ben.carrasco/freeswitch-modules
1
0
Fork 0
mirror of https://github.com/jambonz/freeswitch-modules.git synced 2025-12-19 08:27:44 +00:00
Code Issues Packages Projects Releases Wiki Activity
Files
8bd20703b8d9a32bbbd93068f2c17bfafc11cbfd
freeswitch-modules/mod_audio_fork/lws_glue.cpp
Dave Horton 8bd20703b8 Fix/azure tts no device output (#79) 
* enable audio logging if env AZURE_AUDIO_LOGGING is set

* wip

* per discussion with microsoft, add nullptr to creation of speechSynthesizer to ensure it knows we do not want it to play to device

Signed-off-by: Dave Horton <daveh@beachdognet.com>

* logging

* fix bug in creation of config string

* fix ticket 230 - Microsoft TTS having configuration data as part of audio generation

* azure transcribe, resuse existing cap_cb if azure configuration is changed

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

* clean up azure code for how to re-create gsstream when configuration is changed

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

* fix review comments

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

* fix review comment

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

* fix review comment

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

* wrap function in try catch

---------

Signed-off-by: Dave Horton <daveh@beachdognet.com>
Signed-off-by: Hoan HL <quan.luuhoang8@gmail.com>
Co-authored-by: Hoan HL <quan.luuhoang8@gmail.com>
2024-06-23 14:54:36 -04:00

777 lines
33 KiB
C++
Raw Blame History

#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 "base64.hpp"
#include "parser.hpp"
#include "mod_audio_fork.h"
#include "audio_pipe.hpp"
#include "vector_math.h"
#include <boost/circular_buffer.hpp>
typedef boost::circular_buffer<uint16_t> CircularBuffer_t;
#define RTP_PACKETIZATION_PERIOD 20
#define FRAME_SIZE_8000 320 /*which means each 20ms frame as 320 bytes at 8 khz (1 channel only)*/
#define BUFFER_GROW_SIZE (8192)
namespace {
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 const char* mySubProtocolName = std::getenv("MOD_AUDIO_FORK_SUBPROTOCOL_NAME") ?
std::getenv("MOD_AUDIO_FORK_SUBPROTOCOL_NAME") : "audio.drachtio.org";
static unsigned int nServiceThreads = std::max(1, std::min(requestedNumServiceThreads ? ::atoi(requestedNumServiceThreads) : 1, 5));
static unsigned int idxCallCount = 0;
static uint32_t playCount = 0;
switch_status_t processIncomingBinary(private_t* tech_pvt, switch_core_session_t* session, const char* message, size_t dataLength) {
CircularBuffer_t *cBuffer = (CircularBuffer_t *) tech_pvt->circularBuffer;
uint8_t* data = reinterpret_cast<uint8_t*>(const_cast<char*>(message));
uint16_t* data_uint16 = reinterpret_cast<uint16_t*>(data);
std::vector<uint16_t> pcm_data(data_uint16, data_uint16 + dataLength / sizeof(uint16_t));
// resample if necessary
try {
if (tech_pvt->bidirectional_audio_resampler) {
std::vector<int16_t> in(pcm_data.begin(), pcm_data.end());
std::vector<int16_t> out(dataLength);
spx_uint32_t in_len = pcm_data.size();
spx_uint32_t out_len = out.size();
speex_resampler_process_interleaved_int(tech_pvt->bidirectional_audio_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;
}
// Resize the pcm_data to match the output length from resampler, and then copy the resampled data into it.
pcm_data.resize(out_len);
memcpy(pcm_data.data(), out.data(), out_len * sizeof(int16_t));
}
} catch (const std::exception& e) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error resampling incoming binary message: %s\n", e.what());
return SWITCH_STATUS_FALSE;
} catch (...) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error resampling incoming binary message\n");
return SWITCH_STATUS_FALSE;
}
switch_mutex_lock(tech_pvt->mutex);
try {
// Resize the buffer if necessary
size_t bytesResampled = pcm_data.size() * sizeof(uint16_t);
if (cBuffer->capacity() - cBuffer->size() < bytesResampled / sizeof(uint16_t)) {
// If buffer exceeds some max size, you could return SWITCH_STATUS_FALSE to abort the transfer
// if (cBuffer->size() + std::max(bytesResampled / sizeof(uint16_t), (size_t)BUFFER_GROW_SIZE) > MAX_BUFFER_SIZE) return SWITCH_STATUS_FALSE;
cBuffer->set_capacity(cBuffer->size() + std::max(bytesResampled / sizeof(uint16_t), (size_t)BUFFER_GROW_SIZE));
}
// Push the data into the buffer.
cBuffer->insert(cBuffer->end(), pcm_data.begin(), pcm_data.end());
} catch (const std::exception& e) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error processing incoming binary message: %s\n", e.what());
switch_mutex_unlock(tech_pvt->mutex);
return SWITCH_STATUS_FALSE;
} catch (...) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error processing incoming binary message\n");
switch_mutex_unlock(tech_pvt->mutex);
return SWITCH_STATUS_FALSE;
}
switch_mutex_unlock(tech_pvt->mutex);
return SWITCH_STATUS_SUCCESS;
}
void processIncomingMessage(private_t* tech_pvt, switch_core_session_t* session, const char* message) {
std::string msg = message;
std::string type;
cJSON* json = parse_json(session, msg, type) ;
if (json) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "(%u) processIncomingMessage - received %s message\n", tech_pvt->id, type.c_str());
cJSON* jsonData = cJSON_GetObjectItem(json, "data");
if (0 == type.compare("playAudio") &&
// playAudio is enabled and there is no bidirectional audio from stream is enabled.
tech_pvt->bidirectional_audio_enable &&
!tech_pvt->bidirectional_audio_stream) {
if (jsonData) {
// dont send actual audio bytes in event message
cJSON* jsonFile = NULL;
cJSON* jsonAudio = cJSON_DetachItemFromObject(jsonData, "audioContent");
int validAudio = (jsonAudio && NULL != jsonAudio->valuestring);
const char* szAudioContentType = cJSON_GetObjectCstr(jsonData, "audioContentType");
char fileType[6];
int sampleRate = 16000;
if (0 == strcmp(szAudioContentType, "raw")) {
cJSON* jsonSR = cJSON_GetObjectItem(jsonData, "sampleRate");
sampleRate = jsonSR && jsonSR->valueint ? jsonSR->valueint : 0;
switch(sampleRate) {
case 8000:
strcpy(fileType, ".r8");
break;
case 16000:
strcpy(fileType, ".r16");
break;
case 24000:
strcpy(fileType, ".r24");
break;
case 32000:
strcpy(fileType, ".r32");
break;
case 48000:
strcpy(fileType, ".r48");
break;
case 64000:
strcpy(fileType, ".r64");
break;
default:
strcpy(fileType, ".r16");
break;
}
}
else if (0 == strcmp(szAudioContentType, "wave") || 0 == strcmp(szAudioContentType, "wav")) {
strcpy(fileType, ".wav");
}
else {
validAudio = 0;
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "(%u) processIncomingMessage - unsupported audioContentType: %s\n", tech_pvt->id, szAudioContentType);
}
if (validAudio) {
char szFilePath[256];
std::string rawAudio = drachtio::base64_decode(jsonAudio->valuestring);
switch_snprintf(szFilePath, 256, "%s%s%s_%d.tmp%s", SWITCH_GLOBAL_dirs.temp_dir,
SWITCH_PATH_SEPARATOR, tech_pvt->sessionId, playCount++, fileType);
std::ofstream f(szFilePath, std::ofstream::binary);
f << rawAudio;
f.close();
// add the file to the list of files played for this session, we'll delete when session closes
struct playout* playout = (struct playout *) malloc(sizeof(struct playout));
playout->file = (char *) malloc(strlen(szFilePath) + 1);
strcpy(playout->file, szFilePath);
playout->next = tech_pvt->playout;
tech_pvt->playout = playout;
jsonFile = cJSON_CreateString(szFilePath);
cJSON_AddItemToObject(jsonData, "file", jsonFile);
}
char* jsonString = cJSON_PrintUnformatted(jsonData);
tech_pvt->responseHandler(session, EVENT_PLAY_AUDIO, jsonString);
free(jsonString);
if (jsonAudio) cJSON_Delete(jsonAudio);
}
else {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "(%u) processIncomingMessage - missing data payload in playAudio request\n", tech_pvt->id);
}
}
else if (0 == type.compare("killAudio")) {
tech_pvt->responseHandler(session, EVENT_KILL_AUDIO, NULL);
// kill any current playback on the channel
switch_channel_t *channel = switch_core_session_get_channel(session);
switch_channel_set_flag_value(channel, CF_BREAK, 2);
}
else if (0 == type.compare("transcription")) {
char* jsonString = cJSON_PrintUnformatted(jsonData);
tech_pvt->responseHandler(session, EVENT_TRANSCRIPTION, jsonString);
free(jsonString);
}
else if (0 == type.compare("transfer")) {
char* jsonString = cJSON_PrintUnformatted(jsonData);
tech_pvt->responseHandler(session, EVENT_TRANSFER, jsonString);
free(jsonString);
}
else if (0 == type.compare("disconnect")) {
char* jsonString = cJSON_PrintUnformatted(jsonData);
tech_pvt->responseHandler(session, EVENT_DISCONNECT, jsonString);
free(jsonString);
}
else if (0 == type.compare("error")) {
char* jsonString = cJSON_PrintUnformatted(jsonData);
tech_pvt->responseHandler(session, EVENT_ERROR, jsonString);
free(jsonString);
}
else if (0 == type.compare("json")) {
char* jsonString = cJSON_PrintUnformatted(json);
tech_pvt->responseHandler(session, EVENT_JSON, jsonString);
free(jsonString);
}
else {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "(%u) processIncomingMessage - unsupported msg type %s\n", tech_pvt->id, type.c_str());
}
cJSON_Delete(json);
}
else {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "(%u) processIncomingMessage - could not parse message: %s\n", tech_pvt->id, message);
}
}
static void eventCallback(const char* sessionId, const char* bugname, drachtio::AudioPipe::NotifyEvent_t event, const char* message, const char* binary, size_t len) {
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 drachtio::AudioPipe::CONNECT_SUCCESS:
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_INFO, "connection successful\n");
tech_pvt->responseHandler(session, EVENT_CONNECT_SUCCESS, NULL);
if (strlen(tech_pvt->initialMetadata) > 0) {
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "sending initial metadata %s\n", tech_pvt->initialMetadata);
drachtio::AudioPipe *pAudioPipe = static_cast<drachtio::AudioPipe *>(tech_pvt->pAudioPipe);
pAudioPipe->bufferForSending(tech_pvt->initialMetadata);
}
break;
case drachtio::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, EVENT_CONNECT_FAIL, (char *) json.str().c_str());
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_NOTICE, "connection failed: %s\n", message);
}
break;
case drachtio::AudioPipe::CONNECTION_DROPPED:
// first thing: we can no longer access the AudioPipe
tech_pvt->pAudioPipe = nullptr;
tech_pvt->responseHandler(session, EVENT_DISCONNECT, NULL);
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_NOTICE, "connection dropped from far end\n");
break;
case drachtio::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 drachtio::AudioPipe::MESSAGE:
processIncomingMessage(tech_pvt, session, message);
break;
case drachtio::AudioPipe::BINARY:
processIncomingBinary(tech_pvt, session, binary, len);
break;
}
}
}
switch_core_session_rwunlock(session);
}
}
switch_status_t fork_data_init(private_t *tech_pvt, switch_core_session_t *session, char * host,
unsigned int port, char* path, int sslFlags, int sampling, int desiredSampling, int channels,
char *bugname, char* metadata, int bidirectional_audio_enable,
int bidirectional_audio_stream, int bidirectional_audio_sample_rate, responseHandler_t responseHandler) {
const char* username = nullptr;
const char* password = nullptr;
int err;
int bidirectional_audio_stream_enable = bidirectional_audio_enable + bidirectional_audio_stream;
switch_codec_implementation_t read_impl;
switch_channel_t *channel = switch_core_session_get_channel(session);
switch_core_session_get_read_impl(session, &read_impl);
if (username = switch_channel_get_variable(channel, "MOD_AUDIO_BASIC_AUTH_USERNAME")) {
password = switch_channel_get_variable(channel, "MOD_AUDIO_BASIC_AUTH_PASSWORD");
}
memset(tech_pvt, 0, sizeof(private_t));
strncpy(tech_pvt->sessionId, switch_core_session_get_uuid(session), MAX_SESSION_ID);
strncpy(tech_pvt->host, host, MAX_WS_URL_LEN);
tech_pvt->port = port;
strncpy(tech_pvt->path, path, MAX_PATH_LEN);
tech_pvt->sampling = desiredSampling;
tech_pvt->responseHandler = responseHandler;
tech_pvt->playout = NULL;
tech_pvt->channels = channels;
tech_pvt->id = ++idxCallCount;
tech_pvt->buffer_overrun_notified = 0;
tech_pvt->audio_paused = 0;
tech_pvt->graceful_shutdown = 0;
tech_pvt->circularBuffer = (void *) new CircularBuffer_t(8192);
tech_pvt->bidirectional_audio_enable = bidirectional_audio_enable;
tech_pvt->bidirectional_audio_stream = bidirectional_audio_stream;
tech_pvt->bidirectional_audio_sample_rate = bidirectional_audio_sample_rate;
strncpy(tech_pvt->bugname, bugname, MAX_BUG_LEN);
if (metadata) strncpy(tech_pvt->initialMetadata, metadata, MAX_METADATA_LEN);
size_t buflen = LWS_PRE + (FRAME_SIZE_8000 * desiredSampling / 8000 * channels * 1000 / RTP_PACKETIZATION_PERIOD * nAudioBufferSecs);
drachtio::AudioPipe* ap = new drachtio::AudioPipe(tech_pvt->sessionId, host, port, path, sslFlags,
buflen, read_impl.decoded_bytes_per_packet, username, password, bugname, bidirectional_audio_stream_enable, 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);
}
if (bidirectional_audio_sample_rate && sampling != bidirectional_audio_sample_rate) {
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "(%u) bidirectional audio resampling from %u to %u\n", tech_pvt->id, bidirectional_audio_sample_rate, sampling);
tech_pvt->bidirectional_audio_resampler = speex_resampler_init(channels, bidirectional_audio_sample_rate, sampling, SWITCH_RESAMPLE_QUALITY, &err);
if (0 != err) {
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_ERROR, "Error initializing bidirectional audio resampler: %s.\n", speex_resampler_strerror(err));
return SWITCH_STATUS_FALSE;
}
}
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "(%u) fork_data_init\n", tech_pvt->id);
return SWITCH_STATUS_SUCCESS;
}
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->resampler) {
speex_resampler_destroy(tech_pvt->resampler);
tech_pvt->resampler = nullptr;
}
if (tech_pvt->bidirectional_audio_resampler) {
speex_resampler_destroy(tech_pvt->bidirectional_audio_resampler);
tech_pvt->bidirectional_audio_resampler = nullptr;
}
if (tech_pvt->mutex) {
switch_mutex_destroy(tech_pvt->mutex);
tech_pvt->mutex = nullptr;
}
if (tech_pvt->circularBuffer) {
CircularBuffer_t *cBuffer = (CircularBuffer_t *) tech_pvt->circularBuffer;
delete cBuffer;
tech_pvt->circularBuffer = nullptr;
}
}
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, llevel, "%s\n", line);
}
}
extern "C" {
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);
//- `([^/:]+)` captures the hostname/IP address, match any character except in the set
//- `:?([0-9]*)?` optionally captures a colon and the port number, if it's present.
//- `(/.*)` captures everything else (the path).
std::regex re("([^/:]+):?([0-9]*)?(/.*)?$");
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;
}
switch_status_t fork_init() {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "mod_audio_fork: audio buffer (in secs): %d secs\n", nAudioBufferSecs);
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "mod_audio_fork: sub-protocol: %s\n", mySubProtocolName);
int logs = LLL_ERR | LLL_WARN | LLL_NOTICE ;
//LLL_INFO | LLL_PARSER | LLL_HEADER | LLL_EXT | LLL_CLIENT | LLL_LATENCY | LLL_DEBUG ;
drachtio::AudioPipe::initialize(mySubProtocolName, logs, lws_logger);
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "mod_audio_fork successfully initialized\n");
return SWITCH_STATUS_SUCCESS;
}
switch_status_t fork_cleanup() {
bool cleanup = false;
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "mod_audio_fork unloading..\n");
cleanup = drachtio::AudioPipe::deinitialize();
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "mod_audio_fork unloaded status %d\n", cleanup);
if (cleanup == true) {
return SWITCH_STATUS_SUCCESS;
}
return SWITCH_STATUS_FALSE;
}
switch_status_t fork_session_init(switch_core_session_t *session,
responseHandler_t responseHandler,
uint32_t samples_per_second,
char *host,
unsigned int port,
char *path,
int sampling,
int sslFlags,
int channels,
char *bugname,
char* metadata,
int bidirectional_audio_enable,
int bidirectional_audio_stream,
int bidirectional_audio_sample_rate,
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, host, port, path, sslFlags, samples_per_second, sampling, channels,
bugname, metadata, bidirectional_audio_enable, bidirectional_audio_stream, bidirectional_audio_sample_rate, responseHandler)) {
destroy_tech_pvt(tech_pvt);
return SWITCH_STATUS_FALSE;
}
*ppUserData = tech_pvt;
return SWITCH_STATUS_SUCCESS;
}
switch_status_t fork_session_connect(void **ppUserData) {
private_t *tech_pvt = static_cast<private_t *>(*ppUserData);
drachtio::AudioPipe *pAudioPipe = static_cast<drachtio::AudioPipe*>(tech_pvt->pAudioPipe);
pAudioPipe->connect();
return SWITCH_STATUS_SUCCESS;
}
switch_status_t fork_session_cleanup(switch_core_session_t *session, char *bugname, char* text, int channelIsClosing) {
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, "fork_session_cleanup: 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);
uint32_t id = tech_pvt->id;
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "(%u) fork_session_cleanup\n", id);
if (!tech_pvt) return SWITCH_STATUS_FALSE;
drachtio::AudioPipe *pAudioPipe = static_cast<drachtio::AudioPipe *>(tech_pvt->pAudioPipe);
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);
if (!channelIsClosing) {
switch_core_media_bug_remove(session, &bug);
}
}
}
// delete any temp files
struct playout* playout = tech_pvt->playout;
while (playout) {
std::remove(playout->file);
free(playout->file);
struct playout *tmp = playout;
playout = playout->next;
free(tmp);
}
if (pAudioPipe && text) pAudioPipe->bufferForSending(text);
if (pAudioPipe) pAudioPipe->close();
destroy_tech_pvt(tech_pvt);
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_INFO, "(%u) fork_session_cleanup: connection closed\n", id);
return SWITCH_STATUS_SUCCESS;
}
switch_status_t fork_session_send_text(switch_core_session_t *session, char *bugname, char* text) {
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_ERROR, "fork_session_send_text failed because no bug\n");
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;
drachtio::AudioPipe *pAudioPipe = static_cast<drachtio::AudioPipe *>(tech_pvt->pAudioPipe);
if (pAudioPipe && text) pAudioPipe->bufferForSending(text);
return SWITCH_STATUS_SUCCESS;
}
switch_status_t fork_session_pauseresume(switch_core_session_t *session, char *bugname, int pause) {
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_ERROR, "fork_session_pauseresume failed because no bug\n");
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;
switch_core_media_bug_flush(bug);
tech_pvt->audio_paused = pause;
return SWITCH_STATUS_SUCCESS;
}
switch_status_t fork_session_graceful_shutdown(switch_core_session_t *session, 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_ERROR, "fork_session_graceful_shutdown failed because no bug\n");
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;
tech_pvt->graceful_shutdown = 1;
drachtio::AudioPipe *pAudioPipe = static_cast<drachtio::AudioPipe *>(tech_pvt->pAudioPipe);
if (pAudioPipe) pAudioPipe->do_graceful_shutdown();
return SWITCH_STATUS_SUCCESS;
}
switch_bool_t fork_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 || tech_pvt->audio_paused || tech_pvt->graceful_shutdown) 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;
}
drachtio::AudioPipe *pAudioPipe = static_cast<drachtio::AudioPipe *>(tech_pvt->pAudioPipe);
if (pAudioPipe->getLwsState() != drachtio::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, EVENT_BUFFER_OVERRUN, NULL);
}
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, EVENT_BUFFER_OVERRUN, NULL);
}
break;
}
}
}
}
pAudioPipe->unlockAudioBuffer();
switch_mutex_unlock(tech_pvt->mutex);
}
return SWITCH_TRUE;
}
switch_bool_t dub_speech_frame(switch_media_bug_t *bug, private_t* tech_pvt) {
CircularBuffer_t *cBuffer = (CircularBuffer_t *) tech_pvt->circularBuffer;
if (switch_mutex_trylock(tech_pvt->mutex) == SWITCH_STATUS_SUCCESS) {
switch_frame_t* rframe = switch_core_media_bug_get_write_replace_frame(bug);
int16_t *fp = reinterpret_cast<int16_t*>(rframe->data);
rframe->channels = 1;
rframe->datalen = rframe->samples * sizeof(int16_t);
int16_t data[SWITCH_RECOMMENDED_BUFFER_SIZE];
memset(data, 0, sizeof(data));
int samplesToCopy = std::min(static_cast<int>(cBuffer->size()), static_cast<int>(rframe->samples));
std::copy_n(cBuffer->begin(), samplesToCopy, data);
cBuffer->erase(cBuffer->begin(), cBuffer->begin() + samplesToCopy);
if (samplesToCopy > 0) {
vector_add(fp, data, rframe->samples);
}
vector_normalize(fp, rframe->samples);
switch_core_media_bug_set_write_replace_frame(bug, rframe);
switch_mutex_unlock(tech_pvt->mutex);
}
return SWITCH_TRUE;
}
switch_status_t fork_session_stop_play(switch_core_session_t *session, 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_ERROR, "fork_session_stop_play failed because no bug\n");
return SWITCH_STATUS_FALSE;
}
private_t* tech_pvt = (private_t*) switch_core_media_bug_get_user_data(bug);
CircularBuffer_t *cBuffer = (CircularBuffer_t *) tech_pvt->circularBuffer;
if (switch_mutex_trylock(tech_pvt->mutex) == SWITCH_STATUS_SUCCESS) {
if (cBuffer != nullptr) {
cBuffer->clear();
}
switch_mutex_unlock(tech_pvt->mutex);
}
return SWITCH_STATUS_SUCCESS;
}
}
Reference in New Issue View Git Blame Copy Permalink