mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2025-01-18 05:16:05 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/gst-libs/gst/audio/audio.c
Sebastian Dröge c227f5e77e audio: Add new channel positions and simplify channel expression in the caps 
The available channel positions are all channels from SMPTE 2036-2-2008
(in that order) and DTS Coherent Acoustics, which are basically all 28
channels that currently can appear.

The channels are now expressed in the caps as a channel-mask, which
describes which of the channels are present, and an optional
channel-reorder-map, which must only be used after negotiation for
fixated caps.

For negotiation only the channel-mask and the channel count is relevant
and all elements are expected to handle all reorder maps. Elements that
don't can use the new API to reorder an audio buffer from any order to
another order.

This simplifies negotiation a lot while still having as few reorderings
necassary as possible and still allow all kinds of channel layouts.
2012-01-05 10:27:21 +01:00

1019 lines
29 KiB
C
Raw Blame History

/* 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)
};
G_DEFINE_POINTER_TYPE (GstAudioFormatInfo, gst_audio_format_info);
/**
* gst_audio_format_build_integer:
* @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)) ||
(!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_copy:
* @info: a #GstAudioInfo
*
* Copy a GstAudioInfo structure.
*
* Returns: a new #GstAudioInfo. free with gst_audio_info_free.
*/
GstAudioInfo *
gst_audio_info_copy (const GstAudioInfo * info)
{
return g_slice_dup (GstAudioInfo, info);
}
/**
* gst_audio_info_free:
* @info: a #GstAudioInfo
*
* Free a GstAudioInfo structure previously allocated with gst_audio_info_new()
* or gst_audio_info_copy().
*/
void
gst_audio_info_free (GstAudioInfo * info)
{
g_slice_free (GstAudioInfo, info);
}
G_DEFINE_BOXED_TYPE (GstAudioInfo, gst_audio_info,
(GBoxedCopyFunc) gst_audio_info_copy, (GBoxedFreeFunc) gst_audio_info_free);
/**
* gst_audio_info_new:
*
* Allocate a new #GstAudioInfo that is also initialized with
* gst_audio_info_init().
*
* Returns: a new #GstAudioInfo. free with gst_audio_info_free().
*/
GstAudioInfo *
gst_audio_info_new (void)
{
GstAudioInfo *info;
info = g_slice_new (GstAudioInfo);
gst_audio_info_init (info);
return info;
}
/**
* 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));
info->finfo = &formats[GST_AUDIO_FORMAT_UNKNOWN];
memset (&info->position, 0xff, sizeof (info->position));
}
static const GstAudioChannelPosition default_channel_order[64] = {
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_LFE1,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER,
GST_AUDIO_CHANNEL_POSITION_REAR_CENTER,
GST_AUDIO_CHANNEL_POSITION_LFE2,
GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT,
GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_TOP_CENTER,
GST_AUDIO_CHANNEL_POSITION_TOP_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_TOP_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_TOP_SIDE_LEFT,
GST_AUDIO_CHANNEL_POSITION_TOP_SIDE_RIGHT,
GST_AUDIO_CHANNEL_POSITION_TOP_REAR_CENTER,
GST_AUDIO_CHANNEL_POSITION_BOTTOM_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_BOTTOM_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_BOTTOM_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_WIDE_LEFT,
GST_AUDIO_CHANNEL_POSITION_WIDE_RIGHT,
GST_AUDIO_CHANNEL_POSITION_SURROUND_LEFT,
GST_AUDIO_CHANNEL_POSITION_SURROUND_RIGHT,
GST_AUDIO_CHANNEL_POSITION_INVALID,
GST_AUDIO_CHANNEL_POSITION_INVALID,
GST_AUDIO_CHANNEL_POSITION_INVALID,
GST_AUDIO_CHANNEL_POSITION_INVALID,
GST_AUDIO_CHANNEL_POSITION_INVALID,
GST_AUDIO_CHANNEL_POSITION_INVALID,
GST_AUDIO_CHANNEL_POSITION_INVALID,
GST_AUDIO_CHANNEL_POSITION_INVALID,
GST_AUDIO_CHANNEL_POSITION_INVALID,
GST_AUDIO_CHANNEL_POSITION_INVALID,
GST_AUDIO_CHANNEL_POSITION_INVALID,
GST_AUDIO_CHANNEL_POSITION_INVALID,
GST_AUDIO_CHANNEL_POSITION_INVALID,
GST_AUDIO_CHANNEL_POSITION_INVALID,
GST_AUDIO_CHANNEL_POSITION_INVALID,
GST_AUDIO_CHANNEL_POSITION_INVALID,
GST_AUDIO_CHANNEL_POSITION_INVALID,
GST_AUDIO_CHANNEL_POSITION_INVALID,
GST_AUDIO_CHANNEL_POSITION_INVALID,
GST_AUDIO_CHANNEL_POSITION_INVALID,
GST_AUDIO_CHANNEL_POSITION_INVALID,
GST_AUDIO_CHANNEL_POSITION_INVALID,
GST_AUDIO_CHANNEL_POSITION_INVALID,
GST_AUDIO_CHANNEL_POSITION_INVALID,
GST_AUDIO_CHANNEL_POSITION_INVALID,
GST_AUDIO_CHANNEL_POSITION_INVALID,
GST_AUDIO_CHANNEL_POSITION_INVALID,
GST_AUDIO_CHANNEL_POSITION_INVALID,
GST_AUDIO_CHANNEL_POSITION_INVALID,
GST_AUDIO_CHANNEL_POSITION_INVALID,
GST_AUDIO_CHANNEL_POSITION_INVALID,
GST_AUDIO_CHANNEL_POSITION_INVALID,
GST_AUDIO_CHANNEL_POSITION_INVALID,
GST_AUDIO_CHANNEL_POSITION_INVALID,
GST_AUDIO_CHANNEL_POSITION_INVALID,
GST_AUDIO_CHANNEL_POSITION_INVALID
};
static gboolean
check_valid_channel_positions (const GstAudioChannelPosition * position,
gint channels, gboolean enforce_order, guint64 * channel_mask_out)
{
gint i, j;
guint64 channel_mask = 0;
gboolean none_layout = FALSE;
if (channels == 1 && position[0] == GST_AUDIO_CHANNEL_POSITION_MONO) {
if (channel_mask_out)
*channel_mask_out = 0;
return TRUE;
}
j = 0;
for (i = 0; i < channels; i++) {
while (j < G_N_ELEMENTS (default_channel_order)
&& default_channel_order[j] != position[i])
j++;
if (position[i] == GST_AUDIO_CHANNEL_POSITION_INVALID ||
position[i] == GST_AUDIO_CHANNEL_POSITION_MONO)
return FALSE;
if (position[i] == GST_AUDIO_CHANNEL_POSITION_NONE) {
none_layout = TRUE;
continue;
}
/* Is this in valid channel order? */
if (enforce_order && j == G_N_ELEMENTS (default_channel_order))
return FALSE;
j++;
if ((channel_mask & (G_GUINT64_CONSTANT (1) << position[i])))
return FALSE;
channel_mask |= (G_GUINT64_CONSTANT (1) << position[i]);
}
if (none_layout && channel_mask != 0)
return FALSE;
if (channel_mask_out)
*channel_mask_out = channel_mask;
return TRUE;
}
/**
* gst_audio_info_set_format:
* @info: a #GstAudioInfo
* @format: the format
* @rate: the samplerate
* @channels: the number of channels
* @position: the channel positions
*
* 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 GstAudioChannelPosition * position)
{
const GstAudioFormatInfo *finfo;
g_return_if_fail (info != NULL);
g_return_if_fail (format != GST_AUDIO_FORMAT_UNKNOWN);
g_return_if_fail (channels <= 2 || position != NULL);
finfo = &formats[format];
info->flags = 0;
info->finfo = finfo;
info->rate = rate;
info->channels = channels;
info->bpf = (finfo->width * channels) / 8;
memset (&info->position, 0xff, sizeof (info->position));
if (!position && channels == 1) {
info->position[0] = GST_AUDIO_CHANNEL_POSITION_MONO;
} else if (!position && channels == 2) {
info->position[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
info->position[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
} else {
if (!check_valid_channel_positions (position, channels, TRUE, NULL)) {
g_warning ("Invalid channel positions");
} else {
memcpy (&info->position, position,
info->channels * sizeof (info->position[0]));
}
}
}
/**
* 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;
guint64 channel_mask;
gint i;
GstAudioChannelPosition position[64];
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;
if (!gst_structure_get (str, "channel-mask", GST_TYPE_BITMASK, &channel_mask,
NULL)) {
if (channels == 1) {
position[0] = GST_AUDIO_CHANNEL_POSITION_MONO;
} else if (channels == 2) {
position[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
position[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
} else {
goto no_channel_mask;
}
} else if (channel_mask == 0) {
for (i = 0; i < G_N_ELEMENTS (position); i++)
position[i] = GST_AUDIO_CHANNEL_POSITION_NONE;
} else {
gint j;
j = 0;
for (i = 0; i < 64; i++) {
if ((channel_mask & (G_GUINT64_CONSTANT (1) << i))) {
position[j] = default_channel_order[i];
if (default_channel_order[i] == GST_AUDIO_CHANNEL_POSITION_INVALID)
goto invalid_channel_mask;
j++;
}
}
if (j != channels)
goto invalid_channel_mask;
}
gst_audio_info_set_format (info, format, rate, channels, position);
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;
}
no_channel_mask:
{
GST_ERROR ("no channel-mask property given");
return FALSE;
}
invalid_channel_mask:
{
GST_ERROR ("Invalid channel mask 0x%016" G_GINT64_MODIFIER
"x for %d channels", channel_mask, channels);
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 (const 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 > 1
|| info->position[0] != GST_AUDIO_CHANNEL_POSITION_MONO) {
guint64 channel_mask = 0;
if (!check_valid_channel_positions (info->position, info->channels,
TRUE, &channel_mask))
goto invalid_channel_positions;
if (info->channels == 1
&& info->position[0] == GST_AUDIO_CHANNEL_POSITION_MONO) {
/* Default mono special case */
} else {
gst_caps_set_simple (caps, "channel-mask", GST_TYPE_BITMASK, channel_mask,
NULL);
}
}
return caps;
invalid_channel_positions:
{
GST_ERROR ("Invalid channel positions");
gst_caps_unref (caps);
return NULL;
}
}
/**
* 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 (const 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 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;
}
gboolean
gst_audio_buffer_reorder_channels (GstBuffer * buffer,
GstAudioFormat format, gint channels,
const GstAudioChannelPosition * from, const GstAudioChannelPosition * to)
{
gsize size;
guint8 *data;
gboolean ret;
g_return_val_if_fail (GST_IS_BUFFER (buffer), FALSE);
data = gst_buffer_map (buffer, &size, NULL, GST_MAP_READ | GST_MAP_WRITE);
ret = gst_audio_reorder_channels (data, size, format, channels, from, to);
gst_buffer_unmap (buffer, data, size);
return ret;
}
gboolean
gst_audio_reorder_channels (gpointer data, gsize size, GstAudioFormat format,
gint channels, const GstAudioChannelPosition * from,
const GstAudioChannelPosition * to)
{
const GstAudioFormatInfo *info;
gint i, j, n;
gint reorder_map[64] = { 0, };
guint8 *ptr;
gint bpf, bps;
guint8 tmp[64 * 8];
info = gst_audio_format_get_info (format);
g_return_val_if_fail (data != NULL, FALSE);
g_return_val_if_fail (from != NULL, FALSE);
g_return_val_if_fail (info != NULL && info->width > 0, FALSE);
g_return_val_if_fail (info->width > 0, FALSE);
g_return_val_if_fail (info->width <= 8 * 64, FALSE);
g_return_val_if_fail (size % ((info->width * channels) / 8) != 0, FALSE);
g_return_val_if_fail (channels > 0, FALSE);
g_return_val_if_fail (channels <= 64, FALSE);
g_return_val_if_fail (to != NULL, FALSE);
g_return_val_if_fail (check_valid_channel_positions (from, channels, FALSE,
NULL), FALSE);
g_return_val_if_fail (check_valid_channel_positions (to, channels, FALSE,
NULL), FALSE);
if (size == 0)
return TRUE;
if (memcmp (from, to, channels * sizeof (from[0])) == 0)
return TRUE;
bps = info->width / 8;
bpf = bps * channels;
/* Build reorder map and check compatibility */
for (i = 0; i < channels; i++) {
if (from[i] == GST_AUDIO_CHANNEL_POSITION_NONE
|| to[i] == GST_AUDIO_CHANNEL_POSITION_NONE)
return FALSE;
if (from[i] == GST_AUDIO_CHANNEL_POSITION_INVALID
|| to[i] == GST_AUDIO_CHANNEL_POSITION_INVALID)
return FALSE;
if (from[i] == GST_AUDIO_CHANNEL_POSITION_MONO
|| to[i] == GST_AUDIO_CHANNEL_POSITION_MONO)
return FALSE;
for (j = 0; j < channels; j++) {
if (from[i] == to[j]) {
reorder_map[i] = j;
break;
}
}
/* Not all channels present in both */
if (j == channels)
return FALSE;
}
ptr = data;
n = size / bpf;
for (i = 0; i < n; i++) {
memcpy (tmp, ptr, bpf);
for (j = 0; j < channels; j++)
memcpy (ptr + reorder_map[j] * bps, tmp + j * bps, bps);
ptr += bpf;
}
return TRUE;
}
Reference in a new issue View git blame Copy permalink