| /* $Id$ */ |
| /* |
| * 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__ */ |
| |
| |