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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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235ea5989c
Original commit message from CVS: Make ringbuffer faster and more simple by removing the locks in the playback thread. Add sample accurate playback based on buffer sample offsets. Make the baseaudiosink provide a clock. Parse caps in the base class. Correctly handle seeking, flushing and state changes.
129 lines
4.6 KiB
C
129 lines
4.6 KiB
C
/* GStreamer
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* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
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* Library <2001> Thomas Vander Stichele <thomas@apestaart.org>
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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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#include <gst/gst.h>
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#ifndef __GST_AUDIO_AUDIO_H__
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#define __GST_AUDIO_AUDIO_H__
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G_BEGIN_DECLS
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/* For people that are looking at this source: the purpose of these defines is
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* to make GstCaps a bit easier, in that you don't have to know all of the
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* properties that need to be defined. you can just use these macros. currently
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* (8/01) the only plugins that use these are the passthrough, speed, volume,
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* adder, and [de]interleave plugins. These are for convenience only, and do not
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* specify the 'limits' of GStreamer. you might also use these definitions as a
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* base for making your own caps, if need be.
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*
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* For example, to make a source pad that can output streams of either mono
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* float or any channel int:
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*
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* template = gst_pad_template_new
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* ("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
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* gst_caps_append(gst_caps_new ("sink_int", "audio/x-raw-int",
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* GST_AUDIO_INT_PAD_TEMPLATE_PROPS),
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* gst_caps_new ("sink_float", "audio/x-raw-float",
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* GST_AUDIO_FLOAT_PAD_TEMPLATE_PROPS)),
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* NULL);
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*
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* sinkpad = gst_pad_new_from_template(template, "sink");
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*
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* Andy Wingo, 18 August 2001
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* Thomas, 6 September 2002 */
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#define GST_AUDIO_DEF_RATE 44100
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#define GST_AUDIO_INT_PAD_TEMPLATE_CAPS \
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"audio/x-raw-int, " \
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"rate = (int) [ 1, MAX ], " \
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"channels = (int) [ 1, MAX ], " \
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"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
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"width = (int) { 8, 16, 24, 32 }, " \
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"depth = (int) [ 1, 32 ], " \
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"signed = (boolean) { true, false }"
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/* "standard" int audio is native order, 16 bit stereo. */
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#define GST_AUDIO_INT_STANDARD_PAD_TEMPLATE_CAPS \
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"audio/x-raw-int, " \
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"rate = (int) [ 1, MAX ], " \
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"channels = (int) 2, " \
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"endianness = (int) BYTE_ORDER, " \
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"width = (int) 16, " \
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"depth = (int) 16, " \
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"signed = (boolean) true"
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#define GST_AUDIO_FLOAT_PAD_TEMPLATE_CAPS \
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"audio/x-raw-float, " \
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"rate = (int) [ 1, MAX ], " \
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"channels = (int) [ 1, MAX ], " \
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"endianness = (int) { LITTLE_ENDIAN , BIG_ENDIAN }, " \
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"width = (int) { 32, 64 }, " \
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"buffer-frames = (int) [ 1, MAX]"
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/* "standard" float audio is native order, 32 bit mono. */
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#define GST_AUDIO_FLOAT_STANDARD_PAD_TEMPLATE_CAPS \
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"audio/x-raw-float, " \
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"width = (int) 32, " \
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"rate = (int) [ 1, MAX ], " \
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"channels = (int) 1, " \
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"endianness = (int) BYTE_ORDER, " \
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"buffer-frames = (int) [ 1, MAX]"
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/*
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* this library defines and implements some helper functions for audio
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* handling
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*/
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/* get byte size of audio frame (based on caps of pad */
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int gst_audio_frame_byte_size (GstPad* pad);
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/* get length in frames of buffer */
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long gst_audio_frame_length (GstPad* pad, GstBuffer* buf);
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/* get frame rate based on caps */
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long gst_audio_frame_rate (GstPad *pad);
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/* calculate length in seconds of audio buffer buf based on caps of pad */
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double gst_audio_length (GstPad* pad, GstBuffer* buf);
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/* calculate highest possible sample value based on capabilities of pad */
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long gst_audio_highest_sample_value (GstPad* pad);
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/* check if the buffer size is a whole multiple of the frame size */
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gboolean gst_audio_is_buffer_framed (GstPad* pad, GstBuffer* buf);
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/* functions useful for _getcaps functions */
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typedef enum {
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GST_AUDIO_FIELD_RATE = (1 << 0),
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GST_AUDIO_FIELD_CHANNELS = (1 << 1),
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GST_AUDIO_FIELD_ENDIANNESS = (1 << 2),
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GST_AUDIO_FIELD_WIDTH = (1 << 3),
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GST_AUDIO_FIELD_DEPTH = (1 << 4),
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GST_AUDIO_FIELD_SIGNED = (1 << 5),
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GST_AUDIO_FIELD_BUFFER_FRAMES = (1 << 6)
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} GstAudioFieldFlag;
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void gst_audio_structure_set_int (GstStructure *structure, GstAudioFieldFlag flag);
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G_END_DECLS
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#endif /* __GST_AUDIO_AUDIO_H__ */
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