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|
#ifndef foostreamhfoo
#define foostreamhfoo
/***
This file is part of PulseAudio.
Copyright 2004-2006 Lennart Poettering
Copyright 2006 Pierre Ossman <ossman@cendio.se> for Cendio AB
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 needs to be chosen
* in 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. It is recommended to pass
* (uint32_t) -1 here which will cause the server to
* fill in the maximum possible value.
*
* \li tlength - The target fill level of the 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. If you pass (uint32_t) -1 (which is
* recommended) here the server will choose the longest
* target buffer fill level possible to minimize the
* number of necessary wakeups and maximize drop-out
* safety. This can exceed 2s of buffering. For
* low-latency applications or applications where
* latency matters you should pass a proper value here.
*
* \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. If you pass (uint32_t) -1 here
* (which is recommended) the server will choose the same
* value as tlength here.
*
* \li minreq - Minimum free number of the bytes in the playback
* buffer before the server will request more data. It is
* recommended to fill in (uint32_t) -1 here. This value
* influences how much time the sound server has to move
* data from the per-stream server-side playback buffer
* to the hardware playback buffer.
*
* \li fragsize - Maximum number of bytes that the server will push in
* one chunk for record streams. If you pass (uint32_t)
* -1 (which is recommended) here, the server will
* choose the longest fragment setting possible to
* minimize the number of necessary wakeups and
* maximize drop-out safety. This can exceed 2s of
* buffering. For low-latency applications or
* applications where latency matters you should pass a
* proper value here.
*
* If PA_STREAM_ADJUST_LATENCY is set, then the tlength/fragsize
* parameters will be interpreted slightly differently than described
* above when passed to pa_stream_connect_record() and
* pa_stream_connect_playback(): the overall latency that is comprised
* of both the server side playback buffer length, the hardware
* playback buffer length and additional latencies will be adjusted in
* a way that it matches tlength resp. fragsize. Set
* PA_STREAM_ADJUST_LATENCY if you want to control the overall
* playback latency for your stream. Unset it if you want to control
* only the latency induced by the server-side, rewritable playback
* buffer. The server will try to fulfill the clients latency requests
* as good as possible. However if the underlying hardware cannot
* change the hardware buffer length or only in a limited range, the
* actually resulting latency might be different from what the client
* requested. Thus, for synchronization clients always need to check
* the actual measured latency via pa_stream_get_latency() or a
* similar call, and not make any assumptions. about the latency
* available. The function pa_stream_get_buffer_attr() will always
* return the actual size of the server-side per-stream buffer in
* tlength/fragsize, regardless whether PA_STREAM_ADJUST_LATENCY is
* set or not.
*
* The server-side per-stream 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, PulseAudio 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, PulseAudio 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, PulseAudio can do that automatically for you:
* if PA_STREAM_AUTO_TIMING_UPDATE is passed when connecting the
* stream PulseAudio 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. PulseAudio 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 simpler 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 overall 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 PulseAudio 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
*
* PulseAudio 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 bytes, 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. It is recommended to use pa_stream_new_with_proplist()
* instead and specify some initial properties. */
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 */);
/** Create a new, unconnected stream with the specified name and
* sample type, and specify the the initial stream property
* list. \since 0.9.11 */
pa_stream* pa_stream_new_with_proplist(
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 */,
pa_proplist *p /**< The initial property list */);
/** 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 sink input resp. source output index this stream is
* identified in the server with. This is useful for usage with the
* introspection functions, such as pa_context_get_sink_input_info()
* resp. pa_context_get_source_output_info(). */
uint32_t pa_stream_get_index(pa_stream *s);
/** Return the index of the sink or source this stream is connected to
* in the server. This is useful for usage with the introspection
* functions, such as pa_context_get_sink_info_by_index()
* resp. pa_context_get_source_info_by_index(). Please note that
* streams may be moved between sinks/sources and thus it is
* recommended to use pa_stream_set_moved_callback() to be notified
* about this. This function will return with PA_ERR_NOTSUPPORTED when the
* server is older than 0.9.8. \since 0.9.8 */
uint32_t pa_stream_get_device_index(pa_stream *s);
/** Return the name of the sink or source this stream is connected to
* in the server. This is useful for usage with the introspection
* functions, such as pa_context_get_sink_info_by_name()
* resp. pa_context_get_source_info_by_name(). Please note that
* streams may be moved between sinks/sources and thus it is
* recommended to use pa_stream_set_moved_callback() to be notified
* about this. This function will return with PA_ERR_NOTSUPPORTED when the
* server is older than 0.9.8. \since 0.9.8 */
const char *pa_stream_get_device_name(pa_stream *s);
/** Return 1 if the sink or source this stream is connected to has
* been suspended. This will return 0 if not, and negative on
* error. This function will return with PA_ERR_NOTSUPPORTED when the
* server is older than 0.9.8. \since 0.9.8 */
int pa_stream_is_suspended(pa_stream *s);
/** Return 1 if the this stream has been corked. This will return 0 if
* not, and negative on error. \since 0.9.11 */
int pa_stream_is_corked(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 nbytes /**< The length of the data to write in bytes*/,
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 */
int pa_stream_peek(
pa_stream *p /**< The stream to use */,
const void **data /**< Pointer to pointer that will point to data */,
size_t *nbytes /**< The length of the data read in bytes */);
/** Remove the current fragment on record streams. It is invalid to do this without first
* calling pa_stream_peek(). */
int pa_stream_drop(pa_stream *p);
/** Return the number 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_peek()*/
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.*/
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) */
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) */
void pa_stream_set_underflow_callback(pa_stream *p, pa_stream_notify_cb_t cb, void *userdata);
/** Set the callback function that is called when a the server starts
* playback after an underrun or on initial startup. This only informs
* that audio is flowing again, it is no indication that audio started
* to reach the speakers already. (Only for playback streams). \since
* 0.9.11 */
void pa_stream_set_started_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) */
void pa_stream_set_latency_update_callback(pa_stream *p, pa_stream_notify_cb_t cb, void *userdata);
/** Set the callback function that is called whenever the stream is
* moved to a different sink/source. Use pa_stream_get_device_name()or
* pa_stream_get_device_index() to query the new sink/source. This
* notification is only generated when the server is at least
* 0.9.8. \since 0.9.8 */
void pa_stream_set_moved_callback(pa_stream *p, pa_stream_notify_cb_t cb, void *userdata);
/** Set the callback function that is called whenever the sink/source
* this stream is connected to is suspended or resumed. Use
* pa_stream_is_suspended() to query the new suspend status. Please
* note that the suspend status might also change when the stream is
* moved between devices. Thus if you call this function you very
* likely want to call pa_stream_set_moved_callback, too. This
* notification is only generated when the server is at least
* 0.9.8. \since 0.9.8 */
void pa_stream_set_suspended_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. */
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. */
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. */
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. */
pa_operation* pa_stream_trigger(pa_stream *s, pa_stream_success_cb_t cb, void *userdata);
/** Rename the stream. */
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'. */
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. */
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. */
const pa_timing_info* pa_stream_get_timing_info(pa_stream *s);
/** Return a pointer to the stream's sample specification. */
const pa_sample_spec* pa_stream_get_sample_spec(pa_stream *s);
/** Return a pointer to the stream's channel map. */
const pa_channel_map* pa_stream_get_channel_map(pa_stream *s);
/** Return the per-stream server-side buffer metrics of the
* stream. Only valid after the stream has been connected successfuly
* and if the server is at least PulseAudio 0.9. This will return the
* actual configured buffering metrics, which may differ from what was
* requested during pa_stream_connect_record() or
* pa_stream_connect_playback(). This call will always return the
* actually per-stream server-side buffer metrics, regardless whether
* PA_STREAM_ADJUST_LATENCY is set or not. \since 0.9.0 */
const pa_buffer_attr* pa_stream_get_buffer_attr(pa_stream *s);
/** Change the buffer metrics of the stream during playback. The
* server might have chosen different buffer metrics then
* requested. The selected metrics may be queried with
* pa_stream_get_buffer_attr() as soon as the callback is called. Only
* valid after the stream has been connected successfully and if the
* server is at least PulseAudio 0.9.8. Please be aware of the
* slightly different semantics of the call depending whether
* PA_STREAM_ADJUST_LATENCY is set or not. \since 0.9.8 */
pa_operation *pa_stream_set_buffer_attr(pa_stream *s, const pa_buffer_attr *attr, pa_stream_success_cb_t cb, void *userdata);
/** Change the stream sampling rate during playback. You need to pass
* PA_STREAM_VARIABLE_RATE in the flags parameter of
* pa_stream_connect() if you plan to use this function. Only valid
* after the stream has been connected successfully and if the server
* is at least PulseAudio 0.9.8. \since 0.9.8 */
pa_operation *pa_stream_update_sample_rate(pa_stream *s, uint32_t rate, pa_stream_success_cb_t cb, void *userdata);
/** Update the property list of the sink input/source output of this
* stream, adding new entries. Please note that it is highly
* recommended to set as much properties initially via
* pa_stream_new_with_proplist() as possible instead a posteriori with
* this function, since that information may then be used to route
* this stream to the right device. \since 0.9.11 */
pa_operation *pa_stream_proplist_update(pa_stream *s, pa_update_mode_t mode, pa_proplist *p, pa_stream_success_cb_t cb, void *userdata);
/** Update the property list of the sink input/source output of this
* stream, remove entries. \since 0.9.11 */
pa_operation *pa_stream_proplist_remove(pa_stream *s, const char *const keys[], pa_stream_success_cb_t cb, void *userdata);
/** For record streams connected to a monitor source: monitor only a
* very specific sink input of the sink. Thus function needs to be
* called before pa_stream_connect_record() is called. \since
* 0.9.11 */
int pa_stream_set_monitor_stream(pa_stream *s, uint32_t sink_input_idx);
/** Return what has been set with pa_stream_set_monitor_stream()
* ebfore. \since 0.9.11 */
uint32_t pa_stream_get_monitor_stream(pa_stream *s);
PA_C_DECL_END
#endif
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