core_session_rwunlock(session);
}
}
};
@@ -231,16 +250,12 @@ extern "C" {
auto cancellation = SpeechSynthesisCancellationDetails::FromResult(e.Result);
a->response_code = static_cast<long int>(cancellation->ErrorCode);
a->err_msg = strdup(cancellation->ErrorDetails.c_str());
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error synthsize tex %d with error string: %s.\n", static_cast<int>(cancellation->ErrorCode), cancellation->ErrorDetails.c_str());
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error synthesizing text %d with error string: %s.\n", static_cast<int>(cancellation->ErrorCode), cancellation->ErrorDetails.c_str());
}
a->draining = 1;
};
// switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "azure_speech_feed_tts before sending synthesize request\n");
if (std::strncmp(text, "<speak", 6) == 0) {
speechSynthesizer->SpeakSsmlAsync(text);
} else {
speechSynthesizer->SpeakTextAsync(text);
}
// switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "azure_speech_feed_tts sent synthesize request\n");
std::thread(start_synthesis, speechSynthesizer, text).detach();
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "azure_speech_feed_tts sent synthesize request\n");
return SWITCH_STATUS_SUCCESS;
}
@@ -248,33 +263,31 @@ extern "C" {
CircularBuffer_t *cBuffer = (CircularBuffer_t *) a->circularBuffer;
std::vector<uint16_t> pcm_data;
{
switch_mutex_lock(a->mutex);
if (a->response_code > 0 && a->response_code != 200) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "azure_speech_read_tts, returning failure\n") ;
return SWITCH_STATUS_FALSE;
}
if (a->flushed) {
if (a->response_code > 0 && a->response_code != 200) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "azure_speech_read_tts, returning failure\n") ;
return SWITCH_STATUS_FALSE;
}
if (a->flushed) {
return SWITCH_STATUS_BREAK;
}
switch_mutex_lock(a->mutex);
size_t bufSize = cBuffer->size();
if (cBuffer->empty()) {
switch_mutex_unlock(a->mutex);
if (a->draining) {
return SWITCH_STATUS_BREAK;
}
if (cBuffer->empty()) {
if (a->draining) {
switch_mutex_unlock(a->mutex);
return SWITCH_STATUS_BREAK;
}
/* no audio available yet so send silence */
memset(data, 255, *datalen);
switch_mutex_unlock(a->mutex);
return SWITCH_STATUS_SUCCESS;
}
// azure returned 8000hz 16 bit data, we have to take enough data based on call sample rate.
size_t size = a->samples_rate ?
std::min((*datalen/(2 * a->samples_rate / 8000)), cBuffer->size()) :
std::min((*datalen/2), cBuffer->size());
pcm_data.insert(pcm_data.end(), cBuffer->begin(), cBuffer->begin() + size);
cBuffer->erase(cBuffer->begin(), cBuffer->begin() + size);
switch_mutex_unlock(a->mutex);
/* no audio available yet so send silence */
memset(data, 255, *datalen);
return SWITCH_STATUS_SUCCESS;
}
// azure returned 8000hz 16 bit data, we have to take enough data based on call sample rate.
size_t size = a->samples_rate ?
std::min((*datalen/(2 * a->samples_rate / 8000)), bufSize) :
std::min((*datalen/2), bufSize);
pcm_data.insert(pcm_data.end(), cBuffer->begin(), cBuffer->begin() + size);
cBuffer->erase(cBuffer->begin(), cBuffer->begin() + size);
switch_mutex_unlock(a->mutex);
size_t data_size = pcm_data.size();
@@ -336,6 +349,9 @@ extern "C" {
switch_core_session_t* session = switch_core_session_locate(a->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) {
@@ -357,7 +373,6 @@ extern "C" {
else {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "write_cb: channel not found\n");
}
switch_core_session_rwunlock(session);
}
}

4
mod_deepgram_transcribe/README.md
View File

@@ -25,10 +25,11 @@ Stop transcription on the channel.
| variable | Description |
| --- | ----------- |
| DEEPGRAM_API_KEY | Deepgram API key used to authenticate |
| DEEPGRAM_SPEECH_TIER | https://developers.deepgram.com/documentation/features/tier/ |
| DEEPGRAM_SPEECH_CUSTOM_MODEL | custom model id |
| DEEPGRAM_SPEECH_MODEL | https://developers.deepgram.com/documentation/features/model/ |
| DEEPGRAM_SPEECH_MODEL_VERSION | https://developers.deepgram.com/documentation/features/version/ |
| DEEPGRAM_SPEECH_ENABLE_SMART_FORMAT | https://developers.deepgram.com/docs/smart-format |
| DEEPGRAM_SPEECH_ENABLE_FILLER_WORDS | https://developers.deepgram.com/docs/filler-words |
| DEEPGRAM_SPEECH_ENABLE_AUTOMATIC_PUNCTUATION | https://developers.deepgram.com/documentation/features/punctuate/ |
| DEEPGRAM_SPEECH_PROFANITY_FILTER | https://developers.deepgram.com/documentation/features/profanity-filter/ |
| DEEPGRAM_SPEECH_REDACT | https://developers.deepgram.com/documentation/features/redact/ |
@@ -42,6 +43,7 @@ Stop transcription on the channel.
| DEEPGRAM_SPEECH_REPLACE | https://developers.deepgram.com/documentation/features/replace/ |
| DEEPGRAM_SPEECH_TAG | https://developers.deepgram.com/documentation/features/tag/ |
| DEEPGRAM_SPEECH_ENDPOINTING | https://developers.deepgram.com/documentation/features/endpointing/ |
| DEEPGRAM_SPEECH_UTTERANCE_END_MS | https://developers.deepgram.com/docs/utterance-end |
| DEEPGRAM_SPEECH_VAD_TURNOFF | https://developers.deepgram.com/documentation/features/voice-activity-detection/ |

3
mod_deepgram_transcribe/dg_transcribe_glue.cpp
View File

@@ -184,6 +184,9 @@ namespace {
*
*/
}
if (var = switch_channel_get_variable(channel, "DEEPGRAM_SPEECH_ENABLE_FILLER_WORDS")) {
oss << "&filler_words=true";
}
if (var = switch_channel_get_variable(channel, "DEEPGRAM_SPEECH_ENABLE_AUTOMATIC_PUNCTUATION")) {
oss << "&punctuate=true";
}