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gstreamer/subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioiec61937.c

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/* GStreamer audio helper functions for IEC 61937 payloading
* (c) 2011 Intel Corporation
* 2011 Collabora Multimedia
* 2011 Arun Raghavan <arun.raghavan@collabora.co.uk>
*
* 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:gstaudioiec61937
* @title: GstAudio IEC61937
* @short_description: Utility functions for IEC 61937 payloading
*
* This module contains some helper functions for encapsulating various
* audio formats in IEC 61937 headers and padding.
*
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include <gst/audio/audio.h>
#include "gstaudioiec61937.h"
#define IEC61937_HEADER_SIZE 8
#define IEC61937_PAYLOAD_SIZE_AC3 (1536 * 4)
#define IEC61937_PAYLOAD_SIZE_EAC3 (6144 * 4)
#define IEC61937_PAYLOAD_SIZE_AAC (1024 * 4)
static gint
caps_get_int_field (const GstCaps * caps, const gchar * field)
{
const GstStructure *st;
gint ret = 0;
st = gst_caps_get_structure (caps, 0);
gst_structure_get_int (st, field, &ret);
return ret;
}
static const gchar *
caps_get_string_field (const GstCaps * caps, const gchar * field)
{
const GstStructure *st = gst_caps_get_structure (caps, 0);
return gst_structure_get_string (st, field);
}
/**
* gst_audio_iec61937_frame_size:
* @spec: the ringbufer spec
*
* Calculated the size of the buffer expected by gst_audio_iec61937_payload() for
* payloading type from @spec.
*
* Returns: the size or 0 if the given @type is not supported or cannot be
* payloaded.
*/
guint
gst_audio_iec61937_frame_size (const GstAudioRingBufferSpec * spec)
{
switch (spec->type) {
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3:
return IEC61937_PAYLOAD_SIZE_AC3;
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_EAC3:
/* Check that the parser supports /some/ alignment. Need to be less
* strict about this at checking time since the alignment is dynamically
* set at the moment. */
if (caps_get_string_field (spec->caps, "alignment"))
return IEC61937_PAYLOAD_SIZE_EAC3;
else
return 0;
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DTS:
{
gint dts_frame_size = caps_get_int_field (spec->caps, "frame-size");
gint iec_frame_size = caps_get_int_field (spec->caps, "block-size") * 4;
/* Note: this will also (correctly) fail if either field is missing */
if (iec_frame_size >= (dts_frame_size + IEC61937_HEADER_SIZE))
return iec_frame_size;
else
return 0;
}
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG:
{
int version, layer, channels, frames;
version = caps_get_int_field (spec->caps, "mpegaudioversion");
layer = caps_get_int_field (spec->caps, "layer");
channels = caps_get_int_field (spec->caps, "channels");
/* Bail out if we can't figure out either, if it's MPEG 2.5, or if it's
* MP3 with multichannel audio */
if (!version || !layer || version == 3 || channels > 2)
return 0;
if (version == 1 && layer == 1)
frames = 384;
else if (version == 2 && layer == 1 && spec->info.rate <= 12000)
frames = 768;
else if (version == 2 && layer == 2 && spec->info.rate <= 12000)
frames = 2304;
else {
/* MPEG-1 layer 2,3, MPEG-2 with or without extension,
* MPEG-2 layer 3 low sample freq. */
frames = 1152;
}
return frames * 4;
}
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC:
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC:
{
return IEC61937_PAYLOAD_SIZE_AAC;
}
default:
return 0;
}
}
/**
* gst_audio_iec61937_payload:
* @src: (array length=src_n): a buffer containing the data to payload
* @src_n: size of @src in bytes
* @dst: (array length=dst_n): the destination buffer to store the
* payloaded contents in. Should not overlap with @src
* @dst_n: size of @dst in bytes
* @spec: the ringbufer spec for @src
* @endianness: the expected byte order of the payloaded data
*
* Payloads @src in the form specified by IEC 61937 for the type from @spec and
* stores the result in @dst. @src must contain exactly one frame of data and
* the frame is not checked for errors.
*
* Returns: transfer-full: %TRUE if the payloading was successful, %FALSE
* otherwise.
*/
gboolean
gst_audio_iec61937_payload (const guint8 * src, guint src_n, guint8 * dst,
guint dst_n, const GstAudioRingBufferSpec * spec, gint endianness)
{
guint i, tmp;
#if G_BYTE_ORDER == G_BIG_ENDIAN
guint8 zero = 0, one = 1, two = 2, three = 3, four = 4, five = 5, six = 6,
seven = 7;
#else
/* We need to send the data byte-swapped */
guint8 zero = 1, one = 0, two = 3, three = 2, four = 5, five = 4, six = 7,
seven = 6;
#endif
g_return_val_if_fail (src != NULL, FALSE);
g_return_val_if_fail (dst != NULL, FALSE);
g_return_val_if_fail (src != dst, FALSE);
g_return_val_if_fail (dst_n >= gst_audio_iec61937_frame_size (spec), FALSE);
if (dst_n < src_n + IEC61937_HEADER_SIZE)
return FALSE;
/* Pa, Pb */
dst[zero] = 0xF8;
dst[one] = 0x72;
dst[two] = 0x4E;
dst[three] = 0x1F;
switch (spec->type) {
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3:
{
g_return_val_if_fail (src_n >= 6, FALSE);
/* Pc: bit 13-15 - stream number (0)
* bit 11-12 - reserved (0)
* bit 8-10 - bsmod from AC3 frame */
dst[four] = src[5] & 0x7;
/* Pc: bit 7 - error bit (0)
* bit 5-6 - subdata type (0)
* bit 0-4 - data type (1) */
dst[five] = 1;
/* Pd: bit 15-0 - frame size in bits */
tmp = src_n * 8;
dst[six] = (guint8) (tmp >> 8);
dst[seven] = (guint8) (tmp & 0xff);
break;
}
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_EAC3:
{
if (g_str_equal (caps_get_string_field (spec->caps, "alignment"),
"iec61937"))
return FALSE;
/* Pc: bit 13-15 - stream number (0)
* bit 11-12 - reserved (0)
* bit 8-10 - bsmod from E-AC3 frame if present */
/* FIXME: this works, but nicer if we can put in the actual bsmod */
dst[four] = 0;
/* Pc: bit 7 - error bit (0)
* bit 5-6 - subdata type (0)
* bit 0-4 - data type (21) */
dst[five] = 21;
/* Pd: bit 15-0 - frame size in bytes */
dst[six] = ((guint16) src_n) >> 8;
dst[seven] = ((guint16) src_n) & 0xff;
break;
}
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DTS:
{
int blocksize = caps_get_int_field (spec->caps, "block-size");
g_return_val_if_fail (src_n != 0, FALSE);
if (blocksize == 0)
return FALSE;
/* Pc: bit 13-15 - stream number (0)
* bit 11-12 - reserved (0)
* bit 8-10 - for DTS type I-III (0) */
dst[four] = 0;
/* Pc: bit 7 - error bit (0)
* bit 5-6 - reserved (0)
* bit 0-4 - data type (11 = type I, 12 = type II,
* 13 = type III) */
dst[five] = 11 + (blocksize / 1024);
/* Pd: bit 15-0 - frame size, in bits (for type I-III) */
tmp = src_n * 8;
dst[six] = ((guint16) tmp) >> 8;
dst[seven] = ((guint16) tmp) & 0xff;
break;
}
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG:
{
int version, layer;
version = caps_get_int_field (spec->caps, "mpegaudioversion");
layer = caps_get_int_field (spec->caps, "layer");
g_return_val_if_fail (version > 0 && layer > 0, FALSE);
/* NOTE: multichannel audio (MPEG-2) is not supported */
/* Pc: bit 13-15 - stream number (0)
* bit 11-12 - reserved (0)
* bit 9-10 - 0 - no dynamic range control
* - 2 - dynamic range control exists
* - 1,3 - reserved
* bit 8 - Normal (0) or Karaoke (1) mode */
dst[four] = 0;
/* Pc: bit 7 - error bit (0)
* bit 5-6 - reserved (0)
* bit 0-4 - data type (04 = MPEG 1, Layer 1
* 05 = MPEG 1, Layer 2, 3 / MPEG 2, w/o ext.
* 06 = MPEG 2, with extension
* 08 - MPEG 2 LSF, Layer 1
* 09 - MPEG 2 LSF, Layer 2
* 10 - MPEG 2 LSF, Layer 3
* FIXME: we don't handle type 06 at the moment */
if (version == 1 && layer == 1)
dst[five] = 0x04;
else if ((version == 1 && (layer == 2 || layer == 3)) ||
(version == 2 && spec->info.rate >= 12000))
dst[five] = 0x05;
else if (version == 2 && layer == 1 && spec->info.rate < 12000)
dst[five] = 0x08;
else if (version == 2 && layer == 2 && spec->info.rate < 12000)
dst[five] = 0x09;
else if (version == 2 && layer == 3 && spec->info.rate < 12000)
dst[five] = 0x0A;
else
g_return_val_if_reached (FALSE);
/* Pd: bit 15-0 - frame size in bits */
dst[six] = ((guint16) src_n * 8) >> 8;
dst[seven] = ((guint16) src_n * 8) & 0xff;
break;
}
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC:
/* HACK. disguising MPEG4 AAC as MPEG2 AAC seems to work. */
/* TODO: set the right Pc,Pd for MPEG4 in accordance with IEC61937-6 */
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC:
{
int num_rd_blks;
g_return_val_if_fail (src_n >= 7, FALSE);
num_rd_blks = (src[6] & 0x03) + 1;
/* Pc: bit 13-15 - stream number (0)
* bit 11-12 - reserved (0)
* bit 8-10 - reserved? (0) */
dst[four] = 0;
/* Pc: bit 7 - error bit (0)
* bit 5-6 - reserved (0)
* bit 0-4 - data type (07 = MPEG2 AAC ADTS
* 19 = MPEG2 AAC ADTS half-rate LSF
* 51 = MPEG2 AAC ADTS quater-rate LSF */
if (num_rd_blks == 1)
dst[five] = 0x07;
else if (num_rd_blks == 2)
dst[five] = 0x13;
else if (num_rd_blks == 4)
dst[five] = 0x33;
else
g_return_val_if_reached (FALSE);
/* Pd: bit 15-0 - frame size in bits */
tmp = GST_ROUND_UP_2 (src_n) * 8;
dst[six] = (guint8) (tmp >> 8);
dst[seven] = (guint8) (tmp & 0xff);
break;
}
default:
return FALSE;
}
/* Copy the payload */
i = 8;
if (G_BYTE_ORDER == endianness) {
memcpy (dst + i, src, src_n);
} else {
/* Byte-swapped again */
/* FIXME: orc-ify this */
for (tmp = 1; tmp < src_n; tmp += 2) {
dst[i + tmp - 1] = src[tmp];
dst[i + tmp] = src[tmp - 1];
}
/* Do we have 1 byte remaining? */
if (src_n % 2) {
dst[i + src_n - 1] = 0;
dst[i + src_n] = src[src_n - 1];
i++;
}
}
i += src_n;
/* Zero the rest */
memset (dst + i, 0, dst_n - i);
return TRUE;
}
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