text: /* $Id$ */
  /* 
   * Copyright (C) 2008-2011 Teluu Inc. (http://www.teluu.com)
   * Copyright (C) 2003-2008 Benny Prijono <benny@prijono.org>
   *
   * This program 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 2 of the License, or
   * (at your option) any later version.
   *
   * This program 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 this program; if not, write to the Free Software
   * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA 
   */
  #ifndef __PJMEDIA_CONFIG_H__
  #define __PJMEDIA_CONFIG_H__
  
  /**
   * @file pjmedia/config.h Compile time config
   * @brief Contains some compile time constants.
   */
  #include <ZadarmaPJSIP/pj/config.h>
  
  /**
   * @defgroup PJMEDIA_BASE Base Types and Configurations
   */
  
  /**
   * @defgroup PJMEDIA_CONFIG Compile time configuration
   * @ingroup PJMEDIA_BASE
   * @brief Some compile time configuration settings.
   * @{
   */
  
  /*
   * Include config_auto.h if autoconf is used (PJ_AUTOCONF is set)
   */
  #if defined(PJ_AUTOCONF)
  #   include <ZadarmaPJSIP/pjmedia/config_auto.h>
  #endif
  
  /**
   * Specify whether we prefer to use audio switch board rather than 
   * conference bridge.
   *
   * Audio switch board is a kind of simplified version of conference 
   * bridge, but not really the subset of conference bridge. It has 
   * stricter rules on audio routing among the pjmedia ports and has
   * no audio mixing capability. The power of it is it could work with
   * encoded audio frames where conference brigde couldn't.
   *
   * Default: 0
   */
  #ifndef PJMEDIA_CONF_USE_SWITCH_BOARD
  #   define PJMEDIA_CONF_USE_SWITCH_BOARD    0
  #endif
  
  /**
   * Specify buffer size for audio switch board, in bytes. This buffer will
   * be used for transmitting/receiving audio frame data (and some overheads,
   * i.e: pjmedia_frame structure) among conference ports in the audio
   * switch board. For example, if a port uses PCM format @44100Hz mono
   * and frame time 20ms, the PCM audio data will require 1764 bytes,
   * so with overhead, a safe buffer size will be ~1900 bytes.
   *
   * Default: PJMEDIA_MAX_MTU
   */
  #ifndef PJMEDIA_CONF_SWITCH_BOARD_BUF_SIZE
  #   define PJMEDIA_CONF_SWITCH_BOARD_BUF_SIZE    PJMEDIA_MAX_MTU
  #endif
  
  /**
   * Specify whether the conference bridge uses AGC, an automatic adjustment to
   * avoid dramatic change in the signal level which can cause noise.
   *
   * Default: 1 (enabled)
   */
  #ifndef PJMEDIA_CONF_USE_AGC
  #   define PJMEDIA_CONF_USE_AGC    	    1
  #endif
  
  
  /*
   * Types of sound stream backends.
   */
  
  /**
   * This macro has been deprecated in releasee 1.1. Please see
   * http://trac.pjsip.org/repos/wiki/Audio_Dev_API for more information.
   */
  #if defined(PJMEDIA_SOUND_IMPLEMENTATION)
  #   error PJMEDIA_SOUND_IMPLEMENTATION has been deprecated
  #endif
  
  /**
   * This macro has been deprecated in releasee 1.1. Please see
   * http://trac.pjsip.org/repos/wiki/Audio_Dev_API for more information.
   */
  #if defined(PJMEDIA_PREFER_DIRECT_SOUND)
  #   error PJMEDIA_PREFER_DIRECT_SOUND has been deprecated
  #endif
  
  /**
   * This macro controls whether the legacy sound device API is to be
   * implemented, for applications that still use the old sound device
   * API (sound.h). If this macro is set to non-zero, the sound_legacy.c
   * will be included in the compilation. The sound_legacy.c is an
   * implementation of old sound device (sound.h) using the new Audio
   * Device API.
   *
   * Please see http://trac.pjsip.org/repos/wiki/Audio_Dev_API for more
   * info.
   */
  #ifndef PJMEDIA_HAS_LEGACY_SOUND_API
  #   define PJMEDIA_HAS_LEGACY_SOUND_API	    1
  #endif
  
  /**
   * Specify default sound device latency, in milisecond.
   */
  #ifndef PJMEDIA_SND_DEFAULT_REC_LATENCY
  #   define PJMEDIA_SND_DEFAULT_REC_LATENCY  100
  #endif
  
  /**
   * Specify default sound device latency, in milisecond. 
   *
   * Default is 160ms for Windows Mobile and 140ms for other platforms.
   */
  #ifndef PJMEDIA_SND_DEFAULT_PLAY_LATENCY
  #   if defined(PJ_WIN32_WINCE) && PJ_WIN32_WINCE!=0
  #	define PJMEDIA_SND_DEFAULT_PLAY_LATENCY	    160
  #   else
  #	define PJMEDIA_SND_DEFAULT_PLAY_LATENCY	    140
  #   endif
  #endif
  
  
  /*
   * Types of WSOLA backend algorithm.
   */
  
  /**
   * This denotes implementation of WSOLA using null algorithm. Expansion
   * will generate zero frames, and compression will just discard some
   * samples from the input.
   *
   * This type of implementation may be used as it requires the least
   * processing power.
   */
  #define PJMEDIA_WSOLA_IMP_NULL		    0
  
  /**
   * This denotes implementation of WSOLA using fixed or floating point WSOLA
   * algorithm. This implementation provides the best quality of the result,
   * at the expense of one frame delay and intensive processing power 
   * requirement.
   */
  #define PJMEDIA_WSOLA_IMP_WSOLA		    1
  
  /**
   * This denotes implementation of WSOLA algorithm with faster waveform 
   * similarity calculation. This implementation provides fair quality of 
   * the result with the main advantage of low processing power requirement.
   */
  #define PJMEDIA_WSOLA_IMP_WSOLA_LITE	    2
  
  /**
   * Specify type of Waveform based Similarity Overlap and Add (WSOLA) backend
   * implementation to be used. WSOLA is an algorithm to expand and/or compress 
   * audio frames without changing the pitch, and used by the delaybuf and as PLC
   * backend algorithm.
   *
   * Default is PJMEDIA_WSOLA_IMP_WSOLA
   */
  #ifndef PJMEDIA_WSOLA_IMP
  #   define PJMEDIA_WSOLA_IMP		    PJMEDIA_WSOLA_IMP_WSOLA
  #endif
  
  
  /**
   * Specify the default maximum duration of synthetic audio that is generated
   * by WSOLA. This value should be long enough to cover burst of packet losses. 
   * but not too long, because as the duration increases the quality would 
   * degrade considerably.
   *
   * Note that this limit is only applied when fading is enabled in the WSOLA
   * session.
   *
   * Default: 80
   */
  #ifndef PJMEDIA_WSOLA_MAX_EXPAND_MSEC
  #   define PJMEDIA_WSOLA_MAX_EXPAND_MSEC    80
  #endif
  
  
  /**
   * Specify WSOLA template length, in milliseconds. The longer the template,
   * the smoother signal to be generated at the expense of more computation
   * needed, since the algorithm will have to compare more samples to find
   * the most similar pitch.
   *
   * Default: 5
   */
  #ifndef PJMEDIA_WSOLA_TEMPLATE_LENGTH_MSEC
  #   define PJMEDIA_WSOLA_TEMPLATE_LENGTH_MSEC	5
  #endif
  
  
  /**
   * Specify WSOLA algorithm delay, in milliseconds. The algorithm delay is
   * used to merge synthetic samples with real samples in the transition
   * between real to synthetic and vice versa. The longer the delay, the 
   * smoother signal to be generated, at the expense of longer latency and
   * a slighty more computation.
   *
   * Default: 5
   */
  #ifndef PJMEDIA_WSOLA_DELAY_MSEC
  #   define PJMEDIA_WSOLA_DELAY_MSEC	    5
  #endif
  
  
  /**
   * Set this to non-zero to disable fade-out/in effect in the PLC when it
   * instructs WSOLA to generate synthetic frames. The use of fading may
   * or may not improve the quality of audio, depending on the nature of
   * packet loss and the type of audio input (e.g. speech vs music).
   * Disabling fading also implicitly remove the maximum limit of synthetic
   * audio samples generated by WSOLA (see PJMEDIA_WSOLA_MAX_EXPAND_MSEC).
   *
   * Default: 0
   */
  #ifndef PJMEDIA_WSOLA_PLC_NO_FADING
  #   define PJMEDIA_WSOLA_PLC_NO_FADING	    0
  #endif
  
  
  /**
   * Limit the number of calls by stream to the PLC to generate synthetic
   * frames to this duration. If packets are still lost after this maximum
   * duration, silence will be generated by the stream instead. Since the
   * PLC normally should have its own limit on the maximum duration of
   * synthetic frames to be generated (for PJMEDIA's PLC, the limit is
   * PJMEDIA_WSOLA_MAX_EXPAND_MSEC), we can set this value to a large number
   * to give additional flexibility should the PLC wants to do something
   * clever with the lost frames.
   *
   * Default: 240 ms
   */
  #ifndef PJMEDIA_MAX_PLC_DURATION_MSEC
  #   define PJMEDIA_MAX_PLC_DURATION_MSEC    240
  #endif
  
  
  /**
   * Specify number of sound buffers. Larger number is better for sound
   * stability and to accommodate sound devices that are unable to send frames
   * in timely manner, however it would probably cause more audio delay (and 
   * definitely will take more memory). One individual buffer is normally 10ms
   * or 20 ms long, depending on ptime settings (samples_per_frame value).
   *
   * The setting here currently is used by the conference bridge, the splitter
   * combiner port, and dsound.c.
   *
   * Default: (PJMEDIA_SND_DEFAULT_PLAY_LATENCY+20)/20
   */
  #ifndef PJMEDIA_SOUND_BUFFER_COUNT
  #   define PJMEDIA_SOUND_BUFFER_COUNT	    ((PJMEDIA_SND_DEFAULT_PLAY_LATENCY+20)/20)
  #endif
  
  
  /**
   * Specify which A-law/U-law conversion algorithm to use.
   * By default the conversion algorithm uses A-law/U-law table which gives
   * the best performance, at the expense of 33 KBytes of static data.
   * If this option is disabled, a smaller but slower algorithm will be used.
   */
  #ifndef PJMEDIA_HAS_ALAW_ULAW_TABLE
  #   define PJMEDIA_HAS_ALAW_ULAW_TABLE	    1
  #endif
  
  
  /**
   * Unless specified otherwise, G711 codec is included by default.
   */
  #ifndef PJMEDIA_HAS_G711_CODEC
  #   define PJMEDIA_HAS_G711_CODEC	    1
  #endif
  
  
  /*
   * Warn about obsolete macros.
   *
   * PJMEDIA_HAS_SMALL_FILTER has been deprecated in 0.7.
   */
  #if defined(PJMEDIA_HAS_SMALL_FILTER)
  #   ifdef _MSC_VER
  #	pragma message("Warning: PJMEDIA_HAS_SMALL_FILTER macro is deprecated"\
  		       " and has no effect")
  #   else
  #	warning "PJMEDIA_HAS_SMALL_FILTER macro is deprecated and has no effect"
  #   endif
  #endif
  
  
  /*
   * Warn about obsolete macros.
   *
   * PJMEDIA_HAS_LARGE_FILTER has been deprecated in 0.7.
   */
  #if defined(PJMEDIA_HAS_LARGE_FILTER)
  #   ifdef _MSC_VER
  #	pragma message("Warning: PJMEDIA_HAS_LARGE_FILTER macro is deprecated"\
  		       " and has no effect")
  #   else
  #	warning "PJMEDIA_HAS_LARGE_FILTER macro is deprecated"
  #   endif
  #endif
  
  
  /*
   * These macros are obsolete in 0.7.1 so it will trigger compilation error.
   * Please use PJMEDIA_RESAMPLE_IMP to select the resample implementation
   * to use.
   */
  #ifdef PJMEDIA_HAS_LIBRESAMPLE
  #   error "PJMEDIA_HAS_LIBRESAMPLE macro is deprecated. Use '#define PJMEDIA_RESAMPLE_IMP PJMEDIA_RESAMPLE_LIBRESAMPLE'"
  #endif
  
  #ifdef PJMEDIA_HAS_SPEEX_RESAMPLE
  #   error "PJMEDIA_HAS_SPEEX_RESAMPLE macro is deprecated. Use '#define PJMEDIA_RESAMPLE_IMP PJMEDIA_RESAMPLE_SPEEX'"
  #endif
  
  
  /*
   * Sample rate conversion backends.
   * Select one of these backends in PJMEDIA_RESAMPLE_IMP.
   */
  #define PJMEDIA_RESAMPLE_NONE		    1	/**< No resampling.	    */
  #define PJMEDIA_RESAMPLE_LIBRESAMPLE	    2	/**< Sample rate conversion 
  						     using libresample.  */
  #define PJMEDIA_RESAMPLE_SPEEX		    3	/**< Sample rate conversion 
  						     using Speex. */
  #define PJMEDIA_RESAMPLE_LIBSAMPLERATE	    4	/**< Sample rate conversion 
  						     using libsamplerate 
  						     (a.k.a Secret Rabbit Code)
  						 */
  
  /**
   * Select which resample implementation to use. Currently pjmedia supports:
   *  - #PJMEDIA_RESAMPLE_LIBRESAMPLE, to use libresample-1.7, this is the default
   *    implementation to be used.
   *  - #PJMEDIA_RESAMPLE_LIBSAMPLERATE, to use libsamplerate implementation
   *    (a.k.a. Secret Rabbit Code).
   *  - #PJMEDIA_RESAMPLE_SPEEX, to use sample rate conversion in Speex library.
   *  - #PJMEDIA_RESAMPLE_NONE, to disable sample rate conversion. Any calls to
   *    resample function will return error.
   *
   * Default is PJMEDIA_RESAMPLE_LIBRESAMPLE
   */
  #ifndef PJMEDIA_RESAMPLE_IMP
  #   define PJMEDIA_RESAMPLE_IMP		    PJMEDIA_RESAMPLE_LIBRESAMPLE
  #endif
  
  
  /**
   * Specify whether libsamplerate, when used, should be linked statically
   * into the application. This option is only useful for Visual Studio
   * projects, and when this static linking is enabled
   */
  
  
  /**
   * Default file player/writer buffer size.
   */
  #ifndef PJMEDIA_FILE_PORT_BUFSIZE
  #   define PJMEDIA_FILE_PORT_BUFSIZE		4000
  #endif
  
  
  /**
   * Maximum frame duration (in msec) to be supported.
   * This (among other thing) will affect the size of buffers to be allocated
   * for outgoing packets.
   */
  #ifndef PJMEDIA_MAX_FRAME_DURATION_MS   
  #   define PJMEDIA_MAX_FRAME_DURATION_MS   	200
  #endif
  
  
  /**
   * Max packet size for transmitting direction.
   */
  #ifndef PJMEDIA_MAX_MTU			
  #  define PJMEDIA_MAX_MTU			1500
  #endif
  
  
  /**
   * Max packet size for receiving direction.
   */
  #ifndef PJMEDIA_MAX_MRU			
  #  define PJMEDIA_MAX_MRU			2000
  #endif
  
  
  /**
   * DTMF/telephone-event duration, in timestamp. To specify the duration in
   * milliseconds, use the setting PJMEDIA_DTMF_DURATION_MSEC instead.
   */
  #ifndef PJMEDIA_DTMF_DURATION		
  #  define PJMEDIA_DTMF_DURATION			1600	/* in timestamp */
  #endif
  
  
  /**
   * DTMF/telephone-event duration, in milliseconds. If the value is greater
   * than zero, than this setting will be used instead of PJMEDIA_DTMF_DURATION.
   *
   * Note that for a clockrate of 8 KHz, a dtmf duration of 1600 timestamp
   * units (the default value of PJMEDIA_DTMF_DURATION) is equivalent to 200 ms. 
   */
  #ifndef PJMEDIA_DTMF_DURATION_MSEC		
  #  define PJMEDIA_DTMF_DURATION_MSEC		0
  #endif
  
  
  /**
   * Number of RTP packets received from different source IP address from the
   * remote address required to make the stream switch transmission
   * to the source address.
   */
  #ifndef PJMEDIA_RTP_NAT_PROBATION_CNT	
  #  define PJMEDIA_RTP_NAT_PROBATION_CNT		10
  #endif
  
  
  /**
   * Number of RTCP packets received from different source IP address from the
   * remote address required to make the stream switch RTCP transmission
   * to the source address.
   */
  #ifndef PJMEDIA_RTCP_NAT_PROBATION_CNT
  #  define PJMEDIA_RTCP_NAT_PROBATION_CNT	3
  #endif
  
  
  /**
   * Specify whether RTCP should be advertised in SDP. This setting would
   * affect whether RTCP candidate will be added in SDP when ICE is used.
   * Application might want to disable RTCP advertisement in SDP to
   * reduce the message size.
   *
   * Default: 1 (yes)
   */
  #ifndef PJMEDIA_ADVERTISE_RTCP
  #   define PJMEDIA_ADVERTISE_RTCP		1
  #endif
  
  
  /**
   * Interval to send RTCP packets, in msec
   */
  #ifndef PJMEDIA_RTCP_INTERVAL
  #	define PJMEDIA_RTCP_INTERVAL		5000	/* msec*/
  #endif
  
  
  /**
   * Tell RTCP to ignore the first N packets when calculating the
   * jitter statistics. From experimentation, the first few packets
   * (25 or so) have relatively big jitter, possibly because during
   * this time, the program is also busy setting up the signaling,
   * so they make the average jitter big.
   *
   * Default: 25.
   */
  #ifndef PJMEDIA_RTCP_IGNORE_FIRST_PACKETS
  #   define  PJMEDIA_RTCP_IGNORE_FIRST_PACKETS	25
  #endif
  
  
  /**
   * Specify whether RTCP statistics includes raw jitter statistics.
   * Raw jitter is defined as absolute value of network transit time
   * difference of two consecutive packets; refering to "difference D"
   * term in interarrival jitter calculation in RFC 3550 section 6.4.1.
   *
   * Default: 0 (no).
   */
  #ifndef PJMEDIA_RTCP_STAT_HAS_RAW_JITTER
  #   define PJMEDIA_RTCP_STAT_HAS_RAW_JITTER	0
  #endif
  
  /**
   * Specify the factor with wich RTCP RTT statistics should be normalized 
   * if exceptionally high. For e.g. mobile networks with potentially large
   * fluctuations, this might be unwanted.
   *
   * Use (0) to disable this feature.
   *
   * Default: 3.
   */
  #ifndef PJMEDIA_RTCP_NORMALIZE_FACTOR
  #   define PJMEDIA_RTCP_NORMALIZE_FACTOR	3
  #endif
  
  
  /**
   * Specify whether RTCP statistics includes IP Delay Variation statistics.
   * IPDV is defined as network transit time difference of two consecutive
   * packets. The IPDV statistic can be useful to inspect clock skew existance
   * and level, e.g: when the IPDV mean values were stable in positive numbers,
   * then the remote clock (used in sending RTP packets) is faster than local
   * system clock. Ideally, the IPDV mean values are always equal to 0.
   *
   * Default: 0 (no).
   */
  #ifndef PJMEDIA_RTCP_STAT_HAS_IPDV
  #   define PJMEDIA_RTCP_STAT_HAS_IPDV		0
  #endif
  
  
  /**
   * Specify whether RTCP XR support should be built into PJMEDIA. Disabling
   * this feature will reduce footprint slightly. Note that even when this 
   * setting is enabled, RTCP XR processing will only be performed in stream 
   * if it is enabled on run-time on per stream basis. See  
   * PJMEDIA_STREAM_ENABLE_XR setting for more info.
   *
   * Default: 0 (no).
   */
  #ifndef PJMEDIA_HAS_RTCP_XR
  #   define PJMEDIA_HAS_RTCP_XR			0
  #endif
  
  
  /**
   * The RTCP XR feature is activated and used by stream if \a enable_rtcp_xr
   * field of \a pjmedia_stream_info structure is non-zero. This setting 
   * controls the default value of this field.
   *
   * Default: 0 (disabled)
   */
  #ifndef PJMEDIA_STREAM_ENABLE_XR
  #   define PJMEDIA_STREAM_ENABLE_XR		0
  #endif
  
  
  /**
   * Specify the buffer length for storing any received RTCP SDES text
   * in a stream session. Usually RTCP contains only the mandatory SDES
   * field, i.e: CNAME.
   * 
   * Default: 64 bytes.
   */
  #ifndef PJMEDIA_RTCP_RX_SDES_BUF_LEN
  #   define PJMEDIA_RTCP_RX_SDES_BUF_LEN		64
  #endif
  
  
  /**
   * Specify the maximum number of RTCP Feedback capability definition.
   * 
   * Default: 16
   */
  #ifndef PJMEDIA_RTCP_FB_MAX_CAP
  #   define PJMEDIA_RTCP_FB_MAX_CAP		16
  #endif
  
  
  /**
   * Specify how long (in miliseconds) the stream should suspend the
   * silence detector/voice activity detector (VAD) during the initial
   * period of the session. This feature is useful to open bindings in
   * all NAT routers between local and remote endpoint since most NATs
   * do not allow incoming packet to get in before local endpoint sends
   * outgoing packets.
   *
   * Specify zero to disable this feature.
   *
   * Default: 600 msec (which gives good probability that some RTP 
   *                    packets will reach the destination, but without
   *                    filling up the jitter buffer on the remote end).
   */
  #ifndef PJMEDIA_STREAM_VAD_SUSPEND_MSEC
  #   define PJMEDIA_STREAM_VAD_SUSPEND_MSEC	600
  #endif
  
  /**
   * Perform RTP payload type checking in the stream. Normally the peer
   * MUST send RTP with payload type as we specified in our SDP. Certain
   * agents may not be able to follow this hence the only way to have
   * communication is to disable this check.
   *
   * Default: 1
   */
  #ifndef PJMEDIA_STREAM_CHECK_RTP_PT
  #   define PJMEDIA_STREAM_CHECK_RTP_PT		1
  #endif
  
  /**
   * Reserve some space for application extra data, e.g: SRTP auth tag,
   * in RTP payload, so the total payload length will not exceed the MTU.
   */
  #ifndef PJMEDIA_STREAM_RESV_PAYLOAD_LEN
  #   define PJMEDIA_STREAM_RESV_PAYLOAD_LEN	20
  #endif
  
  
  /**
   * Specify the maximum duration of silence period in the codec, in msec. 
   * This is useful for example to keep NAT binding open in the firewall
   * and to prevent server from disconnecting the call because no 
   * RTP packet is received.
   *
   * This only applies to codecs that use PJMEDIA's VAD (pretty much
   * everything including iLBC, except Speex, which has its own DTX 
   * mechanism).
   *
   * Use (-1) to disable this feature.
   *
   * Default: 5000 ms
   *
   */
  #ifndef PJMEDIA_CODEC_MAX_SILENCE_PERIOD
  #   define PJMEDIA_CODEC_MAX_SILENCE_PERIOD	5000
  #endif
  
  
  /**
   * Suggested or default threshold to be set for fixed silence detection
   * or as starting threshold for adaptive silence detection. The threshold
   * has the range from zero to 0xFFFF.
   */
  #ifndef PJMEDIA_SILENCE_DET_THRESHOLD
  #   define PJMEDIA_SILENCE_DET_THRESHOLD	4
  #endif
  
  
  /**
   * Maximum silence threshold in the silence detector. The silence detector
   * will not cut the audio transmission if the audio level is above this
   * level.
   *
   * Use 0x10000 (or greater) to disable this feature.
   *
   * Default: 0x10000 (disabled)
   */
  #ifndef PJMEDIA_SILENCE_DET_MAX_THRESHOLD
  #   define PJMEDIA_SILENCE_DET_MAX_THRESHOLD	0x10000
  #endif
  
  
  /**
   * Speex Accoustic Echo Cancellation (AEC).
   * By default is enabled.
   */
  #ifndef PJMEDIA_HAS_SPEEX_AEC
  #   define PJMEDIA_HAS_SPEEX_AEC		1
  #endif
  
  
  /**
   * Specify whether Automatic Gain Control (AGC) should also be enabled in
   * Speex AEC.
   *
   * Default: 1 (yes)
   */
  #ifndef PJMEDIA_SPEEX_AEC_USE_AGC
  #   define PJMEDIA_SPEEX_AEC_USE_AGC		1
  #endif
  
  
  /**
   * Specify whether denoise should also be enabled in Speex AEC.
   *
   * Default: 1 (yes)
   */
  #ifndef PJMEDIA_SPEEX_AEC_USE_DENOISE
  #   define PJMEDIA_SPEEX_AEC_USE_DENOISE	1
  #endif
  
  
  /**
   * WebRtc Accoustic Echo Cancellation (AEC).
   * By default is disabled.
   */
  #ifndef PJMEDIA_HAS_WEBRTC_AEC
  #   define PJMEDIA_HAS_WEBRTC_AEC		0
  #endif
  
  /**
   * Specify whether WebRtc EC should use its mobile version AEC.
   *
   * Default: 0 (no)
   */
  #ifndef PJMEDIA_WEBRTC_AEC_USE_MOBILE
  #   define PJMEDIA_WEBRTC_AEC_USE_MOBILE 	0
  #endif
  
  
  /**
   * Maximum number of parameters in SDP fmtp attribute.
   *
   * Default: 16
   */
  #ifndef PJMEDIA_CODEC_MAX_FMTP_CNT
  #   define PJMEDIA_CODEC_MAX_FMTP_CNT		16
  #endif
  
  
  /**
   * This specifies the behavior of the SDP negotiator when responding to an
   * offer, whether it should rather use the codec preference as set by
   * remote, or should it rather use the codec preference as specified by
   * local endpoint.
   *
   * For example, suppose incoming call has codec order "8 0 3", while 
   * local codec order is "3 0 8". If remote codec order is preferable,
   * the selected codec will be 8, while if local codec order is preferable,
   * the selected codec will be 3.
   *
   * If set to non-zero, the negotiator will use the codec order as specified
   * by remote in the offer.
   *
   * Note that this behavior can be changed during run-time by calling
   * pjmedia_sdp_neg_set_prefer_remote_codec_order().
   *
   * Default is 1 (to maintain backward compatibility)
   */
  #ifndef PJMEDIA_SDP_NEG_PREFER_REMOTE_CODEC_ORDER
  #   define PJMEDIA_SDP_NEG_PREFER_REMOTE_CODEC_ORDER	1
  #endif
  
  /**
   * This specifies the behavior of the SDP negotiator when responding to an
   * offer, whether it should answer with multiple formats or not.
   *
   * Note that this behavior can be changed during run-time by calling
   * pjmedia_sdp_neg_set_allow_multiple_codecs().
   *
   * Default is 0 (to maintain backward compatibility)
   */
  #ifndef PJMEDIA_SDP_NEG_ANSWER_MULTIPLE_CODECS
  #   define PJMEDIA_SDP_NEG_ANSWER_MULTIPLE_CODECS	0
  #endif
  
  
  /**
   * This specifies the maximum number of the customized SDP format
   * negotiation callbacks.
   */
  #ifndef PJMEDIA_SDP_NEG_MAX_CUSTOM_FMT_NEG_CB
  #   define PJMEDIA_SDP_NEG_MAX_CUSTOM_FMT_NEG_CB	8
  #endif
  
  
  /**
   * This specifies if the SDP negotiator should rewrite answer payload
   * type numbers to use the same payload type numbers as the remote offer
   * for all matched codecs.
   *
   * Default is 1 (yes)
   */
  #ifndef PJMEDIA_SDP_NEG_ANSWER_SYMMETRIC_PT
  #   define PJMEDIA_SDP_NEG_ANSWER_SYMMETRIC_PT		1
  #endif
  
  
  /**
   * This specifies if the SDP negotiator should compare its content before 
   * incrementing the origin version on the subsequent offer/answer. 
   * If this is set to 1, origin version will only by incremented if the 
   * new offer/answer is different than the previous one. For backward 
   * compatibility and performance this is set to 0.
   *
   * Default is 0 (No)
   */
  #ifndef PJMEDIA_SDP_NEG_COMPARE_BEFORE_INC_VERSION
  #   define PJMEDIA_SDP_NEG_COMPARE_BEFORE_INC_VERSION	0
  #endif
  
  
  /**
   * Support for sending and decoding RTCP port in SDP (RFC 3605).
   * Default is equal to PJMEDIA_ADVERTISE_RTCP setting.
   */
  #ifndef PJMEDIA_HAS_RTCP_IN_SDP
  #   define PJMEDIA_HAS_RTCP_IN_SDP		(PJMEDIA_ADVERTISE_RTCP)
  #endif
  
  
  /**
   * This macro controls whether pjmedia should include SDP
   * bandwidth modifier "TIAS" (RFC3890).
   *
   * Note that there is also a run-time variable to turn this setting
   * on or off, defined in endpoint.c. To access this variable, use
   * the following construct
   *
   \verbatim
      extern pj_bool_t pjmedia_add_bandwidth_tias_in_sdp;
  
      // Do not enable bandwidth information inclusion in sdp
      pjmedia_add_bandwidth_tias_in_sdp = PJ_FALSE;
   \endverbatim
   *
   * Default: 1 (yes)
   */
  #ifndef PJMEDIA_ADD_BANDWIDTH_TIAS_IN_SDP
  #   define PJMEDIA_ADD_BANDWIDTH_TIAS_IN_SDP	1
  #endif
  
  
  /**
   * This macro controls whether pjmedia should include SDP rtpmap 
   * attribute for static payload types. SDP rtpmap for static
   * payload types are optional, although they are normally included
   * for interoperability reason.
   *
   * Note that there is also a run-time variable to turn this setting
   * on or off, defined in endpoint.c. To access this variable, use
   * the following construct
   *
   \verbatim
      extern pj_bool_t pjmedia_add_rtpmap_for_static_pt;
  
      // Do not include rtpmap for static payload types (<96)
      pjmedia_add_rtpmap_for_static_pt = PJ_FALSE;
   \endverbatim
   *
   * Default: 1 (yes)
   */
  #ifndef PJMEDIA_ADD_RTPMAP_FOR_STATIC_PT
  #   define PJMEDIA_ADD_RTPMAP_FOR_STATIC_PT	1
  #endif
  
  
  /**
   * This macro declares the payload type for telephone-event
   * that is advertised by PJMEDIA for outgoing SDP. If this macro
   * is set to zero, telephone events would not be advertised nor
   * supported.
   *
   * If this value is changed to other number, please update the
   * PJMEDIA_RTP_PT_TELEPHONE_EVENTS_STR too.
   */
  #ifndef PJMEDIA_RTP_PT_TELEPHONE_EVENTS
  #   define PJMEDIA_RTP_PT_TELEPHONE_EVENTS	    96
  #endif
  
  
  /**
   * Macro to get the string representation of the telephone-event
   * payload type.
   */
  #ifndef PJMEDIA_RTP_PT_TELEPHONE_EVENTS_STR
  #   define PJMEDIA_RTP_PT_TELEPHONE_EVENTS_STR	    "96"
  #endif
  
  
  /**
   * Maximum tones/digits that can be enqueued in the tone generator.
   */
  #ifndef PJMEDIA_TONEGEN_MAX_DIGITS
  #   define PJMEDIA_TONEGEN_MAX_DIGITS		    32
  #endif
  
  
  /* 
   * Below specifies the various tone generator backend algorithm.
   */
  
  /** 
   * The math's sine(), floating point. This has very good precision 
   * but it's the slowest and requires floating point support and
   * linking with the math library.
   */
  #define PJMEDIA_TONEGEN_SINE			    1
  
  /**
   * Floating point approximation of sine(). This has relatively good
   * precision and much faster than plain sine(), but it requires floating-
   * point support and linking with the math library.
   */
  #define PJMEDIA_TONEGEN_FLOATING_POINT		    2
  
  /**
   * Fixed point using sine signal generated by Cordic algorithm. This
   * algorithm can be tuned to provide balance between precision and
   * performance by tuning the PJMEDIA_TONEGEN_FIXED_POINT_CORDIC_LOOP 
   * setting, and may be suitable for platforms that lack floating-point
   * support.
   */
  #define PJMEDIA_TONEGEN_FIXED_POINT_CORDIC	    3
  
  /**
   * Fast fixed point using some approximation to generate sine waves.
   * The tone generated by this algorithm is not very precise, however
   * the algorithm is very fast.
   */
  #define PJMEDIA_TONEGEN_FAST_FIXED_POINT	    4
  
  
  /**
   * Specify the tone generator algorithm to be used. Please see 
   * http://trac.pjsip.org/repos/wiki/Tone_Generator for the performance
   * analysis results of the various tone generator algorithms.
   *
   * Default value:
   *  - PJMEDIA_TONEGEN_FLOATING_POINT when PJ_HAS_FLOATING_POINT is set
   *  - PJMEDIA_TONEGEN_FIXED_POINT_CORDIC when PJ_HAS_FLOATING_POINT is not set
   */
  #ifndef PJMEDIA_TONEGEN_ALG
  #   if defined(PJ_HAS_FLOATING_POINT) && PJ_HAS_FLOATING_POINT
  #	define PJMEDIA_TONEGEN_ALG	PJMEDIA_TONEGEN_FLOATING_POINT
  #   else
  #	define PJMEDIA_TONEGEN_ALG	PJMEDIA_TONEGEN_FIXED_POINT_CORDIC
  #   endif
  #endif
  
  
  /**
   * Specify the number of calculation loops to generate the tone, when
   * PJMEDIA_TONEGEN_FIXED_POINT_CORDIC algorithm is used. With more calculation
   * loops, the tone signal gets more precise, but this will add more 
   * processing.
   *
   * Valid values are 1 to 28.
   *
   * Default value: 10
   */
  #ifndef PJMEDIA_TONEGEN_FIXED_POINT_CORDIC_LOOP
  #   define PJMEDIA_TONEGEN_FIXED_POINT_CORDIC_LOOP  10
  #endif
  
  
  /**
   * Enable high quality of tone generation, the better quality will cost
   * more CPU load. This is only applied to floating point enabled machines.
   *
   * By default it is enabled when PJ_HAS_FLOATING_POINT is set.
   *
   * This macro has been deprecated in version 1.0-rc3.
   */
  #ifdef PJMEDIA_USE_HIGH_QUALITY_TONEGEN
  #   error   "The PJMEDIA_USE_HIGH_QUALITY_TONEGEN macro is obsolete"
  #endif
  
  
  /**
   * Fade-in duration for the tone, in milliseconds. Set to zero to disable
   * this feature.
   *
   * Default: 1 (msec)
   */
  #ifndef PJMEDIA_TONEGEN_FADE_IN_TIME
  #   define PJMEDIA_TONEGEN_FADE_IN_TIME		    1
  #endif
  
  
  /**
   * Fade-out duration for the tone, in milliseconds. Set to zero to disable
   * this feature.
   *
   * Default: 2 (msec)
   */
  #ifndef PJMEDIA_TONEGEN_FADE_OUT_TIME
  #   define PJMEDIA_TONEGEN_FADE_OUT_TIME	    2
  #endif
  
  
  /**
   * The default tone generator amplitude (1-32767).
   *
   * Default value: 12288
   */
  #ifndef PJMEDIA_TONEGEN_VOLUME
  #   define PJMEDIA_TONEGEN_VOLUME		    12288
  #endif
  
  
  /**
   * Enable support for SRTP media transport. This will require linking
   * with libsrtp from the third_party directory.
   *
   * By default it is enabled.
   */
  #ifndef PJMEDIA_HAS_SRTP
  #   define PJMEDIA_HAS_SRTP			    1
  #endif
  
  
  /**
   * Enable session description for SRTP keying.
   *
   * By default it is enabled.
   */
  #ifndef PJMEDIA_SRTP_HAS_SDES
  #   define PJMEDIA_SRTP_HAS_SDES		    1
  #endif
  
  
  /**
   * Enable DTLS for SRTP keying.
   *
   * Default value: 0 (disabled)
   */
  #ifndef PJMEDIA_SRTP_HAS_DTLS
  #   define PJMEDIA_SRTP_HAS_DTLS		    0
  #endif
  
  
  /**
   * Set OpenSSL ciphers for DTLS-SRTP.
   *
   * Default value: "DEFAULT"
   */
  #ifndef PJMEDIA_SRTP_DTLS_OSSL_CIPHERS
  #   define PJMEDIA_SRTP_DTLS_OSSL_CIPHERS	    "DEFAULT"
  #endif
  
  
  /**
   * Maximum number of SRTP cryptos.
   *
   * Default: 16
   */
  #ifndef PJMEDIA_SRTP_MAX_CRYPTOS
  #   define PJMEDIA_SRTP_MAX_CRYPTOS		    16
  #endif
  
  
  /**
   * Enable AES_CM_256 cryptos in SRTP.
   * Default: enabled.
   */
  #ifndef PJMEDIA_SRTP_HAS_AES_CM_256
  #   define PJMEDIA_SRTP_HAS_AES_CM_256	    	    1
  #endif
  
  
  /**
   * Enable AES_CM_192 cryptos in SRTP.
   * It was reported that this crypto only works among libsrtp backends,
   * so we recommend to disable this.
   *
   * To enable this, you would require OpenSSL which supports it.
   * See https://trac.pjsip.org/repos/ticket/1943 for more info.
   *
   * Default: disabled.
   */
  #ifndef PJMEDIA_SRTP_HAS_AES_CM_192
  #   define PJMEDIA_SRTP_HAS_AES_CM_192	    	    0
  #endif
  
  
  /**
   * Enable AES_CM_128 cryptos in SRTP.
   * Default: enabled.
   */
  #ifndef PJMEDIA_SRTP_HAS_AES_CM_128
  #   define PJMEDIA_SRTP_HAS_AES_CM_128    	    1
  #endif
  
  
  /**
   * Enable AES_GCM_256 cryptos in SRTP.
   *
   * To enable this, you would require OpenSSL which supports it.
   * See https://trac.pjsip.org/repos/ticket/1943 for more info. 
   *
   * Default: disabled.
   */
  #ifndef PJMEDIA_SRTP_HAS_AES_GCM_256
  #   define PJMEDIA_SRTP_HAS_AES_GCM_256	    	    0
  #endif
  
  
  /**
   * Enable AES_GCM_128 cryptos in SRTP.
   *
   * To enable this, you would require OpenSSL which supports it.
   * See https://trac.pjsip.org/repos/ticket/1943 for more info.
   *
   * Default: disabled.
   */
  #ifndef PJMEDIA_SRTP_HAS_AES_GCM_128
  #   define PJMEDIA_SRTP_HAS_AES_GCM_128    	    0
  #endif
  
  
  /**
   * Let the library handle libsrtp initialization and deinitialization.
   * Application may want to disable this and manually perform libsrtp
   * initialization and deinitialization when it needs to use libsrtp
   * before the library is initialized or after the library is shutdown.
   *
   * By default it is enabled.
   */
  #ifndef PJMEDIA_LIBSRTP_AUTO_INIT_DEINIT
  #   define PJMEDIA_LIBSRTP_AUTO_INIT_DEINIT	    1
  #endif
  
  
  /**
   * Enable support to handle codecs with inconsistent clock rate
   * between clock rate in SDP/RTP & the clock rate that is actually used.
   * This happens for example with G.722 and MPEG audio codecs.
   * See:
   *  - G.722      : RFC 3551 4.5.2
   *  - MPEG audio : RFC 3551 4.5.13 & RFC 3119
   *  - OPUS	 : RFC 7587
   *
   * Also when this feature is enabled, some handling will be performed
   * to deal with clock rate incompatibilities of some phones.
   *
   * By default it is enabled.
   */
  #ifndef PJMEDIA_HANDLE_G722_MPEG_BUG
  #   define PJMEDIA_HANDLE_G722_MPEG_BUG		    1
  #endif
  
  
  /* Setting to determine if media transport should switch RTP and RTCP
   * remote address to the source address of the packets it receives.
   *
   * By default it is enabled.
   */
  #ifndef PJMEDIA_TRANSPORT_SWITCH_REMOTE_ADDR
  #   define PJMEDIA_TRANSPORT_SWITCH_REMOTE_ADDR	    1
  #endif
  
  
  /**
   * Transport info (pjmedia_transport_info) contains a socket info and list
   * of transport specific info, since transports can be chained together 
   * (for example, SRTP transport uses UDP transport as the underlying 
   * transport). This constant specifies maximum number of transport specific
   * infos that can be held in a transport info.
   */
  #ifndef PJMEDIA_TRANSPORT_SPECIFIC_INFO_MAXCNT
  #   define PJMEDIA_TRANSPORT_SPECIFIC_INFO_MAXCNT   4
  #endif
  
  
  /**
   * Maximum size in bytes of storage buffer of a transport specific info.
   */
  #ifndef PJMEDIA_TRANSPORT_SPECIFIC_INFO_MAXSIZE
  #   define PJMEDIA_TRANSPORT_SPECIFIC_INFO_MAXSIZE  (36*sizeof(long))
  #endif
  
  
  /**
   * Value to be specified in PJMEDIA_STREAM_ENABLE_KA setting.
   * This indicates that an empty RTP packet should be used as
   * the keep-alive packet.
   */
  #define PJMEDIA_STREAM_KA_EMPTY_RTP		    1
  
  /**
   * Value to be specified in PJMEDIA_STREAM_ENABLE_KA setting.
   * This indicates that a user defined packet should be used
   * as the keep-alive packet. The content of the user-defined
   * packet is specified by PJMEDIA_STREAM_KA_USER_PKT. Default
   * content is a CR-LF packet.
   */
  #define PJMEDIA_STREAM_KA_USER			    2
  
  /**
   * The content of the user defined keep-alive packet. The format
   * of the packet is initializer to pj_str_t structure. Note that
   * the content may contain NULL character.
   */
  #ifndef PJMEDIA_STREAM_KA_USER_PKT
  #   define PJMEDIA_STREAM_KA_USER_PKT	{ "\r\n", 2 }
  #endif
  
  /**
   * Specify another type of keep-alive and NAT hole punching 
   * mechanism (the other type is PJMEDIA_STREAM_VAD_SUSPEND_MSEC
   * and PJMEDIA_CODEC_MAX_SILENCE_PERIOD) to be used by stream. 
   * When this feature is enabled, the stream will initially 
   * transmit one packet to punch a hole in NAT, and periodically
   * transmit keep-alive packets.
   *
   * When this alternative keep-alive mechanism is used, application
   * may disable the other keep-alive mechanisms, i.e: by setting 
   * PJMEDIA_STREAM_VAD_SUSPEND_MSEC to zero and 
   * PJMEDIA_CODEC_MAX_SILENCE_PERIOD to -1.
   *
   * The value of this macro specifies the type of packet used
   * for the keep-alive mechanism. Valid values are
   * PJMEDIA_STREAM_KA_EMPTY_RTP and PJMEDIA_STREAM_KA_USER.
   * 
   * The duration of the keep-alive interval further can be set
   * with PJMEDIA_STREAM_KA_INTERVAL setting.
   *
   * Default: 0 (disabled)
   */
  #ifndef PJMEDIA_STREAM_ENABLE_KA
  #   define PJMEDIA_STREAM_ENABLE_KA		    0
  #endif
  
  
  /**
   * Specify the keep-alive interval of PJMEDIA_STREAM_ENABLE_KA
   * mechanism, in seconds.
   *
   * Default: 5 seconds
   */
  #ifndef PJMEDIA_STREAM_KA_INTERVAL
  #   define PJMEDIA_STREAM_KA_INTERVAL		    5
  #endif
  
  
  /**
   * Specify the number of identical consecutive error that will be ignored when 
   * receiving RTP/RTCP data before the library tries to restart the transport.
   *
   * When receiving RTP/RTCP data, the library will ignore error besides 
   * PJ_EPENDING or PJ_ECANCELLED and continue the loop to receive the data. 
   * If the OS always return error, then the loop will continue non stop.
   * This setting will limit the number of the identical consecutive error, 
   * before the library start to restart the transport. If error still happens
   * after transport restart, then PJMEDIA_EVENT_MEDIA_TP_ERR event will be 
   * publish as a notification.
   *
   * If PJ_ESOCKETSTOP is raised, then transport will be restarted regardless
   * of this setting.
   * 
   * To always ignore the error when receving RTP/RTCP, set this to 0.
   *
   * Default : 20
   */
  #ifndef PJMEDIA_IGNORE_RECV_ERR_CNT
  #   define PJMEDIA_IGNORE_RECV_ERR_CNT		20
  #endif
  
  
  /*
   * .... new stuffs ...
   */
  
  /*
   * Video
   */
  
  /**
   * Top level option to enable/disable video features.
   *
   * Default: 0 (disabled)
   */
  #ifndef PJMEDIA_HAS_VIDEO
  #   define PJMEDIA_HAS_VIDEO				0
  #endif
  
  
  /**
   * Specify if FFMPEG is available. The value here will be used as the default
   * value for other FFMPEG settings below.
   *
   * Default: 0
   */
  #ifndef PJMEDIA_HAS_FFMPEG
  #   define PJMEDIA_HAS_FFMPEG				0
  #endif
  
  /**
   * Specify if FFMPEG libavformat is available.
   *
   * Default: PJMEDIA_HAS_FFMPEG (or detected by configure)
   */
  #ifndef PJMEDIA_HAS_LIBAVFORMAT
  #   define PJMEDIA_HAS_LIBAVFORMAT			PJMEDIA_HAS_FFMPEG
  #endif
  
  /**
   * Specify if FFMPEG libavformat is available.
   *
   * Default: PJMEDIA_HAS_FFMPEG (or detected by configure)
   */
  #ifndef PJMEDIA_HAS_LIBAVCODEC
  #   define PJMEDIA_HAS_LIBAVCODEC			PJMEDIA_HAS_FFMPEG
  #endif
  
  /**
   * Specify if FFMPEG libavutil is available.
   *
   * Default: PJMEDIA_HAS_FFMPEG (or detected by configure)
   */
  #ifndef PJMEDIA_HAS_LIBAVUTIL
  #   define PJMEDIA_HAS_LIBAVUTIL			PJMEDIA_HAS_FFMPEG
  #endif
  
  /**
   * Specify if FFMPEG libswscale is available.
   *
   * Default: PJMEDIA_HAS_FFMPEG (or detected by configure)
   */
  #ifndef PJMEDIA_HAS_LIBSWSCALE
  #   define PJMEDIA_HAS_LIBSWSCALE			PJMEDIA_HAS_FFMPEG
  #endif
  
  /**
   * Specify if FFMPEG libavdevice is available.
   *
   * Default: PJMEDIA_HAS_FFMPEG (or detected by configure)
   */
  #ifndef PJMEDIA_HAS_LIBAVDEVICE
  #   define PJMEDIA_HAS_LIBAVDEVICE			PJMEDIA_HAS_FFMPEG
  #endif
  
  /**
   * Maximum video planes.
   *
   * Default: 4
   */
  #ifndef PJMEDIA_MAX_VIDEO_PLANES
  #   define PJMEDIA_MAX_VIDEO_PLANES			4
  #endif
  
  /**
   * Maximum number of video formats.
   *
   * Default: 32
   */
  #ifndef PJMEDIA_MAX_VIDEO_FORMATS
  #   define PJMEDIA_MAX_VIDEO_FORMATS			32
  #endif
  
  /**
   * Specify the maximum time difference (in ms) for synchronization between
   * two medias. If the synchronization media source is ahead of time
   * greater than this duration, it is considered to make a very large jump
   * and the synchronization will be reset.
   *
   * Default: 20000
   */
  #ifndef PJMEDIA_CLOCK_SYNC_MAX_SYNC_MSEC
  #   define PJMEDIA_CLOCK_SYNC_MAX_SYNC_MSEC         20000
  #endif
  
  /**
   * Maximum video frame size.
   * Default: 128kB
   */
  #ifndef PJMEDIA_MAX_VIDEO_ENC_FRAME_SIZE
  #  define PJMEDIA_MAX_VIDEO_ENC_FRAME_SIZE	    (1<<17)
  #endif
  
  
  /**
   * Specify the maximum duration (in ms) for resynchronization. When a media
   * is late to another media it is supposed to be synchronized to, it is
   * guaranteed to be synchronized again after this duration. While if the
   * media is ahead/early by t ms, it is guaranteed to be synchronized after
   * t + this duration. This timing only applies if there is no additional
   * resynchronization required during the specified duration.
   *
   * Default: 2000
   */
  #ifndef PJMEDIA_CLOCK_SYNC_MAX_RESYNC_DURATION
  #   define PJMEDIA_CLOCK_SYNC_MAX_RESYNC_DURATION 2000
  #endif
  
  
  /**
   * Minimum gap between two consecutive discards in jitter buffer,
   * in milliseconds.
   *
   * Default: 200 ms
   */
  #ifndef PJMEDIA_JBUF_DISC_MIN_GAP
  #   define PJMEDIA_JBUF_DISC_MIN_GAP		    200
  #endif
  
  
  /**
   * Minimum burst level reference used for calculating discard duration
   * in jitter buffer progressive discard algorithm, in frames.
   * 
   * Default: 1 frame
   */
  #ifndef PJMEDIA_JBUF_PRO_DISC_MIN_BURST
  #   define PJMEDIA_JBUF_PRO_DISC_MIN_BURST	    1
  #endif
  
  
  /**
   * Maximum burst level reference used for calculating discard duration
   * in jitter buffer progressive discard algorithm, in frames.
   * 
   * Default: 200 frames
   */
  #ifndef PJMEDIA_JBUF_PRO_DISC_MAX_BURST
  #   define PJMEDIA_JBUF_PRO_DISC_MAX_BURST	    100
  #endif
  
  
  /**
   * Duration for progressive discard algotithm in jitter buffer to discard
   * an excessive frame when burst is equal to or lower than
   * PJMEDIA_JBUF_PRO_DISC_MIN_BURST, in milliseconds.
   *
   * Default: 2000 ms
   */
  #ifndef PJMEDIA_JBUF_PRO_DISC_T1
  #   define PJMEDIA_JBUF_PRO_DISC_T1		    2000
  #endif
  
  
  /**
   * Duration for progressive discard algotithm in jitter buffer to discard
   * an excessive frame when burst is equal to or greater than
   * PJMEDIA_JBUF_PRO_DISC_MAX_BURST, in milliseconds.
   *
   * Default: 10000 ms
   */
  #ifndef PJMEDIA_JBUF_PRO_DISC_T2
  #   define PJMEDIA_JBUF_PRO_DISC_T2		    10000
  #endif
  
  
  /**
   * Video stream will discard old picture from the jitter buffer as soon as
   * new picture is received, to reduce latency.
   *
   * Default: 0
   */
  #ifndef PJMEDIA_VID_STREAM_SKIP_PACKETS_TO_REDUCE_LATENCY
  #   define PJMEDIA_VID_STREAM_SKIP_PACKETS_TO_REDUCE_LATENCY	0
  #endif
  
  
  /**
   * Maximum video payload size. Note that this must not be greater than
   * PJMEDIA_MAX_MTU.
   *
   * Default: (PJMEDIA_MAX_MTU - 20 - (128+16)) if SRTP is enabled, 
   *	    otherwise (PJMEDIA_MAX_MTU - 20). 
   *          Note that (128+16) constant value is taken from libSRTP macro 
   *          SRTP_MAX_TRAILER_LEN.
   */
  #ifndef PJMEDIA_MAX_VID_PAYLOAD_SIZE
  #  if PJMEDIA_HAS_SRTP
  #     define PJMEDIA_MAX_VID_PAYLOAD_SIZE     (PJMEDIA_MAX_MTU - 20 - (128+16))
  #  else
  #     define PJMEDIA_MAX_VID_PAYLOAD_SIZE     (PJMEDIA_MAX_MTU - 20)
  #  endif
  #endif
  
  
  /**
   * Specify target value for socket receive buffer size. It will be
   * applied to RTP socket of media transport using setsockopt(). When
   * transport failed to set the specified size, it will try with lower
   * value until the highest possible is successfully set.
   *
   * Setting this to zero will leave the socket receive buffer size to
   * OS default (e.g: usually 8 KB on desktop platforms).
   *
   * Default: 64 KB when video is enabled, otherwise zero (OS default)
   */
  #ifndef PJMEDIA_TRANSPORT_SO_RCVBUF_SIZE
  #   if PJMEDIA_HAS_VIDEO
  #	define PJMEDIA_TRANSPORT_SO_RCVBUF_SIZE	(64*1024)
  #   else
  #	define PJMEDIA_TRANSPORT_SO_RCVBUF_SIZE	0
  #   endif
  #endif
  
  
  /**
   * Specify target value for socket send buffer size. It will be
   * applied to RTP socket of media transport using setsockopt(). When
   * transport failed to set the specified size, it will try with lower
   * value until the highest possible is successfully set.
   *
   * Setting this to zero will leave the socket send buffer size to
   * OS default (e.g: usually 8 KB on desktop platforms).
   *
   * Default: 64 KB when video is enabled, otherwise zero (OS default)
   */
  #ifndef PJMEDIA_TRANSPORT_SO_SNDBUF_SIZE
  #   if PJMEDIA_HAS_VIDEO
  #	define PJMEDIA_TRANSPORT_SO_SNDBUF_SIZE	(64*1024)
  #   else
  #	define PJMEDIA_TRANSPORT_SO_SNDBUF_SIZE	0
  #   endif
  #endif
  
  
  /**
   * Specify if libyuv is available.
   *
   * Default: 0 (disable)
   */
  #ifndef PJMEDIA_HAS_LIBYUV
  #   define PJMEDIA_HAS_LIBYUV				0
  #endif
  
  
  /**
   * Specify if dtmf flash in RFC 2833 is available.
   */
  #ifndef PJMEDIA_HAS_DTMF_FLASH
  #   define PJMEDIA_HAS_DTMF_FLASH			1
  #endif
  
  
  /**
   * Specify the number of keyframe needed to be sent after the stream is 
   * created. Setting this to 0 will disable it.
   *
   * Default : 5
   */
  #ifndef PJMEDIA_VID_STREAM_START_KEYFRAME_CNT
  #   define PJMEDIA_VID_STREAM_START_KEYFRAME_CNT	5
  #endif
  
  
  /**
   * Specify the interval to send keyframe after the stream is created, in msec.
   *
   * Default : 1000
   */
  #ifndef PJMEDIA_VID_STREAM_START_KEYFRAME_INTERVAL_MSEC
  #   define PJMEDIA_VID_STREAM_START_KEYFRAME_INTERVAL_MSEC  1000
  #endif
  
  
  /**
   * Specify minimum delay of video decoding, in milliseconds. Lower value may
   * degrade video quality significantly in a bad network environment (e.g:
   * with persistent late and out-of-order RTP packets). Note that the value
   * must be lower than jitter buffer maximum delay (configurable via
   * pjmedia_stream_info.jb_max or pjsua_media_config.jb_max).
   *
   * Default : 100
   */
  #ifndef PJMEDIA_VID_STREAM_DECODE_MIN_DELAY_MSEC
  #   define PJMEDIA_VID_STREAM_DECODE_MIN_DELAY_MSEC	    100
  #endif
  
  
  
  /**
   * @}
   */
  
  
  #endif	/* __PJMEDIA_CONFIG_H__ */
  
  
  