/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
*
* 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 (at your option) 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:gstaudio
* @title: GstAudio
* @short_description: Support library for audio elements
*
* This library contains some helper functions for audio elements.
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <string.h>
#include "audio.h"
#include "audio-enumtypes.h"
#ifndef GST_DISABLE_GST_DEBUG
#define GST_CAT_DEFAULT ensure_debug_category()
static GstDebugCategory *
ensure_debug_category (void)
{
static gsize cat_gonce = 0;
if (g_once_init_enter (&cat_gonce)) {
gsize cat_done;
cat_done = (gsize) _gst_debug_category_new ("audio", 0, "audio library");
g_once_init_leave (&cat_gonce, cat_done);
}
return (GstDebugCategory *) cat_gonce;
}
#else
#define ensure_debug_category() /* NOOP */
#endif /* GST_DISABLE_GST_DEBUG */
/**
* gst_audio_buffer_clip:
* @buffer: (transfer full): The buffer to clip.
* @segment: Segment in %GST_FORMAT_TIME or %GST_FORMAT_DEFAULT to which
* the buffer should be clipped.
* @rate: sample rate.
* @bpf: size of one audio frame in bytes. This is the size of one sample *
* number of channels.
*
* Clip the buffer to the given %GstSegment.
*
* After calling this function the caller does not own a reference to
* @buffer anymore.
*
* Returns: (transfer full): %NULL if the buffer is completely outside the configured segment,
* otherwise the clipped buffer is returned.
*
* If the buffer has no timestamp, it is assumed to be inside the segment and
* is not clipped
*/
GstBuffer *
gst_audio_buffer_clip (GstBuffer * buffer, const GstSegment * segment,
gint rate, gint bpf)
{
GstBuffer *ret;
GstAudioMeta *meta;
GstClockTime timestamp = GST_CLOCK_TIME_NONE, duration = GST_CLOCK_TIME_NONE;
guint64 offset = GST_BUFFER_OFFSET_NONE, offset_end = GST_BUFFER_OFFSET_NONE;
gsize trim, size, osize;
gboolean change_duration = TRUE, change_offset = TRUE, change_offset_end =
TRUE;
g_return_val_if_fail (segment->format == GST_FORMAT_TIME ||
segment->format == GST_FORMAT_DEFAULT, buffer);
g_return_val_if_fail (GST_IS_BUFFER (buffer), NULL);
if (!GST_BUFFER_TIMESTAMP_IS_VALID (buffer))
/* No timestamp - assume the buffer is completely in the segment */
return buffer;
/* Get copies of the buffer metadata to change later.
* Calculate the missing values for the calculations,
* they won't be changed later though. */
meta = gst_buffer_get_audio_meta (buffer);
/* these variables measure samples */
trim = 0;
osize = size = meta ? meta->samples : (gst_buffer_get_size (buffer) / bpf);
/* no data, nothing to clip */
if (!size)
return buffer;
timestamp = GST_BUFFER_TIMESTAMP (buffer);
GST_DEBUG ("timestamp %" GST_TIME_FORMAT, GST_TIME_ARGS (timestamp));
if (GST_BUFFER_DURATION_IS_VALID (buffer)) {
duration = GST_BUFFER_DURATION (buffer);
} else {
change_duration = FALSE;
duration = gst_util_uint64_scale (size, GST_SECOND, rate);
}
if (GST_BUFFER_OFFSET_IS_VALID (buffer)) {
offset = GST_BUFFER_OFFSET (buffer);
} else {
change_offset = FALSE;
offset = 0;
}
if (GST_BUFFER_OFFSET_END_IS_VALID (buffer)) {
offset_end = GST_BUFFER_OFFSET_END (buffer);
} else {
change_offset_end = FALSE;
offset_end = offset + size;
}
if (segment->format == GST_FORMAT_TIME) {
/* Handle clipping for GST_FORMAT_TIME */
guint64 start, stop, cstart, cstop, diff;
start = timestamp;
stop = timestamp + duration;
if (gst_segment_clip (segment, GST_FORMAT_TIME,
start, stop, &cstart, &cstop)) {
diff = cstart - start;
if (diff > 0) {
timestamp = cstart;
if (change_duration)
duration -= diff;
diff = gst_util_uint64_scale (diff, rate, GST_SECOND);
if (change_offset)
offset += diff;
trim += diff;
size -= diff;
}
diff = stop - cstop;
if (diff > 0) {
/* duration is always valid if stop is valid */
duration -= diff;
diff = gst_util_uint64_scale (diff, rate, GST_SECOND);
if (change_offset_end)
offset_end -= diff;
size -= diff;
}
} else {
gst_buffer_unref (buffer);
return NULL;
}
} else {
/* Handle clipping for GST_FORMAT_DEFAULT */
guint64 start, stop, cstart, cstop, diff;
g_return_val_if_fail (GST_BUFFER_OFFSET_IS_VALID (buffer), buffer);
start = offset;
stop = offset_end;
if (gst_segment_clip (segment, GST_FORMAT_DEFAULT,
start, stop, &cstart, &cstop)) {
diff = cstart - start;
if (diff > 0) {
offset = cstart;
timestamp = gst_util_uint64_scale (cstart, GST_SECOND, rate);
if (change_duration)
duration -= gst_util_uint64_scale (diff, GST_SECOND, rate);
trim += diff;
size -= diff;
}
diff = stop - cstop;
if (diff > 0) {
offset_end = cstop;
if (change_duration)
duration -= gst_util_uint64_scale (diff, GST_SECOND, rate);
size -= diff;
}
} else {
gst_buffer_unref (buffer);
return NULL;
}
}
if (trim == 0 && size == osize) {
ret = buffer;
if (GST_BUFFER_TIMESTAMP (ret) != timestamp) {
ret = gst_buffer_make_writable (ret);
GST_BUFFER_TIMESTAMP (ret) = timestamp;
}
if (GST_BUFFER_DURATION (ret) != duration) {
ret = gst_buffer_make_writable (ret);
GST_BUFFER_DURATION (ret) = duration;
}
} else {
/* cut out all the samples that are no longer relevant */
GST_DEBUG ("trim %" G_GSIZE_FORMAT " size %" G_GSIZE_FORMAT, trim, size);
ret = gst_audio_buffer_truncate (buffer, bpf, trim, size);
GST_DEBUG ("timestamp %" GST_TIME_FORMAT, GST_TIME_ARGS (timestamp));
if (ret) {
GST_BUFFER_TIMESTAMP (ret) = timestamp;
if (change_duration)
GST_BUFFER_DURATION (ret) = duration;
if (change_offset)
GST_BUFFER_OFFSET (ret) = offset;
if (change_offset_end)
GST_BUFFER_OFFSET_END (ret) = offset_end;
} else {
GST_ERROR ("gst_audio_buffer_truncate failed");
}
}
return ret;
}
/**
* gst_audio_buffer_truncate:
* @buffer: (transfer full): The buffer to truncate.
* @bpf: size of one audio frame in bytes. This is the size of one sample *
* number of channels.
* @trim: the number of samples to remove from the beginning of the buffer
* @samples: the final number of samples that should exist in this buffer or -1
* to use all the remaining samples if you are only removing samples from the
* beginning.
*
* Truncate the buffer to finally have @samples number of samples, removing
* the necessary amount of samples from the end and @trim number of samples
* from the beginning.
*
* After calling this function the caller does not own a reference to
* @buffer anymore.
*
* Returns: (transfer full): the truncated buffer or %NULL if the arguments
* were invalid
*
* Since: 1.16
*/
GstBuffer *
gst_audio_buffer_truncate (GstBuffer * buffer, gint bpf, gsize trim,
gsize samples)
{
GstAudioMeta *meta = NULL;
GstBuffer *ret = NULL;
gsize orig_samples;
gint i;
g_return_val_if_fail (GST_IS_BUFFER (buffer), NULL);
meta = gst_buffer_get_audio_meta (buffer);
orig_samples = meta ? meta->samples : gst_buffer_get_size (buffer) / bpf;
g_return_val_if_fail (trim < orig_samples, NULL);
g_return_val_if_fail (samples == -1 || trim + samples <= orig_samples, NULL);
if (samples == -1)
samples = orig_samples - trim;
/* nothing to truncate */
if (samples == orig_samples)
return buffer;
if (!meta || meta->info.layout == GST_AUDIO_LAYOUT_INTERLEAVED) {
/* interleaved */
ret = gst_buffer_copy_region (buffer, GST_BUFFER_COPY_ALL, trim * bpf,
samples * bpf);
gst_buffer_unref (buffer);
if ((meta = gst_buffer_get_audio_meta (ret)))
meta->samples = samples;
} else {
/* non-interleaved */
ret = gst_buffer_make_writable (buffer);
meta = gst_buffer_get_audio_meta (buffer);
meta->samples = samples;
for (i = 0; i < meta->info.channels; i++) {
meta->offsets[i] += trim * bpf / meta->info.channels;
}
}
return ret;
}