Adding upstream version 14.2.upstream/14.2upstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 808 insertions, 0 deletions

diff --git a/src/pulse/stream.h b/src/pulse/stream.h
new file mode 100644
index 0000000..f9b52d4
--- /dev/null
+++ b/src/pulse/stream.h
@@ -0,0 +1,808 @@
+#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.
+ *
+ * If PA_STREAM_ADJUST_LATENCY is set, then the tlength/fragsize
+ * parameters of the pa_buffer_attr structure will be interpreted
+ * slightly differently than otherwise 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 registers 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 playing when a
+ * server side buffer underrun happens on it while the other
+ * synchronized streams continue playing and hence deviate, you need to
+ * pass a pa_buffer_attr with prebuf set to 0 when connecting.
+ *
+ * \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(const pa_stream *p);
+
+/** Return the context this stream is attached to. */
+pa_context* pa_stream_get_context(const 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(). This returns PA_INVALID_INDEX
+ * on failure. */
+uint32_t pa_stream_get_index(const 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_INVALID_INDEX on failure,
+ * including the being server older than 0.9.8. \since 0.9.8 */
+uint32_t pa_stream_get_device_index(const 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 fail when the server is older than
+ * 0.9.8. \since 0.9.8 */
+const char *pa_stream_get_device_name(const 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(const 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(const 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.
+ *
+ * Returns zero on success. */
+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 to let the server decide */ ,
+        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. Returns zero on success. */
+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 to let the server decide */,
+        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. Returns zero on success. */
+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.
+ *
+ * On success, will return zero and a valid (non-NULL) pointer. If the
+ * return value is non-zero, or the pointer is NULL, this indicates an
+ * error. Callers should also pay careful attention to the returned
+ * length, which may not be the same as that passed in, as mentioned above.
+ *
+ * \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.
+ * Returns zero on success. \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 respectively.
+ * After a successful write call 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 to 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.
+ *
+ * Returns zero on success. */
+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.
+ *
+ * Returns zero on success, negative on error. */
+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(). Returns zero on success. */
+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) -1 is returned on error.
+ */
+size_t pa_stream_writable_size(const pa_stream *p);
+
+/** Return the number of bytes that may be read using pa_stream_peek().
+ *
+ * (size_t) -1 is returned on error. */
+size_t pa_stream_readable_size(const 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(const 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 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(). The returned time is in the sound card
+ * clock domain, which usually runs at a slightly different rate than
+ * the system clock.
+ *
+ * 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().
+ *
+ * Returns zero on success, negative on error. */
+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 returned time is in the sound card clock
+ * domain, which usually runs at a slightly different rate than the
+ * system clock.
+ *
+ * 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
+ * wanting 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 return NULL.
+ *
+ * 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(const 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 than
+ * 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.
+ * Returns zero on success, negative on error. \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(). Returns PA_INVALID_INDEX
+ * on failure. \since 0.9.11 */
+uint32_t pa_stream_get_monitor_stream(const pa_stream *s);
+
+PA_C_DECL_END
+
+#endif