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367 lines
11 KiB
C
367 lines
11 KiB
C
/* GStreamer audio helper functions for IEC 61937 payloading
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* (c) 2011 Intel Corporation
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* 2011 Collabora Multimedia
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* 2011 Arun Raghavan <arun.raghavan@collabora.co.uk>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:gstaudioiec61937
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* @title: GstAudio IEC61937
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* @short_description: Utility functions for IEC 61937 payloading
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*
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* This module contains some helper functions for encapsulating various
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* audio formats in IEC 61937 headers and padding.
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*
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <string.h>
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#include <gst/audio/audio.h>
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#include "gstaudioiec61937.h"
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#define IEC61937_HEADER_SIZE 8
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#define IEC61937_PAYLOAD_SIZE_AC3 (1536 * 4)
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#define IEC61937_PAYLOAD_SIZE_EAC3 (6144 * 4)
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#define IEC61937_PAYLOAD_SIZE_AAC (1024 * 4)
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static gint
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caps_get_int_field (const GstCaps * caps, const gchar * field)
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{
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const GstStructure *st;
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gint ret = 0;
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st = gst_caps_get_structure (caps, 0);
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gst_structure_get_int (st, field, &ret);
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return ret;
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}
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static const gchar *
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caps_get_string_field (const GstCaps * caps, const gchar * field)
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{
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const GstStructure *st = gst_caps_get_structure (caps, 0);
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return gst_structure_get_string (st, field);
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}
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/**
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* gst_audio_iec61937_frame_size:
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* @spec: the ringbufer spec
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*
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* Calculated the size of the buffer expected by gst_audio_iec61937_payload() for
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* payloading type from @spec.
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*
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* Returns: the size or 0 if the given @type is not supported or cannot be
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* payloaded.
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*/
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guint
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gst_audio_iec61937_frame_size (const GstAudioRingBufferSpec * spec)
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{
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switch (spec->type) {
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case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3:
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return IEC61937_PAYLOAD_SIZE_AC3;
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case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_EAC3:
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/* Check that the parser supports /some/ alignment. Need to be less
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* strict about this at checking time since the alignment is dynamically
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* set at the moment. */
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if (caps_get_string_field (spec->caps, "alignment"))
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return IEC61937_PAYLOAD_SIZE_EAC3;
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else
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return 0;
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case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DTS:
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{
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gint dts_frame_size = caps_get_int_field (spec->caps, "frame-size");
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gint iec_frame_size = caps_get_int_field (spec->caps, "block-size") * 4;
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/* Note: this will also (correctly) fail if either field is missing */
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if (iec_frame_size >= (dts_frame_size + IEC61937_HEADER_SIZE))
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return iec_frame_size;
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else
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return 0;
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}
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case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG:
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{
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int version, layer, channels, frames;
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version = caps_get_int_field (spec->caps, "mpegaudioversion");
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layer = caps_get_int_field (spec->caps, "layer");
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channels = caps_get_int_field (spec->caps, "channels");
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/* Bail out if we can't figure out either, if it's MPEG 2.5, or if it's
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* MP3 with multichannel audio */
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if (!version || !layer || version == 3 || channels > 2)
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return 0;
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if (version == 1 && layer == 1)
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frames = 384;
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else if (version == 2 && layer == 1 && spec->info.rate <= 12000)
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frames = 768;
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else if (version == 2 && layer == 2 && spec->info.rate <= 12000)
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frames = 2304;
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else {
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/* MPEG-1 layer 2,3, MPEG-2 with or without extension,
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* MPEG-2 layer 3 low sample freq. */
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frames = 1152;
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}
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return frames * 4;
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}
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case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC:
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case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC:
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{
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return IEC61937_PAYLOAD_SIZE_AAC;
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}
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default:
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return 0;
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}
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}
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/**
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* gst_audio_iec61937_payload:
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* @src: (array length=src_n): a buffer containing the data to payload
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* @src_n: size of @src in bytes
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* @dst: (array length=dst_n): the destination buffer to store the
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* payloaded contents in. Should not overlap with @src
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* @dst_n: size of @dst in bytes
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* @spec: the ringbufer spec for @src
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* @endianness: the expected byte order of the payloaded data
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*
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* Payloads @src in the form specified by IEC 61937 for the type from @spec and
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* stores the result in @dst. @src must contain exactly one frame of data and
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* the frame is not checked for errors.
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*
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* Returns: transfer-full: %TRUE if the payloading was successful, %FALSE
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* otherwise.
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*/
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gboolean
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gst_audio_iec61937_payload (const guint8 * src, guint src_n, guint8 * dst,
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guint dst_n, const GstAudioRingBufferSpec * spec, gint endianness)
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{
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guint i, tmp;
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#if G_BYTE_ORDER == G_BIG_ENDIAN
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guint8 zero = 0, one = 1, two = 2, three = 3, four = 4, five = 5, six = 6,
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seven = 7;
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#else
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/* We need to send the data byte-swapped */
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guint8 zero = 1, one = 0, two = 3, three = 2, four = 5, five = 4, six = 7,
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seven = 6;
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#endif
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g_return_val_if_fail (src != NULL, FALSE);
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g_return_val_if_fail (dst != NULL, FALSE);
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g_return_val_if_fail (src != dst, FALSE);
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g_return_val_if_fail (dst_n >= gst_audio_iec61937_frame_size (spec), FALSE);
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if (dst_n < src_n + IEC61937_HEADER_SIZE)
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return FALSE;
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/* Pa, Pb */
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dst[zero] = 0xF8;
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dst[one] = 0x72;
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dst[two] = 0x4E;
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dst[three] = 0x1F;
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switch (spec->type) {
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case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3:
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{
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g_return_val_if_fail (src_n >= 6, FALSE);
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/* Pc: bit 13-15 - stream number (0)
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* bit 11-12 - reserved (0)
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* bit 8-10 - bsmod from AC3 frame */
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dst[four] = src[5] & 0x7;
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/* Pc: bit 7 - error bit (0)
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* bit 5-6 - subdata type (0)
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* bit 0-4 - data type (1) */
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dst[five] = 1;
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/* Pd: bit 15-0 - frame size in bits */
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tmp = src_n * 8;
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dst[six] = (guint8) (tmp >> 8);
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dst[seven] = (guint8) (tmp & 0xff);
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break;
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}
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case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_EAC3:
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{
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if (g_str_equal (caps_get_string_field (spec->caps, "alignment"),
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"iec61937"))
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return FALSE;
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/* Pc: bit 13-15 - stream number (0)
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* bit 11-12 - reserved (0)
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* bit 8-10 - bsmod from E-AC3 frame if present */
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/* FIXME: this works, but nicer if we can put in the actual bsmod */
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dst[four] = 0;
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/* Pc: bit 7 - error bit (0)
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* bit 5-6 - subdata type (0)
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* bit 0-4 - data type (21) */
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dst[five] = 21;
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/* Pd: bit 15-0 - frame size in bytes */
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dst[six] = ((guint16) src_n) >> 8;
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dst[seven] = ((guint16) src_n) & 0xff;
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break;
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}
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case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DTS:
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{
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int blocksize = caps_get_int_field (spec->caps, "block-size");
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g_return_val_if_fail (src_n != 0, FALSE);
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if (blocksize == 0)
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return FALSE;
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/* Pc: bit 13-15 - stream number (0)
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* bit 11-12 - reserved (0)
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* bit 8-10 - for DTS type I-III (0) */
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dst[four] = 0;
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/* Pc: bit 7 - error bit (0)
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* bit 5-6 - reserved (0)
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* bit 0-4 - data type (11 = type I, 12 = type II,
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* 13 = type III) */
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dst[five] = 11 + (blocksize / 1024);
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/* Pd: bit 15-0 - frame size, in bits (for type I-III) */
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tmp = src_n * 8;
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dst[six] = ((guint16) tmp) >> 8;
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dst[seven] = ((guint16) tmp) & 0xff;
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break;
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}
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case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG:
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{
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int version, layer;
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version = caps_get_int_field (spec->caps, "mpegaudioversion");
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layer = caps_get_int_field (spec->caps, "layer");
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g_return_val_if_fail (version > 0 && layer > 0, FALSE);
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/* NOTE: multichannel audio (MPEG-2) is not supported */
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/* Pc: bit 13-15 - stream number (0)
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* bit 11-12 - reserved (0)
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* bit 9-10 - 0 - no dynamic range control
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* - 2 - dynamic range control exists
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* - 1,3 - reserved
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* bit 8 - Normal (0) or Karaoke (1) mode */
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dst[four] = 0;
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/* Pc: bit 7 - error bit (0)
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* bit 5-6 - reserved (0)
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* bit 0-4 - data type (04 = MPEG 1, Layer 1
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* 05 = MPEG 1, Layer 2, 3 / MPEG 2, w/o ext.
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* 06 = MPEG 2, with extension
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* 08 - MPEG 2 LSF, Layer 1
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* 09 - MPEG 2 LSF, Layer 2
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* 10 - MPEG 2 LSF, Layer 3
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* FIXME: we don't handle type 06 at the moment */
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if (version == 1 && layer == 1)
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dst[five] = 0x04;
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else if ((version == 1 && (layer == 2 || layer == 3)) ||
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(version == 2 && spec->info.rate >= 12000))
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dst[five] = 0x05;
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else if (version == 2 && layer == 1 && spec->info.rate < 12000)
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dst[five] = 0x08;
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else if (version == 2 && layer == 2 && spec->info.rate < 12000)
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dst[five] = 0x09;
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else if (version == 2 && layer == 3 && spec->info.rate < 12000)
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dst[five] = 0x0A;
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else
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g_return_val_if_reached (FALSE);
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/* Pd: bit 15-0 - frame size in bits */
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dst[six] = ((guint16) src_n * 8) >> 8;
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dst[seven] = ((guint16) src_n * 8) & 0xff;
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break;
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}
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case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC:
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/* HACK. disguising MPEG4 AAC as MPEG2 AAC seems to work. */
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/* TODO: set the right Pc,Pd for MPEG4 in accordance with IEC61937-6 */
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case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC:
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{
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int num_rd_blks;
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g_return_val_if_fail (src_n >= 7, FALSE);
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num_rd_blks = (src[6] & 0x03) + 1;
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/* Pc: bit 13-15 - stream number (0)
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* bit 11-12 - reserved (0)
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* bit 8-10 - reserved? (0) */
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dst[four] = 0;
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/* Pc: bit 7 - error bit (0)
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* bit 5-6 - reserved (0)
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* bit 0-4 - data type (07 = MPEG2 AAC ADTS
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* 19 = MPEG2 AAC ADTS half-rate LSF
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* 51 = MPEG2 AAC ADTS quater-rate LSF */
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if (num_rd_blks == 1)
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dst[five] = 0x07;
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else if (num_rd_blks == 2)
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dst[five] = 0x13;
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else if (num_rd_blks == 4)
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dst[five] = 0x33;
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else
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g_return_val_if_reached (FALSE);
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/* Pd: bit 15-0 - frame size in bits */
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tmp = GST_ROUND_UP_2 (src_n) * 8;
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dst[six] = (guint8) (tmp >> 8);
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dst[seven] = (guint8) (tmp & 0xff);
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break;
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}
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default:
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return FALSE;
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}
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/* Copy the payload */
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i = 8;
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if (G_BYTE_ORDER == endianness) {
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memcpy (dst + i, src, src_n);
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} else {
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/* Byte-swapped again */
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/* FIXME: orc-ify this */
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for (tmp = 1; tmp < src_n; tmp += 2) {
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dst[i + tmp - 1] = src[tmp];
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dst[i + tmp] = src[tmp - 1];
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}
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/* Do we have 1 byte remaining? */
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if (src_n % 2) {
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dst[i + src_n - 1] = 0;
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dst[i + src_n] = src[src_n - 1];
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i++;
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}
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}
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i += src_n;
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/* Zero the rest */
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memset (dst + i, 0, dst_n - i);
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return TRUE;
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}
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