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Compile time configuration

Some compile time configuration settings. More...

Macros

#define PJMEDIA_CONF_USE_SWITCH_BOARD   0
 
#define PJMEDIA_CONF_SWITCH_BOARD_BUF_SIZE   PJMEDIA_MAX_MTU
 
#define PJMEDIA_HAS_LEGACY_SOUND_API   1
 
#define PJMEDIA_SND_DEFAULT_REC_LATENCY   100
 
#define PJMEDIA_SND_DEFAULT_PLAY_LATENCY   140
 
#define PJMEDIA_WSOLA_IMP_NULL   0
 
#define PJMEDIA_WSOLA_IMP_WSOLA   1
 
#define PJMEDIA_WSOLA_IMP_WSOLA_LITE   2
 
#define PJMEDIA_WSOLA_IMP   PJMEDIA_WSOLA_IMP_WSOLA
 
#define PJMEDIA_WSOLA_MAX_EXPAND_MSEC   80
 
#define PJMEDIA_WSOLA_TEMPLATE_LENGTH_MSEC   5
 
#define PJMEDIA_WSOLA_DELAY_MSEC   5
 
#define PJMEDIA_WSOLA_PLC_NO_FADING   0
 
#define PJMEDIA_MAX_PLC_DURATION_MSEC   240
 
#define PJMEDIA_SOUND_BUFFER_COUNT   ((PJMEDIA_SND_DEFAULT_PLAY_LATENCY+20)/20)
 
#define PJMEDIA_HAS_ALAW_ULAW_TABLE   1
 
#define PJMEDIA_HAS_G711_CODEC   1
 
#define PJMEDIA_RESAMPLE_NONE   1
 
#define PJMEDIA_RESAMPLE_LIBRESAMPLE   2
 
#define PJMEDIA_RESAMPLE_SPEEX   3
 
#define PJMEDIA_RESAMPLE_LIBSAMPLERATE   4
 
#define PJMEDIA_RESAMPLE_IMP   PJMEDIA_RESAMPLE_LIBRESAMPLE
 
#define PJMEDIA_FILE_PORT_BUFSIZE   4000
 
#define PJMEDIA_MAX_FRAME_DURATION_MS   200
 
#define PJMEDIA_MAX_MTU   1500
 
#define PJMEDIA_MAX_MRU   2000
 
#define PJMEDIA_DTMF_DURATION   1600 /* in timestamp */
 
#define PJMEDIA_RTP_NAT_PROBATION_CNT   10
 
#define PJMEDIA_RTCP_NAT_PROBATION_CNT   3
 
#define PJMEDIA_ADVERTISE_RTCP   1
 
#define PJMEDIA_RTCP_INTERVAL   5000 /* msec*/
 
#define PJMEDIA_RTCP_IGNORE_FIRST_PACKETS   25
 
#define PJMEDIA_RTCP_STAT_HAS_RAW_JITTER   0
 
#define PJMEDIA_RTCP_NORMALIZE_FACTOR   3
 
#define PJMEDIA_RTCP_STAT_HAS_IPDV   0
 
#define PJMEDIA_HAS_RTCP_XR   0
 
#define PJMEDIA_STREAM_ENABLE_XR   0
 
#define PJMEDIA_RTCP_RX_SDES_BUF_LEN   64
 
#define PJMEDIA_STREAM_VAD_SUSPEND_MSEC   600
 
#define PJMEDIA_STREAM_CHECK_RTP_PT   1
 
#define PJMEDIA_STREAM_RESV_PAYLOAD_LEN   20
 
#define PJMEDIA_CODEC_MAX_SILENCE_PERIOD   5000
 
#define PJMEDIA_SILENCE_DET_THRESHOLD   4
 
#define PJMEDIA_SILENCE_DET_MAX_THRESHOLD   0x10000
 
#define PJMEDIA_HAS_SPEEX_AEC   1
 
#define PJMEDIA_SPEEX_AEC_USE_AGC   1
 
#define PJMEDIA_SPEEX_AEC_USE_DENOISE   1
 
#define PJMEDIA_HAS_WEBRTC_AEC   0
 
#define PJMEDIA_WEBRTC_AEC_USE_MOBILE   0
 
#define PJMEDIA_CODEC_MAX_FMTP_CNT   16
 
#define PJMEDIA_SDP_NEG_PREFER_REMOTE_CODEC_ORDER   1
 
#define PJMEDIA_SDP_NEG_ANSWER_MULTIPLE_CODECS   0
 
#define PJMEDIA_SDP_NEG_MAX_CUSTOM_FMT_NEG_CB   8
 
#define PJMEDIA_SDP_NEG_ANSWER_SYMMETRIC_PT   1
 
#define PJMEDIA_HAS_RTCP_IN_SDP   (PJMEDIA_ADVERTISE_RTCP)
 
#define PJMEDIA_ADD_BANDWIDTH_TIAS_IN_SDP   1
 
#define PJMEDIA_ADD_RTPMAP_FOR_STATIC_PT   1
 
#define PJMEDIA_RTP_PT_TELEPHONE_EVENTS   96
 
#define PJMEDIA_RTP_PT_TELEPHONE_EVENTS_STR   "96"
 
#define PJMEDIA_TONEGEN_MAX_DIGITS   32
 
#define PJMEDIA_TONEGEN_SINE   1
 
#define PJMEDIA_TONEGEN_FLOATING_POINT   2
 
#define PJMEDIA_TONEGEN_FIXED_POINT_CORDIC   3
 
#define PJMEDIA_TONEGEN_FAST_FIXED_POINT   4
 
#define PJMEDIA_TONEGEN_ALG   PJMEDIA_TONEGEN_FIXED_POINT_CORDIC
 
#define PJMEDIA_TONEGEN_FIXED_POINT_CORDIC_LOOP   10
 
#define PJMEDIA_TONEGEN_FADE_IN_TIME   1
 
#define PJMEDIA_TONEGEN_FADE_OUT_TIME   2
 
#define PJMEDIA_TONEGEN_VOLUME   12288
 
#define PJMEDIA_SRTP_MAX_CRYPTOS   16
 
#define PJMEDIA_SRTP_HAS_AES_CM_256   1
 
#define PJMEDIA_SRTP_HAS_AES_CM_192   0
 
#define PJMEDIA_SRTP_HAS_AES_CM_128   1
 
#define PJMEDIA_SRTP_HAS_AES_GCM_256   0
 
#define PJMEDIA_SRTP_HAS_AES_GCM_128   0
 
#define PJMEDIA_LIBSRTP_AUTO_INIT_DEINIT   1
 
#define PJMEDIA_HANDLE_G722_MPEG_BUG   1
 
#define PJMEDIA_TRANSPORT_SPECIFIC_INFO_MAXCNT   4
 
#define PJMEDIA_TRANSPORT_SPECIFIC_INFO_MAXSIZE   (36*sizeof(long))
 
#define PJMEDIA_STREAM_KA_EMPTY_RTP   1
 
#define PJMEDIA_STREAM_KA_USER   2
 
#define PJMEDIA_STREAM_KA_USER_PKT   { "\r\n", 2 }
 
#define PJMEDIA_STREAM_KA_INTERVAL   5
 
#define PJMEDIA_HAS_VIDEO   0
 
#define PJMEDIA_HAS_FFMPEG   0
 
#define PJMEDIA_HAS_LIBAVFORMAT   PJMEDIA_HAS_FFMPEG
 
#define PJMEDIA_HAS_LIBAVCODEC   PJMEDIA_HAS_FFMPEG
 
#define PJMEDIA_HAS_LIBAVUTIL   PJMEDIA_HAS_FFMPEG
 
#define PJMEDIA_HAS_LIBSWSCALE   PJMEDIA_HAS_FFMPEG
 
#define PJMEDIA_HAS_LIBAVDEVICE   PJMEDIA_HAS_FFMPEG
 
#define PJMEDIA_MAX_VIDEO_PLANES   4
 
#define PJMEDIA_MAX_VIDEO_FORMATS   32
 
#define PJMEDIA_CLOCK_SYNC_MAX_SYNC_MSEC   20000
 
#define PJMEDIA_MAX_VIDEO_ENC_FRAME_SIZE   (1<<17)
 
#define PJMEDIA_CLOCK_SYNC_MAX_RESYNC_DURATION   2000
 
#define PJMEDIA_JBUF_DISC_MIN_GAP   200
 
#define PJMEDIA_JBUF_PRO_DISC_MIN_BURST   1
 
#define PJMEDIA_JBUF_PRO_DISC_MAX_BURST   100
 
#define PJMEDIA_JBUF_PRO_DISC_T1   2000
 
#define PJMEDIA_JBUF_PRO_DISC_T2   10000
 
#define PJMEDIA_VID_STREAM_SKIP_PACKETS_TO_REDUCE_LATENCY   0
 
#define PJMEDIA_MAX_VID_PAYLOAD_SIZE   (PJMEDIA_MAX_MTU - 100)
 
#define PJMEDIA_TRANSPORT_SO_RCVBUF_SIZE   0
 
#define PJMEDIA_TRANSPORT_SO_SNDBUF_SIZE   0
 
#define PJMEDIA_HAS_LIBYUV   0
 
#define PJMEDIA_HAS_DTMF_FLASH   1
 
#define PJMEDIA_VID_STREAM_START_KEYFRAME_CNT   5
 
#define PJMEDIA_VID_STREAM_START_KEYFRAME_INTERVAL_MSEC   1000
 

Detailed Description

Macro Definition Documentation

#define PJMEDIA_CONF_USE_SWITCH_BOARD   0

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

#define PJMEDIA_CONF_SWITCH_BOARD_BUF_SIZE   PJMEDIA_MAX_MTU

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

#define PJMEDIA_HAS_LEGACY_SOUND_API   1

This macro has been deprecated in releasee 1.1. Please see http://trac.pjsip.org/repos/wiki/Audio_Dev_API for more information. This macro has been deprecated in releasee 1.1. Please see http://trac.pjsip.org/repos/wiki/Audio_Dev_API for more information. 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.

#define PJMEDIA_SND_DEFAULT_REC_LATENCY   100

Specify default sound device latency, in milisecond.

#define PJMEDIA_SND_DEFAULT_PLAY_LATENCY   140

Specify default sound device latency, in milisecond.

Default is 160ms for Windows Mobile and 140ms for other platforms.

#define PJMEDIA_WSOLA_IMP_NULL   0

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_WSOLA   1

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_LITE   2

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   PJMEDIA_WSOLA_IMP_WSOLA

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

#define PJMEDIA_WSOLA_MAX_EXPAND_MSEC   80

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

#define PJMEDIA_WSOLA_TEMPLATE_LENGTH_MSEC   5

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

#define PJMEDIA_WSOLA_DELAY_MSEC   5

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

#define PJMEDIA_WSOLA_PLC_NO_FADING   0

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

#define PJMEDIA_MAX_PLC_DURATION_MSEC   240

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

#define PJMEDIA_SOUND_BUFFER_COUNT   ((PJMEDIA_SND_DEFAULT_PLAY_LATENCY+20)/20)

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

#define PJMEDIA_HAS_ALAW_ULAW_TABLE   1

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.

#define PJMEDIA_HAS_G711_CODEC   1

Unless specified otherwise, G711 codec is included by default.

#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)

#define PJMEDIA_RESAMPLE_IMP   PJMEDIA_RESAMPLE_LIBRESAMPLE

Select which resample implementation to use. Currently pjmedia supports:

Default is PJMEDIA_RESAMPLE_LIBRESAMPLE

#define PJMEDIA_FILE_PORT_BUFSIZE   4000

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.

#define PJMEDIA_MAX_FRAME_DURATION_MS   200

Maximum frame duration (in msec) to be supported. This (among other thing) will affect the size of buffers to be allocated for outgoing packets.

#define PJMEDIA_MAX_MTU   1500

Max packet size for transmitting direction.

#define PJMEDIA_MAX_MRU   2000

Max packet size for receiving direction.

#define PJMEDIA_DTMF_DURATION   1600 /* in timestamp */

DTMF/telephone-event duration, in timestamp.

#define PJMEDIA_RTP_NAT_PROBATION_CNT   10

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.

#define PJMEDIA_RTCP_NAT_PROBATION_CNT   3

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.

#define PJMEDIA_ADVERTISE_RTCP   1

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)

#define PJMEDIA_RTCP_INTERVAL   5000 /* msec*/

Interval to send RTCP packets, in msec

#define PJMEDIA_RTCP_IGNORE_FIRST_PACKETS   25

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.

#define PJMEDIA_RTCP_STAT_HAS_RAW_JITTER   0

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).

#define PJMEDIA_RTCP_NORMALIZE_FACTOR   3

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.

#define PJMEDIA_RTCP_STAT_HAS_IPDV   0

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).

#define PJMEDIA_HAS_RTCP_XR   0

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).

#define PJMEDIA_STREAM_ENABLE_XR   0

The RTCP XR feature is activated and used by stream if enable_rtcp_xr field of pjmedia_stream_info structure is non-zero. This setting controls the default value of this field.

Default: 0 (disabled)

#define PJMEDIA_RTCP_RX_SDES_BUF_LEN   64

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.

#define PJMEDIA_STREAM_VAD_SUSPEND_MSEC   600

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).

#define PJMEDIA_STREAM_CHECK_RTP_PT   1

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

#define PJMEDIA_STREAM_RESV_PAYLOAD_LEN   20

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.

#define PJMEDIA_CODEC_MAX_SILENCE_PERIOD   5000

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

#define PJMEDIA_SILENCE_DET_THRESHOLD   4

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.

#define PJMEDIA_SILENCE_DET_MAX_THRESHOLD   0x10000

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)

#define PJMEDIA_HAS_SPEEX_AEC   1

Speex Accoustic Echo Cancellation (AEC). By default is enabled.

#define PJMEDIA_SPEEX_AEC_USE_AGC   1

Specify whether Automatic Gain Control (AGC) should also be enabled in Speex AEC.

Default: 1 (yes)

#define PJMEDIA_SPEEX_AEC_USE_DENOISE   1

Specify whether denoise should also be enabled in Speex AEC.

Default: 1 (yes)

#define PJMEDIA_HAS_WEBRTC_AEC   0

WebRtc Accoustic Echo Cancellation (AEC). By default is disabled.

#define PJMEDIA_WEBRTC_AEC_USE_MOBILE   0

Specify whether WebRtc EC should use its mobile version AEC.

Default: 0 (no)

#define PJMEDIA_CODEC_MAX_FMTP_CNT   16

Maximum number of parameters in SDP fmtp attribute.

Default: 16

#define PJMEDIA_SDP_NEG_PREFER_REMOTE_CODEC_ORDER   1

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)

#define PJMEDIA_SDP_NEG_ANSWER_MULTIPLE_CODECS   0

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)

#define PJMEDIA_SDP_NEG_MAX_CUSTOM_FMT_NEG_CB   8

This specifies the maximum number of the customized SDP format negotiation callbacks.

#define PJMEDIA_SDP_NEG_ANSWER_SYMMETRIC_PT   1

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)

#define PJMEDIA_HAS_RTCP_IN_SDP   (PJMEDIA_ADVERTISE_RTCP)

Support for sending and decoding RTCP port in SDP (RFC 3605). Default is equal to PJMEDIA_ADVERTISE_RTCP setting.

#define PJMEDIA_ADD_BANDWIDTH_TIAS_IN_SDP   1

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

   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;

Default: 1 (yes)

#define PJMEDIA_ADD_RTPMAP_FOR_STATIC_PT   1

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

   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;

Default: 1 (yes)

#define PJMEDIA_RTP_PT_TELEPHONE_EVENTS   96

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.

#define PJMEDIA_RTP_PT_TELEPHONE_EVENTS_STR   "96"

Macro to get the string representation of the telephone-event payload type.

#define PJMEDIA_TONEGEN_MAX_DIGITS   32

Maximum tones/digits that can be enqueued in the tone generator.

#define PJMEDIA_TONEGEN_SINE   1

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_FLOATING_POINT   2

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_FIXED_POINT_CORDIC   3

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_FAST_FIXED_POINT   4

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_ALG   PJMEDIA_TONEGEN_FIXED_POINT_CORDIC

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
#define PJMEDIA_TONEGEN_FIXED_POINT_CORDIC_LOOP   10

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

#define PJMEDIA_TONEGEN_FADE_IN_TIME   1

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. Fade-in duration for the tone, in milliseconds. Set to zero to disable this feature.

Default: 1 (msec)

#define PJMEDIA_TONEGEN_FADE_OUT_TIME   2

Fade-out duration for the tone, in milliseconds. Set to zero to disable this feature.

Default: 2 (msec)

#define PJMEDIA_TONEGEN_VOLUME   12288

The default tone generator amplitude (1-32767).

Default value: 12288

#define PJMEDIA_SRTP_MAX_CRYPTOS   16

Enable support for SRTP media transport. This will require linking with libsrtp from the third_party directory.

By default it is enabled. Maximum number of SRTP cryptos.

Default: 16

#define PJMEDIA_SRTP_HAS_AES_CM_256   1

Enable AES_CM_256 cryptos in SRTP. Default: enabled.

#define PJMEDIA_SRTP_HAS_AES_CM_192   0

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.

#define PJMEDIA_SRTP_HAS_AES_CM_128   1

Enable AES_CM_128 cryptos in SRTP. Default: enabled.

#define PJMEDIA_SRTP_HAS_AES_GCM_256   0

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.

#define PJMEDIA_SRTP_HAS_AES_GCM_128   0

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.

#define PJMEDIA_LIBSRTP_AUTO_INIT_DEINIT   1

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.

#define PJMEDIA_HANDLE_G722_MPEG_BUG   1

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.

#define PJMEDIA_TRANSPORT_SPECIFIC_INFO_MAXCNT   4

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.

#define PJMEDIA_TRANSPORT_SPECIFIC_INFO_MAXSIZE   (36*sizeof(long))

Maximum size in bytes of storage buffer of a transport specific info.

#define PJMEDIA_STREAM_KA_EMPTY_RTP   1

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_USER   2

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_PKT   { "\r\n", 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.

#define PJMEDIA_STREAM_KA_INTERVAL   5

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) Specify the keep-alive interval of PJMEDIA_STREAM_ENABLE_KA mechanism, in seconds.

Default: 5 seconds

#define PJMEDIA_HAS_VIDEO   0

Top level option to enable/disable video features.

Default: 0 (disabled)

#define PJMEDIA_HAS_FFMPEG   0

Specify if FFMPEG is available. The value here will be used as the default value for other FFMPEG settings below.

Default: 0

#define PJMEDIA_HAS_LIBAVFORMAT   PJMEDIA_HAS_FFMPEG

Specify if FFMPEG libavformat is available.

Default: PJMEDIA_HAS_FFMPEG (or detected by configure)

#define PJMEDIA_HAS_LIBAVCODEC   PJMEDIA_HAS_FFMPEG

Specify if FFMPEG libavformat is available.

Default: PJMEDIA_HAS_FFMPEG (or detected by configure)

#define PJMEDIA_HAS_LIBAVUTIL   PJMEDIA_HAS_FFMPEG

Specify if FFMPEG libavutil is available.

Default: PJMEDIA_HAS_FFMPEG (or detected by configure)

#define PJMEDIA_HAS_LIBSWSCALE   PJMEDIA_HAS_FFMPEG

Specify if FFMPEG libswscale is available.

Default: PJMEDIA_HAS_FFMPEG (or detected by configure)

#define PJMEDIA_HAS_LIBAVDEVICE   PJMEDIA_HAS_FFMPEG

Specify if FFMPEG libavdevice is available.

Default: PJMEDIA_HAS_FFMPEG (or detected by configure)

#define PJMEDIA_MAX_VIDEO_PLANES   4

Maximum video planes.

Default: 4

#define PJMEDIA_MAX_VIDEO_FORMATS   32

Maximum number of video formats.

Default: 32

#define PJMEDIA_CLOCK_SYNC_MAX_SYNC_MSEC   20000

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

#define PJMEDIA_MAX_VIDEO_ENC_FRAME_SIZE   (1<<17)

Maximum video frame size. Default: 128kB

#define PJMEDIA_CLOCK_SYNC_MAX_RESYNC_DURATION   2000

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

#define PJMEDIA_JBUF_DISC_MIN_GAP   200

Minimum gap between two consecutive discards in jitter buffer, in milliseconds.

Default: 200 ms

#define PJMEDIA_JBUF_PRO_DISC_MIN_BURST   1

Minimum burst level reference used for calculating discard duration in jitter buffer progressive discard algorithm, in frames.

Default: 1 frame

#define PJMEDIA_JBUF_PRO_DISC_MAX_BURST   100

Maximum burst level reference used for calculating discard duration in jitter buffer progressive discard algorithm, in frames.

Default: 200 frames

#define PJMEDIA_JBUF_PRO_DISC_T1   2000

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

#define PJMEDIA_JBUF_PRO_DISC_T2   10000

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

#define PJMEDIA_VID_STREAM_SKIP_PACKETS_TO_REDUCE_LATENCY   0

Video stream will discard old picture from the jitter buffer as soon as new picture is received, to reduce latency.

Default: 0

#define PJMEDIA_MAX_VID_PAYLOAD_SIZE   (PJMEDIA_MAX_MTU - 100)

Maximum video payload size. Note that this must not be greater than PJMEDIA_MAX_MTU.

Default: (PJMEDIA_MAX_MTU - 100)

#define PJMEDIA_TRANSPORT_SO_RCVBUF_SIZE   0

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)

#define PJMEDIA_TRANSPORT_SO_SNDBUF_SIZE   0

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)

#define PJMEDIA_HAS_LIBYUV   0

Specify if libyuv is available.

Default: 0 (disable)

#define PJMEDIA_HAS_DTMF_FLASH   1

Specify if dtmf flash in RFC 2833 is available.

#define PJMEDIA_VID_STREAM_START_KEYFRAME_CNT   5

Specify the number of keyframe needed to be sent after the stream is created. Setting this to 0 will disable it.

Default : 5

#define PJMEDIA_VID_STREAM_START_KEYFRAME_INTERVAL_MSEC   1000

Specify the interval to send keyframe after the stream is created, in msec.

Default : 1000

 


PJMEDIA small footprint Open Source media stack
Copyright (C) 2006-2008 Teluu Inc.