diff options
author | David M. Lee <dlee@digium.com> | 2013-01-07 14:24:28 -0600 |
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committer | David M. Lee <dlee@digium.com> | 2013-01-07 14:24:28 -0600 |
commit | f3ab456a17af1c89a6e3be4d20c5944853df1cb0 (patch) | |
tree | d00e1a332cd038a6d906a1ea0ac91e1a4458e617 /pjmedia/include/pjmedia/config.h |
Import pjproject-2.0.1
Diffstat (limited to 'pjmedia/include/pjmedia/config.h')
-rw-r--r-- | pjmedia/include/pjmedia/config.h | 1181 |
1 files changed, 1181 insertions, 0 deletions
diff --git a/pjmedia/include/pjmedia/config.h b/pjmedia/include/pjmedia/config.h new file mode 100644 index 0000000..d5a598f --- /dev/null +++ b/pjmedia/include/pjmedia/config.h @@ -0,0 +1,1181 @@ +/* $Id: config.h 4130 2012-05-17 08:35:51Z nanang $ */ +/* + * 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 + +/* + * 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 to support. + */ +#ifndef PJMEDIA_MAX_MTU +# define PJMEDIA_MAX_MTU 1500 +#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 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 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 + + +/** + * @} + */ + + +#endif /* __PJMEDIA_CONFIG_H__ */ + + |