godot/thirdparty/linuxbsd_headers/pulse/stream.h

832 lines
42 KiB
C++

#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 2.1 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, see <http://www.gnu.org/licenses/>.
***/
#include <sys/types.h>
#include <pulse/sample.h>
#include <pulse/format.h>
#include <pulse/channelmap.h>
#include <pulse/volume.h>
#include <pulse/def.h>
#include <pulse/cdecl.h>
#include <pulse/operation.h>
#include <pulse/context.h>
#include <pulse/proplist.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() or pa_stream_new_extended(). pa_stream_new() is for PCM
* streams only, while pa_stream_new_extended() can be used for both PCM and
* compressed audio streams. At this point the application must specify what
* stream format(s) it supports. See \ref sample and \ref channelmap for more
* information on the stream format parameters. FIXME: Those references only
* talk about PCM parameters, we should also have an overview page for how the
* pa_format_info based stream format configuration works. Bug filed:
* https://bugs.freedesktop.org/show_bug.cgi?id=72265
*
* 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.
*
* Note: there is a user-controllable slider in mixer applications such as
* pavucontrol corresponding to each of the created streams. Multiple
* (especially identically named) volume sliders for the same application might
* confuse the user. Also, the server supports only a limited number of
* simultaneous streams. Because of this, it is not always appropriate to
* create multiple streams in one application that needs to output multiple
* sounds. The rough guideline is: if there is no use case that would require
* separate user-initiated volume changes for each stream, perform the mixing
* inside the application.
*
* \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 number of free 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 client's 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. Subtracting 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 immediately 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 or record buffer. 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 especially 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 Synchronizing 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_playback(). 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
*
* See also \subpage streams
*/
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 nbytes, void *userdata);
/** A generic notification callback */
typedef void (*pa_stream_notify_cb_t)(pa_stream *p, void *userdata);
/** A callback for asynchronous meta/policy event messages. Well known
* event names are PA_STREAM_EVENT_REQUEST_CORK and
* PA_STREAM_EVENT_REQUEST_UNCORK. The set of defined events can be
* extended at any time. Also, server modules may introduce additional
* message types so make sure that your callback function ignores messages
* it doesn't know. \since 0.9.15 */
typedef void (*pa_stream_event_cb_t)(pa_stream *p, const char *name, pa_proplist *pl, 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 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 */);
/** Create a new, unconnected stream with the specified name, the set of formats
* this client can provide, and an initial list of properties. While
* connecting, the server will select the most appropriate format which the
* client must then provide. \since 1.0 */
pa_stream *pa_stream_new_extended(
pa_context *c /**< The context to create this stream in */,
const char *name /**< A name for this stream */,
pa_format_info * const * formats /**< The list of formats that can be provided */,
unsigned int n_formats /**< The number of formats being passed in */,
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 with the
* introspection functions such as pa_context_get_sink_input_info()
* or 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 with the introspection
* functions such as pa_context_get_sink_info_by_index() or
* 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 with the introspection
* functions such as pa_context_get_sink_info_by_name()
* or 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 a negative value 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 a negative value on error. \since 0.9.11 */
int pa_stream_is_corked(pa_stream *s);
/** Connect the stream to a sink. It is strongly recommended to pass
* NULL in both \a dev and \a volume and to set neither
* PA_STREAM_START_MUTED nor PA_STREAM_START_UNMUTED -- unless these
* options are directly dependent on user input or configuration.
*
* If you follow this rule then the sound server will have the full
* flexibility to choose the device, volume and mute status
* automatically, based on server-side policies, heuristics and stored
* information from previous uses. Also the server may choose to
* reconfigure audio devices to make other sinks/sources or
* capabilities available to be able to accept the stream.
*
* Before 0.9.20 it was not defined whether the \a volume parameter was
* interpreted relative to the sink's current volume or treated as
* an absolute device volume. Since 0.9.20 it is an absolute volume when
* the sink is in flat volume mode, and relative otherwise, thus
* making sure the volume passed here has always the same semantics as
* the volume passed to pa_context_set_sink_input_volume(). It is possible
* to figure out whether flat volume mode is in effect for a given sink
* by calling pa_context_get_sink_info_by_name().
*
* Since 5.0, it's possible to specify a single-channel volume even if the
* stream has multiple channels. In that case the same volume is applied to all
* channels. */
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 */,
const 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);
/** Prepare writing data to the server (for playback streams). This
* function may be used to optimize the number of memory copies when
* doing playback ("zero-copy"). It is recommended to call this
* function before each call to pa_stream_write().
*
* Pass in the address to a pointer and an address of the number of
* bytes you want to write. On return the two values will contain a
* pointer where you can place the data to write and the maximum number
* of bytes you can write. \a *nbytes can be smaller or have the same
* value as you passed in. You need to be able to handle both cases.
* Accessing memory beyond the returned \a *nbytes value is invalid.
* Accessing the memory returned after the following pa_stream_write()
* or pa_stream_cancel_write() is invalid.
*
* On invocation only \a *nbytes needs to be initialized, on return both
* *data and *nbytes will be valid. If you place (size_t) -1 in *nbytes
* on invocation the memory size will be chosen automatically (which is
* recommended to do). After placing your data in the memory area
* returned, call pa_stream_write() with \a data set to an address
* within this memory area and an \a nbytes value that is smaller or
* equal to what was returned by this function to actually execute the
* write.
*
* An invocation of pa_stream_write() should follow "quickly" on
* pa_stream_begin_write(). It is not recommended letting an unbounded
* amount of time pass after calling pa_stream_begin_write() and
* before calling pa_stream_write(). If you want to cancel a
* previously called pa_stream_begin_write() without calling
* pa_stream_write() use pa_stream_cancel_write(). Calling
* pa_stream_begin_write() twice without calling pa_stream_write() or
* pa_stream_cancel_write() in between will return exactly the same
* \a data pointer and \a nbytes values. \since 0.9.16 */
int pa_stream_begin_write(
pa_stream *p,
void **data,
size_t *nbytes);
/** Reverses the effect of pa_stream_begin_write() dropping all data
* that has already been placed in the memory area returned by
* pa_stream_begin_write(). Only valid to call if
* pa_stream_begin_write() was called before and neither
* pa_stream_cancel_write() nor pa_stream_write() have been called
* yet. Accessing the memory previously returned by
* pa_stream_begin_write() after this call is invalid. Any further
* explicit freeing of the memory area is not necessary. \since
* 0.9.16 */
int pa_stream_cancel_write(
pa_stream *p);
/** Write some data to the server (for playback streams).
* If \a free_cb is non-NULL this routine is called when all data has
* been written out. 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 may freely seek around in the output buffer. For
* most applications it is typical to pass 0 and PA_SEEK_RELATIVE
* as values for the arguments \a offset and \a seek. After the write
* call succeeded the write index will be at the position after where
* this chunk of data has been written to.
*
* As an optimization for avoiding needless memory copies you may call
* pa_stream_begin_write() before this call and then place your audio
* data directly in the memory area returned by that call. Then, pass
* a pointer to that memory area to pa_stream_write(). After the
* invocation of pa_stream_write() the memory area may no longer be
* accessed. Any further explicit freeing of the memory area is not
* necessary. It is OK to write the memory area returned by
* pa_stream_begin_write() only partially with this call, skipping
* bytes both at the end and at the beginning of the reserved memory
* area.*/
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, must be in multiples of the stream's sample spec frame size */,
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, must be in multiples of the stream's sample spec frame size */,
pa_seek_mode_t seek /**< Seek mode, must be PA_SEEK_RELATIVE for upload streams */);
/** Function does exactly the same as pa_stream_write() with the difference
* that free_cb_data is passed to free_cb instead of data. \since 6.0 */
int pa_stream_write_ext_free(
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 */,
void *free_cb_data /**< Argument passed to free_cb function */,
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 streams).
* If there is data at the current read index, \a data will point to
* the actual data and \a nbytes will contain the size of the data in
* bytes (which can be less or more than a complete fragment).
*
* If there is no data at the current read index, it means that either
* the buffer is empty or it contains a hole (that is, the write index
* is ahead of the read index but there's no data where the read index
* points at). If the buffer is empty, \a data will be NULL and
* \a nbytes will be 0. If there is a hole, \a data will be NULL and
* \a nbytes will contain the length of the hole.
*
* Use pa_stream_drop() to actually remove the data from the buffer
* and move the read index forward. pa_stream_drop() should not be
* called if the buffer is empty, but it should be called if there is
* a hole. */
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 requested by the server that have not yet
* been written.
*
* It is possible to write more than this amount, up to the stream's
* buffer_attr.maxlength bytes. This is usually not desirable, though, as
* it would increase stream latency to be higher than requested
* (buffer_attr.tlength).
*/
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
* playback buffer is empty after playing all the audio in the buffer.
* Please note that only one drain operation per stream may be issued
* at a time. */
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. */
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);
/** Return at what position the latest underflow occurred, or -1 if this information is not
* known (e.g.\ if no underflow has occurred, or server is older than 1.0).
* Can be used inside the underflow callback to get information about the current underflow.
* (Only for playback streams) \since 1.0 */
int64_t pa_stream_get_underflow_index(pa_stream *p);
/** 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. */
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);
/** Set the callback function that is called whenever a meta/policy
* control event is received. \since 0.9.15 */
void pa_stream_set_event_callback(pa_stream *p, pa_stream_event_cb_t cb, void *userdata);
/** Set the callback function that is called whenever the buffer
* attributes on the server side change. Please note that the buffer
* attributes can change when moving a stream to a different
* sink/source too, hence if you use this callback you should use
* pa_stream_set_moved_callback() as well. \since 0.9.15 */
void pa_stream_set_buffer_attr_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. If \a b is 1 the stream is
* paused. If \a b is 0 the stream is resumed. The pause/resume operation
* is executed as quickly as possible. If a cork is very quickly
* followed by an uncork or the other way round, this might not
* actually have any effect on the stream that is output. You can use
* pa_stream_is_corked() to find out whether the stream is currently
* paused or not. Normally a stream will be created in uncorked
* state. If you pass PA_STREAM_START_CORKED as a flag when connecting
* the stream, it will be created in corked state. */
pa_operation* pa_stream_cork(pa_stream *s, int b, pa_stream_success_cb_t cb, void *userdata);
/** Flush the playback or record buffer of this stream. This discards any audio data
* in the buffer. Most of the time you're better off using the parameter
* \a seek of pa_stream_write() instead of this function. */
pa_operation* pa_stream_flush(pa_stream *s, pa_stream_success_cb_t cb, void *userdata);
/** Reenable prebuffering if 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 temporarily if specified in the pa_buffer_attr structure.
* 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 received. 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 (the returned value is always greater
* or equal to the value returned by the last call). This behaviour
* can be disabled by using PA_STREAM_NOT_MONOTONIC. This may be
* desirable to better deal with bad estimations of transport
* latencies, but may have strange effects if the application is not
* able to deal with time going 'backwards'.
*
* The time interpolator activated by PA_STREAM_INTERPOLATE_TIMING
* favours 'smooth' time graphs over accurate ones to improve the
* smoothness of UI operations that are tied to the audio clock. If
* accuracy is more important to you, you might need to estimate your
* timing based on the data from pa_stream_get_timing_info() yourself
* or not work with interpolated timing at all and instead always
* query the server side for the most up to date timing with
* pa_stream_update_timing_info().
*
* If no timing information has been
* received yet this call will return -PA_ERR_NODATA. For more details
* see pa_stream_get_timing_info(). */
int pa_stream_get_time(pa_stream *s, pa_usec_t *r_usec);
/** Determine the total stream latency. This function is based on
* pa_stream_get_time().
*
* The latency is stored in \a *r_usec. 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 \a *negative is set to 1.
*
* If no timing information has been received yet, this call will
* return -PA_ERR_NODATA. On success, it will return 0.
*
* For more details see pa_stream_get_timing_info() and
* pa_stream_get_time(). */
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
* refers 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 timing information has been received before (i.e. by
* requesting pa_stream_update_timing_info() or by using
* PA_STREAM_AUTO_TIMING_UPDATE), 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 received. */
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 a pointer to the stream's format. \since 1.0 */
const pa_format_info* pa_stream_get_format_info(pa_stream *s);
/** Return the per-stream server-side buffer metrics of the
* stream. Only valid after the stream has been connected successfully
* 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
* actual 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_playback() 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 many properties initially via
* pa_stream_new_with_proplist() as possible instead a posteriori with
* this function, since that information may 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. This 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 the sink input index previously set with
* pa_stream_set_monitor_stream().
* \since 0.9.11 */
uint32_t pa_stream_get_monitor_stream(pa_stream *s);
PA_C_DECL_END
#endif