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gstreamer/gst-libs/gst/audio/audio.c
Wim Taymans f04b8fd8af audio/video add descriptions 
Add a description to the audio and video format info in case we want to use this
later.
2011-09-06 16:46:48 +02:00

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/* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/**
* SECTION: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"
#include <gst/gststructure.h>
#define SINT (GST_AUDIO_FORMAT_FLAG_INTEGER | GST_AUDIO_FORMAT_FLAG_SIGNED)
#define UINT (GST_AUDIO_FORMAT_FLAG_INTEGER)
#define MAKE_FORMAT(str,desc,flags,end,width,depth,silent) \
{ GST_AUDIO_FORMAT_ ##str, G_STRINGIFY(str), desc, flags, end, width, depth, silent }
#define SILENT_0 { 0, 0, 0, 0, 0, 0, 0, 0 }
#define SILENT_U8 { 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80 }
#define SILENT_U16LE { 0x00, 0x80, 0x00, 0x80, 0x00, 0x80, 0x00, 0x80 }
#define SILENT_U16BE { 0x80, 0x00, 0x80, 0x00, 0x80, 0x00, 0x80, 0x00 }
#define SILENT_U24_32LE { 0x00, 0x00, 0x80, 0x00, 0x00, 0x00, 0x80, 0x00 }
#define SILENT_U24_32BE { 0x00, 0x80, 0x00, 0x00, 0x00, 0x80, 0x00, 0x00 }
#define SILENT_U32LE { 0x00, 0x00, 0x00, 0x80, 0x00, 0x00, 0x00, 0x80 }
#define SILENT_U32BE { 0x80, 0x00, 0x00, 0x00, 0x80, 0x00, 0x00, 0x00 }
#define SILENT_U24LE { 0x00, 0x00, 0x80, 0x00, 0x00, 0x80 }
#define SILENT_U24BE { 0x80, 0x00, 0x00, 0x80, 0x00, 0x00 }
#define SILENT_U20LE { 0x00, 0x00, 0x08, 0x00, 0x00, 0x08 }
#define SILENT_U20BE { 0x08, 0x00, 0x00, 0x08, 0x00, 0x00 }
#define SILENT_U18LE { 0x00, 0x00, 0x02, 0x00, 0x00, 0x02 }
#define SILENT_U18BE { 0x02, 0x00, 0x00, 0x02, 0x00, 0x00 }
static GstAudioFormatInfo formats[] = {
{GST_AUDIO_FORMAT_UNKNOWN, "UNKNOWN", 0, 0, 0, 0},
/* 8 bit */
MAKE_FORMAT (S8, "8-bit signed PCM audio", SINT, 0, 8, 8, SILENT_0),
MAKE_FORMAT (U8, "8-bit unsigned PCM audio", UINT, 0, 8, 8, SILENT_U8),
/* 16 bit */
MAKE_FORMAT (S16LE, "16-bit signed PCM audio", SINT, G_LITTLE_ENDIAN, 16, 16,
SILENT_0),
MAKE_FORMAT (S16BE, "16-bit signed PCM audio", SINT, G_BIG_ENDIAN, 16, 16,
SILENT_0),
MAKE_FORMAT (U16LE, "16-bit unsigned PCM audio", UINT, G_LITTLE_ENDIAN, 16,
16, SILENT_U16LE),
MAKE_FORMAT (U16BE, "16-bit unsigned PCM audio", UINT, G_BIG_ENDIAN, 16, 16,
SILENT_U16BE),
/* 24 bit in low 3 bytes of 32 bits */
MAKE_FORMAT (S24_32LE, "24-bit signed PCM audio", SINT, G_LITTLE_ENDIAN, 32,
24, SILENT_0),
MAKE_FORMAT (S24_32BE, "24-bit signed PCM audio", SINT, G_BIG_ENDIAN, 32, 24,
SILENT_0),
MAKE_FORMAT (U24_32LE, "24-bit unsigned PCM audio", UINT, G_LITTLE_ENDIAN, 32,
24, SILENT_U24_32LE),
MAKE_FORMAT (U24_32BE, "24-bit unsigned PCM audio", UINT, G_BIG_ENDIAN, 32,
24, SILENT_U24_32BE),
/* 32 bit */
MAKE_FORMAT (S32LE, "32-bit signed PCM audio", SINT, G_LITTLE_ENDIAN, 32, 32,
SILENT_0),
MAKE_FORMAT (S32BE, "32-bit signed PCM audio", SINT, G_BIG_ENDIAN, 32, 32,
SILENT_0),
MAKE_FORMAT (U32LE, "32-bit unsigned PCM audio", UINT, G_LITTLE_ENDIAN, 32,
32, SILENT_U32LE),
MAKE_FORMAT (U32BE, "32-bit unsigned PCM audio", UINT, G_BIG_ENDIAN, 32, 32,
SILENT_U32BE),
/* 24 bit in 3 bytes */
MAKE_FORMAT (S24LE, "24-bit signed PCM audio", SINT, G_LITTLE_ENDIAN, 24, 24,
SILENT_0),
MAKE_FORMAT (S24BE, "24-bit signed PCM audio", SINT, G_BIG_ENDIAN, 24, 24,
SILENT_0),
MAKE_FORMAT (U24LE, "24-bit unsigned PCM audio", UINT, G_LITTLE_ENDIAN, 24,
24, SILENT_U24LE),
MAKE_FORMAT (U24BE, "24-bit unsigned PCM audio", UINT, G_BIG_ENDIAN, 24, 24,
SILENT_U24BE),
/* 20 bit in 3 bytes */
MAKE_FORMAT (S20LE, "20-bit signed PCM audio", SINT, G_LITTLE_ENDIAN, 24, 20,
SILENT_0),
MAKE_FORMAT (S20BE, "20-bit signed PCM audio", SINT, G_BIG_ENDIAN, 24, 20,
SILENT_0),
MAKE_FORMAT (U20LE, "20-bit unsigned PCM audio", UINT, G_LITTLE_ENDIAN, 24,
20, SILENT_U20LE),
MAKE_FORMAT (U20BE, "20-bit unsigned PCM audio", UINT, G_BIG_ENDIAN, 24, 20,
SILENT_U20BE),
/* 18 bit in 3 bytes */
MAKE_FORMAT (S18LE, "18-bit signed PCM audio", SINT, G_LITTLE_ENDIAN, 24, 18,
SILENT_0),
MAKE_FORMAT (S18BE, "18-bit signed PCM audio", SINT, G_BIG_ENDIAN, 24, 18,
SILENT_0),
MAKE_FORMAT (U18LE, "18-bit unsigned PCM audio", UINT, G_LITTLE_ENDIAN, 24,
18, SILENT_U18LE),
MAKE_FORMAT (U18BE, "18-bit unsigned PCM audio", UINT, G_BIG_ENDIAN, 24, 18,
SILENT_U18BE),
/* float */
MAKE_FORMAT (F32LE, "32-bit floating-point audio",
GST_AUDIO_FORMAT_FLAG_FLOAT, G_LITTLE_ENDIAN, 32, 32,
SILENT_0),
MAKE_FORMAT (F32BE, "32-bit floating-point audio",
GST_AUDIO_FORMAT_FLAG_FLOAT, G_BIG_ENDIAN, 32, 32,
SILENT_0),
MAKE_FORMAT (F64LE, "64-bit floating-point audio",
GST_AUDIO_FORMAT_FLAG_FLOAT, G_LITTLE_ENDIAN, 64, 64,
SILENT_0),
MAKE_FORMAT (F64BE, "64-bit floating-point audio",
GST_AUDIO_FORMAT_FLAG_FLOAT, G_BIG_ENDIAN, 64, 64,
SILENT_0)
};
/**
* gst_audio_format_build_int:
* @sign: signed or unsigned format
* @endianness: G_LITTLE_ENDIAN or G_BIG_ENDIAN
* @width: amount of bits used per sample
* @depth: amount of used bits in @width
*
* Construct a #GstAudioFormat with given parameters.
*
* Returns: a #GstAudioFormat or GST_AUDIO_FORMAT_UNKNOWN when no audio format
* exists with the given parameters.
*/
GstAudioFormat
gst_audio_format_build_integer (gboolean sign, gint endianness,
gint width, gint depth)
{
gint i, e;
for (i = 0; i < G_N_ELEMENTS (formats); i++) {
GstAudioFormatInfo *finfo = &formats[i];
/* must be int */
if (!GST_AUDIO_FORMAT_INFO_IS_INTEGER (finfo))
continue;
/* width and depth must match */
if (width != GST_AUDIO_FORMAT_INFO_WIDTH (finfo))
continue;
if (depth != GST_AUDIO_FORMAT_INFO_DEPTH (finfo))
continue;
/* if there is endianness, it must match */
e = GST_AUDIO_FORMAT_INFO_ENDIANNESS (finfo);
if (e && e != endianness)
continue;
/* check sign */
if (sign && !GST_AUDIO_FORMAT_INFO_IS_SIGNED (finfo))
continue;
return GST_AUDIO_FORMAT_INFO_FORMAT (finfo);
}
return GST_AUDIO_FORMAT_UNKNOWN;
}
/**
* gst_audio_format_from_string:
* @format: a format string
*
* Convert the @format string to its #GstAudioFormat.
*
* Returns: the #GstAudioFormat for @format or GST_AUDIO_FORMAT_UNKNOWN when the
* string is not a known format.
*/
GstAudioFormat
gst_audio_format_from_string (const gchar * format)
{
guint i;
for (i = 0; i < G_N_ELEMENTS (formats); i++) {
if (strcmp (GST_AUDIO_FORMAT_INFO_NAME (&formats[i]), format) == 0)
return GST_AUDIO_FORMAT_INFO_FORMAT (&formats[i]);
}
return GST_AUDIO_FORMAT_UNKNOWN;
}
const gchar *
gst_audio_format_to_string (GstAudioFormat format)
{
g_return_val_if_fail (format != GST_AUDIO_FORMAT_UNKNOWN, NULL);
if (format >= G_N_ELEMENTS (formats))
return NULL;
return GST_AUDIO_FORMAT_INFO_NAME (&formats[format]);
}
/**
* gst_audio_format_get_info:
* @format: a #GstAudioFormat
*
* Get the #GstAudioFormatInfo for @format
*
* Returns: The #GstAudioFormatInfo for @format.
*/
const GstAudioFormatInfo *
gst_audio_format_get_info (GstAudioFormat format)
{
g_return_val_if_fail (format != GST_AUDIO_FORMAT_UNKNOWN, NULL);
g_return_val_if_fail (format < G_N_ELEMENTS (formats), NULL);
return &formats[format];
}
/**
* gst_audio_format_fill_silence:
* @info: a #GstAudioFormatInfo
* @dest: a destination to fill
* @length: the length to fill
*
* Fill @length bytes in @dest with silence samples for @info.
*/
void
gst_audio_format_fill_silence (const GstAudioFormatInfo * info,
gpointer dest, gsize length)
{
guint8 *dptr = dest;
g_return_if_fail (info != NULL);
g_return_if_fail (dest != NULL);
if (info->flags & GST_AUDIO_FORMAT_FLAG_FLOAT ||
info->flags & GST_AUDIO_FORMAT_FLAG_SIGNED) {
/* float or signed always 0 */
memset (dest, 0, length);
} else {
gint i, j, bps = info->width >> 3;
switch (bps) {
case 1:
memset (dest, info->silence[0], length);
break;
default:
for (i = 0; i < length; i += bps) {
for (j = 0; j < bps; j++)
*dptr++ = info->silence[j];
}
break;
}
}
}
/**
* gst_audio_info_init:
* @info: a #GstAudioInfo
*
* Initialize @info with default values.
*/
void
gst_audio_info_init (GstAudioInfo * info)
{
g_return_if_fail (info != NULL);
memset (info, 0, sizeof (GstAudioInfo));
}
/**
* gst_audio_info_set_format:
* @info: a #GstAudioInfo
* @format: the format
* @rate: the samplerate
* @channels: the number of channels
*
* Set the default info for the audio info of @format and @rate and @channels.
*/
void
gst_audio_info_set_format (GstAudioInfo * info, GstAudioFormat format,
gint rate, gint channels)
{
const GstAudioFormatInfo *finfo;
g_return_if_fail (info != NULL);
g_return_if_fail (format != GST_AUDIO_FORMAT_UNKNOWN);
finfo = &formats[format];
info->flags = 0;
info->finfo = finfo;
info->rate = rate;
info->channels = channels;
info->bpf = (finfo->width * channels) / 8;
}
/**
* gst_audio_info_from_caps:
* @info: a #GstAudioInfo
* @caps: a #GstCaps
*
* Parse @caps and update @info.
*
* Returns: TRUE if @caps could be parsed
*/
gboolean
gst_audio_info_from_caps (GstAudioInfo * info, const GstCaps * caps)
{
GstStructure *str;
const gchar *s;
GstAudioFormat format;
gint rate, channels;
const GValue *pos_val_arr, *pos_val_entry;
gint i;
g_return_val_if_fail (info != NULL, FALSE);
g_return_val_if_fail (caps != NULL, FALSE);
g_return_val_if_fail (gst_caps_is_fixed (caps), FALSE);
GST_DEBUG ("parsing caps %" GST_PTR_FORMAT, caps);
str = gst_caps_get_structure (caps, 0);
if (!gst_structure_has_name (str, "audio/x-raw"))
goto wrong_name;
if (!(s = gst_structure_get_string (str, "format")))
goto no_format;
format = gst_audio_format_from_string (s);
if (format == GST_AUDIO_FORMAT_UNKNOWN)
goto unknown_format;
if (!gst_structure_get_int (str, "rate", &rate))
goto no_rate;
if (!gst_structure_get_int (str, "channels", &channels))
goto no_channels;
gst_audio_info_set_format (info, format, rate, channels);
pos_val_arr = gst_structure_get_value (str, "channel-positions");
if (pos_val_arr) {
guint max_pos = MAX (channels, 64);
for (i = 0; i < max_pos; i++) {
pos_val_entry = gst_value_array_get_value (pos_val_arr, i);
info->position[i] = g_value_get_enum (pos_val_entry);
}
} else {
info->flags |= GST_AUDIO_FLAG_DEFAULT_POSITIONS;
/* FIXME, set default positions */
}
return TRUE;
/* ERROR */
wrong_name:
{
GST_ERROR ("wrong name, expected audio/x-raw");
return FALSE;
}
no_format:
{
GST_ERROR ("no format given");
return FALSE;
}
unknown_format:
{
GST_ERROR ("unknown format given");
return FALSE;
}
no_rate:
{
GST_ERROR ("no rate property given");
return FALSE;
}
no_channels:
{
GST_ERROR ("no channels property given");
return FALSE;
}
}
/**
* gst_audio_info_to_caps:
* @info: a #GstAudioInfo
*
* Convert the values of @info into a #GstCaps.
*
* Returns: (transfer full): the new #GstCaps containing the
* info of @info.
*/
GstCaps *
gst_audio_info_to_caps (GstAudioInfo * info)
{
GstCaps *caps;
const gchar *format;
g_return_val_if_fail (info != NULL, NULL);
g_return_val_if_fail (info->finfo != NULL, NULL);
g_return_val_if_fail (info->finfo->format != GST_AUDIO_FORMAT_UNKNOWN, NULL);
format = gst_audio_format_to_string (info->finfo->format);
g_return_val_if_fail (format != NULL, NULL);
caps = gst_caps_new_simple ("audio/x-raw",
"format", G_TYPE_STRING, format,
"rate", G_TYPE_INT, info->rate,
"channels", G_TYPE_INT, info->channels, NULL);
if (info->channels > 2) {
GValue pos_val_arr = { 0 }
, pos_val_entry = {
0};
gint i, max_pos;
GstStructure *str;
/* build gvaluearray from positions */
g_value_init (&pos_val_arr, GST_TYPE_ARRAY);
g_value_init (&pos_val_entry, GST_TYPE_AUDIO_CHANNEL_POSITION);
max_pos = MAX (info->channels, 64);
for (i = 0; i < max_pos; i++) {
g_value_set_enum (&pos_val_entry, info->position[i]);
gst_value_array_append_value (&pos_val_arr, &pos_val_entry);
}
g_value_unset (&pos_val_entry);
/* add to structure */
str = gst_caps_get_structure (caps, 0);
gst_structure_set_value (str, "channel-positions", &pos_val_arr);
g_value_unset (&pos_val_arr);
}
return caps;
}
/**
* gst_audio_format_convert:
* @info: a #GstAudioInfo
* @src_format: #GstFormat of the @src_value
* @src_value: value to convert
* @dest_format: #GstFormat of the @dest_value
* @dest_value: pointer to destination value
*
* Converts among various #GstFormat types. This function handles
* GST_FORMAT_BYTES, GST_FORMAT_TIME, and GST_FORMAT_DEFAULT. For
* raw audio, GST_FORMAT_DEFAULT corresponds to audio frames. This
* function can be used to handle pad queries of the type GST_QUERY_CONVERT.
*
* Returns: TRUE if the conversion was successful.
*/
gboolean
gst_audio_info_convert (GstAudioInfo * info,
GstFormat src_fmt, gint64 src_val, GstFormat dest_fmt, gint64 * dest_val)
{
gboolean res = TRUE;
gint bpf, rate;
GST_DEBUG ("converting value %" G_GINT64_FORMAT " from %s (%d) to %s (%d)",
src_val, gst_format_get_name (src_fmt), src_fmt,
gst_format_get_name (dest_fmt), dest_fmt);
if (src_fmt == dest_fmt || src_val == -1) {
*dest_val = src_val;
goto done;
}
/* get important info */
bpf = GST_AUDIO_INFO_BPF (info);
rate = GST_AUDIO_INFO_RATE (info);
if (bpf == 0 || rate == 0) {
GST_DEBUG ("no rate or bpf configured");
res = FALSE;
goto done;
}
switch (src_fmt) {
case GST_FORMAT_BYTES:
switch (dest_fmt) {
case GST_FORMAT_TIME:
*dest_val = GST_FRAMES_TO_CLOCK_TIME (src_val / bpf, rate);
break;
case GST_FORMAT_DEFAULT:
*dest_val = src_val / bpf;
break;
default:
res = FALSE;
break;
}
break;
case GST_FORMAT_DEFAULT:
switch (dest_fmt) {
case GST_FORMAT_TIME:
*dest_val = GST_FRAMES_TO_CLOCK_TIME (src_val, rate);
break;
case GST_FORMAT_BYTES:
*dest_val = src_val * bpf;
break;
default:
res = FALSE;
break;
}
break;
case GST_FORMAT_TIME:
switch (dest_fmt) {
case GST_FORMAT_DEFAULT:
*dest_val = GST_CLOCK_TIME_TO_FRAMES (src_val, rate);
break;
case GST_FORMAT_BYTES:
*dest_val = GST_CLOCK_TIME_TO_FRAMES (src_val, rate);
*dest_val *= bpf;
break;
default:
res = FALSE;
break;
}
break;
default:
res = FALSE;
break;
}
done:
GST_DEBUG ("ret=%d result %" G_GINT64_FORMAT, res, *dest_val);
return res;
}
/**
* gst_audio_buffer_clip:
* @buffer: 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
* * channels.
*
* Clip the the buffer to the given %GstSegment.
*
* After calling this function the caller does not own a reference to
* @buffer anymore.
*
* Returns: %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
*
* Since: 0.10.14
*/
GstBuffer *
gst_audio_buffer_clip (GstBuffer * buffer, GstSegment * segment, gint rate,
gint bpf)
{
GstBuffer *ret;
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;
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. */
trim = 0;
size = gst_buffer_get_size (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 / bpf, 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 / bpf;
}
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 * bpf;
size -= diff * bpf;
}
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 * bpf;
}
} 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 * bpf;
size -= diff * bpf;
}
diff = stop - cstop;
if (diff > 0) {
offset_end = cstop;
if (change_duration)
duration -= gst_util_uint64_scale (diff, GST_SECOND, rate);
size -= diff * bpf;
}
} else {
gst_buffer_unref (buffer);
return NULL;
}
}
/* Get a writable buffer and apply all changes */
GST_DEBUG ("trim %" G_GSIZE_FORMAT " size %" G_GSIZE_FORMAT, trim, size);
ret = gst_buffer_copy_region (buffer, GST_BUFFER_COPY_ALL, trim, size);
gst_buffer_unref (buffer);
GST_DEBUG ("timestamp %" GST_TIME_FORMAT, GST_TIME_ARGS (timestamp));
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;
return ret;
}
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