831 lines
42 KiB
C++
831 lines
42 KiB
C++
#ifndef foostreamhfoo
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#define foostreamhfoo
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/***
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This file is part of PulseAudio.
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Copyright 2004-2006 Lennart Poettering
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Copyright 2006 Pierre Ossman <ossman@cendio.se> for Cendio AB
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PulseAudio is free software; you can redistribute it and/or modify
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it under the terms of the GNU Lesser General Public License as published
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by the Free Software Foundation; either version 2.1 of the License,
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or (at your option) any later version.
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PulseAudio is distributed in the hope that it will be useful, but
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WITHOUT ANY WARRANTY; without even the implied warranty of
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MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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General Public License for more details.
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You should have received a copy of the GNU Lesser General Public License
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along with PulseAudio; if not, see <http://www.gnu.org/licenses/>.
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***/
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#include <sys/types.h>
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#include <pulse/sample.h>
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#include <pulse/format.h>
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#include <pulse/channelmap.h>
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#include <pulse/volume.h>
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#include <pulse/def.h>
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#include <pulse/cdecl.h>
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#include <pulse/operation.h>
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#include <pulse/context.h>
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#include <pulse/proplist.h>
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/** \page streams Audio Streams
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*
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* \section overv_sec Overview
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*
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* Audio streams form the central functionality of the sound server. Data is
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* routed, converted and mixed from several sources before it is passed along
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* to a final output. Currently, there are three forms of audio streams:
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*
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* \li Playback streams - Data flows from the client to the server.
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* \li Record streams - Data flows from the server to the client.
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* \li Upload streams - Similar to playback streams, but the data is stored in
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* the sample cache. See \ref scache for more information
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* about controlling the sample cache.
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*
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* \section create_sec Creating
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*
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* To access a stream, a pa_stream object must be created using
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* pa_stream_new() or pa_stream_new_extended(). pa_stream_new() is for PCM
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* streams only, while pa_stream_new_extended() can be used for both PCM and
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* compressed audio streams. At this point the application must specify what
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* stream format(s) it supports. See \ref sample and \ref channelmap for more
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* information on the stream format parameters. FIXME: Those references only
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* talk about PCM parameters, we should also have an overview page for how the
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* pa_format_info based stream format configuration works. Bug filed:
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* https://bugs.freedesktop.org/show_bug.cgi?id=72265
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*
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* This first step will only create a client-side object, representing the
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* stream. To use the stream, a server-side object must be created and
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* associated with the local object. Depending on which type of stream is
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* desired, a different function is needed:
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*
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* \li Playback stream - pa_stream_connect_playback()
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* \li Record stream - pa_stream_connect_record()
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* \li Upload stream - pa_stream_connect_upload() (see \ref scache)
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*
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* Similar to how connections are done in contexts, connecting a stream will
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* not generate a pa_operation object. Also like contexts, the application
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* should register a state change callback, using
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* pa_stream_set_state_callback(), and wait for the stream to enter an active
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* state.
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*
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* Note: there is a user-controllable slider in mixer applications such as
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* pavucontrol corresponding to each of the created streams. Multiple
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* (especially identically named) volume sliders for the same application might
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* confuse the user. Also, the server supports only a limited number of
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* simultaneous streams. Because of this, it is not always appropriate to
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* create multiple streams in one application that needs to output multiple
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* sounds. The rough guideline is: if there is no use case that would require
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* separate user-initiated volume changes for each stream, perform the mixing
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* inside the application.
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*
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* \subsection bufattr_subsec Buffer Attributes
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*
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* Playback and record streams always have a server-side buffer as
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* part of the data flow. The size of this buffer needs to be chosen
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* in a compromise between low latency and sensitivity for buffer
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* overflows/underruns.
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*
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* The buffer metrics may be controlled by the application. They are
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* described with a pa_buffer_attr structure which contains a number
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* of fields:
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*
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* \li maxlength - The absolute maximum number of bytes that can be
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* stored in the buffer. If this value is exceeded
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* then data will be lost. It is recommended to pass
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* (uint32_t) -1 here which will cause the server to
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* fill in the maximum possible value.
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*
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* \li tlength - The target fill level of the playback buffer. The
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* server will only send requests for more data as long
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* as the buffer has less than this number of bytes of
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* data. If you pass (uint32_t) -1 (which is
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* recommended) here the server will choose the longest
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* target buffer fill level possible to minimize the
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* number of necessary wakeups and maximize drop-out
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* safety. This can exceed 2s of buffering. For
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* low-latency applications or applications where
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* latency matters you should pass a proper value here.
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*
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* \li prebuf - Number of bytes that need to be in the buffer before
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* playback will commence. Start of playback can be
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* forced using pa_stream_trigger() even though the
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* prebuffer size hasn't been reached. If a buffer
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* underrun occurs, this prebuffering will be again
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* enabled. If the playback shall never stop in case of a
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* buffer underrun, this value should be set to 0. In
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* that case the read index of the output buffer
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* overtakes the write index, and hence the fill level of
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* the buffer is negative. If you pass (uint32_t) -1 here
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* (which is recommended) the server will choose the same
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* value as tlength here.
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*
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* \li minreq - Minimum number of free bytes in the playback
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* buffer before the server will request more data. It is
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* recommended to fill in (uint32_t) -1 here. This value
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* influences how much time the sound server has to move
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* data from the per-stream server-side playback buffer
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* to the hardware playback buffer.
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*
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* \li fragsize - Maximum number of bytes that the server will push in
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* one chunk for record streams. If you pass (uint32_t)
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* -1 (which is recommended) here, the server will
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* choose the longest fragment setting possible to
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* minimize the number of necessary wakeups and
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* maximize drop-out safety. This can exceed 2s of
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* buffering. For low-latency applications or
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* applications where latency matters you should pass a
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* proper value here.
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*
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* If PA_STREAM_ADJUST_LATENCY is set, then the tlength/fragsize
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* parameters will be interpreted slightly differently than described
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* above when passed to pa_stream_connect_record() and
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* pa_stream_connect_playback(): the overall latency that is comprised
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* of both the server side playback buffer length, the hardware
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* playback buffer length and additional latencies will be adjusted in
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* a way that it matches tlength resp. fragsize. Set
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* PA_STREAM_ADJUST_LATENCY if you want to control the overall
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* playback latency for your stream. Unset it if you want to control
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* only the latency induced by the server-side, rewritable playback
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* buffer. The server will try to fulfill the client's latency requests
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* as good as possible. However if the underlying hardware cannot
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* change the hardware buffer length or only in a limited range, the
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* actually resulting latency might be different from what the client
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* requested. Thus, for synchronization clients always need to check
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* the actual measured latency via pa_stream_get_latency() or a
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* similar call, and not make any assumptions about the latency
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* available. The function pa_stream_get_buffer_attr() will always
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* return the actual size of the server-side per-stream buffer in
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* tlength/fragsize, regardless whether PA_STREAM_ADJUST_LATENCY is
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* set or not.
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*
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* The server-side per-stream playback buffers are indexed by a write and a read
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* index. The application writes to the write index and the sound
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* device reads from the read index. The read index is increased
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* monotonically, while the write index may be freely controlled by
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* the application. Subtracting the read index from the write index
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* will give you the current fill level of the buffer. The read/write
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* indexes are 64bit values and measured in bytes, they will never
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* wrap. The current read/write index may be queried using
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* pa_stream_get_timing_info() (see below for more information). In
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* case of a buffer underrun the read index is equal or larger than
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* the write index. Unless the prebuf value is 0, PulseAudio will
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* temporarily pause playback in such a case, and wait until the
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* buffer is filled up to prebuf bytes again. If prebuf is 0, the
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* read index may be larger than the write index, in which case
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* silence is played. If the application writes data to indexes lower
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* than the read index, the data is immediately lost.
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*
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* \section transfer_sec Transferring Data
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*
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* Once the stream is up, data can start flowing between the client and the
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* server. Two different access models can be used to transfer the data:
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*
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* \li Asynchronous - The application register a callback using
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* pa_stream_set_write_callback() and
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* pa_stream_set_read_callback() to receive notifications
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* that data can either be written or read.
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* \li Polled - Query the library for available data/space using
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* pa_stream_writable_size() and pa_stream_readable_size() and
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* transfer data as needed. The sizes are stored locally, in the
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* client end, so there is no delay when reading them.
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*
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* It is also possible to mix the two models freely.
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*
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* Once there is data/space available, it can be transferred using either
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* pa_stream_write() for playback, or pa_stream_peek() / pa_stream_drop() for
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* record. Make sure you do not overflow the playback buffers as data will be
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* dropped.
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*
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* \section bufctl_sec Buffer Control
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*
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* The transfer buffers can be controlled through a number of operations:
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*
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* \li pa_stream_cork() - Start or stop the playback or recording.
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* \li pa_stream_trigger() - Start playback immediately and do not wait for
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* the buffer to fill up to the set trigger level.
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* \li pa_stream_prebuf() - Reenable the playback trigger level.
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* \li pa_stream_drain() - Wait for the playback buffer to go empty. Will
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* return a pa_operation object that will indicate when
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* the buffer is completely drained.
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* \li pa_stream_flush() - Drop all data from the playback or record buffer. Do not
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* wait for it to finish playing.
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*
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* \section seek_modes Seeking in the Playback Buffer
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*
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* A client application may freely seek in the playback buffer. To
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* accomplish that the pa_stream_write() function takes a seek mode
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* and an offset argument. The seek mode is one of:
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*
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* \li PA_SEEK_RELATIVE - seek relative to the current write index
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* \li PA_SEEK_ABSOLUTE - seek relative to the beginning of the playback buffer, (i.e. the first that was ever played in the stream)
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* \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
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* \li PA_SEEK_RELATIVE_END - seek relative to the last byte ever written.
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*
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* If an application just wants to append some data to the output
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* buffer, PA_SEEK_RELATIVE and an offset of 0 should be used.
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*
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* After a call to pa_stream_write() the write index will be left at
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* the position right after the last byte of the written data.
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*
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* \section latency_sec Latency
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*
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* A major problem with networked audio is the increased latency caused by
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* the network. To remedy this, PulseAudio supports an advanced system of
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* monitoring the current latency.
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*
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* To get the raw data needed to calculate latencies, call
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* pa_stream_get_timing_info(). This will give you a pa_timing_info
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* structure that contains everything that is known about the server
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* side buffer transport delays and the backend active in the
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* server. (Besides other things it contains the write and read index
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* values mentioned above.)
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*
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* This structure is updated every time a
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* pa_stream_update_timing_info() operation is executed. (i.e. before
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* the first call to this function the timing information structure is
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* not available!) Since it is a lot of work to keep this structure
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* up-to-date manually, PulseAudio can do that automatically for you:
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* if PA_STREAM_AUTO_TIMING_UPDATE is passed when connecting the
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* stream PulseAudio will automatically update the structure every
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* 100ms and every time a function is called that might invalidate the
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* previously known timing data (such as pa_stream_write() or
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* pa_stream_flush()). Please note however, that there always is a
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* short time window when the data in the timing information structure
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* is out-of-date. PulseAudio tries to mark these situations by
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* setting the write_index_corrupt and read_index_corrupt fields
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* accordingly.
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*
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* The raw timing data in the pa_timing_info structure is usually hard
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* to deal with. Therefore a simpler interface is available:
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* you can call pa_stream_get_time() or pa_stream_get_latency(). The
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* former will return the current playback time of the hardware since
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* the stream has been started. The latter returns the overall time a sample
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* that you write now takes to be played by the hardware. These two
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* functions base their calculations on the same data that is returned
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* by pa_stream_get_timing_info(). Hence the same rules for keeping
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* the timing data up-to-date apply here. In case the write or read
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* index is corrupted, these two functions will fail with
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* -PA_ERR_NODATA set.
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*
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* Since updating the timing info structure usually requires a full
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* network round trip and some applications monitor the timing very
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* often PulseAudio offers a timing interpolation system. If
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* PA_STREAM_INTERPOLATE_TIMING is passed when connecting the stream,
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* pa_stream_get_time() and pa_stream_get_latency() will try to
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* interpolate the current playback time/latency by estimating the
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* number of samples that have been played back by the hardware since
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* the last regular timing update. It is especially useful to combine
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* this option with PA_STREAM_AUTO_TIMING_UPDATE, which will enable
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* you to monitor the current playback time/latency very precisely and
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* very frequently without requiring a network round trip every time.
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*
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* \section flow_sec Overflow and underflow
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*
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* Even with the best precautions, buffers will sometime over - or
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* underflow. To handle this gracefully, the application can be
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* notified when this happens. Callbacks are registered using
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* pa_stream_set_overflow_callback() and
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* pa_stream_set_underflow_callback().
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*
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* \section sync_streams Synchronizing Multiple Playback Streams
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*
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* PulseAudio allows applications to fully synchronize multiple
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* playback streams that are connected to the same output device. That
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* means the streams will always be played back sample-by-sample
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* synchronously. If stream operations like pa_stream_cork() are
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* issued on one of the synchronized streams, they are simultaneously
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* issued on the others.
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*
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* To synchronize a stream to another, just pass the "master" stream
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* as last argument to pa_stream_connect_playback(). To make sure that
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* the freshly created stream doesn't start playback right-away, make
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* sure to pass PA_STREAM_START_CORKED and -- after all streams have
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* been created -- uncork them all with a single call to
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* pa_stream_cork() for the master stream.
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*
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* To make sure that a particular stream doesn't stop to play when a
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* server side buffer underrun happens on it while the other
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* synchronized streams continue playing and hence deviate, you need to
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* pass a "prebuf" pa_buffer_attr of 0 when connecting it.
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*
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* \section disc_sec Disconnecting
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*
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* When a stream has served is purpose it must be disconnected with
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* pa_stream_disconnect(). If you only unreference it, then it will live on
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* and eat resources both locally and on the server until you disconnect the
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* context.
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*
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*/
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/** \file
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* Audio streams for input, output and sample upload
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*
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* See also \subpage streams
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*/
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PA_C_DECL_BEGIN
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/** An opaque stream for playback or recording */
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typedef struct pa_stream pa_stream;
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/** A generic callback for operation completion */
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typedef void (*pa_stream_success_cb_t) (pa_stream*s, int success, void *userdata);
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/** A generic request callback */
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typedef void (*pa_stream_request_cb_t)(pa_stream *p, size_t nbytes, void *userdata);
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/** A generic notification callback */
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typedef void (*pa_stream_notify_cb_t)(pa_stream *p, void *userdata);
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/** A callback for asynchronous meta/policy event messages. Well known
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* event names are PA_STREAM_EVENT_REQUEST_CORK and
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* PA_STREAM_EVENT_REQUEST_UNCORK. The set of defined events can be
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* extended at any time. Also, server modules may introduce additional
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* message types so make sure that your callback function ignores messages
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* it doesn't know. \since 0.9.15 */
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typedef void (*pa_stream_event_cb_t)(pa_stream *p, const char *name, pa_proplist *pl, void *userdata);
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/** Create a new, unconnected stream with the specified name and
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* sample type. It is recommended to use pa_stream_new_with_proplist()
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* instead and specify some initial properties. */
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pa_stream* pa_stream_new(
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pa_context *c /**< The context to create this stream in */,
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const char *name /**< A name for this stream */,
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const pa_sample_spec *ss /**< The desired sample format */,
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const pa_channel_map *map /**< The desired channel map, or NULL for default */);
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/** Create a new, unconnected stream with the specified name and
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* sample type, and specify the initial stream property
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* list. \since 0.9.11 */
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pa_stream* pa_stream_new_with_proplist(
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pa_context *c /**< The context to create this stream in */,
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const char *name /**< A name for this stream */,
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const pa_sample_spec *ss /**< The desired sample format */,
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const pa_channel_map *map /**< The desired channel map, or NULL for default */,
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pa_proplist *p /**< The initial property list */);
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/** Create a new, unconnected stream with the specified name, the set of formats
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* this client can provide, and an initial list of properties. While
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* connecting, the server will select the most appropriate format which the
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* client must then provide. \since 1.0 */
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pa_stream *pa_stream_new_extended(
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pa_context *c /**< The context to create this stream in */,
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const char *name /**< A name for this stream */,
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pa_format_info * const * formats /**< The list of formats that can be provided */,
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unsigned int n_formats /**< The number of formats being passed in */,
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pa_proplist *p /**< The initial property list */);
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/** Decrease the reference counter by one. */
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void pa_stream_unref(pa_stream *s);
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/** Increase the reference counter by one. */
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pa_stream *pa_stream_ref(pa_stream *s);
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/** Return the current state of the stream. */
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pa_stream_state_t pa_stream_get_state(pa_stream *p);
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/** Return the context this stream is attached to. */
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pa_context* pa_stream_get_context(pa_stream *p);
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/** Return the sink input resp.\ source output index this stream is
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* identified in the server with. This is useful with the
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* introspection functions such as pa_context_get_sink_input_info()
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* or pa_context_get_source_output_info(). */
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uint32_t pa_stream_get_index(pa_stream *s);
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/** Return the index of the sink or source this stream is connected to
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* in the server. This is useful with the introspection
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* functions such as pa_context_get_sink_info_by_index() or
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* pa_context_get_source_info_by_index().
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*
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* Please note that streams may be moved between sinks/sources and thus
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* it is recommended to use pa_stream_set_moved_callback() to be notified
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* about this. This function will return with -PA_ERR_NOTSUPPORTED when the
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* server is older than 0.9.8. \since 0.9.8 */
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uint32_t pa_stream_get_device_index(pa_stream *s);
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/** Return the name of the sink or source this stream is connected to
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* in the server. This is useful with the introspection
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* functions such as pa_context_get_sink_info_by_name()
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* or pa_context_get_source_info_by_name().
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*
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* Please note that streams may be moved between sinks/sources and thus
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* it is recommended to use pa_stream_set_moved_callback() to be notified
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* about this. This function will return with -PA_ERR_NOTSUPPORTED when the
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* server is older than 0.9.8. \since 0.9.8 */
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const char *pa_stream_get_device_name(pa_stream *s);
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|
|
/** 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
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* stream, adding new entries. Please note that it is highly
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* recommended to set as many properties initially via
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* pa_stream_new_with_proplist() as possible instead a posteriori with
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* this function, since that information may be used to route
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* this stream to the right device. \since 0.9.11 */
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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);
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/** Update the property list of the sink input/source output of this
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* stream, remove entries. \since 0.9.11 */
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pa_operation *pa_stream_proplist_remove(pa_stream *s, const char *const keys[], pa_stream_success_cb_t cb, void *userdata);
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/** For record streams connected to a monitor source: monitor only a
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* very specific sink input of the sink. This function needs to be
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* called before pa_stream_connect_record() is called. \since
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* 0.9.11 */
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int pa_stream_set_monitor_stream(pa_stream *s, uint32_t sink_input_idx);
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/** Return the sink input index previously set with
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* pa_stream_set_monitor_stream().
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* \since 0.9.11 */
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uint32_t pa_stream_get_monitor_stream(pa_stream *s);
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PA_C_DECL_END
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#endif
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