mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-18 15:51:11 +00:00
33196cdd2c
Remove the _ in front of the endianness prefix. Remove the _3 postfix for the 24 bits formats. Add a _32 postfix after the formats that occupy extra space beyond their natural size. The result is that the GST_AUDIO_NE() macro can simply append the endianness after all formats and that we only specify a different sample width when it is different from the natural size of the sample. This makes things more consistent and follows the pulseaudio conventions instead of the alsa ones.
672 lines
19 KiB
C
672 lines
19 KiB
C
/* 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,flags,end,width,depth,silent) \
|
|
{ GST_AUDIO_FORMAT_ ##str, G_STRINGIFY(str), 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, SINT, 0, 8, 8, SILENT_0),
|
|
MAKE_FORMAT (U8, UINT, 0, 8, 8, SILENT_U8),
|
|
/* 16 bit */
|
|
MAKE_FORMAT (S16LE, SINT, G_LITTLE_ENDIAN, 16, 16, SILENT_0),
|
|
MAKE_FORMAT (S16BE, SINT, G_BIG_ENDIAN, 16, 16, SILENT_0),
|
|
MAKE_FORMAT (U16LE, UINT, G_LITTLE_ENDIAN, 16, 16, SILENT_U16LE),
|
|
MAKE_FORMAT (U16BE, UINT, G_BIG_ENDIAN, 16, 16, SILENT_U16BE),
|
|
/* 24 bit in low 3 bytes of 32 bits */
|
|
MAKE_FORMAT (S24_32LE, SINT, G_LITTLE_ENDIAN, 32, 24, SILENT_0),
|
|
MAKE_FORMAT (S24_32BE, SINT, G_BIG_ENDIAN, 32, 24, SILENT_0),
|
|
MAKE_FORMAT (U24_32LE, UINT, G_LITTLE_ENDIAN, 32, 24, SILENT_U24_32LE),
|
|
MAKE_FORMAT (U24_32BE, UINT, G_BIG_ENDIAN, 32, 24, SILENT_U24_32BE),
|
|
/* 32 bit */
|
|
MAKE_FORMAT (S32LE, SINT, G_LITTLE_ENDIAN, 32, 32, SILENT_0),
|
|
MAKE_FORMAT (S32BE, SINT, G_BIG_ENDIAN, 32, 32, SILENT_0),
|
|
MAKE_FORMAT (U32LE, UINT, G_LITTLE_ENDIAN, 32, 32, SILENT_U32LE),
|
|
MAKE_FORMAT (U32BE, UINT, G_BIG_ENDIAN, 32, 32, SILENT_U32BE),
|
|
/* 24 bit in 3 bytes */
|
|
MAKE_FORMAT (S24LE, SINT, G_LITTLE_ENDIAN, 24, 24, SILENT_0),
|
|
MAKE_FORMAT (S24BE, SINT, G_BIG_ENDIAN, 24, 24, SILENT_0),
|
|
MAKE_FORMAT (U24LE, UINT, G_LITTLE_ENDIAN, 24, 24, SILENT_U24LE),
|
|
MAKE_FORMAT (U24BE, UINT, G_BIG_ENDIAN, 24, 24, SILENT_U24BE),
|
|
/* 20 bit in 3 bytes */
|
|
MAKE_FORMAT (S20LE, SINT, G_LITTLE_ENDIAN, 24, 20, SILENT_0),
|
|
MAKE_FORMAT (S20BE, SINT, G_BIG_ENDIAN, 24, 20, SILENT_0),
|
|
MAKE_FORMAT (U20LE, UINT, G_LITTLE_ENDIAN, 24, 20, SILENT_U20LE),
|
|
MAKE_FORMAT (U20BE, UINT, G_BIG_ENDIAN, 24, 20, SILENT_U20BE),
|
|
/* 18 bit in 3 bytes */
|
|
MAKE_FORMAT (S18LE, SINT, G_LITTLE_ENDIAN, 24, 18, SILENT_0),
|
|
MAKE_FORMAT (S18BE, SINT, G_BIG_ENDIAN, 24, 18, SILENT_0),
|
|
MAKE_FORMAT (U18LE, UINT, G_LITTLE_ENDIAN, 24, 18, SILENT_U18LE),
|
|
MAKE_FORMAT (U18BE, UINT, G_BIG_ENDIAN, 24, 18, SILENT_U18BE),
|
|
/* float */
|
|
MAKE_FORMAT (F32LE, GST_AUDIO_FORMAT_FLAG_FLOAT, G_LITTLE_ENDIAN, 32, 32,
|
|
SILENT_0),
|
|
MAKE_FORMAT (F32BE, GST_AUDIO_FORMAT_FLAG_FLOAT, G_BIG_ENDIAN, 32, 32,
|
|
SILENT_0),
|
|
MAKE_FORMAT (F64LE, GST_AUDIO_FORMAT_FLAG_FLOAT, G_LITTLE_ENDIAN, 64, 64,
|
|
SILENT_0),
|
|
MAKE_FORMAT (F64BE, 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;
|
|
}
|