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,
+ 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
#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
*
*
* \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
+ * 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.
- * \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 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.
- * \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
+ * 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
* accordingly.
*
* The raw timing data in the pa_timing_info structure is usually hard
- * to deal with. Therefore a more simplistic interface is available:
+ * 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 time a sample
+ * 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
* 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
+ * 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.
* 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
+ * 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
*/
/** \file
- * Audio streams for input, output and sample upload */
+ * Audio streams for input, output and sample upload
+ *
+ * See also \subpage streams
+ */
PA_C_DECL_BEGIN
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);
+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. */
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);
* not, and negative on error. \since 0.9.11 */
int pa_stream_is_corked(pa_stream *s);
-/** Connect the stream to a sink */
+/** Connect the stream to a sink. It is strongly recommended to pass
+ * NULL in both dev and volume and not to set either
+ * PA_STREAM_START_MUTED nor PA_STREAM_START_UNMUTED -- unless these
+ * options are directly dependant 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 'volume' parameter was
+ * interpreted relative to the sink's current volume or treated as
+ * 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(). */
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 */,
+ 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 */
/** 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
+/** 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. On return *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 *nbytes value is invalid. Acessing the
+ * memory returned after the following pa_stream_write() or
+ * pa_stream_cancel_write() is invalid. On invocation only *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 data set to an address within this memory
+ * area and an 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
+ * pointer/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 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 may 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.*/
+ * offset and seek should be useful. 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 */,
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 */
+/** Read the next fragment from the buffer (for recording streams).
+ * data will point to the actual data and nbytes will contain the size
+ * of the data in bytes (which can be less or more than a complete
+ * fragment). Use pa_stream_drop() to actually remove the data from
+ * the buffer. If no data is available this 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 */,
/** 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 */
+/** 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
/** 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 startet
+ * 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);
* 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. */
+/** Set the callback function that is called whenver 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 whenver 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 b is 1 the stream is
+ * paused. If 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 flag during connection
+ * of 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 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. */
+/** Flush the playback buffer of this stream. This discards any audio
+ * in the buffer. 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
/** 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
+ * 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_MONOTONIC. 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'. */
+ * 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 on the server side for the most up to date timing with
+ * pa_stream_update_timing_info().
+ *
+ * If no timing information has been
+ * recieved 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);
/** 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. */
+ * 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.
+ *
+ * If no timing information has been recieved yet this call will
+ * return PA_ERR_NODATA. 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
* 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. */
+ * 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 recieved. */
const pa_timing_info* pa_stream_get_timing_info(pa_stream *s);
/** Return a pointer to the stream's sample specification. */
/** Return a pointer to the stream's channel map. */
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
- * PulseAudio 0.9. \since 0.9.0 */
+/** 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 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
* 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. \since 0.9.8 */
+ * 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
+ * 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);
* 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 */
+/** 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