mediastreamer2 2.7.3
Typedefs | Functions
Creating typical VoIP audio streams.
Mediastreamer2's high level APIs

Audio streaming API - easily run audio streams from soundcard or wav files to RTP. More...

Typedefs

typedef struct _AudioStream AudioStream
typedef VideoStream VideoPreview

Functions

MS2_PUBLIC int audio_stream_start_full (AudioStream *stream, RtpProfile *profile, const char *remip, int remport, int rem_rtcp_port, int payload, int jitt_comp, const char *infile, const char *outfile, MSSndCard *playcard, MSSndCard *captcard, bool_t use_ec)
MS2_PUBLIC AudioStreamaudio_stream_new (int locport, bool_t ipv6)
MS2_PUBLIC int audio_stream_start_now (AudioStream *stream, RtpProfile *prof, const char *remip, int remport, int rem_rtcp_port, int payload_type, int jitt_comp, MSSndCard *playcard, MSSndCard *captcard, bool_t echo_cancel)
MS2_PUBLIC void audio_stream_iterate (AudioStream *stream)

Detailed Description

Audio streaming API - easily run audio streams from soundcard or wav files to RTP.


Typedef Documentation

typedef struct _AudioStream AudioStream

The AudioStream holds all resources to create and run typical VoIP audiostream.

typedef VideoStream VideoPreview

Small API to display a local preview window.


Function Documentation

MS2_PUBLIC int audio_stream_start_full ( AudioStream stream,
RtpProfile *  profile,
const char *  remip,
int  remport,
int  rem_rtcp_port,
int  payload,
int  jitt_comp,
const char *  infile,
const char *  outfile,
MSSndCard playcard,
MSSndCard captcard,
bool_t  use_ec 
)

Starts an audio stream from/to local wav files or soundcards.

This method starts the processing of the audio stream, that is playing from wav file or soundcard, voice processing, encoding, sending through RTP, receiving from RTP, decoding, voice processing and wav file recording or soundcard playback.

Parameters:
streaman AudioStream previously created with audio_stream_new().
profa RtpProfile containing all PayloadType possible during the audio session.
remipremote IP address where to send the encoded audio.
remportremote IP port where to send the encoded audio
rem_rtcp_portremote port for RTCP.
payload_typepayload type index to use for the sending stream. This index must point to a valid PayloadType in the RtpProfile.
jitt_compNominal jitter buffer size in milliseconds.
infilepath to wav file to play out (can be NULL)
outfilepath to wav file to record into (can be NULL)
playcardThe soundcard to be used for playback (can be NULL)
captcardThe soundcard to be used for catpure. (can be NULL)
echo_cancelwhether echo cancellation is to be performed.
Returns:
0 if sucessful, -1 otherwise.
MS2_PUBLIC AudioStream* audio_stream_new ( int  locport,
bool_t  ipv6 
)

Creates an AudioStream object listening on a RTP port.

Parameters:
locportthe local UDP port to listen for RTP packets.
ipv6TRUE if ipv6 must be used.
Returns:
a new AudioStream.
MS2_PUBLIC int audio_stream_start_now ( AudioStream stream,
RtpProfile *  prof,
const char *  remip,
int  remport,
int  rem_rtcp_port,
int  payload_type,
int  jitt_comp,
MSSndCard playcard,
MSSndCard captcard,
bool_t  echo_cancel 
)

Starts an audio stream from local soundcards.

This method starts the processing of the audio stream, that is capture from soundcard, voice processing, encoding, sending through RTP, receiving from RTP, decoding, voice processing and soundcard playback.

Parameters:
streaman AudioStream previously created with audio_stream_new().
profa RtpProfile containing all PayloadType possible during the audio session.
remipremote IP address where to send the encoded audio.
remportremote IP port where to send the encoded audio
rem_rtcp_portremote port for RTCP.
payload_typepayload type index to use for the sending stream. This index must point to a valid PayloadType in the RtpProfile.
jitt_compNominal jitter buffer size in milliseconds.
playcardThe soundcard to be used for playback
captcardThe soundcard to be used for catpure.
echo_cancelwhether echo cancellation is to be performed.
MS2_PUBLIC void audio_stream_iterate ( AudioStream stream)

Executes background low priority tasks related to audio processing (RTP statistics analysis). It should be called periodically, for example with an interval of 100 ms or so.

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