mirror of
https://github.com/jambonz/freeswitch-modules.git
synced 2025-12-19 06:47:43 +00:00
f828171b3b
* support jambonz transcribe with multiple sampling rate * wip Signed-off-by: Hoan HL <quan.luuhoang8@gmail.com> --------- Signed-off-by: Hoan HL <quan.luuhoang8@gmail.com>
512 lines
21 KiB
C++
512 lines
21 KiB
C++
#include <switch.h>
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#include <switch_json.h>
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#include <string.h>
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#include <string>
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#include <mutex>
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#include <thread>
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#include <list>
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#include <algorithm>
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#include <functional>
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#include <cassert>
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#include <cstdlib>
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#include <fstream>
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#include <sstream>
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#include <regex>
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#include "mod_jambonz_transcribe.h"
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#include "simple_buffer.h"
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#include "parser.hpp"
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#include "audio_pipe.hpp"
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#define RTP_PACKETIZATION_PERIOD 20
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#define FRAME_SIZE_8000 320 /*which means each 20ms frame as 320 bytes at 8 khz (1 channel only)*/
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namespace {
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static bool hasDefaultCredentials = false;
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static const char* defaultApiKey = nullptr;
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static const char *requestedBufferSecs = std::getenv("MOD_AUDIO_FORK_BUFFER_SECS");
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static int nAudioBufferSecs = std::max(1, std::min(requestedBufferSecs ? ::atoi(requestedBufferSecs) : 2, 5));
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static const char *requestedNumServiceThreads = std::getenv("MOD_AUDIO_FORK_SERVICE_THREADS");
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static unsigned int nServiceThreads = std::max(1, std::min(requestedNumServiceThreads ? ::atoi(requestedNumServiceThreads) : 1, 5));
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static unsigned int idxCallCount = 0;
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static uint32_t playCount = 0;
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static int parse_ws_uri(switch_channel_t *channel, const char* szServerUri, char* host, char *path, unsigned int* pPort, int* pSslFlags) {
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int i = 0, offset;
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char server[MAX_WS_URL_LEN + MAX_PATH_LEN];
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char *saveptr;
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int flags = LCCSCF_USE_SSL;
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if (switch_true(switch_channel_get_variable(channel, "MOD_AUDIO_FORK_ALLOW_SELFSIGNED"))) {
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switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "parse_ws_uri - allowing self-signed certs\n");
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flags |= LCCSCF_ALLOW_SELFSIGNED;
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}
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if (switch_true(switch_channel_get_variable(channel, "MOD_AUDIO_FORK_SKIP_SERVER_CERT_HOSTNAME_CHECK"))) {
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switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "parse_ws_uri - skipping hostname check\n");
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flags |= LCCSCF_SKIP_SERVER_CERT_HOSTNAME_CHECK;
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}
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if (switch_true(switch_channel_get_variable(channel, "MOD_AUDIO_FORK_ALLOW_EXPIRED"))) {
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switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "parse_ws_uri - allowing expired certs\n");
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flags |= LCCSCF_ALLOW_EXPIRED;
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}
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// get the scheme
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strncpy(server, szServerUri, MAX_WS_URL_LEN + MAX_PATH_LEN);
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if (0 == strncmp(server, "https://", 8) || 0 == strncmp(server, "HTTPS://", 8)) {
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*pSslFlags = flags;
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offset = 8;
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*pPort = 443;
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}
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else if (0 == strncmp(server, "wss://", 6) || 0 == strncmp(server, "WSS://", 6)) {
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*pSslFlags = flags;
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offset = 6;
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*pPort = 443;
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}
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else if (0 == strncmp(server, "http://", 7) || 0 == strncmp(server, "HTTP://", 7)) {
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offset = 7;
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*pSslFlags = 0;
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*pPort = 80;
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}
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else if (0 == strncmp(server, "ws://", 5) || 0 == strncmp(server, "WS://", 5)) {
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offset = 5;
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*pSslFlags = 0;
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*pPort = 80;
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}
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else {
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switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "parse_ws_uri - error parsing uri %s: invalid scheme\n", szServerUri);;
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return 0;
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}
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std::string strHost(server + offset);
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std::regex re("^(.+?):?(\\d+)?(/.*)?$");
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std::smatch matches;
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if(std::regex_search(strHost, matches, re)) {
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/*
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for (int i = 0; i < matches.length(); i++) {
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switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "parse_ws_uri - %d: %s\n", i, matches[i].str().c_str());
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}
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*/
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strncpy(host, matches[1].str().c_str(), MAX_WS_URL_LEN);
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if (matches[2].str().length() > 0) {
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*pPort = atoi(matches[2].str().c_str());
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}
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if (matches[3].str().length() > 0) {
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strncpy(path, matches[3].str().c_str(), MAX_PATH_LEN);
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}
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else {
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strcpy(path, "/");
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}
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} else {
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switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "parse_ws_uri - invalid format %s\n", strHost.c_str());
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return 0;
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}
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switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "parse_ws_uri - host %s, path %s\n", host, path);
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return 1;
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}
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static void reaper(private_t *tech_pvt) {
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std::shared_ptr<jambonz::AudioPipe> pAp;
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pAp.reset((jambonz::AudioPipe *)tech_pvt->pAudioPipe);
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tech_pvt->pAudioPipe = nullptr;
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std::thread t([pAp, tech_pvt]{
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pAp->finish();
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pAp->waitForClose();
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switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "%s (%u) got remote close\n", tech_pvt->sessionId, tech_pvt->id);
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});
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t.detach();
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}
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static void destroy_tech_pvt(private_t *tech_pvt) {
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switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "%s (%u) destroy_tech_pvt\n", tech_pvt->sessionId, tech_pvt->id);
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if (tech_pvt) {
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if (tech_pvt->mutex) {
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switch_mutex_destroy(tech_pvt->mutex);
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tech_pvt->mutex = nullptr;
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}
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if (tech_pvt->resampler) {
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speex_resampler_destroy(tech_pvt->resampler);
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tech_pvt->resampler = NULL;
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}
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}
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}
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static void sendStartMessage(switch_channel_t *channel, private_t* tech_pvt) {
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auto *pAudioPipe = static_cast<jambonz::AudioPipe*>(tech_pvt->pAudioPipe);
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const char* var;
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bool hasOptions = false;
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if (nullptr == pAudioPipe) return;
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cJSON* json = cJSON_CreateObject();
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cJSON_AddStringToObject(json, "type", "start");
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cJSON_AddStringToObject(json, "language", tech_pvt->language);
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cJSON_AddStringToObject(json, "format", "raw");
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cJSON_AddStringToObject(json, "encoding", "LINEAR16");
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cJSON_AddBoolToObject(json, "interimResults", tech_pvt->interim);
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cJSON_AddNumberToObject(json, "sampleRateHz", tech_pvt->sampling);
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if (var = switch_channel_get_variable(channel, "JAMBONZ_STT_OPTIONS")) {
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cJSON* jOptions = cJSON_Parse(var);
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if (jOptions) {
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hasOptions = true;
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cJSON_AddItemToObject(json, "options", jOptions);
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}
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}
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if (!hasOptions) {
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cJSON* jOptions = cJSON_CreateObject();
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cJSON_AddItemToObject(json, "options", jOptions);
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}
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char* jsonString = cJSON_PrintUnformatted(json);
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/* send */
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pAudioPipe->bufferForSending(jsonString);
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/* cleanup */
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free(jsonString);
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cJSON_Delete(json);
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}
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static void eventCallback(const char* sessionId, const char* bugname, jambonz::AudioPipe::NotifyEvent_t event, const char* message, bool finished) {
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switch_core_session_t* session = switch_core_session_locate(sessionId);
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if (session) {
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switch_channel_t *channel = switch_core_session_get_channel(session);
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switch_media_bug_t *bug = (switch_media_bug_t*) switch_channel_get_private(channel, bugname);
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if (bug) {
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private_t* tech_pvt = (private_t*) switch_core_media_bug_get_user_data(bug);
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if (tech_pvt) {
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switch (event) {
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case jambonz::AudioPipe::CONNECT_SUCCESS:
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switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_INFO, "connection successful\n");
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tech_pvt->responseHandler(session, TRANSCRIBE_EVENT_CONNECT_SUCCESS, NULL, tech_pvt->bugname, finished);
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sendStartMessage(channel, tech_pvt);
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break;
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case jambonz::AudioPipe::CONNECT_FAIL:
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{
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// first thing: we can no longer access the AudioPipe
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std::stringstream json;
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json << "{\"reason\":\"" << message << "\"}";
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tech_pvt->pAudioPipe = nullptr;
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tech_pvt->responseHandler(session, TRANSCRIBE_EVENT_CONNECT_FAIL, (char *) json.str().c_str(), tech_pvt->bugname, finished);
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switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_NOTICE, "connection failed: %s\n", message);
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}
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break;
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case jambonz::AudioPipe::CONNECTION_DROPPED:
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// first thing: we can no longer access the AudioPipe
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tech_pvt->pAudioPipe = nullptr;
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tech_pvt->responseHandler(session, TRANSCRIBE_EVENT_DISCONNECT, NULL, tech_pvt->bugname, finished);
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switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "connection dropped from far end\n");
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break;
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case jambonz::AudioPipe::CONNECTION_CLOSED_GRACEFULLY:
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// first thing: we can no longer access the AudioPipe
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tech_pvt->pAudioPipe = nullptr;
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switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "connection closed gracefully\n");
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break;
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case jambonz::AudioPipe::MESSAGE:
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{
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cJSON* jMessage = cJSON_Parse(message);
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if (!jMessage) {
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switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_ERROR, "discarding incoming message (not json): %s\n", message);
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break;
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}
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const char* type = cJSON_GetStringValue(cJSON_GetObjectItem(jMessage, "type"));
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if (type && 0 == strcmp(type, "error")) {
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tech_pvt->responseHandler(session, TRANSCRIBE_EVENT_ERROR, message, tech_pvt->bugname, finished);
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}
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else if (type && 0 == strcmp(type, "transcription")) {
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tech_pvt->responseHandler(session, TRANSCRIBE_EVENT_RESULTS, message, tech_pvt->bugname, finished);
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}
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else {
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switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_ERROR, "discarding incoming message: %s\n", message);
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}
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cJSON_Delete(jMessage);
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}
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break;
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default:
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switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_NOTICE, "got unexpected msg from jambonz %d:%s\n", event, message);
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break;
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}
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}
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}
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switch_core_session_rwunlock(session);
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}
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}
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switch_status_t fork_data_init(private_t *tech_pvt, switch_core_session_t *session,
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int sampling, int desiredSampling, int channels, char *lang, int interim,
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char* bugname, responseHandler_t responseHandler) {
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int err;
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switch_codec_implementation_t read_impl;
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switch_channel_t *channel = switch_core_session_get_channel(session);
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char host[MAX_WS_URL_LEN], path[MAX_PATH_LEN];
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unsigned int port;
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int sslFlags;
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const char* url = switch_channel_get_variable(channel, "JAMBONZ_STT_URL");
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if (!parse_ws_uri(channel, url, &host[0], &path[0], &port, &sslFlags)) {
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switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_ERROR, "invalid websocket uri: %s\n", url);
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return SWITCH_STATUS_FALSE;
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}
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switch_core_session_get_read_impl(session, &read_impl);
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memset(tech_pvt, 0, sizeof(private_t));
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switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "host: %s, port: %d, path: %s\n", host, port, path);
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strncpy(tech_pvt->sessionId, switch_core_session_get_uuid(session), MAX_SESSION_ID);
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strncpy(tech_pvt->host, host, MAX_WS_URL_LEN);
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strncpy(tech_pvt->path, path, MAX_PATH_LEN);
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tech_pvt->port = port;
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tech_pvt->sslFlags = sslFlags;
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strncpy(tech_pvt->language, lang, MAX_LANG_LEN);
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tech_pvt->interim = interim;
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tech_pvt->sampling = desiredSampling;
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tech_pvt->responseHandler = responseHandler;
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tech_pvt->channels = channels;
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tech_pvt->id = ++idxCallCount;
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tech_pvt->buffer_overrun_notified = 0;
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strncpy(tech_pvt->bugname, bugname, MAX_BUG_LEN);
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size_t buflen = LWS_PRE + (FRAME_SIZE_8000 * desiredSampling / 8000 * channels * 1000 / RTP_PACKETIZATION_PERIOD * nAudioBufferSecs);
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const char* apiKey = switch_channel_get_variable(channel, "JAMBONZ_STT_API_KEY");
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if (!apiKey && defaultApiKey) apiKey = defaultApiKey;
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else if (!apiKey) {
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switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_ERROR, "no jambonz api key provided\n");
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return SWITCH_STATUS_FALSE;
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}
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jambonz::AudioPipe* ap = new jambonz::AudioPipe(tech_pvt->sessionId, bugname, tech_pvt->host, tech_pvt->port, tech_pvt->path,
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tech_pvt->sslFlags, buflen, read_impl.decoded_bytes_per_packet, apiKey, eventCallback);
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if (!ap) {
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switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_ERROR, "Error allocating AudioPipe\n");
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return SWITCH_STATUS_FALSE;
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}
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tech_pvt->pAudioPipe = static_cast<void *>(ap);
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switch_mutex_init(&tech_pvt->mutex, SWITCH_MUTEX_NESTED, switch_core_session_get_pool(session));
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if (desiredSampling != sampling) {
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switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "(%u) resampling from %u to %u\n", tech_pvt->id, sampling, desiredSampling);
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tech_pvt->resampler = speex_resampler_init(channels, sampling, desiredSampling, SWITCH_RESAMPLE_QUALITY, &err);
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if (0 != err) {
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switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_ERROR, "Error initializing resampler: %s.\n", speex_resampler_strerror(err));
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return SWITCH_STATUS_FALSE;
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}
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}
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else {
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switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "(%u) no resampling needed for this call\n", tech_pvt->id);
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}
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switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "(%u) fork_data_init\n", tech_pvt->id);
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return SWITCH_STATUS_SUCCESS;
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}
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void lws_logger(int level, const char *line) {
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switch_log_level_t llevel = SWITCH_LOG_DEBUG;
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switch (level) {
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case LLL_ERR: llevel = SWITCH_LOG_ERROR; break;
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case LLL_WARN: llevel = SWITCH_LOG_WARNING; break;
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case LLL_NOTICE: llevel = SWITCH_LOG_NOTICE; break;
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case LLL_INFO: llevel = SWITCH_LOG_INFO; break;
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break;
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}
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switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "%s\n", line);
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}
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}
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extern "C" {
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switch_status_t jb_transcribe_init() {
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switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "mod_jambonz_transcribe: audio buffer (in secs): %d secs\n", nAudioBufferSecs);
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int logs = LLL_ERR | LLL_WARN | LLL_NOTICE ;
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// | LLL_INFO | LLL_PARSER | LLL_HEADER | LLL_EXT | LLL_CLIENT | LLL_LATENCY | LLL_DEBUG ;
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jambonz::AudioPipe::initialize(logs, lws_logger);
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switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "AudioPipe::initialize completed\n");
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const char* apiKey = std::getenv("JAMBONZ_STT_API_KEY");
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if (NULL == apiKey) {
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switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE,
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"\"JAMBONZ_STT_API_KEY\" env var not set; authentication will expect channel variables of same names to be set\n");
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}
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else {
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hasDefaultCredentials = true;
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defaultApiKey = apiKey;
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}
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return SWITCH_STATUS_SUCCESS;
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}
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switch_status_t jb_transcribe_cleanup() {
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bool cleanup = false;
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cleanup = jambonz::AudioPipe::deinitialize();
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if (cleanup == true) {
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return SWITCH_STATUS_SUCCESS;
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}
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return SWITCH_STATUS_FALSE;
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}
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switch_status_t jb_transcribe_session_init(switch_core_session_t *session,
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responseHandler_t responseHandler, uint32_t samples_per_second, int desiredSampling, uint32_t channels,
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char* lang, int interim, char* bugname, void **ppUserData)
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{
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int err;
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// allocate per-session data structure
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private_t* tech_pvt = (private_t *) switch_core_session_alloc(session, sizeof(private_t));
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if (!tech_pvt) {
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switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_ERROR, "error allocating memory!\n");
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return SWITCH_STATUS_FALSE;
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}
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if (SWITCH_STATUS_SUCCESS != fork_data_init(tech_pvt, session, samples_per_second, desiredSampling, channels, lang, interim, bugname, responseHandler)) {
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destroy_tech_pvt(tech_pvt);
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return SWITCH_STATUS_FALSE;
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}
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*ppUserData = tech_pvt;
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jambonz::AudioPipe *pAudioPipe = static_cast<jambonz::AudioPipe *>(tech_pvt->pAudioPipe);
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switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "connecting now\n");
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pAudioPipe->connect();
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switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "connection in progress\n");
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return SWITCH_STATUS_SUCCESS;
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}
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switch_status_t jb_transcribe_session_stop(switch_core_session_t *session,int channelIsClosing, char* bugname) {
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switch_channel_t *channel = switch_core_session_get_channel(session);
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switch_media_bug_t *bug = (switch_media_bug_t*) switch_channel_get_private(channel, bugname);
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if (!bug) {
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switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "jb_transcribe_session_stop: no bug %s - websocket conection already closed\n", bugname);
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return SWITCH_STATUS_FALSE;
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}
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private_t* tech_pvt = (private_t*) switch_core_media_bug_get_user_data(bug);
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if (!tech_pvt) return SWITCH_STATUS_FALSE;
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// close connection and get final responses
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|
switch_mutex_lock(tech_pvt->mutex);
|
|
{
|
|
// get the bug again, now that we are under lock
|
|
switch_media_bug_t *bug = (switch_media_bug_t*) switch_channel_get_private(channel, bugname);
|
|
if (bug) {
|
|
switch_channel_set_private(channel, bugname, NULL);
|
|
uint32_t id = tech_pvt->id;
|
|
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "(%u) jb_transcribe_session_stop\n", id);
|
|
if (!channelIsClosing) {
|
|
switch_core_media_bug_remove(session, &bug);
|
|
}
|
|
jambonz::AudioPipe *pAudioPipe = static_cast<jambonz::AudioPipe *>(tech_pvt->pAudioPipe);
|
|
if (pAudioPipe) reaper(tech_pvt);
|
|
switch_mutex_unlock(tech_pvt->mutex);
|
|
destroy_tech_pvt(tech_pvt);
|
|
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "(%u) jb_transcribe_session_stop, bug removed\n", id);
|
|
return SWITCH_STATUS_SUCCESS;
|
|
}
|
|
else {
|
|
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "jb_transcribe_session_stop: race condition, previous close completed\n");
|
|
}
|
|
switch_mutex_unlock(tech_pvt->mutex);
|
|
}
|
|
return SWITCH_STATUS_FALSE;
|
|
}
|
|
|
|
switch_bool_t jb_transcribe_frame(switch_core_session_t *session, switch_media_bug_t *bug) {
|
|
private_t* tech_pvt = (private_t*) switch_core_media_bug_get_user_data(bug);
|
|
size_t inuse = 0;
|
|
bool dirty = false;
|
|
char *p = (char *) "{\"msg\": \"buffer overrun\"}";
|
|
|
|
if (!tech_pvt) return SWITCH_TRUE;
|
|
|
|
if (switch_mutex_trylock(tech_pvt->mutex) == SWITCH_STATUS_SUCCESS) {
|
|
if (!tech_pvt->pAudioPipe) {
|
|
switch_mutex_unlock(tech_pvt->mutex);
|
|
return SWITCH_TRUE;
|
|
}
|
|
jambonz::AudioPipe *pAudioPipe = static_cast<jambonz::AudioPipe *>(tech_pvt->pAudioPipe);
|
|
if (pAudioPipe->getLwsState() != jambonz::AudioPipe::LWS_CLIENT_CONNECTED) {
|
|
switch_mutex_unlock(tech_pvt->mutex);
|
|
return SWITCH_TRUE;
|
|
}
|
|
|
|
pAudioPipe->lockAudioBuffer();
|
|
size_t available = pAudioPipe->binarySpaceAvailable();
|
|
if (NULL == tech_pvt->resampler) {
|
|
switch_frame_t frame = { 0 };
|
|
frame.data = pAudioPipe->binaryWritePtr();
|
|
frame.buflen = available;
|
|
while (true) {
|
|
|
|
// check if buffer would be overwritten; dump packets if so
|
|
if (available < pAudioPipe->binaryMinSpace()) {
|
|
if (!tech_pvt->buffer_overrun_notified) {
|
|
tech_pvt->buffer_overrun_notified = 1;
|
|
tech_pvt->responseHandler(session, TRANSCRIBE_EVENT_BUFFER_OVERRUN, NULL, tech_pvt->bugname, 0);
|
|
}
|
|
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_ERROR, "(%u) dropping packets!\n",
|
|
tech_pvt->id);
|
|
pAudioPipe->binaryWritePtrResetToZero();
|
|
|
|
frame.data = pAudioPipe->binaryWritePtr();
|
|
frame.buflen = available = pAudioPipe->binarySpaceAvailable();
|
|
}
|
|
|
|
switch_status_t rv = switch_core_media_bug_read(bug, &frame, SWITCH_TRUE);
|
|
if (rv != SWITCH_STATUS_SUCCESS) break;
|
|
if (frame.datalen) {
|
|
pAudioPipe->binaryWritePtrAdd(frame.datalen);
|
|
frame.buflen = available = pAudioPipe->binarySpaceAvailable();
|
|
frame.data = pAudioPipe->binaryWritePtr();
|
|
dirty = true;
|
|
}
|
|
}
|
|
}
|
|
else {
|
|
uint8_t data[SWITCH_RECOMMENDED_BUFFER_SIZE];
|
|
switch_frame_t frame = { 0 };
|
|
frame.data = data;
|
|
frame.buflen = SWITCH_RECOMMENDED_BUFFER_SIZE;
|
|
while (switch_core_media_bug_read(bug, &frame, SWITCH_TRUE) == SWITCH_STATUS_SUCCESS) {
|
|
if (frame.datalen) {
|
|
spx_uint32_t out_len = available >> 1; // space for samples which are 2 bytes
|
|
spx_uint32_t in_len = frame.samples;
|
|
|
|
speex_resampler_process_interleaved_int(tech_pvt->resampler,
|
|
(const spx_int16_t *) frame.data,
|
|
(spx_uint32_t *) &in_len,
|
|
(spx_int16_t *) ((char *) pAudioPipe->binaryWritePtr()),
|
|
&out_len);
|
|
|
|
if (out_len > 0) {
|
|
// bytes written = num samples * 2 * num channels
|
|
size_t bytes_written = out_len << tech_pvt->channels;
|
|
pAudioPipe->binaryWritePtrAdd(bytes_written);
|
|
available = pAudioPipe->binarySpaceAvailable();
|
|
dirty = true;
|
|
}
|
|
if (available < pAudioPipe->binaryMinSpace()) {
|
|
if (!tech_pvt->buffer_overrun_notified) {
|
|
tech_pvt->buffer_overrun_notified = 1;
|
|
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_ERROR, "(%u) dropping packets!\n",
|
|
tech_pvt->id);
|
|
tech_pvt->responseHandler(session, TRANSCRIBE_EVENT_BUFFER_OVERRUN, NULL, tech_pvt->bugname, 0);
|
|
}
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
pAudioPipe->unlockAudioBuffer();
|
|
switch_mutex_unlock(tech_pvt->mutex);
|
|
}
|
|
return SWITCH_TRUE;
|
|
}
|
|
}
|