/* $Id$ */ /* * Copyright (C) 2008-2011 Teluu Inc. (http://www.teluu.com) * Copyright (C) 2003-2008 Benny Prijono * * 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 /** * @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 #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 regular RTCP packets, in msec. */ #ifndef PJMEDIA_RTCP_INTERVAL # define PJMEDIA_RTCP_INTERVAL 5000 /* msec*/ #endif /** * Minimum interval between two consecutive outgoing RTCP-FB packets, * such as Picture Loss Indication, in msec. */ #ifndef PJMEDIA_RTCP_FB_INTERVAL # define PJMEDIA_RTCP_FB_INTERVAL 50 /* 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 start 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. */ #ifndef PJMEDIA_RTP_PT_TELEPHONE_EVENTS # define PJMEDIA_RTP_PT_TELEPHONE_EVENTS 120 #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 the minimum interval to send video keyframe, in msec. * * Default : 1000 */ #ifndef PJMEDIA_VID_STREAM_MIN_KEYFRAME_INTERVAL_MSEC # define PJMEDIA_VID_STREAM_MIN_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__ */