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93daa1435a
Since we started depending on GLib 2.44, we can be sure this macro is defined (it will be a no-op on compilers that don't support it). For plugins we should just start using `G_DECLARE_FINAL_TYPE` which means we no longer need the macro there, but for most types in base/gst-libs we don't want to break ABI, which means it's better to just keep it like it is (and use the `#ifdef` instead).
411 lines
15 KiB
C
411 lines
15 KiB
C
/* GStreamer
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* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
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* 2005 Wim Taymans <wim@fluendo.com>
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*
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* gstaudioringbuffer.h:
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but 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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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#ifndef __GST_AUDIO_AUDIO_H__
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#include <gst/audio/audio.h>
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#endif
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#ifndef __GST_AUDIO_RING_BUFFER_H__
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#define __GST_AUDIO_RING_BUFFER_H__
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G_BEGIN_DECLS
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#define GST_TYPE_AUDIO_RING_BUFFER (gst_audio_ring_buffer_get_type())
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#define GST_AUDIO_RING_BUFFER(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_RING_BUFFER,GstAudioRingBuffer))
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#define GST_AUDIO_RING_BUFFER_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_RING_BUFFER,GstAudioRingBufferClass))
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#define GST_AUDIO_RING_BUFFER_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_AUDIO_RING_BUFFER, GstAudioRingBufferClass))
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#define GST_AUDIO_RING_BUFFER_CAST(obj) ((GstAudioRingBuffer *)obj)
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#define GST_IS_AUDIO_RING_BUFFER(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_RING_BUFFER))
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#define GST_IS_AUDIO_RING_BUFFER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_RING_BUFFER))
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typedef struct _GstAudioRingBuffer GstAudioRingBuffer;
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typedef struct _GstAudioRingBufferClass GstAudioRingBufferClass;
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typedef struct _GstAudioRingBufferSpec GstAudioRingBufferSpec;
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/**
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* GstAudioRingBufferCallback:
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* @rbuf: a #GstAudioRingBuffer
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* @data: (array length=len): target to fill
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* @len: amount to fill
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* @user_data: user data
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*
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* This function is set with gst_audio_ring_buffer_set_callback() and is
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* called to fill the memory at @data with @len bytes of samples.
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*/
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typedef void (*GstAudioRingBufferCallback) (GstAudioRingBuffer *rbuf, guint8* data, guint len, gpointer user_data);
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/**
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* GstAudioRingBufferState:
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* @GST_AUDIO_RING_BUFFER_STATE_STOPPED: The ringbuffer is stopped
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* @GST_AUDIO_RING_BUFFER_STATE_PAUSED: The ringbuffer is paused
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* @GST_AUDIO_RING_BUFFER_STATE_STARTED: The ringbuffer is started
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* @GST_AUDIO_RING_BUFFER_STATE_ERROR: The ringbuffer has encountered an
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* error after it has been started, e.g. because the device was
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* disconnected (Since: 1.2)
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*
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* The state of the ringbuffer.
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*/
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typedef enum {
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GST_AUDIO_RING_BUFFER_STATE_STOPPED,
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GST_AUDIO_RING_BUFFER_STATE_PAUSED,
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GST_AUDIO_RING_BUFFER_STATE_STARTED,
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GST_AUDIO_RING_BUFFER_STATE_ERROR
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} GstAudioRingBufferState;
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/**
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* GstAudioRingBufferFormatType:
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* @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW: samples in linear or float
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* @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MU_LAW: samples in mulaw
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* @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_A_LAW: samples in alaw
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* @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IMA_ADPCM: samples in ima adpcm
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* @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG: samples in mpeg audio (but not AAC) format
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* @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_GSM: samples in gsm format
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* @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IEC958: samples in IEC958 frames (e.g. AC3)
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* @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3: samples in AC3 format
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* @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_EAC3: samples in EAC3 format
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* @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DTS: samples in DTS format
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* @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC: samples in MPEG-2 AAC ADTS format
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* @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC: samples in MPEG-4 AAC ADTS format
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* @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC_RAW: samples in MPEG-2 AAC raw format (Since: 1.12)
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* @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC_RAW: samples in MPEG-4 AAC raw format (Since: 1.12)
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* @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_FLAC: samples in FLAC format (Since: 1.12)
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*
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* The format of the samples in the ringbuffer.
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*/
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typedef enum
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{
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GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW,
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GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MU_LAW,
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GST_AUDIO_RING_BUFFER_FORMAT_TYPE_A_LAW,
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GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IMA_ADPCM,
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GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG,
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GST_AUDIO_RING_BUFFER_FORMAT_TYPE_GSM,
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GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IEC958,
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GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3,
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GST_AUDIO_RING_BUFFER_FORMAT_TYPE_EAC3,
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GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DTS,
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GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC,
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GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC,
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GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC_RAW,
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GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC_RAW,
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GST_AUDIO_RING_BUFFER_FORMAT_TYPE_FLAC
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} GstAudioRingBufferFormatType;
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/**
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* GstAudioRingBufferSpec:
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* @caps: The caps that generated the Spec.
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* @type: the sample type
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* @info: the #GstAudioInfo
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* @latency_time: the latency in microseconds
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* @buffer_time: the total buffer size in microseconds
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* @segsize: the size of one segment in bytes
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* @segtotal: the total number of segments
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* @seglatency: number of segments queued in the lower level device,
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* defaults to segtotal
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*
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* The structure containing the format specification of the ringbuffer.
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*/
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struct _GstAudioRingBufferSpec
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{
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/*< public >*/
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/* in */
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GstCaps *caps; /* the caps of the buffer */
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/* in/out */
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GstAudioRingBufferFormatType type;
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GstAudioInfo info;
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guint64 latency_time; /* the required/actual latency time, this is the
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* actual the size of one segment and the
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* minimum possible latency we can achieve. */
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guint64 buffer_time; /* the required/actual time of the buffer, this is
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* the total size of the buffer and maximum
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* latency we can compensate for. */
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gint segsize; /* size of one buffer segment in bytes, this value
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* should be chosen to match latency_time as
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* well as possible. */
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gint segtotal; /* total number of segments, this value is the
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* number of segments of @segsize and should be
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* chosen so that it matches buffer_time as
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* close as possible. */
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/* ABI added 0.10.20 */
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gint seglatency; /* number of segments queued in the lower
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* level device, defaults to segtotal. */
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/*< private >*/
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gpointer _gst_reserved[GST_PADDING];
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};
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#define GST_AUDIO_RING_BUFFER_GET_COND(buf) (&(((GstAudioRingBuffer *)buf)->cond))
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#define GST_AUDIO_RING_BUFFER_WAIT(buf) (g_cond_wait (GST_AUDIO_RING_BUFFER_GET_COND (buf), GST_OBJECT_GET_LOCK (buf)))
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#define GST_AUDIO_RING_BUFFER_SIGNAL(buf) (g_cond_signal (GST_AUDIO_RING_BUFFER_GET_COND (buf)))
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#define GST_AUDIO_RING_BUFFER_BROADCAST(buf)(g_cond_broadcast (GST_AUDIO_RING_BUFFER_GET_COND (buf)))
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/**
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* GstAudioRingBuffer:
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* @cond: used to signal start/stop/pause/resume actions
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* @open: boolean indicating that the ringbuffer is open
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* @acquired: boolean indicating that the ringbuffer is acquired
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* @memory: data in the ringbuffer
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* @size: size of data in the ringbuffer
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* @spec: format and layout of the ringbuffer data
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* @samples_per_seg: number of samples in one segment
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* @empty_seg: pointer to memory holding one segment of silence samples
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* @state: state of the buffer
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* @segdone: readpointer in the ringbuffer
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* @segbase: segment corresponding to segment 0 (unused)
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* @waiting: is a reader or writer waiting for a free segment
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*
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* The ringbuffer base class structure.
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*/
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struct _GstAudioRingBuffer {
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GstObject object;
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/*< public >*/ /* with LOCK */
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GCond cond;
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gboolean open;
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gboolean acquired;
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guint8 *memory;
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gsize size;
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/*< private >*/
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GstClockTime *timestamps;
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/*< public >*/ /* with LOCK */
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GstAudioRingBufferSpec spec;
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gint samples_per_seg;
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guint8 *empty_seg;
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/*< public >*/ /* ATOMIC */
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gint state;
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gint segdone;
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gint segbase;
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gint waiting;
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/*< private >*/
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GstAudioRingBufferCallback callback;
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gpointer cb_data;
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gboolean need_reorder;
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/* gst[channel_reorder_map[i]] = device[i] */
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gint channel_reorder_map[64];
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gboolean flushing;
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/* ATOMIC */
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gint may_start;
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gboolean active;
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GDestroyNotify cb_data_notify;
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/*< private >*/
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gpointer _gst_reserved[GST_PADDING - 1];
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};
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/**
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* GstAudioRingBufferClass:
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* @parent_class: parent class
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* @open_device: open the device, don't set any params or allocate anything
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* @acquire: allocate the resources for the ringbuffer using the given spec
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* @release: free resources of the ringbuffer
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* @close_device: close the device
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* @start: start processing of samples
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* @pause: pause processing of samples
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* @resume: resume processing of samples after pause
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* @stop: stop processing of samples
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* @delay: get number of frames queued in device
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* @activate: activate the thread that starts pulling and monitoring the
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* consumed segments in the device.
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* @commit: write samples into the ringbuffer
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* @clear_all: clear the entire ringbuffer.
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*
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* The vmethods that subclasses can override to implement the ringbuffer.
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*/
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struct _GstAudioRingBufferClass {
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GstObjectClass parent_class;
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/*< public >*/
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gboolean (*open_device) (GstAudioRingBuffer *buf);
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gboolean (*acquire) (GstAudioRingBuffer *buf, GstAudioRingBufferSpec *spec);
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gboolean (*release) (GstAudioRingBuffer *buf);
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gboolean (*close_device) (GstAudioRingBuffer *buf);
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gboolean (*start) (GstAudioRingBuffer *buf);
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gboolean (*pause) (GstAudioRingBuffer *buf);
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gboolean (*resume) (GstAudioRingBuffer *buf);
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gboolean (*stop) (GstAudioRingBuffer *buf);
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guint (*delay) (GstAudioRingBuffer *buf);
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/* ABI added */
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gboolean (*activate) (GstAudioRingBuffer *buf, gboolean active);
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guint (*commit) (GstAudioRingBuffer * buf, guint64 *sample,
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guint8 * data, gint in_samples,
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gint out_samples, gint * accum);
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void (*clear_all) (GstAudioRingBuffer * buf);
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/*< private >*/
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gpointer _gst_reserved[GST_PADDING];
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};
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GST_AUDIO_API
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GType gst_audio_ring_buffer_get_type(void);
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/* callback stuff */
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GST_AUDIO_API
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void gst_audio_ring_buffer_set_callback (GstAudioRingBuffer *buf,
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GstAudioRingBufferCallback cb,
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gpointer user_data);
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GST_AUDIO_API
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void gst_audio_ring_buffer_set_callback_full (GstAudioRingBuffer *buf,
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GstAudioRingBufferCallback cb,
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gpointer user_data,
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GDestroyNotify notify);
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GST_AUDIO_API
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gboolean gst_audio_ring_buffer_parse_caps (GstAudioRingBufferSpec *spec, GstCaps *caps);
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GST_AUDIO_API
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void gst_audio_ring_buffer_debug_spec_caps (GstAudioRingBufferSpec *spec);
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GST_AUDIO_API
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void gst_audio_ring_buffer_debug_spec_buff (GstAudioRingBufferSpec *spec);
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GST_AUDIO_API
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gboolean gst_audio_ring_buffer_convert (GstAudioRingBuffer * buf, GstFormat src_fmt,
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gint64 src_val, GstFormat dest_fmt,
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gint64 * dest_val);
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/* device state */
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GST_AUDIO_API
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gboolean gst_audio_ring_buffer_open_device (GstAudioRingBuffer *buf);
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GST_AUDIO_API
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gboolean gst_audio_ring_buffer_close_device (GstAudioRingBuffer *buf);
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GST_AUDIO_API
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gboolean gst_audio_ring_buffer_device_is_open (GstAudioRingBuffer *buf);
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/* allocate resources */
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GST_AUDIO_API
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gboolean gst_audio_ring_buffer_acquire (GstAudioRingBuffer *buf, GstAudioRingBufferSpec *spec);
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GST_AUDIO_API
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gboolean gst_audio_ring_buffer_release (GstAudioRingBuffer *buf);
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GST_AUDIO_API
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gboolean gst_audio_ring_buffer_is_acquired (GstAudioRingBuffer *buf);
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/* set the device channel positions */
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GST_AUDIO_API
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void gst_audio_ring_buffer_set_channel_positions (GstAudioRingBuffer *buf, const GstAudioChannelPosition *position);
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/* activating */
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GST_AUDIO_API
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gboolean gst_audio_ring_buffer_activate (GstAudioRingBuffer *buf, gboolean active);
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GST_AUDIO_API
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gboolean gst_audio_ring_buffer_is_active (GstAudioRingBuffer *buf);
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/* flushing */
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GST_AUDIO_API
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void gst_audio_ring_buffer_set_flushing (GstAudioRingBuffer *buf, gboolean flushing);
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GST_AUDIO_API
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gboolean gst_audio_ring_buffer_is_flushing (GstAudioRingBuffer *buf);
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/* playback/pause */
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GST_AUDIO_API
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gboolean gst_audio_ring_buffer_start (GstAudioRingBuffer *buf);
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GST_AUDIO_API
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gboolean gst_audio_ring_buffer_pause (GstAudioRingBuffer *buf);
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GST_AUDIO_API
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gboolean gst_audio_ring_buffer_stop (GstAudioRingBuffer *buf);
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/* get status */
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GST_AUDIO_API
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guint gst_audio_ring_buffer_delay (GstAudioRingBuffer *buf);
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GST_AUDIO_API
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guint64 gst_audio_ring_buffer_samples_done (GstAudioRingBuffer *buf);
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GST_AUDIO_API
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void gst_audio_ring_buffer_set_sample (GstAudioRingBuffer *buf, guint64 sample);
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/* clear all segments */
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GST_AUDIO_API
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void gst_audio_ring_buffer_clear_all (GstAudioRingBuffer *buf);
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/* commit samples */
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GST_AUDIO_API
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guint gst_audio_ring_buffer_commit (GstAudioRingBuffer * buf, guint64 *sample,
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guint8 * data, gint in_samples,
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gint out_samples, gint * accum);
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/* read samples */
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GST_AUDIO_API
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guint gst_audio_ring_buffer_read (GstAudioRingBuffer *buf, guint64 sample,
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guint8 *data, guint len, GstClockTime *timestamp);
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/* Set timestamp on buffer */
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GST_AUDIO_API
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void gst_audio_ring_buffer_set_timestamp (GstAudioRingBuffer * buf, gint readseg, GstClockTime
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timestamp);
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/* mostly protected */
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/* not yet implemented
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gboolean gst_audio_ring_buffer_prepare_write (GstAudioRingBuffer *buf, gint *segment, guint8 **writeptr, gint *len);
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*/
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GST_AUDIO_API
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gboolean gst_audio_ring_buffer_prepare_read (GstAudioRingBuffer *buf, gint *segment,
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guint8 **readptr, gint *len);
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GST_AUDIO_API
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void gst_audio_ring_buffer_clear (GstAudioRingBuffer *buf, gint segment);
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GST_AUDIO_API
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void gst_audio_ring_buffer_advance (GstAudioRingBuffer *buf, guint advance);
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GST_AUDIO_API
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void gst_audio_ring_buffer_may_start (GstAudioRingBuffer *buf, gboolean allowed);
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G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstAudioRingBuffer, gst_object_unref)
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G_END_DECLS
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#endif /* __GST_AUDIO_RING_BUFFER_H__ */
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