gstreamer/gst/audioresample/resample.c

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/* Resampling library
* Copyright (C) <2001> David A. Schleef <ds@schleef.org>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif
#include <string.h>
#include <math.h>
#include <stdio.h>
#include <stdlib.h>
#include <limits.h>
#include <liboil/liboil.h>
#include "resample.h"
#include "buffer.h"
#include "debug.h"
void resample_scale_ref (ResampleState * r);
void resample_scale_functable (ResampleState * r);
GST_DEBUG_CATEGORY (libaudioresample_debug);
void
resample_init (void)
{
static int inited = 0;
if (!inited) {
oil_init ();
inited = 1;
GST_DEBUG_CATEGORY_INIT (libaudioresample_debug, "libaudioresample", 0,
"audio resampling library");
}
}
ResampleState *
resample_new (void)
{
ResampleState *r;
r = malloc (sizeof (ResampleState));
memset (r, 0, sizeof (ResampleState));
r->filter_length = 16;
r->i_start = 0;
if (r->filter_length & 1) {
r->o_start = 0;
} else {
r->o_start = r->o_inc * 0.5;
}
r->queue = audioresample_buffer_queue_new ();
r->out_tmp = malloc (10000 * sizeof (double));
r->need_reinit = 1;
return r;
}
void
resample_free (ResampleState * r)
{
if (r->buffer) {
free (r->buffer);
}
if (r->ft) {
functable_free (r->ft);
}
if (r->queue) {
audioresample_buffer_queue_free (r->queue);
}
if (r->out_tmp) {
free (r->out_tmp);
}
free (r);
}
static void
resample_buffer_free (AudioresampleBuffer * buffer, void *priv)
{
if (buffer->priv2) {
((void (*)(void *)) buffer->priv2) (buffer->priv);
}
}
/*
* free_func: a function that frees the given closure. If NULL, caller is
* responsible for freeing.
*/
void
resample_add_input_data (ResampleState * r, void *data, int size,
void (*free_func) (void *), void *closure)
{
AudioresampleBuffer *buffer;
RESAMPLE_DEBUG ("data %p size %d", data, size);
buffer = audioresample_buffer_new_with_data (data, size);
buffer->free = resample_buffer_free;
ext/gnomevfs/: Fix URI interface implementation return type. Original commit message from CVS: 2006-10-10 Zaheer Abbas Merali <zaheerabbas at merali dot org> Patch by: Josep Torre Valles <josep@fluendo.com> * ext/gnomevfs/gstgnomevfssink.c: * ext/gnomevfs/gstgnomevfssrc.c: Fix URI interface implementation return type. * ext/pango/gsttextoverlay.c: (gst_text_overlay_set_property): Fix what looks like a copy/paste issue when assigning values. * gst-libs/gst/audio/gstaudiofiltertemplate.c: (gst_audio_filter_template_get_type): Cast to prevent Forte warnings. * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create): Fix URI interface implementation return type. gst_pad_query_position requires a signed integer pointer as 3rd parameter, GstClockTime is unsigned. * gst/audioconvert/audioconvert.c: Fix integer overflow when treated as signed. * gst/audioresample/resample.c: (resample_add_input_data): Cast to prevent warnings on Forte. * gst/ffmpegcolorspace/imgconvert.c: (build_rgb_palette): Fix integer overflow when treated as signed. * gst/ffmpegcolorspace/imgconvert_template.h: Fix integer overflow when treated as signed. RGBA_OUT shifts bits. * gst/playback/gstdecodebin.c: (queue_filled_cb), (cleanup_decodebin): Who initialises a guint to -1! Cast function pointers to prevent warnings on Forte. * gst/playback/gstplaybasebin.c: (queue_deadlock_check), (queue_threshold_reached): Cast function pointers correctly to prevent warnings on Forte. * gst/playback/gststreaminfo.c: (gst_stream_info_dispose): Cast function pointers correctly to prevent warnings on Forte. * gst/subparse/gstssaparse.c: (gst_ssa_parse_setcaps): Obvious change to unsigned, 0xEF > max signed char. * gst/tcp/gstmultifdsink.c: (get_buffers_max), (count_burst_unit): GstClockTime is unsigned, initialise correctly. * gst/tcp/gsttcp.c: (gst_tcp_socket_write): Cast so pointer arithemetic doesn't cause warnings on Forte. * gst/videorate/gstvideorate.c: Use correct return value. * tests/examples/seek/scrubby.c: GstClockTime is unsigned, initialise correctly.
2006-10-10 12:49:02 +00:00
buffer->priv2 = (void *) free_func;
buffer->priv = closure;
audioresample_buffer_queue_push (r->queue, buffer);
}
void
resample_input_flush (ResampleState * r)
{
RESAMPLE_DEBUG ("flush");
audioresample_buffer_queue_flush (r->queue);
r->buffer_filled = 0;
r->need_reinit = 1;
}
void
resample_input_pushthrough (ResampleState * r)
{
AudioresampleBuffer *buffer;
int filter_bytes;
int buffer_filled;
if (r->sample_size == 0)
return;
filter_bytes = r->filter_length * r->sample_size;
buffer_filled = r->buffer_filled;
RESAMPLE_DEBUG ("pushthrough filter_bytes %d, filled %d",
filter_bytes, buffer_filled);
/* if we have no pending samples, we don't need to do anything. */
if (buffer_filled <= 0)
return;
/* send filter_length/2 number of samples so we can get to the
* last queued samples */
buffer = audioresample_buffer_new_and_alloc (filter_bytes / 2);
memset (buffer->data, 0, buffer->length);
Printf format fixes. Original commit message from CVS: * ext/alsa/gstalsadeviceprobe.c: (gst_alsa_device_property_probe_get_values): * ext/alsa/gstalsasink.c: (set_hwparams): * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_elem_pad), (gst_ogg_chain_new_stream), (gst_ogg_demux_read_chain): * ext/ogg/gstoggmux.c: (gst_ogg_mux_send_headers), (gst_ogg_mux_process_best_pad): * ext/ogg/gstoggparse.c: (gst_ogg_parse_new_stream), (gst_ogg_parse_chain): * ext/ogg/gstogmparse.c: (gst_ogm_parse_stream_header): * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet): * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_setup), (gst_vorbis_enc_buffer_check_discontinuous): * ext/vorbis/vorbisparse.c: (vorbis_parse_src_query): * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_render): * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_handle_track_seek): * gst-libs/gst/rtp/gstbasertpdepayload.c: (gst_base_rtp_depayload_push_full): * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push): * gst/audioresample/resample.c: (resample_input_pushthrough): * gst/playback/gstplaybasebin.c: (queue_out_of_data): * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_handle_clients): * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset), (wavpack_type_find): * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_create): * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy), (gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new): * tests/check/elements/volume.c: (GST_START_TEST): Printf format fixes.
2006-10-05 15:55:20 +00:00
RESAMPLE_DEBUG ("pushthrough %u", buffer->length);
audioresample_buffer_queue_push (r->queue, buffer);
}
void
resample_input_eos (ResampleState * r)
{
RESAMPLE_DEBUG ("EOS");
resample_input_pushthrough (r);
r->eos = 1;
}
int
resample_get_output_size_for_input (ResampleState * r, int size)
{
int outsize;
double outd;
int avail;
int filter_bytes;
int buffer_filled;
if (r->sample_size == 0)
return 0;
filter_bytes = r->filter_length * r->sample_size;
buffer_filled = filter_bytes / 2 - r->buffer_filled / 2;
avail =
audioresample_buffer_queue_get_depth (r->queue) + size - buffer_filled;
RESAMPLE_DEBUG ("avail %d, o_rate %f, i_rate %f, filter_bytes %d, filled %d",
avail, r->o_rate, r->i_rate, filter_bytes, buffer_filled);
if (avail <= 0)
return 0;
outd = (double) avail *r->o_rate / r->i_rate;
outsize = (int) floor (outd);
/* round off for sample size */
outsize -= outsize % r->sample_size;
return outsize;
}
int
resample_get_input_size_for_output (ResampleState * r, int size)
{
int outsize;
double outd;
int avail;
if (r->sample_size == 0)
return 0;
avail = size;
RESAMPLE_DEBUG ("size %d, o_rate %f, i_rate %f", avail, r->o_rate, r->i_rate);
outd = (double) avail *r->i_rate / r->o_rate;
outsize = (int) ceil (outd);
/* round off for sample size */
outsize -= outsize % r->sample_size;
return outsize;
}
int
resample_get_output_size (ResampleState * r)
{
return resample_get_output_size_for_input (r, 0);
}
int
resample_get_output_data (ResampleState * r, void *data, int size)
{
r->o_buf = data;
r->o_size = size;
if (size == 0)
return 0;
switch (r->method) {
case 0:
resample_scale_ref (r);
break;
case 1:
resample_scale_functable (r);
break;
default:
break;
}
return size - r->o_size;
}
void
resample_set_filter_length (ResampleState * r, int length)
{
r->filter_length = length;
r->need_reinit = 1;
}
void
resample_set_input_rate (ResampleState * r, double rate)
{
r->i_rate = rate;
r->need_reinit = 1;
}
void
resample_set_output_rate (ResampleState * r, double rate)
{
r->o_rate = rate;
r->need_reinit = 1;
}
void
resample_set_n_channels (ResampleState * r, int n_channels)
{
r->n_channels = n_channels;
r->sample_size = r->n_channels * resample_format_size (r->format);
r->need_reinit = 1;
}
void
resample_set_format (ResampleState * r, ResampleFormat format)
{
r->format = format;
r->sample_size = r->n_channels * resample_format_size (r->format);
r->need_reinit = 1;
}
void
resample_set_method (ResampleState * r, int method)
{
r->method = method;
r->need_reinit = 1;
}
int
resample_format_size (ResampleFormat format)
{
switch (format) {
case RESAMPLE_FORMAT_S16:
return 2;
case RESAMPLE_FORMAT_S32:
case RESAMPLE_FORMAT_F32:
return 4;
case RESAMPLE_FORMAT_F64:
return 8;
}
return 0;
}