Big documentation update. Describe the client API in a more tutorial like

manner.


git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@667 fefdeb5f-60dc-0310-8127-8f9354f1896f
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Pierre Ossman 2006-04-09 19:31:09 +00:00
parent a74253606f
commit a6ce5c4b1d
12 changed files with 757 additions and 52 deletions

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#include <polyp/cdecl.h>
#include <polyp/operation.h>
/** \page streams Audio streams
*
* \section overv_sec Overview
*
* Audio streams form the central functionality of the sound server. Data is
* routed, converted and mixed from several sources before it is passed along
* to a final output. Currently, there are three forms of audio streams:
*
* \li Playback streams - Data flows from the client to the server.
* \li Record streams - Data flows from the server to the client.
* \li Upload streams - Similar to playback streams, but the data is stored in
* the sample cache. See \ref scache for more information
* about controlling the sample cache.
*
* \section create_sec Creating
*
* To access a stream, a pa_stream object must be created using
* pa_stream_new(). At this point the audio sample format and mapping of
* channels must be specified. See \ref sample and \ref channelmap for more
* information about those structures.
*
* This first step will only create a client-side object, representing the
* stream. To use the stream, a server-side object must be created and
* associated with the local object. Depending on which type of stream is
* desired, a different function is needed:
*
* \li Playback stream - pa_stream_connect_playback()
* \li Record stream - pa_stream_connect_record()
* \li Upload stream - pa_stream_connect_upload() (see \ref scache)
*
* Similar to how connections are done in contexts, connecting a stream will
* not generate a pa_operation object. Also like contexts, the application
* should register a state change callback, using
* pa_stream_set_state_callback(), and wait for the stream to enter an active
* state.
*
* \subsection bufattr_subsec Buffer attributes
*
* Playback and record streams always have a buffer as part of the data flow.
* The size of this buffer strikes a compromise between low latency and
* sensitivity for buffer overflows/underruns.
*
* The buffer is described with a pa_buffer_attr structure which contains a
* number of field:
*
* \li maxlength - The absolute maximum number of bytes that can be stored in
* the buffer. If this value is exceeded then data will be
* lost.
* \li tlength - The target length of a playback buffer. The server will only
* send requests for more data as long as the buffer has less
* than this number of bytes of data.
* \li prebuf - Number of bytes that need to be in the buffer before playback
* will commence. Start of playback can be forced using
* pa_stream_trigger() even though the prebuffer size hasn't been
* reached.
* \li minreq - Minimum free number of the bytes in the playback buffer before
* the server will request more data.
* \li fragsize - Maximum number of bytes that the server will push in one
* chunk for record streams.
*
* \section transfer_sec Transferring data
*
* Once the stream is up, data can start flowing between the client and the
* server. Two different access models can be used to transfer the data:
*
* \li Asynchronous - The application register a callback using
* pa_stream_set_write_callback() and
* pa_stream_set_read_callback() to receive notifications
* that data can either be written or read.
* \li Polled - Query the library for available data/space using
* pa_stream_writable_size() and pa_stream_readable_size() and
* transfer data as needed. The sizes are stored locally, in the
* client end, so there is no delay when reading them.
*
* It is also possible to mix the two models freely.
*
* Once there is data/space available, it can be transferred using either
* pa_stream_write() for playback, or pa_stream_peek() / pa_stream_drop() for
* record. Make sure you do not overflow the playback buffers as data will be
* dropped.
*
* \section bufctl_sec Buffer control
*
* The transfer buffers can be controlled through a number of operations:
*
* \li pa_stream_cork() - Start or stop the playback or recording.
* \li pa_stream_trigger() - Start playback immediatly and do not wait for
* the buffer to fill up to the set trigger level.
* \li pa_stream_prebuf() - Reenable the playback trigger level.
* \li pa_stream_drain() - Wait for the playback buffer to go empty. Will
* return a pa_operation object that will indicate when
* the buffer is completely drained.
* \li pa_stream_flush() - Drop all data from the playback buffer and do not
* wait for it to finish playing.
*
* \section latency_sec Latency
*
* A major problem with networked audio is the increased latency caused by
* the network. To remedy this, Polypaudio supports an advanced system of
* monitoring the current latency.
*
* To get the raw data needed to calculate latencies, call
* pa_stream_get_timing_info(). This will give you a pa_timing_info structure
* that contains everything that is known about buffers, transport delays
* and the backend active in the server.
*
* If a more simplistic interface is prefered, you can call
* pa_stream_get_time() or pa_stream_get_latency(). These will do all the
* necessary calculations for you.
*
* The latency information is constantly updated from the server. Be aware
* that between updates, old data will be returned. If you specify the flag
* PA_STREAM_INTERPOLATE_TIMING when creating the stream, pa_stream_get_time()
* and pa_stream_get_latency() will calculate the latency between updates
* based on the time elapsed.
*
* \section flow_sec Overflow and underflow
*
* Even with the best precautions, buffers will sometime over- or underflow.
* To handle this gracefully, the application can be notified when this
* happens. Callbacks are registered using pa_stream_set_overflow_callback()
* and pa_stream_set_underflow_callback().
*
* \section disc_sec Disconnecting
*
* When a stream has served is purpose it must be disconnected with
* pa_stream_disconnect(). If you only unreference it, then it will live on
* and eat resources both locally and on the server until you disconnect the
* context.
*/
/** \file
* Audio streams for input, output and sample upload */