/* GStreamer audio helper functions for IEC 61937 payloading * (c) 2011 Intel Corporation * 2011 Collabora Multimedia * 2011 Arun Raghavan * * 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 #include #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; }