/* GStreamer AC3 parser * Copyright (C) 2009 Tim-Philipp Müller <tim centricular net> * Copyright (C) 2009 Mark Nauwelaerts <mnauw users sf net> * Copyright (C) 2009 Nokia Corporation. All rights reserved. * Contact: Stefan Kost <stefan.kost@nokia.com> * * 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:element-ac3parse * @title: ac3parse * @short_description: AC3 parser * @see_also: #GstAmrParse, #GstAACParse * * This is an AC3 parser. * * ## Example launch line * |[ * gst-launch-1.0 filesrc location=abc.ac3 ! ac3parse ! a52dec ! audioresample ! audioconvert ! autoaudiosink * ]| * */ /* TODO: * - audio/ac3 to audio/x-private1-ac3 is not implemented (done in the muxer) * - should accept framed and unframed input (needs decodebin fixes first) */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include <string.h> #include "gstaudioparserselements.h" #include "gstac3parse.h" #include <gst/base/base.h> #include <gst/pbutils/pbutils.h> GST_DEBUG_CATEGORY_STATIC (ac3_parse_debug); #define GST_CAT_DEFAULT ac3_parse_debug static const struct { const guint bit_rate; /* nominal bit rate */ const guint frame_size[3]; /* frame size for 32kHz, 44kHz, and 48kHz */ } frmsizcod_table[38] = { { 32, { 64, 69, 96}}, { 32, { 64, 70, 96}}, { 40, { 80, 87, 120}}, { 40, { 80, 88, 120}}, { 48, { 96, 104, 144}}, { 48, { 96, 105, 144}}, { 56, { 112, 121, 168}}, { 56, { 112, 122, 168}}, { 64, { 128, 139, 192}}, { 64, { 128, 140, 192}}, { 80, { 160, 174, 240}}, { 80, { 160, 175, 240}}, { 96, { 192, 208, 288}}, { 96, { 192, 209, 288}}, { 112, { 224, 243, 336}}, { 112, { 224, 244, 336}}, { 128, { 256, 278, 384}}, { 128, { 256, 279, 384}}, { 160, { 320, 348, 480}}, { 160, { 320, 349, 480}}, { 192, { 384, 417, 576}}, { 192, { 384, 418, 576}}, { 224, { 448, 487, 672}}, { 224, { 448, 488, 672}}, { 256, { 512, 557, 768}}, { 256, { 512, 558, 768}}, { 320, { 640, 696, 960}}, { 320, { 640, 697, 960}}, { 384, { 768, 835, 1152}}, { 384, { 768, 836, 1152}}, { 448, { 896, 975, 1344}}, { 448, { 896, 976, 1344}}, { 512, { 1024, 1114, 1536}}, { 512, { 1024, 1115, 1536}}, { 576, { 1152, 1253, 1728}}, { 576, { 1152, 1254, 1728}}, { 640, { 1280, 1393, 1920}}, { 640, { 1280, 1394, 1920}} }; static const guint fscod_rates[4] = { 48000, 44100, 32000, 0 }; static const guint acmod_chans[8] = { 2, 1, 2, 3, 3, 4, 4, 5 }; static const guint numblks[4] = { 1, 2, 3, 6 }; static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-ac3, framed = (boolean) true, " " channels = (int) [ 1, 6 ], rate = (int) [ 8000, 48000 ], " " alignment = (string) { iec61937, frame}; " "audio/x-eac3, framed = (boolean) true, " " channels = (int) [ 1, 6 ], rate = (int) [ 8000, 48000 ], " " alignment = (string) { iec61937, frame}; ")); static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-ac3; " "audio/x-eac3; " "audio/ac3; " "audio/x-private1-ac3")); static void gst_ac3_parse_finalize (GObject * object); static gboolean gst_ac3_parse_start (GstBaseParse * parse); static gboolean gst_ac3_parse_stop (GstBaseParse * parse); static GstFlowReturn gst_ac3_parse_handle_frame (GstBaseParse * parse, GstBaseParseFrame * frame, gint * skipsize); static GstFlowReturn gst_ac3_parse_pre_push_frame (GstBaseParse * parse, GstBaseParseFrame * frame); static gboolean gst_ac3_parse_src_event (GstBaseParse * parse, GstEvent * event); static GstCaps *gst_ac3_parse_get_sink_caps (GstBaseParse * parse, GstCaps * filter); static gboolean gst_ac3_parse_set_sink_caps (GstBaseParse * parse, GstCaps * caps); #define gst_ac3_parse_parent_class parent_class G_DEFINE_TYPE (GstAc3Parse, gst_ac3_parse, GST_TYPE_BASE_PARSE); GST_ELEMENT_REGISTER_DEFINE (ac3parse, "ac3parse", GST_RANK_PRIMARY + 1, GST_TYPE_AC3_PARSE); static void gst_ac3_parse_class_init (GstAc3ParseClass * klass) { GObjectClass *object_class = G_OBJECT_CLASS (klass); GstElementClass *element_class = GST_ELEMENT_CLASS (klass); GstBaseParseClass *parse_class = GST_BASE_PARSE_CLASS (klass); GST_DEBUG_CATEGORY_INIT (ac3_parse_debug, "ac3parse", 0, "AC3 audio stream parser"); object_class->finalize = gst_ac3_parse_finalize; gst_element_class_add_static_pad_template (element_class, &sink_template); gst_element_class_add_static_pad_template (element_class, &src_template); gst_element_class_set_static_metadata (element_class, "AC3 audio stream parser", "Codec/Parser/Converter/Audio", "AC3 parser", "Tim-Philipp Müller <tim centricular net>"); parse_class->start = GST_DEBUG_FUNCPTR (gst_ac3_parse_start); parse_class->stop = GST_DEBUG_FUNCPTR (gst_ac3_parse_stop); parse_class->handle_frame = GST_DEBUG_FUNCPTR (gst_ac3_parse_handle_frame); parse_class->pre_push_frame = GST_DEBUG_FUNCPTR (gst_ac3_parse_pre_push_frame); parse_class->src_event = GST_DEBUG_FUNCPTR (gst_ac3_parse_src_event); parse_class->get_sink_caps = GST_DEBUG_FUNCPTR (gst_ac3_parse_get_sink_caps); parse_class->set_sink_caps = GST_DEBUG_FUNCPTR (gst_ac3_parse_set_sink_caps); } static void gst_ac3_parse_reset (GstAc3Parse * ac3parse) { ac3parse->channels = -1; ac3parse->sample_rate = -1; ac3parse->blocks = -1; ac3parse->eac = FALSE; ac3parse->sent_codec_tag = FALSE; g_atomic_int_set (&ac3parse->align, GST_AC3_PARSE_ALIGN_NONE); } static void gst_ac3_parse_init (GstAc3Parse * ac3parse) { gst_base_parse_set_min_frame_size (GST_BASE_PARSE (ac3parse), 8); gst_ac3_parse_reset (ac3parse); ac3parse->baseparse_chainfunc = GST_BASE_PARSE_SINK_PAD (GST_BASE_PARSE (ac3parse))->chainfunc; GST_PAD_SET_ACCEPT_INTERSECT (GST_BASE_PARSE_SINK_PAD (ac3parse)); GST_PAD_SET_ACCEPT_TEMPLATE (GST_BASE_PARSE_SINK_PAD (ac3parse)); } static void gst_ac3_parse_finalize (GObject * object) { G_OBJECT_CLASS (parent_class)->finalize (object); } static gboolean gst_ac3_parse_start (GstBaseParse * parse) { GstAc3Parse *ac3parse = GST_AC3_PARSE (parse); GST_DEBUG_OBJECT (parse, "starting"); gst_ac3_parse_reset (ac3parse); return TRUE; } static gboolean gst_ac3_parse_stop (GstBaseParse * parse) { GST_DEBUG_OBJECT (parse, "stopping"); return TRUE; } static void gst_ac3_parse_set_alignment (GstAc3Parse * ac3parse, gboolean eac) { GstCaps *caps; GstStructure *st; const gchar *str = NULL; int i; if (G_LIKELY (!eac)) goto done; caps = gst_pad_get_allowed_caps (GST_BASE_PARSE_SRC_PAD (ac3parse)); if (!caps) goto done; for (i = 0; i < gst_caps_get_size (caps); i++) { st = gst_caps_get_structure (caps, i); if (!g_str_equal (gst_structure_get_name (st), "audio/x-eac3")) continue; if ((str = gst_structure_get_string (st, "alignment"))) { if (g_str_equal (str, "iec61937")) { g_atomic_int_set (&ac3parse->align, GST_AC3_PARSE_ALIGN_IEC61937); GST_DEBUG_OBJECT (ac3parse, "picked iec61937 alignment"); } else if (g_str_equal (str, "frame") == 0) { g_atomic_int_set (&ac3parse->align, GST_AC3_PARSE_ALIGN_FRAME); GST_DEBUG_OBJECT (ac3parse, "picked frame alignment"); } else { g_atomic_int_set (&ac3parse->align, GST_AC3_PARSE_ALIGN_FRAME); GST_WARNING_OBJECT (ac3parse, "unknown alignment: %s", str); } break; } } if (caps) gst_caps_unref (caps); done: /* default */ if (ac3parse->align == GST_AC3_PARSE_ALIGN_NONE) { g_atomic_int_set (&ac3parse->align, GST_AC3_PARSE_ALIGN_FRAME); GST_DEBUG_OBJECT (ac3parse, "picked syncframe alignment"); } } static gboolean gst_ac3_parse_frame_header_ac3 (GstAc3Parse * ac3parse, GstBuffer * buf, gint skip, guint * frame_size, guint * rate, guint * chans, guint * blks, guint * sid) { GstBitReader bits; GstMapInfo map; guint8 fscod, frmsizcod, bsid, acmod, lfe_on, rate_scale; gboolean ret = FALSE; GST_LOG_OBJECT (ac3parse, "parsing ac3"); gst_buffer_map (buf, &map, GST_MAP_READ); gst_bit_reader_init (&bits, map.data, map.size); gst_bit_reader_skip_unchecked (&bits, skip * 8); gst_bit_reader_skip_unchecked (&bits, 16 + 16); fscod = gst_bit_reader_get_bits_uint8_unchecked (&bits, 2); frmsizcod = gst_bit_reader_get_bits_uint8_unchecked (&bits, 6); if (G_UNLIKELY (fscod == 3 || frmsizcod >= G_N_ELEMENTS (frmsizcod_table))) { GST_DEBUG_OBJECT (ac3parse, "bad fscod=%d frmsizcod=%d", fscod, frmsizcod); goto cleanup; } bsid = gst_bit_reader_get_bits_uint8_unchecked (&bits, 5); gst_bit_reader_skip_unchecked (&bits, 3); /* bsmod */ acmod = gst_bit_reader_get_bits_uint8_unchecked (&bits, 3); /* spec not quite clear here: decoder should decode if less than 8, * but seemingly only defines 6 and 8 cases */ /* Files with 9 and 10 happen, and seem to comply with the <= 8 format, so let them through. The spec says nothing about 9 and 10 */ if (bsid > 10) { GST_DEBUG_OBJECT (ac3parse, "unexpected bsid=%d", bsid); goto cleanup; } else if (bsid != 8 && bsid != 6) { GST_DEBUG_OBJECT (ac3parse, "undefined bsid=%d", bsid); } if ((acmod & 0x1) && (acmod != 0x1)) /* 3 front channels */ gst_bit_reader_skip_unchecked (&bits, 2); if ((acmod & 0x4)) /* if a surround channel exists */ gst_bit_reader_skip_unchecked (&bits, 2); if (acmod == 0x2) /* if in 2/0 mode */ gst_bit_reader_skip_unchecked (&bits, 2); lfe_on = gst_bit_reader_get_bits_uint8_unchecked (&bits, 1); /* 6/8->0, 9->1, 10->2, see http://matroska.org/technical/specs/codecid/index.html */ rate_scale = (CLAMP (bsid, 8, 10) - 8); if (frame_size) *frame_size = frmsizcod_table[frmsizcod].frame_size[fscod] * 2; if (rate) *rate = fscod_rates[fscod] >> rate_scale; if (chans) *chans = acmod_chans[acmod] + lfe_on; if (blks) *blks = 6; if (sid) *sid = 0; ret = TRUE; cleanup: gst_buffer_unmap (buf, &map); return ret; } static gboolean gst_ac3_parse_frame_header_eac3 (GstAc3Parse * ac3parse, GstBuffer * buf, gint skip, guint * frame_size, guint * rate, guint * chans, guint * blks, guint * sid) { GstBitReader bits; GstMapInfo map; guint16 frmsiz, sample_rate, blocks; guint8 strmtyp, fscod, fscod2, acmod, lfe_on, strmid, numblkscod; gboolean ret = FALSE; GST_LOG_OBJECT (ac3parse, "parsing e-ac3"); gst_buffer_map (buf, &map, GST_MAP_READ); gst_bit_reader_init (&bits, map.data, map.size); gst_bit_reader_skip_unchecked (&bits, skip * 8); gst_bit_reader_skip_unchecked (&bits, 16); strmtyp = gst_bit_reader_get_bits_uint8_unchecked (&bits, 2); /* strmtyp */ if (G_UNLIKELY (strmtyp == 3)) { GST_DEBUG_OBJECT (ac3parse, "bad strmtyp %d", strmtyp); goto cleanup; } strmid = gst_bit_reader_get_bits_uint8_unchecked (&bits, 3); /* substreamid */ frmsiz = gst_bit_reader_get_bits_uint16_unchecked (&bits, 11); /* frmsiz */ fscod = gst_bit_reader_get_bits_uint8_unchecked (&bits, 2); /* fscod */ if (fscod == 3) { fscod2 = gst_bit_reader_get_bits_uint8_unchecked (&bits, 2); /* fscod2 */ if (G_UNLIKELY (fscod2 == 3)) { GST_DEBUG_OBJECT (ac3parse, "invalid fscod2"); goto cleanup; } sample_rate = fscod_rates[fscod2] / 2; blocks = 6; } else { numblkscod = gst_bit_reader_get_bits_uint8_unchecked (&bits, 2); /* numblkscod */ sample_rate = fscod_rates[fscod]; blocks = numblks[numblkscod]; } acmod = gst_bit_reader_get_bits_uint8_unchecked (&bits, 3); /* acmod */ lfe_on = gst_bit_reader_get_bits_uint8_unchecked (&bits, 1); /* lfeon */ gst_bit_reader_skip_unchecked (&bits, 5); /* bsid */ if (frame_size) *frame_size = (frmsiz + 1) * 2; if (rate) *rate = sample_rate; if (chans) *chans = acmod_chans[acmod] + lfe_on; if (blks) *blks = blocks; if (sid) *sid = (strmtyp & 0x1) << 3 | strmid; ret = TRUE; cleanup: gst_buffer_unmap (buf, &map); return ret; } static gboolean gst_ac3_parse_frame_header (GstAc3Parse * parse, GstBuffer * buf, gint skip, guint * framesize, guint * rate, guint * chans, guint * blocks, guint * sid, gboolean * eac) { GstBitReader bits; guint16 sync; guint8 bsid; GstMapInfo map; gboolean ret = FALSE; gst_buffer_map (buf, &map, GST_MAP_READ); gst_bit_reader_init (&bits, map.data, map.size); GST_MEMDUMP_OBJECT (parse, "AC3 frame sync", map.data, MIN (map.size, 16)); gst_bit_reader_skip_unchecked (&bits, skip * 8); sync = gst_bit_reader_get_bits_uint16_unchecked (&bits, 16); gst_bit_reader_skip_unchecked (&bits, 16 + 8); bsid = gst_bit_reader_peek_bits_uint8_unchecked (&bits, 5); if (G_UNLIKELY (sync != 0x0b77)) goto cleanup; GST_LOG_OBJECT (parse, "bsid = %d", bsid); if (bsid <= 10) { if (eac) *eac = FALSE; ret = gst_ac3_parse_frame_header_ac3 (parse, buf, skip, framesize, rate, chans, blocks, sid); goto cleanup; } else if (bsid <= 16) { if (eac) *eac = TRUE; ret = gst_ac3_parse_frame_header_eac3 (parse, buf, skip, framesize, rate, chans, blocks, sid); goto cleanup; } else { GST_DEBUG_OBJECT (parse, "unexpected bsid %d", bsid); ret = FALSE; goto cleanup; } GST_DEBUG_OBJECT (parse, "unexpected bsid %d", bsid); cleanup: gst_buffer_unmap (buf, &map); return ret; } static GstFlowReturn gst_ac3_parse_handle_frame (GstBaseParse * parse, GstBaseParseFrame * frame, gint * skipsize) { GstAc3Parse *ac3parse = GST_AC3_PARSE (parse); GstBuffer *buf = frame->buffer; GstByteReader reader; gint off; gboolean lost_sync, draining, eac, more = FALSE; guint frmsiz, blocks, sid; guint rate, chans; gboolean update_rate = FALSE; gint framesize = 0; gint have_blocks = 0; GstMapInfo map; gboolean ret = FALSE; GstFlowReturn res = GST_FLOW_OK; gst_buffer_map (buf, &map, GST_MAP_READ); if (G_UNLIKELY (map.size < 8)) { *skipsize = 1; goto cleanup; } gst_byte_reader_init (&reader, map.data, map.size); off = gst_byte_reader_masked_scan_uint32 (&reader, 0xffff0000, 0x0b770000, 0, map.size); GST_LOG_OBJECT (parse, "possible sync at buffer offset %d", off); /* didn't find anything that looks like a sync word, skip */ if (off < 0) { *skipsize = map.size - 3; goto cleanup; } /* possible frame header, but not at offset 0? skip bytes before sync */ if (off > 0) { *skipsize = off; goto cleanup; } /* make sure the values in the frame header look sane */ if (!gst_ac3_parse_frame_header (ac3parse, buf, 0, &frmsiz, &rate, &chans, &blocks, &sid, &eac)) { *skipsize = off + 2; goto cleanup; } GST_LOG_OBJECT (parse, "size: %u, blocks: %u, rate: %u, chans: %u", frmsiz, blocks, rate, chans); framesize = frmsiz; if (G_UNLIKELY (g_atomic_int_get (&ac3parse->align) == GST_AC3_PARSE_ALIGN_NONE)) gst_ac3_parse_set_alignment (ac3parse, eac); GST_LOG_OBJECT (parse, "got frame"); lost_sync = GST_BASE_PARSE_LOST_SYNC (parse); draining = GST_BASE_PARSE_DRAINING (parse); if (g_atomic_int_get (&ac3parse->align) == GST_AC3_PARSE_ALIGN_IEC61937) { /* We need 6 audio blocks from each substream, so we keep going forwards * till we have it */ g_assert (blocks > 0); GST_LOG_OBJECT (ac3parse, "Need %d frames before pushing", 6 / blocks); if (sid != 0) { /* We need the first substream to be the one with id 0 */ GST_LOG_OBJECT (ac3parse, "Skipping till we find sid 0"); *skipsize = off + 2; goto cleanup; } framesize = 0; /* Loop till we have 6 blocks per substream */ for (have_blocks = 0; !more && have_blocks < 6; have_blocks += blocks) { /* Loop till we get one frame from each substream */ do { framesize += frmsiz; if (!gst_byte_reader_skip (&reader, frmsiz) || map.size < (framesize + 6)) { more = TRUE; break; } if (!gst_ac3_parse_frame_header (ac3parse, buf, framesize, &frmsiz, NULL, NULL, NULL, &sid, &eac)) { *skipsize = off + 2; goto cleanup; } } while (sid); } /* We're now at the next frame, so no need to skip if resyncing */ frmsiz = 0; } if (lost_sync && !draining) { guint16 word = 0; GST_DEBUG_OBJECT (ac3parse, "resyncing; checking next frame syncword"); if (more || !gst_byte_reader_skip (&reader, frmsiz) || !gst_byte_reader_get_uint16_be (&reader, &word)) { GST_DEBUG_OBJECT (ac3parse, "... but not sufficient data"); gst_base_parse_set_min_frame_size (parse, framesize + 8); *skipsize = 0; goto cleanup; } else { if (word != 0x0b77) { GST_DEBUG_OBJECT (ac3parse, "0x%x not OK", word); *skipsize = off + 2; goto cleanup; } else { /* ok, got sync now, let's assume constant frame size */ gst_base_parse_set_min_frame_size (parse, framesize); } } } /* expect to have found a frame here */ g_assert (framesize); ret = TRUE; /* arrange for metadata setup */ if (G_UNLIKELY (sid)) { /* dependent frame, no need to (ac)count for or consider further */ GST_LOG_OBJECT (parse, "sid: %d", sid); frame->flags |= GST_BASE_PARSE_FRAME_FLAG_NO_FRAME; /* TODO maybe also mark as DELTA_UNIT, * if that does not surprise baseparse elsewhere */ /* occupies same time space as previous base frame */ if (G_LIKELY (GST_BUFFER_TIMESTAMP (buf) >= GST_BUFFER_DURATION (buf))) GST_BUFFER_TIMESTAMP (buf) -= GST_BUFFER_DURATION (buf); /* only shortcut if we already arranged for caps */ if (G_LIKELY (ac3parse->sample_rate > 0)) goto cleanup; } if (G_UNLIKELY (ac3parse->sample_rate != rate || ac3parse->channels != chans || ac3parse->eac != eac)) { GstCaps *caps = gst_caps_new_simple (eac ? "audio/x-eac3" : "audio/x-ac3", "framed", G_TYPE_BOOLEAN, TRUE, "rate", G_TYPE_INT, rate, "channels", G_TYPE_INT, chans, NULL); gst_caps_set_simple (caps, "alignment", G_TYPE_STRING, g_atomic_int_get (&ac3parse->align) == GST_AC3_PARSE_ALIGN_IEC61937 ? "iec61937" : "frame", NULL); gst_pad_set_caps (GST_BASE_PARSE_SRC_PAD (parse), caps); gst_caps_unref (caps); ac3parse->sample_rate = rate; ac3parse->channels = chans; ac3parse->eac = eac; update_rate = TRUE; } if (G_UNLIKELY (ac3parse->blocks != blocks)) { ac3parse->blocks = blocks; update_rate = TRUE; } if (G_UNLIKELY (update_rate)) gst_base_parse_set_frame_rate (parse, rate, 256 * blocks, 2, 2); cleanup: gst_buffer_unmap (buf, &map); if (ret && framesize <= map.size) { res = gst_base_parse_finish_frame (parse, frame, framesize); } return res; } /* * MPEG-PS private1 streams add a 2 bytes "Audio Substream Headers" for each * buffer (not each frame) with the offset of the next frame's start. * * Buffer 1: * ------------------------------------------- * |firstAccUnit|AC3SyncWord|xxxxxxxxxxxxxxxxx * ------------------------------------------- * Buffer 2: * ------------------------------------------- * |firstAccUnit|xxxxxx|AC3SyncWord|xxxxxxxxxx * ------------------------------------------- * * These 2 bytes can be dropped safely as they do not include any timing * information, only the offset to the start of the next frame. * * From http://stnsoft.com/DVD/ass-hdr.html: * "FirstAccUnit offset to frame which corresponds to PTS value offset 0 is the * last byte of FirstAccUnit, ie add the offset of byte 2 to get the AU's offset * The value 0000 indicates there is no first access unit" * */ static GstFlowReturn gst_ac3_parse_chain_priv (GstPad * pad, GstObject * parent, GstBuffer * buf) { GstAc3Parse *ac3parse = GST_AC3_PARSE (parent); GstFlowReturn ret; gsize size; guint8 data[2]; gint offset; gint len; GstBuffer *subbuf; gint first_access; size = gst_buffer_get_size (buf); if (size < 2) goto not_enough_data; gst_buffer_extract (buf, 0, data, 2); first_access = (data[0] << 8) | data[1]; /* Skip the first_access header */ offset = 2; if (first_access > 1) { /* Length of data before first_access */ len = first_access - 1; if (len <= 0 || offset + len > size) goto bad_first_access_parameter; subbuf = gst_buffer_copy_region (buf, GST_BUFFER_COPY_ALL, offset, len); GST_BUFFER_DTS (subbuf) = GST_CLOCK_TIME_NONE; GST_BUFFER_PTS (subbuf) = GST_CLOCK_TIME_NONE; ret = ac3parse->baseparse_chainfunc (pad, parent, subbuf); if (ret != GST_FLOW_OK && ret != GST_FLOW_NOT_LINKED) { gst_buffer_unref (buf); goto done; } offset += len; len = size - offset; if (len > 0) { subbuf = gst_buffer_copy_region (buf, GST_BUFFER_COPY_ALL, offset, len); GST_BUFFER_PTS (subbuf) = GST_BUFFER_PTS (buf); GST_BUFFER_DTS (subbuf) = GST_BUFFER_DTS (buf); ret = ac3parse->baseparse_chainfunc (pad, parent, subbuf); } gst_buffer_unref (buf); } else { /* first_access = 0 or 1, so if there's a timestamp it applies to the first byte */ subbuf = gst_buffer_copy_region (buf, GST_BUFFER_COPY_ALL, offset, size - offset); GST_BUFFER_PTS (subbuf) = GST_BUFFER_PTS (buf); GST_BUFFER_DTS (subbuf) = GST_BUFFER_DTS (buf); gst_buffer_unref (buf); ret = ac3parse->baseparse_chainfunc (pad, parent, subbuf); } done: return ret; /* ERRORS */ not_enough_data: { GST_ELEMENT_ERROR (GST_ELEMENT (ac3parse), STREAM, FORMAT, (NULL), ("Insufficient data in buffer. Can't determine first_acess")); gst_buffer_unref (buf); return GST_FLOW_ERROR; } bad_first_access_parameter: { GST_ELEMENT_ERROR (GST_ELEMENT (ac3parse), STREAM, FORMAT, (NULL), ("Bad first_access parameter (%d) in buffer", first_access)); gst_buffer_unref (buf); return GST_FLOW_ERROR; } } static GstFlowReturn gst_ac3_parse_pre_push_frame (GstBaseParse * parse, GstBaseParseFrame * frame) { GstAc3Parse *ac3parse = GST_AC3_PARSE (parse); if (!ac3parse->sent_codec_tag) { GstTagList *taglist; GstCaps *caps; /* codec tag */ caps = gst_pad_get_current_caps (GST_BASE_PARSE_SRC_PAD (parse)); if (G_UNLIKELY (caps == NULL)) { if (GST_PAD_IS_FLUSHING (GST_BASE_PARSE_SRC_PAD (parse))) { GST_INFO_OBJECT (parse, "Src pad is flushing"); return GST_FLOW_FLUSHING; } else { GST_INFO_OBJECT (parse, "Src pad is not negotiated!"); return GST_FLOW_NOT_NEGOTIATED; } } taglist = gst_tag_list_new_empty (); gst_pb_utils_add_codec_description_to_tag_list (taglist, GST_TAG_AUDIO_CODEC, caps); gst_caps_unref (caps); gst_base_parse_merge_tags (parse, taglist, GST_TAG_MERGE_REPLACE); gst_tag_list_unref (taglist); /* also signals the end of first-frame processing */ ac3parse->sent_codec_tag = TRUE; } frame->flags |= GST_BASE_PARSE_FRAME_FLAG_CLIP; return GST_FLOW_OK; } static gboolean gst_ac3_parse_src_event (GstBaseParse * parse, GstEvent * event) { GstAc3Parse *ac3parse = GST_AC3_PARSE (parse); if (G_UNLIKELY (GST_EVENT_TYPE (event) == GST_EVENT_CUSTOM_UPSTREAM) && gst_event_has_name (event, "ac3parse-set-alignment")) { const GstStructure *st = gst_event_get_structure (event); const gchar *align = gst_structure_get_string (st, "alignment"); if (g_str_equal (align, "iec61937")) { GST_DEBUG_OBJECT (ac3parse, "Switching to iec61937 alignment"); g_atomic_int_set (&ac3parse->align, GST_AC3_PARSE_ALIGN_IEC61937); } else if (g_str_equal (align, "frame")) { GST_DEBUG_OBJECT (ac3parse, "Switching to frame alignment"); g_atomic_int_set (&ac3parse->align, GST_AC3_PARSE_ALIGN_FRAME); } else { g_atomic_int_set (&ac3parse->align, GST_AC3_PARSE_ALIGN_FRAME); GST_WARNING_OBJECT (ac3parse, "Got unknown alignment request (%s) " "reverting to frame alignment.", gst_structure_get_string (st, "alignment")); } gst_event_unref (event); return TRUE; } return GST_BASE_PARSE_CLASS (parent_class)->src_event (parse, event); } static void remove_fields (GstCaps * caps) { guint i, n; n = gst_caps_get_size (caps); for (i = 0; i < n; i++) { GstStructure *s = gst_caps_get_structure (caps, i); gst_structure_remove_field (s, "framed"); gst_structure_remove_field (s, "alignment"); } } static GstCaps * extend_caps (GstCaps * caps, gboolean add_private) { guint i, n; GstCaps *ncaps = gst_caps_new_empty (); n = gst_caps_get_size (caps); for (i = 0; i < n; i++) { GstStructure *s = gst_caps_get_structure (caps, i); if (add_private && !gst_structure_has_name (s, "audio/x-private1-ac3")) { GstStructure *ns = gst_structure_copy (s); gst_structure_set_name (ns, "audio/x-private1-ac3"); gst_caps_append_structure (ncaps, ns); } else if (!add_private && gst_structure_has_name (s, "audio/x-private1-ac3")) { GstStructure *ns = gst_structure_copy (s); gst_structure_set_name (ns, "audio/x-ac3"); gst_caps_append_structure (ncaps, ns); ns = gst_structure_copy (s); gst_structure_set_name (ns, "audio/x-eac3"); gst_caps_append_structure (ncaps, ns); } else if (!add_private) { gst_caps_append_structure (ncaps, gst_structure_copy (s)); } } if (add_private) { gst_caps_append (caps, ncaps); } else { gst_caps_unref (caps); caps = ncaps; } return caps; } static GstCaps * gst_ac3_parse_get_sink_caps (GstBaseParse * parse, GstCaps * filter) { GstCaps *peercaps, *templ; GstCaps *res; templ = gst_pad_get_pad_template_caps (GST_BASE_PARSE_SINK_PAD (parse)); if (filter) { GstCaps *fcopy = gst_caps_copy (filter); /* Remove the fields we convert */ remove_fields (fcopy); /* we do not ask downstream to handle x-private1-ac3 */ fcopy = extend_caps (fcopy, FALSE); peercaps = gst_pad_peer_query_caps (GST_BASE_PARSE_SRC_PAD (parse), fcopy); gst_caps_unref (fcopy); } else peercaps = gst_pad_peer_query_caps (GST_BASE_PARSE_SRC_PAD (parse), NULL); if (peercaps) { /* Remove the framed and alignment field. We can convert * between different alignments. */ peercaps = gst_caps_make_writable (peercaps); remove_fields (peercaps); /* also allow for x-private1-ac3 input */ peercaps = extend_caps (peercaps, TRUE); res = gst_caps_intersect_full (peercaps, templ, GST_CAPS_INTERSECT_FIRST); gst_caps_unref (peercaps); gst_caps_unref (templ); } else { res = templ; } if (filter) { GstCaps *intersection; intersection = gst_caps_intersect_full (filter, res, GST_CAPS_INTERSECT_FIRST); gst_caps_unref (res); res = intersection; } return res; } static gboolean gst_ac3_parse_set_sink_caps (GstBaseParse * parse, GstCaps * caps) { GstStructure *s; GstAc3Parse *ac3parse = GST_AC3_PARSE (parse); s = gst_caps_get_structure (caps, 0); if (gst_structure_has_name (s, "audio/x-private1-ac3")) { gst_pad_set_chain_function (parse->sinkpad, gst_ac3_parse_chain_priv); } else { gst_pad_set_chain_function (parse->sinkpad, ac3parse->baseparse_chainfunc); } return TRUE; }