diff --git a/jni/pjproject-android/.svn/pristine/2b/2bddd83a96a4908cc8c20571c83292561af64187.svn-base b/jni/pjproject-android/.svn/pristine/2b/2bddd83a96a4908cc8c20571c83292561af64187.svn-base
new file mode 100644
index 0000000..9eec63e
--- /dev/null
+++ b/jni/pjproject-android/.svn/pristine/2b/2bddd83a96a4908cc8c20571c83292561af64187.svn-base
@@ -0,0 +1,1279 @@
+/* $Id$ */
+/* 
+ * 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 <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 <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
+
+
+/*
+ * 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 experimental 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.
+ */
+#ifndef PJMEDIA_DTMF_DURATION		
+#  define PJMEDIA_DTMF_DURATION			1600	/* in timestamp */
+#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 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
+
+
+/**
+ * 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
+
+
+/**
+ * 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 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
+ *
+ * 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
+
+
+/**
+ * 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
+
+
+/*
+ * .... 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
+
+/**
+ * Specify if FFMPEG libavcore is available.
+ *
+ * Default: PJMEDIA_HAS_FFMPEG (or detected by configure)
+ */
+#ifndef PJMEDIA_HAS_LIBAVCORE
+#   define PJMEDIA_HAS_LIBAVCORE			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 - 100)
+ */
+#ifndef PJMEDIA_MAX_VID_PAYLOAD_SIZE			
+#  define PJMEDIA_MAX_VID_PAYLOAD_SIZE		(PJMEDIA_MAX_MTU - 100)
+#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
+
+
+/**
+ * @}
+ */
+
+
+#endif	/* __PJMEDIA_CONFIG_H__ */
+
+