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gstreamer/gst-libs/gst/audio/audio.c
Tim-Philipp Müller 63a3d360dc audio: add GstAudioFormat, GstAudioFormatInfo and GstAudioInfo 
Same as in 0.11, but with caps parsing/serialising for 0.10 style
caps. Add setting default channel positions.
2011-08-27 14:47:01 +01:00

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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 "audio.h"
#include "audio-enumtypes.h"
#include <gst/gststructure.h>
#include <string.h>
/**
* gst_audio_frame_byte_size:
* @pad: the #GstPad to get the caps from
*
* Calculate byte size of an audio frame.
*
* Returns: the byte size, or 0 if there was an error
*/
int
gst_audio_frame_byte_size (GstPad * pad)
{
/* FIXME: this should be moved closer to the gstreamer core
* and be implemented for every mime type IMO
*/
int width = 0;
int channels = 0;
const GstCaps *caps = NULL;
GstStructure *structure;
/* get caps of pad */
caps = GST_PAD_CAPS (pad);
if (caps == NULL) {
/* ERROR: could not get caps of pad */
g_warning ("gstaudio: could not get caps of pad %s:%s\n",
GST_DEBUG_PAD_NAME (pad));
return 0;
}
structure = gst_caps_get_structure (caps, 0);
gst_structure_get_int (structure, "width", &width);
gst_structure_get_int (structure, "channels", &channels);
return (width / 8) * channels;
}
/**
* gst_audio_frame_length:
* @pad: the #GstPad to get the caps from
* @buf: the #GstBuffer
*
* Calculate length of buffer in frames.
*
* Returns: 0 if there's an error, or the number of frames if everything's ok
*/
long
gst_audio_frame_length (GstPad * pad, GstBuffer * buf)
{
/* FIXME: this should be moved closer to the gstreamer core
* and be implemented for every mime type IMO
*/
int frame_byte_size = 0;
frame_byte_size = gst_audio_frame_byte_size (pad);
if (frame_byte_size == 0)
/* error */
return 0;
/* FIXME: this function assumes the buffer size to be a whole multiple
* of the frame byte size
*/
return GST_BUFFER_SIZE (buf) / frame_byte_size;
}
/**
* gst_audio_duration_from_pad_buffer:
* @pad: the #GstPad to get the caps from
* @buf: the #GstBuffer
*
* Calculate length in nanoseconds of audio buffer @buf based on capabilities of
* @pad.
*
* Returns: the length.
*/
GstClockTime
gst_audio_duration_from_pad_buffer (GstPad * pad, GstBuffer * buf)
{
long bytes = 0;
int width = 0;
int channels = 0;
int rate = 0;
GstClockTime length;
const GstCaps *caps = NULL;
GstStructure *structure;
g_assert (GST_IS_BUFFER (buf));
/* get caps of pad */
caps = GST_PAD_CAPS (pad);
if (caps == NULL) {
/* ERROR: could not get caps of pad */
g_warning ("gstaudio: could not get caps of pad %s:%s\n",
GST_DEBUG_PAD_NAME (pad));
length = GST_CLOCK_TIME_NONE;
} else {
structure = gst_caps_get_structure (caps, 0);
bytes = GST_BUFFER_SIZE (buf);
gst_structure_get_int (structure, "width", &width);
gst_structure_get_int (structure, "channels", &channels);
gst_structure_get_int (structure, "rate", &rate);
g_assert (bytes != 0);
g_assert (width != 0);
g_assert (channels != 0);
g_assert (rate != 0);
length = (bytes * 8 * GST_SECOND) / (rate * channels * width);
}
return length;
}
/**
* gst_audio_is_buffer_framed:
* @pad: the #GstPad to get the caps from
* @buf: the #GstBuffer
*
* Check if the buffer size is a whole multiple of the frame size.
*
* Returns: %TRUE if buffer size is multiple.
*/
gboolean
gst_audio_is_buffer_framed (GstPad * pad, GstBuffer * buf)
{
if (GST_BUFFER_SIZE (buf) % gst_audio_frame_byte_size (pad) == 0)
return TRUE;
else
return FALSE;
}
/* _getcaps helper functions
* sets structure fields to default for audio type
* flag determines which structure fields to set to default
* keep these functions in sync with the templates in audio.h
*/
/* private helper function
* sets a list on the structure
* pass in structure, fieldname for the list, type of the list values,
* number of list values, and each of the values, terminating with NULL
*/
static void
_gst_audio_structure_set_list (GstStructure * structure,
const gchar * fieldname, GType type, int number, ...)
{
va_list varargs;
GValue value = { 0 };
GArray *array;
int j;
g_return_if_fail (structure != NULL);
g_value_init (&value, GST_TYPE_LIST);
array = g_value_peek_pointer (&value);
va_start (varargs, number);
for (j = 0; j < number; ++j) {
int i;
gboolean b;
GValue list_value = { 0 };
switch (type) {
case G_TYPE_INT:
i = va_arg (varargs, int);
g_value_init (&list_value, G_TYPE_INT);
g_value_set_int (&list_value, i);
break;
case G_TYPE_BOOLEAN:
b = va_arg (varargs, gboolean);
g_value_init (&list_value, G_TYPE_BOOLEAN);
g_value_set_boolean (&list_value, b);
break;
default:
g_warning
("_gst_audio_structure_set_list: LIST of given type not implemented.");
}
g_array_append_val (array, list_value);
}
gst_structure_set_value (structure, fieldname, &value);
va_end (varargs);
}
/**
* gst_audio_structure_set_int:
* @structure: a #GstStructure
* @flag: a set of #GstAudioFieldFlag
*
* Do not use anymore.
*
* Deprecated: use gst_structure_set()
*/
#ifndef GST_REMOVE_DEPRECATED
#ifdef GST_DISABLE_DEPRECATED
typedef enum
{
GST_AUDIO_FIELD_RATE = (1 << 0),
GST_AUDIO_FIELD_CHANNELS = (1 << 1),
GST_AUDIO_FIELD_ENDIANNESS = (1 << 2),
GST_AUDIO_FIELD_WIDTH = (1 << 3),
GST_AUDIO_FIELD_DEPTH = (1 << 4),
GST_AUDIO_FIELD_SIGNED = (1 << 5),
} GstAudioFieldFlag;
void
gst_audio_structure_set_int (GstStructure * structure, GstAudioFieldFlag flag);
#endif /* GST_DISABLE_DEPRECATED */
void
gst_audio_structure_set_int (GstStructure * structure, GstAudioFieldFlag flag)
{
/* was added here:
* http://webcvs.freedesktop.org/gstreamer/gst-plugins-base/gst-libs/gst/audio/audio.c?r1=1.16&r2=1.17
* but it is not used
*/
if (flag & GST_AUDIO_FIELD_RATE)
gst_structure_set (structure, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT,
NULL);
if (flag & GST_AUDIO_FIELD_CHANNELS)
gst_structure_set (structure, "channels", GST_TYPE_INT_RANGE, 1, G_MAXINT,
NULL);
if (flag & GST_AUDIO_FIELD_ENDIANNESS)
_gst_audio_structure_set_list (structure, "endianness", G_TYPE_INT, 2,
G_LITTLE_ENDIAN, G_BIG_ENDIAN, NULL);
if (flag & GST_AUDIO_FIELD_WIDTH)
_gst_audio_structure_set_list (structure, "width", G_TYPE_INT, 3, 8, 16, 32,
NULL);
if (flag & GST_AUDIO_FIELD_DEPTH)
gst_structure_set (structure, "depth", GST_TYPE_INT_RANGE, 1, 32, NULL);
if (flag & GST_AUDIO_FIELD_SIGNED)
_gst_audio_structure_set_list (structure, "signed", G_TYPE_BOOLEAN, 2, TRUE,
FALSE, NULL);
}
#endif /* GST_REMOVE_DEPRECATED */
#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_U16_LE { 0x00, 0x80, 0x00, 0x80, 0x00, 0x80, 0x00, 0x80 }
#define SILENT_U16_BE { 0x80, 0x00, 0x80, 0x00, 0x80, 0x00, 0x80, 0x00 }
#define SILENT_U24_LE { 0x00, 0x00, 0x80, 0x00, 0x00, 0x00, 0x80, 0x00 }
#define SILENT_U24_BE { 0x00, 0x80, 0x00, 0x00, 0x00, 0x80, 0x00, 0x00 }
#define SILENT_U32_LE { 0x00, 0x00, 0x00, 0x80, 0x00, 0x00, 0x00, 0x80 }
#define SILENT_U32_BE { 0x80, 0x00, 0x00, 0x00, 0x80, 0x00, 0x00, 0x00 }
#define SILENT_U24_3LE { 0x00, 0x00, 0x80, 0x00, 0x00, 0x80 }
#define SILENT_U24_3BE { 0x80, 0x00, 0x00, 0x80, 0x00, 0x00 }
#define SILENT_U20_3LE { 0x00, 0x00, 0x08, 0x00, 0x00, 0x08 }
#define SILENT_U20_3BE { 0x08, 0x00, 0x00, 0x08, 0x00, 0x00 }
#define SILENT_U18_3LE { 0x00, 0x00, 0x02, 0x00, 0x00, 0x02 }
#define SILENT_U18_3BE { 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 (S16_LE, SINT, G_LITTLE_ENDIAN, 16, 16, SILENT_0),
MAKE_FORMAT (S16_BE, SINT, G_BIG_ENDIAN, 16, 16, SILENT_0),
MAKE_FORMAT (U16_LE, UINT, G_LITTLE_ENDIAN, 16, 16, SILENT_U16_LE),
MAKE_FORMAT (U16_BE, UINT, G_BIG_ENDIAN, 16, 16, SILENT_U16_BE),
/* 24 bit in low 3 bytes of 32 bits */
MAKE_FORMAT (S24_LE, SINT, G_LITTLE_ENDIAN, 32, 24, SILENT_0),
MAKE_FORMAT (S24_BE, SINT, G_BIG_ENDIAN, 32, 24, SILENT_0),
MAKE_FORMAT (U24_LE, UINT, G_LITTLE_ENDIAN, 32, 24, SILENT_U24_LE),
MAKE_FORMAT (U24_BE, UINT, G_BIG_ENDIAN, 32, 24, SILENT_U24_BE),
/* 32 bit */
MAKE_FORMAT (S32_LE, SINT, G_LITTLE_ENDIAN, 32, 32, SILENT_0),
MAKE_FORMAT (S32_BE, SINT, G_BIG_ENDIAN, 32, 32, SILENT_0),
MAKE_FORMAT (U32_LE, UINT, G_LITTLE_ENDIAN, 32, 32, SILENT_U32_LE),
MAKE_FORMAT (U32_BE, UINT, G_BIG_ENDIAN, 32, 32, SILENT_U32_BE),
/* 24 bit in 3 bytes */
MAKE_FORMAT (S24_3LE, SINT, G_LITTLE_ENDIAN, 24, 24, SILENT_0),
MAKE_FORMAT (S24_3BE, SINT, G_BIG_ENDIAN, 24, 24, SILENT_0),
MAKE_FORMAT (U24_3LE, UINT, G_LITTLE_ENDIAN, 24, 24, SILENT_U24_3LE),
MAKE_FORMAT (U24_3BE, UINT, G_BIG_ENDIAN, 24, 24, SILENT_U24_3BE),
/* 20 bit in 3 bytes */
MAKE_FORMAT (S20_3LE, SINT, G_LITTLE_ENDIAN, 24, 20, SILENT_0),
MAKE_FORMAT (S20_3BE, SINT, G_BIG_ENDIAN, 24, 20, SILENT_0),
MAKE_FORMAT (U20_3LE, UINT, G_LITTLE_ENDIAN, 24, 20, SILENT_U20_3LE),
MAKE_FORMAT (U20_3BE, UINT, G_BIG_ENDIAN, 24, 20, SILENT_U20_3BE),
/* 18 bit in 3 bytes */
MAKE_FORMAT (S18_3LE, SINT, G_LITTLE_ENDIAN, 24, 18, SILENT_0),
MAKE_FORMAT (S18_3BE, SINT, G_BIG_ENDIAN, 24, 18, SILENT_0),
MAKE_FORMAT (U18_3LE, UINT, G_LITTLE_ENDIAN, 24, 18, SILENT_U18_3LE),
MAKE_FORMAT (U18_3BE, UINT, G_BIG_ENDIAN, 24, 18, SILENT_U18_3BE),
/* float */
MAKE_FORMAT (F32_LE, GST_AUDIO_FORMAT_FLAG_FLOAT, G_LITTLE_ENDIAN, 32, 32,
SILENT_0),
MAKE_FORMAT (F32_BE, GST_AUDIO_FORMAT_FLAG_FLOAT, G_BIG_ENDIAN, 32, 32,
SILENT_0),
MAKE_FORMAT (F64_LE, GST_AUDIO_FORMAT_FLAG_FLOAT, G_LITTLE_ENDIAN, 64, 64,
SILENT_0),
MAKE_FORMAT (F64_BE, GST_AUDIO_FORMAT_FLAG_FLOAT, G_BIG_ENDIAN, 64, 64,
SILENT_0)
};
static GstAudioFormat
gst_audio_format_from_caps_structure (const GstStructure * s)
{
gint endianness, width, depth;
guint i;
if (gst_structure_has_name (s, "audio/x-raw-int")) {
gboolean sign;
if (!gst_structure_get_boolean (s, "signed", &sign))
goto missing_field_signed;
if (!gst_structure_get_int (s, "endianness", &endianness))
goto missing_field_endianness;
if (!gst_structure_get_int (s, "width", &width))
goto missing_field_width;
if (!gst_structure_get_int (s, "depth", &depth))
goto missing_field_depth;
for (i = 0; i < G_N_ELEMENTS (formats); i++) {
if (GST_AUDIO_FORMAT_INFO_IS_INTEGER (&formats[i]) &&
sign == GST_AUDIO_FORMAT_INFO_IS_SIGNED (&formats[i]) &&
GST_AUDIO_FORMAT_INFO_ENDIANNESS (&formats[i]) == endianness &&
GST_AUDIO_FORMAT_INFO_WIDTH (&formats[i]) == width &&
GST_AUDIO_FORMAT_INFO_DEPTH (&formats[i]) == depth) {
return GST_AUDIO_FORMAT_INFO_FORMAT (&formats[i]);
}
}
} else if (gst_structure_has_name (s, "audio/x-raw-float")) {
/* fallbacks are for backwards compatibility (is this needed at all?) */
if (!gst_structure_get_int (s, "endianness", &endianness)) {
GST_WARNING ("float audio caps without endianness %" GST_PTR_FORMAT, s);
endianness = G_BYTE_ORDER;
}
if (!gst_structure_get_int (s, "width", &width)) {
GST_WARNING ("float audio caps without width %" GST_PTR_FORMAT, s);
width = 32;
}
for (i = 0; i < G_N_ELEMENTS (formats); i++) {
if (GST_AUDIO_FORMAT_INFO_IS_FLOAT (&formats[i]) &&
GST_AUDIO_FORMAT_INFO_ENDIANNESS (&formats[i]) == endianness &&
GST_AUDIO_FORMAT_INFO_WIDTH (&formats[i]) == width) {
return GST_AUDIO_FORMAT_INFO_FORMAT (&formats[i]);
}
}
}
/* no match */
return GST_AUDIO_FORMAT_UNKNOWN;
missing_field_signed:
{
GST_ERROR ("missing 'signed' field in audio caps %" GST_PTR_FORMAT, s);
return GST_AUDIO_FORMAT_UNKNOWN;
}
missing_field_endianness:
{
GST_ERROR ("missing 'endianness' field in audio caps %" GST_PTR_FORMAT, s);
return GST_AUDIO_FORMAT_UNKNOWN;
}
missing_field_depth:
{
GST_ERROR ("missing 'depth' field in audio caps %" GST_PTR_FORMAT, s);
return GST_AUDIO_FORMAT_UNKNOWN;
}
missing_field_width:
{
GST_ERROR ("missing 'width' field in audio caps %" GST_PTR_FORMAT, s);
return GST_AUDIO_FORMAT_UNKNOWN;
}
}
static 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;
}
/* from multichannel.c */
void priv_gst_audio_info_fill_default_channel_positions (GstAudioInfo * info);
/**
* gst_audio_info_from_caps:
* @info: a #GstAudioInfo
* @caps: a #GstCaps
*
* Parse @caps and update @info.
*
* Returns: TRUE if @caps could be parsed
*
* Since: 0.10.36
*/
gboolean
gst_audio_info_from_caps (GstAudioInfo * info, const GstCaps * caps)
{
GstStructure *str;
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);
format = gst_audio_format_from_caps_structure (str);
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) {
if (channels <= G_N_ELEMENTS (info->position)) {
for (i = 0; i < channels; i++) {
pos_val_entry = gst_value_array_get_value (pos_val_arr, i);
info->position[i] = g_value_get_enum (pos_val_entry);
}
} else {
/* for that many channels, the positions are always NONE */
for (i = 0; i < G_N_ELEMENTS (info->position); i++)
info->position[i] = GST_AUDIO_CHANNEL_POSITION_NONE;
info->flags |= GST_AUDIO_FLAG_DEFAULT_POSITIONS;
}
} else {
info->flags |= GST_AUDIO_FLAG_DEFAULT_POSITIONS;
priv_gst_audio_info_fill_default_channel_positions (info);
}
return TRUE;
/* ERROR */
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.
*
* Since: 0.10.36
*/
GstCaps *
gst_audio_info_to_caps (GstAudioInfo * info)
{
GstCaps *caps;
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);
if (GST_AUDIO_FORMAT_INFO_IS_INTEGER (info->finfo)) {
caps = gst_caps_new_simple ("audio/x-raw-int",
"width", G_TYPE_INT, GST_AUDIO_INFO_WIDTH (info),
"depth", G_TYPE_INT, GST_AUDIO_INFO_DEPTH (info),
"endianness", G_TYPE_INT,
GST_AUDIO_FORMAT_INFO_ENDIANNESS (info->finfo), "signed",
G_TYPE_BOOLEAN, GST_AUDIO_FORMAT_INFO_IS_SIGNED (info->finfo), "rate",
G_TYPE_INT, GST_AUDIO_INFO_RATE (info), "channels", G_TYPE_INT,
GST_AUDIO_INFO_CHANNELS (info), NULL);
} else if (GST_AUDIO_FORMAT_INFO_IS_FLOAT (info->finfo)) {
caps = gst_caps_new_simple ("audio/x-raw-float",
"width", G_TYPE_INT, GST_AUDIO_INFO_WIDTH (info),
"endianness", G_TYPE_INT,
GST_AUDIO_FORMAT_INFO_ENDIANNESS (info->finfo), "rate", G_TYPE_INT,
GST_AUDIO_INFO_RATE (info), "channels", G_TYPE_INT,
GST_AUDIO_INFO_CHANNELS (info), NULL);
} else {
GST_ERROR ("unknown audio format, neither integer nor float");
return NULL;
}
if (info->channels > 2) {
GValue pos_val_arr = { 0 }
, pos_val_entry = {
0};
GstStructure *str;
gint i;
/* build gvaluearray from positions */
g_value_init (&pos_val_arr, GST_TYPE_ARRAY);
g_value_init (&pos_val_entry, GST_TYPE_AUDIO_CHANNEL_POSITION);
for (i = 0; i < info->channels; i++) {
/* if we have many many channels, all positions are NONE */
if (info->channels <= 64)
g_value_set_enum (&pos_val_entry, info->position[i]);
else
g_value_set_enum (&pos_val_entry, GST_AUDIO_CHANNEL_POSITION_NONE);
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.
*
* Since: 0.10.36
*/
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.
* @frame_size: size of one audio frame in bytes.
*
* 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 frame_size)
{
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;
guint8 *data;
guint 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. */
data = GST_BUFFER_DATA (buffer);
size = GST_BUFFER_SIZE (buffer);
timestamp = GST_BUFFER_TIMESTAMP (buffer);
if (GST_BUFFER_DURATION_IS_VALID (buffer)) {
duration = GST_BUFFER_DURATION (buffer);
} else {
change_duration = FALSE;
duration = gst_util_uint64_scale (size / frame_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 / frame_size;
}
if (segment->format == GST_FORMAT_TIME) {
/* Handle clipping for GST_FORMAT_TIME */
gint64 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;
data += diff * frame_size;
size -= diff * frame_size;
}
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 * frame_size;
}
} else {
gst_buffer_unref (buffer);
return NULL;
}
} else {
/* Handle clipping for GST_FORMAT_DEFAULT */
gint64 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);
data += diff * frame_size;
size -= diff * frame_size;
}
diff = stop - cstop;
if (diff > 0) {
offset_end = cstop;
if (change_duration)
duration -= gst_util_uint64_scale (diff, GST_SECOND, rate);
size -= diff * frame_size;
}
} else {
gst_buffer_unref (buffer);
return NULL;
}
}
/* Get a metadata writable buffer and apply all changes */
ret = gst_buffer_make_metadata_writable (buffer);
GST_BUFFER_TIMESTAMP (ret) = timestamp;
GST_BUFFER_SIZE (ret) = size;
GST_BUFFER_DATA (ret) = data;
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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