Creating typical VoIP audio streams.

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 AudioStream *	audio_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)

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Detailed Description

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

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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.

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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:

stream	an AudioStream previously created with audio_stream_new().
prof	a RtpProfile containing all PayloadType possible during the audio session.
remip	remote IP address where to send the encoded audio.
remport	remote IP port where to send the encoded audio
rem_rtcp_port	remote port for RTCP.
payload_type	payload type index to use for the sending stream. This index must point to a valid PayloadType in the RtpProfile.
jitt_comp	Nominal jitter buffer size in milliseconds.
infile	path to wav file to play out (can be NULL)
outfile	path to wav file to record into (can be NULL)
playcard	The soundcard to be used for playback (can be NULL)
captcard	The soundcard to be used for catpure. (can be NULL)
echo_cancel	whether 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:

locport	the local UDP port to listen for RTP packets.
ipv6	TRUE 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:

stream	an AudioStream previously created with audio_stream_new().
prof	a RtpProfile containing all PayloadType possible during the audio session.
remip	remote IP address where to send the encoded audio.
remport	remote IP port where to send the encoded audio
rem_rtcp_port	remote port for RTCP.
payload_type	payload type index to use for the sending stream. This index must point to a valid PayloadType in the RtpProfile.
jitt_comp	Nominal jitter buffer size in milliseconds.
playcard	The soundcard to be used for playback
captcard	The soundcard to be used for catpure.
echo_cancel	whether 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.