#if HAVE_CRT
 #define _CRTDBG_MAP_ALLOC 
 #include <stdlib.h> 
 #include <crtdbg.h>
 #endif //HAVE_CRT
 /*

 * Copyright (C) 2020, University of the Basque Country (UPV/EHU)

 * Contact for licensing options: <licensing-mcpttclient(at)mcopenplatform(dot)com>
 *
 * The original file was part of Open Source Doubango Framework
 * Copyright (C) 2010-2011 Mamadou Diop.
 * Copyright (C) 2012 Doubango Telecom <http://doubango.org>
 *
 * This file is part of Open Source Doubango Framework.
 *
 * DOUBANGO is free software: you can redistribute it and/or modify
 * it under the terms of the GNU General Public License as published by
 * the Free Software Foundation, either version 3 of the License, or
 * (at your option) any later version.
 *
 * DOUBANGO is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 * GNU General Public License for more details.
 *
 * You should have received a copy of the GNU General Public License
 * along with DOUBANGO.
 *
 */
 
 
 /**@file tdav_session_audio.c
 * @brief Audio Session plugin.
 *
 * @author Mamadou Diop <diopmamadou(at)doubango.org>
 * @contributors: See $(DOUBANGO_HOME)\contributors.txt
 */

 #include <tinydav.h>

 #include "tinydav/audio/tdav_session_audio.h"
 
 //#include "tinydav/codecs/dtmf/tdav_codec_dtmf.h"
 #include "tinydav/audio/tdav_consumer_audio.h"
 
 #include "tinymedia/tmedia_resampler.h"
 #include "tinymedia/tmedia_denoise.h"
 #include "tinymedia/tmedia_jitterbuffer.h"
 #include "tinymedia/tmedia_consumer.h"
 #include "tinymedia/tmedia_producer.h"
 #include "tinymedia/tmedia_defaults.h"
 
 #include "tinyrtp/trtp_manager.h"
 #include "tinyrtp/rtp/trtp_rtp_packet.h"
 
 #include "tsk_timer.h"
 #include "tsk_memory.h"
 #include "tsk_debug.h"
 
 #define TDAV_AUDIO_RESAMPLER_DEFAULT_QUALITY 5
 
 static int _tdav_session_audio_dtmfe_timercb(const void* arg, tsk_timer_id_t timer_id);
 static struct tdav_session_audio_dtmfe_s* _tdav_session_audio_dtmfe_create(const tdav_session_audio_t* session, uint8_t event, uint16_t duration, uint32_t seq, uint32_t timestamp, uint8_t format, tsk_bool_t M, tsk_bool_t E);
 static void _tdav_session_audio_apply_gain(void* buffer, int len, int bps, int gain);
 static tmedia_resampler_t* _tdav_session_audio_resampler_create(int32_t bytes_per_sample, uint32_t in_freq, uint32_t out_freq, uint32_t frame_duration, uint32_t in_channels, uint32_t out_channels, uint32_t quality, void** resampler_buffer, tsk_size_t *resampler_buffer_size);
 
 /* DTMF event object */
 typedef struct tdav_session_audio_dtmfe_s
 {
 	TSK_DECLARE_OBJECT;
 
 	tsk_timer_id_t timer_id;
 	trtp_rtp_packet_t* packet;
 
 	const tdav_session_audio_t* session;
 }
 tdav_session_audio_dtmfe_t;
 extern const tsk_object_def_t *tdav_session_audio_dtmfe_def_t;
 
 // RTP/RTCP callback (From the network to the consumer)
 static int tdav_session_audio_rtp_cb(const void* callback_data, const struct trtp_rtp_packet_s* packet)
 {
 	tdav_session_audio_t* audio = (tdav_session_audio_t*)callback_data;
 	tmedia_codec_t* codec = tsk_null;
 	tdav_session_av_t* base = (tdav_session_av_t*)callback_data;
 	int ret = -1;
 
 	if (!audio || !packet || !packet->header) {
 		TSK_DEBUG_ERROR("Invalid parameter");
 		goto bail;
 	}
 

 	if (TMEDIA_SESSION(audio)->ro_held) {
 		return 0;
 	}
 
 	if (audio->is_started && base->consumer && base->consumer->is_started) {
 		tsk_size_t out_size = 0;
 
 		// Find the codec to use to decode the RTP payload
 		if (!audio->decoder.codec || audio->decoder.payload_type != packet->header->payload_type) {
 			tsk_istr_t format;
 			TSK_OBJECT_SAFE_FREE(audio->decoder.codec);
 			tsk_itoa(packet->header->payload_type, &format);
 			if (!(audio->decoder.codec = tmedia_codec_find_by_format(TMEDIA_SESSION(audio)->neg_codecs, format)) || !audio->decoder.codec->plugin || !audio->decoder.codec->plugin->decode){
 				TSK_DEBUG_ERROR("%s is not a valid payload for this session", format);
 				ret = -2;
 				goto bail;
 			}
 			audio->decoder.payload_type = packet->header->payload_type;
 		}
 		// ref() the codec to be able to use it short time after stop(SAFE_FREE(codec))
 		if (!(codec = tsk_object_ref(TSK_OBJECT(audio->decoder.codec)))) {
 			TSK_DEBUG_ERROR("Failed to get decoder codec");
 			goto bail;
 		}
 
 		// Open codec if not already done
 		if (!TMEDIA_CODEC(codec)->opened) {
 			tsk_safeobj_lock(base);
 			if ((ret = tmedia_codec_open(codec))) {
 				tsk_safeobj_unlock(base);
 				TSK_DEBUG_ERROR("Failed to open [%s] codec", codec->plugin->desc);
 				TSK_OBJECT_SAFE_FREE(audio->decoder.codec);
 				goto bail;
 			}
 			tsk_safeobj_unlock(base);
 		}
 		// Decode data
 		out_size = codec->plugin->decode(codec, packet->payload.data, packet->payload.size, &audio->decoder.buffer, &audio->decoder.buffer_size, packet->header);
 		if (out_size && audio->is_started) { // check "is_started" again ...to be sure stop() not called by another thread 
 			void* buffer = audio->decoder.buffer;
 			tsk_size_t size = out_size;
 
 			// resample if needed
 			if ((base->consumer->audio.out.rate && base->consumer->audio.out.rate != codec->in.rate) || (base->consumer->audio.out.channels && base->consumer->audio.out.channels != TMEDIA_CODEC_AUDIO(codec)->in.channels)) {
 				tsk_size_t resampler_result_size = 0;
 				int bytesPerSample = (base->consumer->audio.bits_per_sample >> 3);
 
 				if (!audio->decoder.resampler.instance) {
 					TSK_DEBUG_INFO("Create audio resampler(%s) for consumer: rate=%d->%d, channels=%d->%d, bytesPerSample=%d",
 						codec->plugin->desc,
 						codec->in.rate, base->consumer->audio.out.rate,
 						TMEDIA_CODEC_AUDIO(codec)->in.channels, base->consumer->audio.out.channels,
 						bytesPerSample);
 					audio->decoder.resampler.instance = _tdav_session_audio_resampler_create(
 						bytesPerSample,
 						codec->in.rate, base->consumer->audio.out.rate,
 						base->consumer->audio.ptime,
 						TMEDIA_CODEC_AUDIO(codec)->in.channels, base->consumer->audio.out.channels,
 						TDAV_AUDIO_RESAMPLER_DEFAULT_QUALITY,
 						&audio->decoder.resampler.buffer, &audio->decoder.resampler.buffer_size
 						);
 				}
 				if (!audio->decoder.resampler.instance) {
 					TSK_DEBUG_ERROR("No resampler to handle data");
 					ret = -5;
 					goto bail;
 				}
 				if (!(resampler_result_size = tmedia_resampler_process(audio->decoder.resampler.instance, buffer, size / bytesPerSample, audio->decoder.resampler.buffer, audio->decoder.resampler.buffer_size / bytesPerSample))){
 					TSK_DEBUG_ERROR("Failed to process audio resampler input buffer");
 					ret = -6;
 					goto bail;
 				}
 
 				buffer = audio->decoder.resampler.buffer;
 				size = audio->decoder.resampler.buffer_size;
 			}
 
 			// adjust the gain
 			if (base->consumer->audio.gain) {
 				_tdav_session_audio_apply_gain(buffer, (int)size, base->consumer->audio.bits_per_sample, base->consumer->audio.gain);
 			}
 			// consume the frame
 			tmedia_consumer_consume(base->consumer, buffer, size, packet->header);
 		}
 	}
 	else {
 		TSK_DEBUG_INFO("Session audio not ready");
 	}
 
 	// everything is ok
 	ret = 0;
 
 bail:
 	tsk_object_unref(TSK_OBJECT(codec));
 	return ret;
 }
 
 // Producer callback (From the producer to the network). Will encode() data before sending
 static int tdav_session_audio_producer_enc_cb(const void* callback_data, const void* buffer, tsk_size_t size)
 {
 	int ret = 0;
 

 	tdav_session_audio_t* audio = (tdav_session_audio_t*)callback_data;
 	tdav_session_av_t* base = (tdav_session_av_t*)callback_data;
 
 	if (!audio) {
 		TSK_DEBUG_ERROR("Null session");
 		return 0;
 	}
 
 	// do nothing if session is held
 	// when the session is held the end user will get feedback he also has possibilities to put the consumer and producer on pause
 	if (TMEDIA_SESSION(audio)->lo_held) {
 		return 0;
 	}
 
 	// get best negotiated codec if not already done
 	// the encoder codec could be null when session is renegotiated without re-starting (e.g. hold/resume)
 	if (!audio->encoder.codec) {
 		const tmedia_codec_t* codec;
 		tsk_safeobj_lock(base);
 		if (!(codec = tdav_session_av_get_best_neg_codec(base))) {
 			TSK_DEBUG_ERROR("No codec matched");
 			tsk_safeobj_unlock(base);
 			return -2;
 		}
 		audio->encoder.codec = tsk_object_ref(TSK_OBJECT(codec));
 		tsk_safeobj_unlock(base);
 	}
 
 	if (audio->is_started && base->rtp_manager && base->rtp_manager->is_started) {
 		/* encode */
 		tsk_size_t out_size = 0;
 
 		// Open codec if not already done
 		if (!audio->encoder.codec->opened) {
 			tsk_safeobj_lock(base);
 			if ((ret = tmedia_codec_open(audio->encoder.codec))) {
 				tsk_safeobj_unlock(base);
 				TSK_DEBUG_ERROR("Failed to open [%s] codec", audio->encoder.codec->plugin->desc);
 				return -4;
 			}
 			tsk_safeobj_unlock(base);
 		}
 		// check if we're sending DTMF or not
 		if (audio->is_sending_dtmf_events) {
 			if (base->rtp_manager) {
 				// increment the timestamp
 				base->rtp_manager->rtp.timestamp += TMEDIA_CODEC_PCM_FRAME_SIZE_AUDIO_ENCODING(audio->encoder.codec)/*duration*/;
 			}
 			TSK_DEBUG_INFO("Skiping audio frame as we're sending DTMF...");
 			return 0;
 		}
 
 		// resample if needed
 		if (base->producer->audio.rate != audio->encoder.codec->out.rate || base->producer->audio.channels != TMEDIA_CODEC_AUDIO(audio->encoder.codec)->out.channels){
 			tsk_size_t resampler_result_size = 0;
 			int bytesPerSample = (base->producer->audio.bits_per_sample >> 3);
 
 			if (!audio->encoder.resampler.instance){
 				TSK_DEBUG_INFO("Create audio resampler(%s) for producer: rate=%d->%d, channels=%d->%d, bytesPerSample=%d",
 					audio->encoder.codec->plugin->desc,
 					base->producer->audio.rate, audio->encoder.codec->out.rate,
 					base->producer->audio.channels, TMEDIA_CODEC_AUDIO(audio->encoder.codec)->out.channels,
 					bytesPerSample);
 				audio->encoder.resampler.instance = _tdav_session_audio_resampler_create(
 					bytesPerSample,
 					base->producer->audio.rate, audio->encoder.codec->out.rate,
 					base->producer->audio.ptime,
 					base->producer->audio.channels, TMEDIA_CODEC_AUDIO(audio->encoder.codec)->out.channels,
 					TDAV_AUDIO_RESAMPLER_DEFAULT_QUALITY,
 					&audio->encoder.resampler.buffer, &audio->encoder.resampler.buffer_size
 					);
 			}
 			if (!audio->encoder.resampler.instance){
 				TSK_DEBUG_ERROR("No resampler to handle data");
 				ret = -1;
 				goto done;
 			}
 			if (!(resampler_result_size = tmedia_resampler_process(audio->encoder.resampler.instance, buffer, size / bytesPerSample, audio->encoder.resampler.buffer, audio->encoder.resampler.buffer_size / bytesPerSample))){
 				TSK_DEBUG_ERROR("Failed to process audio resampler input buffer");
 				ret = -1;
 				goto done;
 			}
 
 			buffer = audio->encoder.resampler.buffer;
 			size = audio->encoder.resampler.buffer_size;
 		}
 
 		// Denoise (VAD, AGC, Noise suppression, ...)
 		// Must be done after resampling
 		if (audio->denoise){
 			tsk_bool_t silence_or_noise = tsk_false;
 			if (audio->denoise->echo_supp_enabled){
 				ret = tmedia_denoise_process_record(TMEDIA_DENOISE(audio->denoise), (void*)buffer, (uint32_t)size, &silence_or_noise);
 			}
 		}
 		// adjust the gain
 		// Must be done after resampling
 		if (base->producer->audio.gain){
 			_tdav_session_audio_apply_gain((void*)buffer, (int)size, base->producer->audio.bits_per_sample, base->producer->audio.gain);
 		}
 
 		// Encode data
 		if ((audio->encoder.codec = tsk_object_ref(audio->encoder.codec))){ /* Thread safeness (SIP reINVITE or UPDATE could update the encoder) */
 			out_size = audio->encoder.codec->plugin->encode(audio->encoder.codec, buffer, size, &audio->encoder.buffer, &audio->encoder.buffer_size);
 			if (out_size){
 				trtp_manager_send_rtp(base->rtp_manager, audio->encoder.buffer, out_size, TMEDIA_CODEC_FRAME_DURATION_AUDIO_ENCODING(audio->encoder.codec), tsk_false/*Marker*/, tsk_true/*lastPacket*/);
 			}
 			tsk_object_unref(audio->encoder.codec);
 		}
 		else{
 			TSK_DEBUG_WARN("No encoder");
 		}
 	}
 
 done:
 	return ret;
 }
 
 
 /* ============ Plugin interface ================= */
 
 static int tdav_session_audio_set(tmedia_session_t* self, const tmedia_param_t* param)
 {
 	int ret = 0;
 	tdav_session_audio_t* audio;
 
 	if (!self){
 		TSK_DEBUG_ERROR("Invalid parameter");
 		return -1;
 	}
 
 	if (tdav_session_av_set(TDAV_SESSION_AV(self), param) == tsk_true){
 		return 0;
 	}
 
 	audio = (tdav_session_audio_t*)self;
 
 	if (param->plugin_type == tmedia_ppt_consumer){
 		TSK_DEBUG_ERROR("Not expected consumer_set(%s)", param->key);
 	}
 	else if (param->plugin_type == tmedia_ppt_producer){
 		TSK_DEBUG_ERROR("Not expected producer_set(%s)", param->key);
 	}
 	else{
 		if (param->value_type == tmedia_pvt_int32){
 			if (tsk_striequals(param->key, "echo-supp")){
 				if (audio->denoise){
 					audio->denoise->echo_supp_enabled = (TSK_TO_INT32((uint8_t*)param->value) != 0);
 				}
 			}
 			else if (tsk_striequals(param->key, "echo-tail")){
 				if (audio->denoise){
 					return tmedia_denoise_set(audio->denoise, param);
 				}
 			}else if (tsk_striequals(param->key, "speech")){
 				audio->speech=TSK_TO_UINT32(((uint8_t*)param->value));
 			}

 		}
 	}
 
 	return ret;
 }
 
 static int tdav_session_audio_get(tmedia_session_t* self, tmedia_param_t* param)
 {
 	if (!self || !param){
 		TSK_DEBUG_ERROR("Invalid parameter");
 		return -1;
 	}
 
 	if (tdav_session_av_get(TDAV_SESSION_AV(self), param) == tsk_true){
 		return 0;
 	}
 
 	// the codec information is held by the session even if the user is authorized to request it for the consumer/producer
 	if (tsk_striequals("codec", param->key) && param->value_type == tmedia_pvt_pobject){
 		if (param->plugin_type == tmedia_ppt_consumer){
 			TSK_DEBUG_ERROR("Not implemented");
 			return -4;
 		}
 		else if (param->plugin_type == tmedia_ppt_producer){
 			const tmedia_codec_t* codec;
 			if (!(codec = TDAV_SESSION_AUDIO(self)->encoder.codec)){
 				codec = tdav_session_av_get_best_neg_codec((const tdav_session_av_t*)self);
 			}
 			*((tsk_object_t**)param->value) = tsk_object_ref(TSK_OBJECT(codec));
 			return 0;
 		}
 	}
 
 	TSK_DEBUG_ERROR("(%s) not expected param key for plugin_type=%d", param->key, param->plugin_type);
 	return -2;
 }
 
 static int tdav_session_audio_prepare(tmedia_session_t* self)
 {
 	tdav_session_av_t* base = (tdav_session_av_t*)(self);
 	int ret;
 
 	if ((ret = tdav_session_av_prepare(base))){
 		TSK_DEBUG_ERROR("tdav_session_av_prepare(audio) failed");
 		return ret;
 	}
 
 	if (base->rtp_manager){
 		ret = trtp_manager_set_rtp_callback(base->rtp_manager, tdav_session_audio_rtp_cb, base);
 	}
 
 	return ret;
 }
 
 static int tdav_session_audio_start(tmedia_session_t* self)
 {
 	int ret;
 	tdav_session_audio_t* audio;
 	const tmedia_codec_t* codec;
 	tdav_session_av_t* base;
 
 	if (!self){
 		TSK_DEBUG_ERROR("Invalid parameter");
 		return -1;
 	}
 
 	audio = (tdav_session_audio_t*)self;
 	base = (tdav_session_av_t*)self;
 

 
     if (!(codec = tdav_session_av_get_best_neg_codec(base))){

 		TSK_DEBUG_ERROR("No codec matched");
 		return -2;
 	}
 

 	TSK_OBJECT_SAFE_FREE(audio->encoder.codec);
 	audio->encoder.codec = tsk_object_ref((tsk_object_t*)codec);
 	if ((ret = tdav_session_av_start(base, codec))){
 		TSK_DEBUG_ERROR("tdav_session_av_start(audio) failed");
 		return ret;
 	}
 

 	if (base->rtp_manager){
 		/* Denoise (AEC, Noise Suppression, AGC)
 		* tmedia_denoise_process_record() is called after resampling and before encoding which means sampling rate is equal to codec's rate
 		* tmedia_denoise_echo_playback() is called before playback which means sampling rate is equal to consumer's rate
 		*/
 		if (audio->denoise){
 			uint32_t record_frame_size_samples = TMEDIA_CODEC_PCM_FRAME_SIZE_AUDIO_ENCODING(audio->encoder.codec);
 			uint32_t record_sampling_rate = TMEDIA_CODEC_RATE_ENCODING(audio->encoder.codec);
 			uint32_t record_channels = TMEDIA_CODEC_CHANNELS_AUDIO_ENCODING(audio->encoder.codec);
 
 			uint32_t playback_frame_size_samples = (base->consumer && base->consumer->audio.ptime && base->consumer->audio.out.rate && base->consumer->audio.out.channels)
 				? ((base->consumer->audio.ptime * base->consumer->audio.out.rate) / 1000) * base->consumer->audio.out.channels
 				: TMEDIA_CODEC_PCM_FRAME_SIZE_AUDIO_DECODING(audio->encoder.codec);
 			uint32_t playback_sampling_rate = (base->consumer && base->consumer->audio.out.rate)
 				? base->consumer->audio.out.rate
 				: TMEDIA_CODEC_RATE_DECODING(audio->encoder.codec);
 			uint32_t playback_channels = (base->consumer && base->consumer->audio.out.channels)
 				? base->consumer->audio.out.channels
 				: TMEDIA_CODEC_CHANNELS_AUDIO_DECODING(audio->encoder.codec);
 
 			TSK_DEBUG_INFO("Audio denoiser to be opened(record_frame_size_samples=%u, record_sampling_rate=%u, record_channels=%u, playback_frame_size_samples=%u, playback_sampling_rate=%u, playback_channels=%u)",
 				record_frame_size_samples, record_sampling_rate, record_channels, playback_frame_size_samples, playback_sampling_rate, playback_channels);
 
 			// close()
 			tmedia_denoise_close(audio->denoise);
 			// open() with new values
 			tmedia_denoise_open(audio->denoise,
 				record_frame_size_samples, record_sampling_rate, TSK_CLAMP(1, record_channels, 2),
 				playback_frame_size_samples, playback_sampling_rate, TSK_CLAMP(1, playback_channels, 2));
 		}
 	}
 
 	audio->is_started = (ret == 0);
 
 	return ret;
 }
 
 static int tdav_session_audio_stop(tmedia_session_t* self)
 {
 	tdav_session_audio_t* audio = TDAV_SESSION_AUDIO(self);
 	tdav_session_av_t* base = TDAV_SESSION_AV(self);
 	int ret = tdav_session_av_stop(base);
 	audio->is_started = tsk_false;
 	TSK_OBJECT_SAFE_FREE(audio->encoder.codec);
 	TSK_OBJECT_SAFE_FREE(audio->decoder.codec);
 
 	// close the jitter buffer and denoiser to be sure it will be reopened and reinitialized if reINVITE or UPDATE
 	// this is a "must" when the initial and updated sessions use codecs with different rate
 	if (audio->jitterbuffer && audio->jitterbuffer->opened) {
 		ret = tmedia_jitterbuffer_close(audio->jitterbuffer);
 	}
 	if (audio->denoise && audio->denoise->opened) {
 		ret = tmedia_denoise_close(audio->denoise);
 	}
 	return ret;
 }
 
 static int tdav_session_audio_send_dtmf(tmedia_session_t* self, uint8_t event)
 {
 	tdav_session_audio_t* audio;
 	tdav_session_av_t* base;
 	tmedia_codec_t* codec;
 	int ret, rate = 8000, ptime = 20;
 	uint16_t duration;
 	tdav_session_audio_dtmfe_t *dtmfe, *copy;
 	int format = 101;
 
 	if (!self){
 		TSK_DEBUG_ERROR("Invalid parameter");
 		return -1;
 	}
 
 	audio = (tdav_session_audio_t*)self;
 	base = (tdav_session_av_t*)self;
 
 	// Find the DTMF codec to use to use the RTP payload
 	if ((codec = tmedia_codec_find_by_format(TMEDIA_SESSION(audio)->codecs, TMEDIA_CODEC_FORMAT_DTMF))){
 		rate = (int)codec->out.rate;
 		format = atoi(codec->neg_format ? codec->neg_format : codec->format);
 		TSK_OBJECT_SAFE_FREE(codec);
 	}
 
 	/* do we have an RTP manager? */
 	if (!base->rtp_manager){
 		TSK_DEBUG_ERROR("No RTP manager associated to this session");
 		return -2;
 	}
 
 	/* Create Events list */
 	if (!audio->dtmf_events){
 		audio->dtmf_events = tsk_list_create();
 	}
 
 	/* Create global reference to the timer manager */
 	if (!audio->timer.handle_mgr_global){
 		if (!(audio->timer.handle_mgr_global = tsk_timer_mgr_global_ref())){
 			TSK_DEBUG_ERROR("Failed to create Global Timer Manager");
 			return -3;
 		}
 	}
 
 	/* Start the timer manager */
 	if (!audio->timer.started){
 		if ((ret = tsk_timer_manager_start(audio->timer.handle_mgr_global))){
 			TSK_DEBUG_ERROR("Failed to start Global Timer Manager");
 			return ret;
 		}
 		audio->timer.started = tsk_true;
 	}
 
 
 	/*	RFC 4733 - 5.  Examples
 
 	+-------+-----------+------+--------+------+--------+--------+------+
 	|  Time | Event     |   M  |  Time- |  Seq |  Event |  Dura- |   E  |
 	|  (ms) |           |  bit |  stamp |   No |  Code  |   tion |  bit |
 	+-------+-----------+------+--------+------+--------+--------+------+
 	|     0 | "9"       |      |        |      |        |        |      |
 	|       | starts    |      |        |      |        |        |      |
 	|    50 | RTP       |  "1" |      0 |    1 |    9   |    400 |  "0" |
 	|       | packet 1  |      |        |      |        |        |      |
 	|       | sent      |      |        |      |        |        |      |
 	|   100 | RTP       |  "0" |      0 |    2 |    9   |    800 |  "0" |
 	|       | packet 2  |      |        |      |        |        |      |
 	|       | sent      |      |        |      |        |        |      |
 	|   150 | RTP       |  "0" |      0 |    3 |    9   |   1200 |  "0" |
 	|       | packet 3  |      |        |      |        |        |      |
 	|       | sent      |      |        |      |        |        |      |
 	|   200 | RTP       |  "0" |      0 |    4 |    9   |   1600 |  "0" |
 	|       | packet 4  |      |        |      |        |        |      |
 	|       | sent      |      |        |      |        |        |      |
 	|   200 | "9" ends  |      |        |      |        |        |      |
 	|   250 | RTP       |  "0" |      0 |    5 |    9   |   1600 |  "1" |
 	|       | packet 4  |      |        |      |        |        |      |
 	|       | first     |      |        |      |        |        |      |
 	|       | retrans-  |      |        |      |        |        |      |
 	|       | mission   |      |        |      |        |        |      |
 	|   300 | RTP       |  "0" |      0 |    6 |    9   |   1600 |  "1" |
 	|       | packet 4  |      |        |      |        |        |      |
 	|       | second    |      |        |      |        |        |      |
 	|       | retrans-  |      |        |      |        |        |      |
 	|       | mission   |      |        |      |        |        |      |
 	=====================================================================
 	|   880 | First "1" |      |        |      |        |        |      |
 	|       | starts    |      |        |      |        |        |      |
 	|   930 | RTP       |  "1" |   7040 |    7 |    1   |    400 |  "0" |
 	|       | packet 5  |      |        |      |        |        |      |
 	|       | sent      |      |        |      |        |        |      |
 	*/
 
 	// ref()(thread safeness)
 	audio = tsk_object_ref(audio);
 
 	// says we're sending DTMF digits to avoid mixing with audio (SRTP won't let this happen because of senquence numbers)
 	// flag will be turned OFF when the list is empty
 	audio->is_sending_dtmf_events = tsk_true;
 
 	duration = TMEDIA_CODEC_PCM_FRAME_SIZE_AUDIO_ENCODING(audio->encoder.codec);
 
 	// lock() list
 	tsk_list_lock(audio->dtmf_events);
 
 	copy = dtmfe = _tdav_session_audio_dtmfe_create(audio, event, duration * 1, ++base->rtp_manager->rtp.seq_num, base->rtp_manager->rtp.timestamp, (uint8_t)format, tsk_true, tsk_false);
 	tsk_list_push_back_data(audio->dtmf_events, (void**)&dtmfe);
 	tsk_timer_mgr_global_schedule(ptime * 0, _tdav_session_audio_dtmfe_timercb, copy);
 	copy = dtmfe = _tdav_session_audio_dtmfe_create(audio, event, duration * 2, ++base->rtp_manager->rtp.seq_num, base->rtp_manager->rtp.timestamp, (uint8_t)format, tsk_false, tsk_false);
 	tsk_list_push_back_data(audio->dtmf_events, (void**)&dtmfe);
 	tsk_timer_mgr_global_schedule(ptime * 1, _tdav_session_audio_dtmfe_timercb, copy);
 	copy = dtmfe = _tdav_session_audio_dtmfe_create(audio, event, duration * 3, ++base->rtp_manager->rtp.seq_num, base->rtp_manager->rtp.timestamp, (uint8_t)format, tsk_false, tsk_false);
 	tsk_list_push_back_data(audio->dtmf_events, (void**)&dtmfe);
 	tsk_timer_mgr_global_schedule(ptime * 2, _tdav_session_audio_dtmfe_timercb, copy);
 	copy = dtmfe = _tdav_session_audio_dtmfe_create(audio, event, duration * 4, ++base->rtp_manager->rtp.seq_num, base->rtp_manager->rtp.timestamp, (uint8_t)format, tsk_false, tsk_false);
 	tsk_list_push_back_data(audio->dtmf_events, (void**)&dtmfe);
 	tsk_timer_mgr_global_schedule(ptime * 3, _tdav_session_audio_dtmfe_timercb, copy);
 
 	copy = dtmfe = _tdav_session_audio_dtmfe_create(audio, event, duration * 4, ++base->rtp_manager->rtp.seq_num, base->rtp_manager->rtp.timestamp, (uint8_t)format, tsk_false, tsk_true);
 	tsk_list_push_back_data(audio->dtmf_events, (void**)&dtmfe);
 	tsk_timer_mgr_global_schedule(ptime * 4, _tdav_session_audio_dtmfe_timercb, copy);
 	copy = dtmfe = _tdav_session_audio_dtmfe_create(audio, event, duration * 4, ++base->rtp_manager->rtp.seq_num, base->rtp_manager->rtp.timestamp, (uint8_t)format, tsk_false, tsk_true);
 	tsk_list_push_back_data(audio->dtmf_events, (void**)&dtmfe);
 	tsk_timer_mgr_global_schedule(ptime * 5, _tdav_session_audio_dtmfe_timercb, copy);
 
 	// unlock() list
 	tsk_list_unlock(audio->dtmf_events);
 
 	// increment timestamp
 	base->rtp_manager->rtp.timestamp += duration;
 
 	// unref()(thread safeness)
 	audio = tsk_object_unref(audio);
 
 	return 0;
 }
 
 static uint32_t tdav_session_audio_get_ssrc(tmedia_session_t* self)
 {
 	tdav_session_av_t* av;
 	
 	if (!self){
 		TSK_DEBUG_ERROR("Invalid parameter");
 		return -1;
 	}
 
 	av = (tdav_session_av_t*)self;
 	return av->rtp_manager->rtp.ssrc.local;
 }
 
 static uint16_t tdav_session_audio_get_last_seqnum(tmedia_session_t* self)
 {
 	tdav_session_av_t* av;
 	
 	if (!self){
 		TSK_DEBUG_ERROR("Invalid parameter");
 		return -1;
 	}
 
 	av = (tdav_session_av_t*)self;
 	return av->rtp_manager->rtp.seq_num;
 }
 
 static int tdav_session_audio_pause(tmedia_session_t* self)
 {
 	return tdav_session_av_pause(TDAV_SESSION_AV(self));
 }
 
 static const tsdp_header_M_t* tdav_session_audio_get_lo(tmedia_session_t* self)
 {
 	tsk_bool_t updated = tsk_false;
 	const tsdp_header_M_t* ret;
 	tdav_session_av_t* base = TDAV_SESSION_AV(self);
 	tmedia_session_t* base1 = TMEDIA_SESSION(self);
 	
 	if (!(ret = tdav_session_av_get_lo(base, &updated))){
 		TSK_DEBUG_ERROR("tdav_session_av_get_lo(audio) failed");
 		return tsk_null;

 	}else if(TDAV_SESSION_AUDIO(self)->speech > 0){//MCPTT

 		tsdp_header_M_add_headers(base1->M.lo,
 				    TSDP_HEADER_I_VA_ARGS("speech"),
 					tsk_null);
 	}
 
 	if (updated){
 		tsk_safeobj_lock(base);
 		TSK_OBJECT_SAFE_FREE(TDAV_SESSION_AUDIO(self)->encoder.codec);
 		tsk_safeobj_unlock(base);
 	}
 
 	return ret;
 }
 
 static int tdav_session_audio_set_ro(tmedia_session_t* self, const tsdp_header_M_t* m)
 {
 	int ret;
 	tsk_bool_t updated = tsk_false;
 	tdav_session_av_t* base = TDAV_SESSION_AV(self);
 
 	if ((ret = tdav_session_av_set_ro(base, m, &updated))){
 		TSK_DEBUG_ERROR("tdav_session_av_set_ro(audio) failed");
 		return ret;
 	}
 
 	if (updated) {
 		tsk_safeobj_lock(base);
 		// reset audio jitter buffer (new Offer probably comes with new seq_nums or timestamps)
 		if (base->consumer) {
 			ret = tdav_consumer_audio_reset(TDAV_CONSUMER_AUDIO(base->consumer));
 		}
 		// destroy encoder to force requesting new one
 		TSK_OBJECT_SAFE_FREE(TDAV_SESSION_AUDIO(self)->encoder.codec);
 		tsk_safeobj_unlock(base);
 	}
 
 	return ret;
 }
 
 /* apply gain */
 static void _tdav_session_audio_apply_gain(void* buffer, int len, int bps, int gain)
 {
 	register int i;
 	int max_val;
 
 	max_val = (1 << (bps - 1 - gain)) - 1;
 
 	if (bps == 8) {
 		int8_t *buff = buffer;
 		for (i = 0; i < len; i++) {
 			if (buff[i] > -max_val && buff[i] < max_val)
 				buff[i] = buff[i] << gain;
 		}
 	}
 	else if (bps == 16) {
 		int16_t *buff = buffer;
 		for (i = 0; i < len / 2; i++) {
 			if (buff[i] > -max_val && buff[i] < max_val)
 				buff[i] = buff[i] << gain;
 		}
 	}
 }
 
 
 /* Internal function used to create new DTMF event */
 static tdav_session_audio_dtmfe_t* _tdav_session_audio_dtmfe_create(const tdav_session_audio_t* session, uint8_t event, uint16_t duration, uint32_t seq, uint32_t timestamp, uint8_t format, tsk_bool_t M, tsk_bool_t E)
 {
 	tdav_session_audio_dtmfe_t* dtmfe;
 	const tdav_session_av_t* base = (const tdav_session_av_t*)session;
 	static uint8_t volume = 10;
 	static uint32_t ssrc = 0x5234A8;
 
 	uint8_t pay[4] = { 0 };
 
 	/* RFC 4733 - 2.3.  Payload Format
 	0                   1                   2                   3
 	0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
 	+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 	|     event     |E|R| volume    |          duration             |
 	+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 	*/
 
 	if (!(dtmfe = tsk_object_new(tdav_session_audio_dtmfe_def_t))){
 		TSK_DEBUG_ERROR("Failed to create new DTMF event");
 		return tsk_null;
 	}
 	dtmfe->session = session;
 
 	if (!(dtmfe->packet = trtp_rtp_packet_create((session && base->rtp_manager) ? base->rtp_manager->rtp.ssrc.local : ssrc, seq, timestamp, format, M))){
 		TSK_DEBUG_ERROR("Failed to create DTMF RTP packet");
 		TSK_OBJECT_SAFE_FREE(dtmfe);
 		return tsk_null;
 	}
 
 	pay[0] = event;
 	pay[1] |= ((E << 7) | (volume & 0x3F));
 	pay[2] = (duration >> 8);
 	pay[3] = (duration & 0xFF);
 
 	/* set data */
 	#if HAVE_CRT //Debug memory
 	if ((dtmfe->packet->payload.data = calloc(sizeof(pay), sizeof(uint8_t)))){
 		
 	#else
 	if ((dtmfe->packet->payload.data = tsk_calloc(sizeof(pay), sizeof(uint8_t)))){
 		
 	#endif //HAVE_CRT
 		memcpy(dtmfe->packet->payload.data, pay, sizeof(pay));
 		dtmfe->packet->payload.size = sizeof(pay);
 	}
 
 	return dtmfe;
 }
 
 static int _tdav_session_audio_dtmfe_timercb(const void* arg, tsk_timer_id_t timer_id)
 {
 	tdav_session_audio_dtmfe_t* dtmfe = (tdav_session_audio_dtmfe_t*)arg;
 	tdav_session_audio_t *audio;
 
 	if (!dtmfe || !dtmfe->session || !dtmfe->session->dtmf_events){
 		TSK_DEBUG_ERROR("Invalid parameter");
 		return -1;
 	}
 
 	/* Send the data */
 	TSK_DEBUG_INFO("Sending DTMF event...");
 	trtp_manager_send_rtp_packet(TDAV_SESSION_AV(dtmfe->session)->rtp_manager, dtmfe->packet, tsk_false);
 
 
 	audio = tsk_object_ref(TSK_OBJECT(dtmfe->session));
 	tsk_list_lock(audio->dtmf_events);
 	/* Remove and delete the event from the queue */
 	tsk_list_remove_item_by_data(audio->dtmf_events, dtmfe);
 	/* Check if there are pending events */
 	audio->is_sending_dtmf_events = !TSK_LIST_IS_EMPTY(audio->dtmf_events);
 	tsk_list_unlock(audio->dtmf_events);
 	tsk_object_unref(audio);
 
 	return 0;
 }
 
 static tmedia_resampler_t* _tdav_session_audio_resampler_create(int32_t bytes_per_sample, uint32_t in_freq, uint32_t out_freq, uint32_t frame_duration, uint32_t in_channels, uint32_t out_channels, uint32_t quality, void** resampler_buffer, tsk_size_t *resampler_buffer_size)
 {
 	uint32_t resampler_buff_size;
 	tmedia_resampler_t* resampler;
 	int ret;
 
 	if (out_channels > 2 || in_channels > 2) {
 		TSK_DEBUG_ERROR("Invalid parameter: out_channels=%u, in_channels=%u", out_channels, in_channels);
 		return tsk_null;
 	}
 
 	resampler_buff_size = (((out_freq * frame_duration) / 1000) * bytes_per_sample) << (out_channels == 2 ? 1 : 0);
 
 	if (!(resampler = tmedia_resampler_create())) {
 		TSK_DEBUG_ERROR("Failed to create audio resampler");
 		return tsk_null;
 	}
 	else {
 		if ((ret = tmedia_resampler_open(resampler, in_freq, out_freq, frame_duration, in_channels, out_channels, quality, 16))) {
 			TSK_DEBUG_ERROR("Failed to open audio resampler (%d, %d, %d, %d, %d,%d) with retcode=%d", in_freq, out_freq, frame_duration, in_channels, out_channels, quality, ret);
 			TSK_OBJECT_SAFE_FREE(resampler);
 			goto done;
 		}
 	}
 	// create temp resampler buffer
 	if ((*resampler_buffer = tsk_realloc(*resampler_buffer, resampler_buff_size))) {
 		*resampler_buffer_size = resampler_buff_size;
 	}
 	else {
 		*resampler_buffer_size = 0;
 		TSK_DEBUG_ERROR("Failed to allocate resampler buffer with size = %d", resampler_buff_size);
 		TSK_OBJECT_SAFE_FREE(resampler);
 		goto done;
 	}
 done:
 	return resampler;
 }
 
 //=================================================================================================
 //	Session Audio Plugin object definition
 //
 /* constructor */
 static tsk_object_t* tdav_session_audio_ctor(tsk_object_t * self, va_list * app)
 {
 	tdav_session_audio_t *audio = self;
 	if (audio){
 		int ret;
 		tdav_session_av_t *base = TDAV_SESSION_AV(self);
 
 		/* init() base */
 		if ((ret = tdav_session_av_init(base, tmedia_audio)) != 0){
 			TSK_DEBUG_ERROR("tdav_session_av_init(audio) failed");
 			return tsk_null;
 		}
 
 		/* init() self */
 		if (base->producer){
 			tmedia_producer_set_enc_callback(base->producer, tdav_session_audio_producer_enc_cb, audio);
 		}
 		if (base->consumer){
 			// It's important to create the denoiser and jitter buffer here as dynamic plugins (from shared libs) don't have access to the registry
 			if (!(audio->denoise = tmedia_denoise_create())){
 				TSK_DEBUG_WARN("No Audio denoiser found");
 			}
 			else{
 				// IMPORTANT: This means that the consumer must be child of "tdav_consumer_audio_t" object
 				tdav_consumer_audio_set_denoise(TDAV_CONSUMER_AUDIO(base->consumer), audio->denoise);
 			}
 
 			if (!(audio->jitterbuffer = tmedia_jitterbuffer_create(tmedia_audio))){
 				TSK_DEBUG_ERROR("Failed to create jitter buffer");
 			}
 			else{
 				ret = tmedia_jitterbuffer_init(TMEDIA_JITTER_BUFFER(audio->jitterbuffer));
 				tdav_consumer_audio_set_jitterbuffer(TDAV_CONSUMER_AUDIO(base->consumer), audio->jitterbuffer);
 			}
 		}
 	}
 	return self;
 }
 /* destructor */
 static tsk_object_t* tdav_session_audio_dtor(tsk_object_t * self)
 {
 	tdav_session_audio_t *audio = self;
 	TSK_DEBUG_INFO("*** tdav_session_audio_t destroyed ***");
 	if (audio){
 		tdav_session_audio_stop((tmedia_session_t*)audio);
 		// Do it in this order (deinit self first)
 
 		/* Timer manager */
 		if (audio->timer.started){
 			if (audio->dtmf_events){
 				/* Cancel all events */
 				tsk_list_item_t* item;
 				tsk_list_foreach(item, audio->dtmf_events){
 					tsk_timer_mgr_global_cancel(((tdav_session_audio_dtmfe_t*)item->data)->timer_id);
 				}
 			}
 		}
 
 		tsk_timer_mgr_global_unref(&audio->timer.handle_mgr_global);
 
 		/* CleanUp the DTMF events */
 		TSK_OBJECT_SAFE_FREE(audio->dtmf_events);
 
 		TSK_OBJECT_SAFE_FREE(audio->denoise);
 		TSK_OBJECT_SAFE_FREE(audio->jitterbuffer);
 
 		TSK_OBJECT_SAFE_FREE(audio->encoder.codec);
 		TSK_FREE(audio->encoder.buffer);
 		TSK_OBJECT_SAFE_FREE(audio->decoder.codec);
 		TSK_FREE(audio->decoder.buffer);
 
 		// free resamplers
 		TSK_FREE(audio->encoder.resampler.buffer);
 		TSK_OBJECT_SAFE_FREE(audio->encoder.resampler.instance);
 		TSK_FREE(audio->decoder.resampler.buffer);
 		TSK_OBJECT_SAFE_FREE(audio->decoder.resampler.instance);
 
 		/* deinit base */
 		tdav_session_av_deinit(TDAV_SESSION_AV(self));
 
 		TSK_DEBUG_INFO("*** Audio session destroyed ***");
 	}
 
 	return self;
 }
 /* object definition */
 static const tsk_object_def_t tdav_session_audio_def_s =
 {
 	sizeof(tdav_session_audio_t),
 	tdav_session_audio_ctor,
 	tdav_session_audio_dtor,
 	tmedia_session_cmp,
 };
 /* plugin definition*/
 static const tmedia_session_plugin_def_t tdav_session_audio_plugin_def_s =
 {
 	&tdav_session_audio_def_s,
 
 	tmedia_audio,
 	"audio",
 
 	tdav_session_audio_set,
 	tdav_session_audio_get,
 	tdav_session_audio_prepare,
 	tdav_session_audio_start,
 	tdav_session_audio_pause,
 	tdav_session_audio_stop,
 
 	/* Audio part */
 	{
 		tdav_session_audio_send_dtmf,
 		tdav_session_audio_get_ssrc,
 		tdav_session_audio_get_last_seqnum,
 	},
 
 	tdav_session_audio_get_lo,
 	tdav_session_audio_set_ro
 };
 const tmedia_session_plugin_def_t *tdav_session_audio_plugin_def_t = &tdav_session_audio_plugin_def_s;
 static const tmedia_session_plugin_def_t tdav_session_bfcpaudio_plugin_def_s =
 {
 	&tdav_session_audio_def_s,
 
 	tmedia_bfcp_audio,
 	"audio",
 
 	tdav_session_audio_set,
 	tdav_session_audio_get,
 	tdav_session_audio_prepare,
 	tdav_session_audio_start,
 	tdav_session_audio_pause,
 	tdav_session_audio_stop,
 
 	/* Audio part */
 	{
 		tdav_session_audio_send_dtmf,
 		tdav_session_audio_get_ssrc,
 		tdav_session_audio_get_last_seqnum,
 	},
 
 	tdav_session_audio_get_lo,
 	tdav_session_audio_set_ro
 };
 const tmedia_session_plugin_def_t *tdav_session_bfcpaudio_plugin_def_t = &tdav_session_bfcpaudio_plugin_def_s;
 
 static const tmedia_session_plugin_def_t tdav_session_mbmsaudio_plugin_def_s =
 {
 	&tdav_session_audio_def_s,
 
 	tmedia_mbms_audio,
 	"audio",
 
 	tdav_session_audio_set,
 	tdav_session_audio_get,
 	tdav_session_audio_prepare,
 	tdav_session_audio_start,
 	tdav_session_audio_pause,
 	tdav_session_audio_stop,
 
 	/* Audio part */
 	{
 		tdav_session_audio_send_dtmf,
 		tdav_session_audio_get_ssrc,
 		tdav_session_audio_get_last_seqnum,
 	},
 
 	tdav_session_audio_get_lo,
 	tdav_session_audio_set_ro
 };
 const tmedia_session_plugin_def_t *tdav_session_mbmsaudio_plugin_def_t = &tdav_session_mbmsaudio_plugin_def_s;
 
 
 //=================================================================================================
 //	DTMF event object definition
 //
 static tsk_object_t* tdav_session_audio_dtmfe_ctor(tsk_object_t * self, va_list * app)
 {
 	tdav_session_audio_dtmfe_t *event = self;
 	if (event){
 		event->timer_id = TSK_INVALID_TIMER_ID;
 	}
 	return self;
 }
 
 static tsk_object_t* tdav_session_audio_dtmfe_dtor(tsk_object_t * self)
 {
 	tdav_session_audio_dtmfe_t *event = self;
 	if (event){
 		TSK_OBJECT_SAFE_FREE(event->packet);
 	}
 
 	return self;
 }
 
 static int tdav_session_audio_dtmfe_cmp(const tsk_object_t *_e1, const tsk_object_t *_e2)
 {
 	int ret;
 	tsk_subsat_int32_ptr(_e1, _e2, &ret);
 	return ret;
 }
 
 static const tsk_object_def_t tdav_session_audio_dtmfe_def_s =
 {
 	sizeof(tdav_session_audio_dtmfe_t),
 	tdav_session_audio_dtmfe_ctor,
 	tdav_session_audio_dtmfe_dtor,
 	tdav_session_audio_dtmfe_cmp,
 };
 const tsk_object_def_t *tdav_session_audio_dtmfe_def_t = &tdav_session_audio_dtmfe_def_s;