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+#ifndef foostreamhfoo
+#define foostreamhfoo
+
+/* $Id$ */
+
+/***
+ This file is part of PulseAudio.
+
+ PulseAudio is free software; you can redistribute it and/or modify
+ it under the terms of the GNU Lesser General Public License as published
+ by the Free Software Foundation; either version 2 of the License,
+ or (at your option) any later version.
+
+ PulseAudio 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 Lesser General Public License
+ along with PulseAudio; if not, write to the Free Software
+ Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
+ USA.
+***/
+
+#include <sys/types.h>
+
+#include <pulse/sample.h>
+#include <pulse/channelmap.h>
+#include <pulse/volume.h>
+#include <pulse/def.h>
+#include <pulse/cdecl.h>
+#include <pulse/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 server side 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 metrics may be controlled by the application. They are
+ * described with a pa_buffer_attr structure which contains a number
+ * of fields:
+ *
+ * \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. If a buffer underrun occurs, this prebuffering will be
+ * again enabled. If the playback shall never stop in case of a buffer
+ * underrun, this value should be set to 0. In that case the read
+ * index of the output buffer overtakes the write index, and hence the
+ * fill level of the buffer is negative.
+ * \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.
+ *
+ * The server side playback buffers are indexed by a write and a read
+ * index. The application writes to the write index and the sound
+ * device reads from the read index. The read index is increased
+ * monotonically, while the write index may be freely controlled by
+ * the application. Substracting the read index from the write index
+ * will give you the current fill level of the buffer. The read/write
+ * indexes are 64bit values and measured in bytes, they will never
+ * wrap. The current read/write index may be queried using
+ * pa_stream_get_timing_info() (see below for more information). In
+ * case of a buffer underrun the read index is equal or larger than
+ * the write index. Unless the prebuf value is 0, Polypaudio will
+ * temporarily pause playback in such a case, and wait until the
+ * buffer is filled up to prebuf bytes again. If prebuf is 0, the
+ * read index may be larger than the write index, in which case
+ * silence is played. If the application writes data to indexes lower
+ * than the read index, the data is immediately lost.
+ *
+ * \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 seek_modes Seeking in the Playback Buffer
+ *
+ * A client application may freely seek in the playback buffer. To
+ * accomplish that the pa_stream_write() function takes a seek mode
+ * and an offset argument. The seek mode is one of:
+ *
+ * \li PA_SEEK_RELATIVE - seek relative to the current write index
+ * \li PA_SEEK_ABSOLUTE - seek relative to the beginning of the playback buffer, (i.e. the first that was ever played in the stream)
+ * \li PA_SEEK_RELATIVE_ON_READ - seek relative to the current read index. Use this to write data to the output buffer that should be played as soon as possible
+ * \li PA_SEEK_RELATIVE_END - seek relative to the last byte ever written.
+ *
+ * If an application just wants to append some data to the output
+ * buffer, PA_SEEK_RELATIVE and an offset of 0 should be used.
+ *
+ * After a call to pa_stream_write() the write index will be left at
+ * the position right after the last byte of the written data.
+ *
+ * \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 the server
+ * side buffer transport delays and the backend active in the
+ * server. (Besides other things it contains the write and read index
+ * values mentioned above.)
+ *
+ * This structure is updated every time a
+ * pa_stream_update_timing_info() operation is executed. (i.e. before
+ * the first call to this function the timing information structure is
+ * not available!) Since it is a lot of work to keep this structure
+ * up-to-date manually, Polypaudio can do that automatically for you:
+ * if PA_STREAM_AUTO_TIMING_UPDATE is passed when connecting the
+ * stream Polypaudio will automatically update the structure every
+ * 100ms and every time a function is called that might invalidate the
+ * previously known timing data (such as pa_stream_write() or
+ * pa_stream_flush()). Please note however, that there always is a
+ * short time window when the data in the timing information structure
+ * is out-of-date. Polypaudio tries to mark these situations by
+ * setting the write_index_corrupt and read_index_corrupt fields
+ * accordingly.
+ *
+ * The raw timing data in the pa_timing_info structure is usually hard
+ * to deal with. Therefore a more simplistic interface is available:
+ * you can call pa_stream_get_time() or pa_stream_get_latency(). The
+ * former will return the current playback time of the hardware since
+ * the stream has been started. The latter returns the time a sample
+ * that you write now takes to be played by the hardware. These two
+ * functions base their calculations on the same data that is returned
+ * by pa_stream_get_timing_info(). Hence the same rules for keeping
+ * the timing data up-to-date apply here. In case the write or read
+ * index is corrupted, these two functions will fail with
+ * PA_ERR_NODATA set.
+ *
+ * Since updating the timing info structure usually requires a full
+ * network round trip and some applications monitor the timing very
+ * often Polypaudio offers a timing interpolation system. If
+ * PA_STREAM_INTERPOLATE_TIMING is passed when connecting the stream,
+ * pa_stream_get_time() and pa_stream_get_latency() will try to
+ * interpolate the current playback time/latency by estimating the
+ * number of samples that have been played back by the hardware since
+ * the last regular timing update. It is espcially useful to combine
+ * this option with PA_STREAM_AUTO_TIMING_UPDATE, which will enable
+ * you to monitor the current playback time/latency very precisely and
+ * very frequently without requiring a network round trip every time.
+ *
+ * \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 sync_streams Sychronizing Multiple Playback Streams
+ *
+ * Polypaudio allows applications to fully synchronize multiple
+ * playback streams that are connected to the same output device. That
+ * means the streams will always be played back sample-by-sample
+ * synchronously. If stream operations like pa_stream_cork() are
+ * issued on one of the synchronized streams, they are simultaneously
+ * issued on the others.
+ *
+ * To synchronize a stream to another, just pass the "master" stream
+ * as last argument to pa_stream_connect_playack(). To make sure that
+ * the freshly created stream doesn't start playback right-away, make
+ * sure to pass PA_STREAM_START_CORKED and - after all streams have
+ * been created - uncork them all with a single call to
+ * pa_stream_cork() for the master stream.
+ *
+ * To make sure that a particular stream doesn't stop to play when a
+ * server side buffer underrun happens on it while the other
+ * synchronized streams continue playing and hence deviate you need to
+ * pass a "prebuf" pa_buffer_attr of 0 when connecting it.
+ *
+ * \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 */
+
+PA_C_DECL_BEGIN
+
+/** An opaque stream for playback or recording */
+typedef struct pa_stream pa_stream;
+
+/** A generic callback for operation completion */
+typedef void (*pa_stream_success_cb_t) (pa_stream*s, int success, void *userdata);
+
+/** A generic request callback */
+typedef void (*pa_stream_request_cb_t)(pa_stream *p, size_t length, void *userdata);
+
+/** A generic notification callback */
+typedef void (*pa_stream_notify_cb_t)(pa_stream *p, void *userdata);
+
+/** Create a new, unconnected stream with the specified name and sample type */
+pa_stream* pa_stream_new(
+ pa_context *c /**< The context to create this stream in */,
+ const char *name /**< A name for this stream */,
+ const pa_sample_spec *ss /**< The desired sample format */,
+ const pa_channel_map *map /**< The desired channel map, or NULL for default */);
+
+/** Decrease the reference counter by one */
+void pa_stream_unref(pa_stream *s);
+
+/** Increase the reference counter by one */
+pa_stream *pa_stream_ref(pa_stream *s);
+
+/** Return the current state of the stream */
+pa_stream_state_t pa_stream_get_state(pa_stream *p);
+
+/** Return the context this stream is attached to */
+pa_context* pa_stream_get_context(pa_stream *p);
+
+/** Return the device (sink input or source output) index this stream is connected to */
+uint32_t pa_stream_get_index(pa_stream *s);
+
+/** Connect the stream to a sink */
+int pa_stream_connect_playback(
+ pa_stream *s /**< The stream to connect to a sink */,
+ const char *dev /**< Name of the sink to connect to, or NULL for default */ ,
+ const pa_buffer_attr *attr /**< Buffering attributes, or NULL for default */,
+ pa_stream_flags_t flags /**< Additional flags, or 0 for default */,
+ pa_cvolume *volume /**< Initial volume, or NULL for default */,
+ pa_stream *sync_stream /**< Synchronize this stream with the specified one, or NULL for a standalone stream*/);
+
+/** Connect the stream to a source */
+int pa_stream_connect_record(
+ pa_stream *s /**< The stream to connect to a source */ ,
+ const char *dev /**< Name of the source to connect to, or NULL for default */,
+ const pa_buffer_attr *attr /**< Buffer attributes, or NULL for default */,
+ pa_stream_flags_t flags /**< Additional flags, or 0 for default */);
+
+/** Disconnect a stream from a source/sink */
+int pa_stream_disconnect(pa_stream *s);
+
+/** Write some data to the server (for playback sinks), if free_cb is
+ * non-NULL this routine is called when all data has been written out
+ * and an internal reference to the specified data is kept, the data
+ * is not copied. If NULL, the data is copied into an internal
+ * buffer. The client my freely seek around in the output buffer. For
+ * most applications passing 0 and PA_SEEK_RELATIVE as arguments for
+ * offset and seek should be useful.*/
+int pa_stream_write(
+ pa_stream *p /**< The stream to use */,
+ const void *data /**< The data to write */,
+ size_t length /**< The length of the data to write */,
+ pa_free_cb_t free_cb /**< A cleanup routine for the data or NULL to request an internal copy */,
+ int64_t offset, /**< Offset for seeking, must be 0 for upload streams */
+ pa_seek_mode_t seek /**< Seek mode, must be PA_SEEK_RELATIVE for upload streams */);
+
+/** Read the next fragment from the buffer (for recording).
+ * data will point to the actual data and length will contain the size
+ * of the data in bytes (which can be less than a complete framgnet).
+ * Use pa_stream_drop() to actually remove the data from the
+ * buffer. If no data is available will return a NULL pointer \since 0.8 */
+int pa_stream_peek(
+ pa_stream *p /**< The stream to use */,
+ const void **data /**< Pointer to pointer that will point to data */,
+ size_t *length /**< The length of the data read */);
+
+/** Remove the current fragment on record streams. It is invalid to do this without first
+ * calling pa_stream_peek(). \since 0.8 */
+int pa_stream_drop(pa_stream *p);
+
+/** Return the nember of bytes that may be written using pa_stream_write() */
+size_t pa_stream_writable_size(pa_stream *p);
+
+/** Return the number of bytes that may be read using pa_stream_read() \since 0.8 */
+size_t pa_stream_readable_size(pa_stream *p);
+
+/** Drain a playback stream. Use this for notification when the buffer is empty */
+pa_operation* pa_stream_drain(pa_stream *s, pa_stream_success_cb_t cb, void *userdata);
+
+/** Request a timing info structure update for a stream. Use
+ * pa_stream_get_timing_info() to get access to the raw timing data,
+ * or pa_stream_get_time() or pa_stream_get_latency() to get cleaned
+ * up values. */
+pa_operation* pa_stream_update_timing_info(pa_stream *p, pa_stream_success_cb_t cb, void *userdata);
+
+/** Set the callback function that is called whenever the state of the stream changes */
+void pa_stream_set_state_callback(pa_stream *s, pa_stream_notify_cb_t cb, void *userdata);
+
+/** Set the callback function that is called when new data may be
+ * written to the stream. */
+void pa_stream_set_write_callback(pa_stream *p, pa_stream_request_cb_t cb, void *userdata);
+
+/** Set the callback function that is called when new data is available from the stream.
+ * Return the number of bytes read. \since 0.8 */
+void pa_stream_set_read_callback(pa_stream *p, pa_stream_request_cb_t cb, void *userdata);
+
+/** Set the callback function that is called when a buffer overflow happens. (Only for playback streams) \since 0.8 */
+void pa_stream_set_overflow_callback(pa_stream *p, pa_stream_notify_cb_t cb, void *userdata);
+
+/** Set the callback function that is called when a buffer underflow happens. (Only for playback streams) \since 0.8 */
+void pa_stream_set_underflow_callback(pa_stream *p, pa_stream_notify_cb_t cb, void *userdata);
+
+/** Set the callback function that is called whenever a latency information update happens. Useful on PA_STREAM_AUTO_TIMING_UPDATE streams only. (Only for playback streams) \since 0.8.2 */
+void pa_stream_set_latency_update_callback(pa_stream *p, pa_stream_notify_cb_t cb, void *userdata);
+
+/** Pause (or resume) playback of this stream temporarily. Available on both playback and recording streams. \since 0.3 */
+pa_operation* pa_stream_cork(pa_stream *s, int b, pa_stream_success_cb_t cb, void *userdata);
+
+/** Flush the playback buffer of this stream. Most of the time you're
+ * better off using the parameter delta of pa_stream_write() instead of this
+ * function. Available on both playback and recording streams. \since 0.3 */
+pa_operation* pa_stream_flush(pa_stream *s, pa_stream_success_cb_t cb, void *userdata);
+
+/** Reenable prebuffering as specified in the pa_buffer_attr
+ * structure. Available for playback streams only. \since 0.6 */
+pa_operation* pa_stream_prebuf(pa_stream *s, pa_stream_success_cb_t cb, void *userdata);
+
+/** Request immediate start of playback on this stream. This disables
+ * prebuffering as specified in the pa_buffer_attr
+ * structure, temporarily. Available for playback streams only. \since 0.3 */
+pa_operation* pa_stream_trigger(pa_stream *s, pa_stream_success_cb_t cb, void *userdata);
+
+/** Rename the stream. \since 0.5 */
+pa_operation* pa_stream_set_name(pa_stream *s, const char *name, pa_stream_success_cb_t cb, void *userdata);
+
+/** Return the current playback/recording time. This is based on the
+ * data in the timing info structure returned by
+ * pa_stream_get_timing_info(). This function will usually only return
+ * new data if a timing info update has been recieved. Only if timing
+ * interpolation has been requested (PA_STREAM_INTERPOLATE_TIMING)
+ * the data from the last timing update is used for an estimation of
+ * the current playback/recording time based on the local time that
+ * passed since the timing info structure has been acquired. The time
+ * value returned by this function is guaranteed to increase
+ * monotonically. (that means: the returned value is always greater or
+ * equal to the value returned on the last call) This behaviour can
+ * be disabled by using PA_STREAM_NOT_MONOTONOUS. This may be
+ * desirable to deal better with bad estimations of transport
+ * latencies, but may have strange effects if the application is not
+ * able to deal with time going 'backwards'. \since 0.6 */
+int pa_stream_get_time(pa_stream *s, pa_usec_t *r_usec);
+
+/** Return the total stream latency. This function is based on
+ * pa_stream_get_time(). In case the stream is a monitoring stream the
+ * result can be negative, i.e. the captured samples are not yet
+ * played. In this case *negative is set to 1. \since 0.6 */
+int pa_stream_get_latency(pa_stream *s, pa_usec_t *r_usec, int *negative);
+
+/** Return the latest raw timing data structure. The returned pointer
+ * points to an internal read-only instance of the timing
+ * structure. The user should make a copy of this structure if he
+ * wants to modify it. An in-place update to this data structure may
+ * be requested using pa_stream_update_timing_info(). If no
+ * pa_stream_update_timing_info() call was issued before, this
+ * function will fail with PA_ERR_NODATA. Please note that the
+ * write_index member field (and only this field) is updated on each
+ * pa_stream_write() call, not just when a timing update has been
+ * recieved. \since 0.8 */
+const pa_timing_info* pa_stream_get_timing_info(pa_stream *s);
+
+/** Return a pointer to the stream's sample specification. \since 0.6 */
+const pa_sample_spec* pa_stream_get_sample_spec(pa_stream *s);
+
+/** Return a pointer to the stream's channel map. \since 0.8 */
+const pa_channel_map* pa_stream_get_channel_map(pa_stream *s);
+
+/** Return the buffer metrics of the stream. Only valid after the
+ * stream has been connected successfuly and if the server is at least
+ * Polypaudio 0.9. \since 0.9.0 */
+const pa_buffer_attr* pa_stream_get_buffer_attr(pa_stream *s);
+
+PA_C_DECL_END
+
+#endif