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Original commit message from CVS: * ext/a52dec/gsta52dec.c: (gst_a52dec_channels), (gst_a52dec_push), (gst_a52dec_reneg), (gst_a52dec_loop), (plugin_init): * ext/alsa/gstalsa.c: (gst_alsa_get_caps): * ext/alsa/gstalsaplugin.c: (plugin_init): * ext/dts/gstdtsdec.c: (gst_dtsdec_channels), (gst_dtsdec_renegotiate), (gst_dtsdec_loop), (plugin_init): * ext/faad/gstfaad.c: (gst_faad_init), (gst_faad_chanpos_from_gst), (gst_faad_chanpos_to_gst), (gst_faad_sinkconnect), (gst_faad_srcgetcaps), (gst_faad_srcconnect), (gst_faad_chain), (gst_faad_change_state), (plugin_init): * ext/faad/gstfaad.h: * ext/vorbis/vorbis.c: (plugin_init): * ext/vorbis/vorbisdec.c: (vorbis_dec_chain): * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/audio.c: (plugin_init): * gst-libs/gst/audio/multichannel.c: (gst_audio_check_channel_positions), (gst_audio_get_channel_positions), (gst_audio_set_channel_positions), (gst_audio_set_structure_channel_positions_list), (add_list_to_struct), (gst_audio_set_caps_channel_positions_list), (gst_audio_fixate_channel_positions): * gst-libs/gst/audio/multichannel.h: * gst-libs/gst/audio/testchannels.c: (main): * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_class_init), (gst_audio_convert_init), (gst_audio_convert_dispose), (gst_audio_convert_getcaps), (gst_audio_convert_parse_caps), (gst_audio_convert_link), (gst_audio_convert_fixate), (gst_audio_convert_channels): * gst/audioconvert/plugin.c: (plugin_init): Surround sound support.
280 lines
7.8 KiB
C
280 lines
7.8 KiB
C
/* GStreamer
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* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
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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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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#include "audio.h"
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#include "multichannel-enumtypes.h"
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#include <gst/gststructure.h>
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int
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gst_audio_frame_byte_size (GstPad * pad)
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{
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/* calculate byte size of an audio frame
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* this should be moved closer to the gstreamer core
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* and be implemented for every mime type IMO
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* returns -1 if there's an error (to avoid division by zero),
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* or the byte size if everything's ok
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*/
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int width = 0;
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int channels = 0;
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const GstCaps *caps = NULL;
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GstStructure *structure;
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/* get caps of pad */
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caps = GST_PAD_CAPS (pad);
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if (caps == NULL) {
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/* ERROR: could not get caps of pad */
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g_warning ("gstaudio: could not get caps of pad %s:%s\n",
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GST_ELEMENT_NAME (gst_pad_get_parent (pad)), GST_PAD_NAME (pad));
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return 0;
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}
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structure = gst_caps_get_structure (caps, 0);
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gst_structure_get_int (structure, "width", &width);
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gst_structure_get_int (structure, "channels", &channels);
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return (width / 8) * channels;
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}
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long
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gst_audio_frame_length (GstPad * pad, GstBuffer * buf)
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/* calculate length of buffer in frames
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* this should be moved closer to the gstreamer core
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* and be implemented for every mime type IMO
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* returns 0 if there's an error, or the number of frames if everything's ok
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*/
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{
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int frame_byte_size = 0;
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frame_byte_size = gst_audio_frame_byte_size (pad);
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if (frame_byte_size == 0)
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/* error */
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return 0;
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/* FIXME: this function assumes the buffer size to be a whole multiple
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* of the frame byte size
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*/
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return GST_BUFFER_SIZE (buf) / frame_byte_size;
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}
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long
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gst_audio_frame_rate (GstPad * pad)
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/*
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* calculate frame rate (based on caps of pad)
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* returns 0 if failed, rate if success
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*/
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{
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const GstCaps *caps = NULL;
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gint rate;
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GstStructure *structure;
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/* get caps of pad */
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caps = GST_PAD_CAPS (pad);
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if (caps == NULL) {
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/* ERROR: could not get caps of pad */
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g_warning ("gstaudio: could not get caps of pad %s:%s\n",
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GST_ELEMENT_NAME (gst_pad_get_parent (pad)), GST_PAD_NAME (pad));
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return 0;
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} else {
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structure = gst_caps_get_structure (caps, 0);
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gst_structure_get_int (structure, "rate", &rate);
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return rate;
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}
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}
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double
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gst_audio_length (GstPad * pad, GstBuffer * buf)
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{
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/* calculate length in seconds
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* of audio buffer buf
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* based on capabilities of pad
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*/
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long bytes = 0;
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int width = 0;
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int channels = 0;
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int rate = 0;
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double length;
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const GstCaps *caps = NULL;
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GstStructure *structure;
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g_assert (GST_IS_BUFFER (buf));
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/* get caps of pad */
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caps = GST_PAD_CAPS (pad);
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if (caps == NULL) {
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/* ERROR: could not get caps of pad */
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g_warning ("gstaudio: could not get caps of pad %s:%s\n",
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GST_ELEMENT_NAME (gst_pad_get_parent (pad)), GST_PAD_NAME (pad));
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length = 0.0;
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} else {
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structure = gst_caps_get_structure (caps, 0);
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bytes = GST_BUFFER_SIZE (buf);
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gst_structure_get_int (structure, "width", &width);
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gst_structure_get_int (structure, "channels", &channels);
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gst_structure_get_int (structure, "rate", &rate);
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g_assert (bytes != 0);
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g_assert (width != 0);
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g_assert (channels != 0);
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g_assert (rate != 0);
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length = (bytes * 8.0) / (double) (rate * channels * width);
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}
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/* g_print ("DEBUG: audio: returning length of %f\n", length); */
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return length;
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}
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long
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gst_audio_highest_sample_value (GstPad * pad)
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/* calculate highest possible sample value
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* based on capabilities of pad
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*/
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{
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gboolean is_signed = FALSE;
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gint width = 0;
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const GstCaps *caps = NULL;
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GstStructure *structure;
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caps = GST_PAD_CAPS (pad);
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if (caps == NULL) {
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g_warning ("gstaudio: could not get caps of pad %s:%s\n",
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GST_ELEMENT_NAME (gst_pad_get_parent (pad)), GST_PAD_NAME (pad));
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}
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structure = gst_caps_get_structure (caps, 0);
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gst_structure_get_int (structure, "width", &width);
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gst_structure_get_boolean (structure, "signed", &is_signed);
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if (is_signed)
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--width;
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/* example : 16 bit, signed : samples between -32768 and 32767 */
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return ((long) (1 << width));
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}
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gboolean
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gst_audio_is_buffer_framed (GstPad * pad, GstBuffer * buf)
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/* check if the buffer size is a whole multiple of the frame size */
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{
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if (GST_BUFFER_SIZE (buf) % gst_audio_frame_byte_size (pad) == 0)
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return TRUE;
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else
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return FALSE;
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}
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/* _getcaps helper functions
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* sets structure fields to default for audio type
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* flag determines which structure fields to set to default
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* keep these functions in sync with the templates in audio.h
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*/
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/* private helper function
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* sets a list on the structure
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* pass in structure, fieldname for the list, type of the list values,
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* number of list values, and each of the values, terminating with NULL
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*/
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static void
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_gst_audio_structure_set_list (GstStructure * structure,
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const gchar * fieldname, GType type, int number, ...)
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{
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va_list varargs;
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GValue value = { 0 };
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GArray *array;
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int j;
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g_return_if_fail (structure != NULL);
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g_value_init (&value, GST_TYPE_LIST);
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array = g_value_peek_pointer (&value);
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va_start (varargs, number);
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for (j = 0; j < number; ++j) {
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int i;
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gboolean b;
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GValue list_value = { 0 };
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switch (type) {
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case G_TYPE_INT:
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i = va_arg (varargs, int);
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g_value_init (&list_value, G_TYPE_INT);
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g_value_set_int (&list_value, i);
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break;
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case G_TYPE_BOOLEAN:
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b = va_arg (varargs, gboolean);
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g_value_init (&list_value, G_TYPE_BOOLEAN);
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g_value_set_boolean (&list_value, b);
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break;
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default:
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g_warning
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("_gst_audio_structure_set_list: LIST of given type not implemented.");
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}
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g_array_append_val (array, list_value);
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}
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gst_structure_set_value (structure, fieldname, &value);
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va_end (varargs);
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}
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void
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gst_audio_structure_set_int (GstStructure * structure, GstAudioFieldFlag flag)
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{
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if (flag & GST_AUDIO_FIELD_RATE)
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gst_structure_set (structure, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT,
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NULL);
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if (flag & GST_AUDIO_FIELD_CHANNELS)
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gst_structure_set (structure, "channels", GST_TYPE_INT_RANGE, 1, G_MAXINT,
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NULL);
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if (flag & GST_AUDIO_FIELD_ENDIANNESS)
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_gst_audio_structure_set_list (structure, "endianness", G_TYPE_INT, 2,
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G_LITTLE_ENDIAN, G_BIG_ENDIAN, NULL);
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if (flag & GST_AUDIO_FIELD_WIDTH)
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_gst_audio_structure_set_list (structure, "width", G_TYPE_INT, 3, 8, 16, 32,
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NULL);
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if (flag & GST_AUDIO_FIELD_DEPTH)
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gst_structure_set (structure, "depth", GST_TYPE_INT_RANGE, 1, 32, NULL);
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if (flag & GST_AUDIO_FIELD_SIGNED)
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_gst_audio_structure_set_list (structure, "signed", G_TYPE_BOOLEAN, 2, TRUE,
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FALSE, NULL);
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if (flag & GST_AUDIO_FIELD_BUFFER_FRAMES)
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gst_structure_set (structure, "buffer-frames", GST_TYPE_INT_RANGE, 1,
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G_MAXINT, NULL);
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}
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static gboolean
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plugin_init (GstPlugin * plugin)
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{
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gst_audio_channel_position_get_type ();
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return TRUE;
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}
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GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
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GST_VERSION_MINOR,
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"gstaudio",
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"Support services for audio plugins",
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plugin_init, VERSION, GST_LICENSE, GST_PACKAGE, GST_ORIGIN);
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