/* GStreamer DCA parser * Copyright (C) 2010 Tim-Philipp Müller * * 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-dcaparse * @title: dcaparse * @short_description: DCA (DTS Coherent Acoustics) parser * @see_also: #GstAmrParse, #GstAACParse, #GstAc3Parse * * This is a DCA (DTS Coherent Acoustics) parser. * * ## Example launch line * |[ * gst-launch-1.0 filesrc location=abc.dts ! dcaparse ! dtsdec ! audioresample ! audioconvert ! autoaudiosink * ]| * */ /* TODO: * - should accept framed and unframed input (needs decodebin fixes first) * - seeking in raw .dts files doesn't seem to work, but duration estimate ok * * - if frames have 'odd' durations, the frame durations (plus timestamps) * aren't adjusted up occasionally to make up for rounding error gaps. * (e.g. if 512 samples per frame @ 48kHz = 10.666666667 ms/frame) */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include #include "gstaudioparserselements.h" #include "gstdcaparse.h" #include #include GST_DEBUG_CATEGORY_STATIC (dca_parse_debug); #define GST_CAT_DEFAULT dca_parse_debug static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-dts," " framed = (boolean) true," " channels = (int) [ 1, 8 ]," " rate = (int) [ 8000, 192000 ]," " depth = (int) { 14, 16 }," " endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " " block-size = (int) [ 1, MAX], " " frame-size = (int) [ 1, MAX]")); static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-dts; " "audio/x-private1-dts")); static void gst_dca_parse_finalize (GObject * object); static gboolean gst_dca_parse_start (GstBaseParse * parse); static gboolean gst_dca_parse_stop (GstBaseParse * parse); static GstFlowReturn gst_dca_parse_handle_frame (GstBaseParse * parse, GstBaseParseFrame * frame, gint * skipsize); static GstFlowReturn gst_dca_parse_pre_push_frame (GstBaseParse * parse, GstBaseParseFrame * frame); static GstCaps *gst_dca_parse_get_sink_caps (GstBaseParse * parse, GstCaps * filter); static gboolean gst_dca_parse_set_sink_caps (GstBaseParse * parse, GstCaps * caps); #define gst_dca_parse_parent_class parent_class G_DEFINE_TYPE (GstDcaParse, gst_dca_parse, GST_TYPE_BASE_PARSE); GST_ELEMENT_REGISTER_DEFINE (dcaparse, "dcaparse", GST_RANK_PRIMARY + 1, GST_TYPE_DCA_PARSE); static void gst_dca_parse_class_init (GstDcaParseClass * klass) { GstBaseParseClass *parse_class = GST_BASE_PARSE_CLASS (klass); GstElementClass *element_class = GST_ELEMENT_CLASS (klass); GObjectClass *object_class = G_OBJECT_CLASS (klass); GST_DEBUG_CATEGORY_INIT (dca_parse_debug, "dcaparse", 0, "DCA audio stream parser"); object_class->finalize = gst_dca_parse_finalize; parse_class->start = GST_DEBUG_FUNCPTR (gst_dca_parse_start); parse_class->stop = GST_DEBUG_FUNCPTR (gst_dca_parse_stop); parse_class->handle_frame = GST_DEBUG_FUNCPTR (gst_dca_parse_handle_frame); parse_class->pre_push_frame = GST_DEBUG_FUNCPTR (gst_dca_parse_pre_push_frame); parse_class->get_sink_caps = GST_DEBUG_FUNCPTR (gst_dca_parse_get_sink_caps); parse_class->set_sink_caps = GST_DEBUG_FUNCPTR (gst_dca_parse_set_sink_caps); 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, "DTS Coherent Acoustics audio stream parser", "Codec/Parser/Audio", "DCA parser", "Tim-Philipp Müller "); } static void gst_dca_parse_reset (GstDcaParse * dcaparse) { dcaparse->channels = -1; dcaparse->rate = -1; dcaparse->depth = -1; dcaparse->endianness = -1; dcaparse->block_size = -1; dcaparse->frame_size = -1; dcaparse->last_sync = 0; dcaparse->sent_codec_tag = FALSE; } static void gst_dca_parse_init (GstDcaParse * dcaparse) { gst_base_parse_set_min_frame_size (GST_BASE_PARSE (dcaparse), DCA_MIN_FRAMESIZE); gst_dca_parse_reset (dcaparse); dcaparse->baseparse_chainfunc = GST_BASE_PARSE_SINK_PAD (GST_BASE_PARSE (dcaparse))->chainfunc; GST_PAD_SET_ACCEPT_INTERSECT (GST_BASE_PARSE_SINK_PAD (dcaparse)); GST_PAD_SET_ACCEPT_TEMPLATE (GST_BASE_PARSE_SINK_PAD (dcaparse)); } static void gst_dca_parse_finalize (GObject * object) { G_OBJECT_CLASS (parent_class)->finalize (object); } static gboolean gst_dca_parse_start (GstBaseParse * parse) { GstDcaParse *dcaparse = GST_DCA_PARSE (parse); GST_DEBUG_OBJECT (parse, "starting"); gst_dca_parse_reset (dcaparse); return TRUE; } static gboolean gst_dca_parse_stop (GstBaseParse * parse) { GST_DEBUG_OBJECT (parse, "stopping"); return TRUE; } static gboolean gst_dca_parse_parse_header (GstDcaParse * dcaparse, const GstByteReader * reader, guint * frame_size, guint * sample_rate, guint * channels, guint * depth, gint * endianness, guint * num_blocks, guint * samples_per_block, gboolean * terminator) { static const int sample_rates[16] = { 0, 8000, 16000, 32000, 0, 0, 11025, 22050, 44100, 0, 0, 12000, 24000, 48000, 96000, 192000 }; static const guint8 channels_table[16] = { 1, 2, 2, 2, 2, 3, 3, 4, 4, 5, 6, 6, 6, 7, 8, 8 }; GstByteReader r = *reader; guint16 hdr[8]; guint32 marker; guint chans, lfe, i; if (gst_byte_reader_get_remaining (&r) < (4 + sizeof (hdr))) return FALSE; marker = gst_byte_reader_peek_uint32_be_unchecked (&r); /* raw big endian or 14-bit big endian */ if (marker == 0x7FFE8001 || marker == 0x1FFFE800) { for (i = 0; i < G_N_ELEMENTS (hdr); ++i) hdr[i] = gst_byte_reader_get_uint16_be_unchecked (&r); } else /* raw little endian or 14-bit little endian */ if (marker == 0xFE7F0180 || marker == 0xFF1F00E8) { for (i = 0; i < G_N_ELEMENTS (hdr); ++i) hdr[i] = gst_byte_reader_get_uint16_le_unchecked (&r); } else { return FALSE; } GST_LOG_OBJECT (dcaparse, "dts sync marker 0x%08x at offset %u", marker, gst_byte_reader_get_pos (reader)); /* 14-bit mode */ if (marker == 0x1FFFE800 || marker == 0xFF1F00E8) { if ((hdr[2] & 0xFFF0) != 0x07F0) return FALSE; /* discard top 2 bits (2 void), shift in 2 */ hdr[0] = (hdr[0] << 2) | ((hdr[1] >> 12) & 0x0003); /* discard top 4 bits (2 void, 2 shifted into hdr[0]), shift in 4 etc. */ hdr[1] = (hdr[1] << 4) | ((hdr[2] >> 10) & 0x000F); hdr[2] = (hdr[2] << 6) | ((hdr[3] >> 8) & 0x003F); hdr[3] = (hdr[3] << 8) | ((hdr[4] >> 6) & 0x00FF); hdr[4] = (hdr[4] << 10) | ((hdr[5] >> 4) & 0x03FF); hdr[5] = (hdr[5] << 12) | ((hdr[6] >> 2) & 0x0FFF); hdr[6] = (hdr[6] << 14) | ((hdr[7] >> 0) & 0x3FFF); g_assert (hdr[0] == 0x7FFE && hdr[1] == 0x8001); } GST_LOG_OBJECT (dcaparse, "frame header: %04x%04x%04x%04x", hdr[2], hdr[3], hdr[4], hdr[5]); *terminator = (hdr[2] & 0x80) ? FALSE : TRUE; *samples_per_block = ((hdr[2] >> 10) & 0x1f) + 1; *num_blocks = ((hdr[2] >> 2) & 0x7F) + 1; *frame_size = (((hdr[2] & 0x03) << 12) | (hdr[3] >> 4)) + 1; chans = ((hdr[3] & 0x0F) << 2) | (hdr[4] >> 14); *sample_rate = sample_rates[(hdr[4] >> 10) & 0x0F]; lfe = (hdr[5] >> 9) & 0x03; GST_TRACE_OBJECT (dcaparse, "frame size %u, num_blocks %u, rate %u, " "samples per block %u", *frame_size, *num_blocks, *sample_rate, *samples_per_block); if (*num_blocks < 6 || *frame_size < 96 || *sample_rate == 0) return FALSE; if (marker == 0x1FFFE800 || marker == 0xFF1F00E8) *frame_size = (*frame_size * 16) / 14; /* FIXME: round up? */ if (chans < G_N_ELEMENTS (channels_table)) *channels = channels_table[chans] + ((lfe) ? 1 : 0); else return FALSE; if (depth) *depth = (marker == 0x1FFFE800 || marker == 0xFF1F00E8) ? 14 : 16; if (endianness) *endianness = (marker == 0xFE7F0180 || marker == 0xFF1F00E8) ? G_LITTLE_ENDIAN : G_BIG_ENDIAN; GST_TRACE_OBJECT (dcaparse, "frame size %u, channels %u, rate %u, " "num_blocks %u, samples_per_block %u", *frame_size, *channels, *sample_rate, *num_blocks, *samples_per_block); return TRUE; } static gint gst_dca_parse_find_sync (GstDcaParse * dcaparse, GstByteReader * reader, gsize bufsize, guint32 * sync) { guint32 best_sync = 0; guint best_offset = G_MAXUINT; gint off; /* FIXME: verify syncs via _parse_header() here already */ /* Raw little endian */ off = gst_byte_reader_masked_scan_uint32 (reader, 0xffffffff, 0xfe7f0180, 0, bufsize); if (off >= 0 && off < best_offset) { best_offset = off; best_sync = 0xfe7f0180; } /* Raw big endian */ off = gst_byte_reader_masked_scan_uint32 (reader, 0xffffffff, 0x7ffe8001, 0, bufsize); if (off >= 0 && off < best_offset) { best_offset = off; best_sync = 0x7ffe8001; } /* FIXME: check next 2 bytes as well for 14-bit formats (but then don't * forget to adjust the *skipsize= in _check_valid_frame() */ /* 14-bit little endian */ off = gst_byte_reader_masked_scan_uint32 (reader, 0xffffffff, 0xff1f00e8, 0, bufsize); if (off >= 0 && off < best_offset) { best_offset = off; best_sync = 0xff1f00e8; } /* 14-bit big endian */ off = gst_byte_reader_masked_scan_uint32 (reader, 0xffffffff, 0x1fffe800, 0, bufsize); if (off >= 0 && off < best_offset) { best_offset = off; best_sync = 0x1fffe800; } if (best_offset == G_MAXUINT) return -1; *sync = best_sync; return best_offset; } static GstFlowReturn gst_dca_parse_handle_frame (GstBaseParse * parse, GstBaseParseFrame * frame, gint * skipsize) { GstDcaParse *dcaparse = GST_DCA_PARSE (parse); GstBuffer *buf = frame->buffer; GstByteReader r; gboolean parser_in_sync; gboolean terminator; guint32 sync = 0; guint size = 0, rate, chans, num_blocks, samples_per_block, depth; gint block_size; gint endianness; gint off = -1; GstMapInfo map; GstFlowReturn ret = GST_FLOW_EOS; gsize extra_size = 0; gst_buffer_map (buf, &map, GST_MAP_READ); if (G_UNLIKELY (map.size < 16)) { *skipsize = 1; goto cleanup; } parser_in_sync = !GST_BASE_PARSE_LOST_SYNC (parse); gst_byte_reader_init (&r, map.data, map.size); if (G_LIKELY (parser_in_sync && dcaparse->last_sync != 0)) { off = gst_byte_reader_masked_scan_uint32 (&r, 0xffffffff, dcaparse->last_sync, 0, map.size); } if (G_UNLIKELY (off < 0)) { off = gst_dca_parse_find_sync (dcaparse, &r, map.size, &sync); } /* didn't find anything that looks like a sync word, skip */ if (off < 0) { *skipsize = map.size - 3; GST_DEBUG_OBJECT (dcaparse, "no sync, skipping %d bytes", *skipsize); goto cleanup; } GST_LOG_OBJECT (parse, "possible sync %08x at buffer offset %d", sync, off); /* 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_dca_parse_parse_header (dcaparse, &r, &size, &rate, &chans, &depth, &endianness, &num_blocks, &samples_per_block, &terminator)) { *skipsize = 4; goto cleanup; } GST_LOG_OBJECT (parse, "got frame, sync %08x, size %u, rate %d, channels %d", sync, size, rate, chans); dcaparse->last_sync = sync; /* FIXME: Don't look for a second syncword, there are streams out there * that consistently contain garbage between every frame so we never ever * find a second consecutive syncword. * See https://bugzilla.gnome.org/show_bug.cgi?id=738237 */ #if 0 parser_draining = GST_BASE_PARSE_DRAINING (parse); if (!parser_in_sync && !parser_draining) { /* check for second frame to be sure */ GST_DEBUG_OBJECT (dcaparse, "resyncing; checking next frame syncword"); if (map.size >= (size + 16)) { guint s2, r2, c2, n2, s3; gboolean t; GST_MEMDUMP ("buf", map.data, size + 16); gst_byte_reader_init (&r, map.data, map.size); gst_byte_reader_skip_unchecked (&r, size); if (!gst_dca_parse_parse_header (dcaparse, &r, &s2, &r2, &c2, NULL, NULL, &n2, &s3, &t)) { GST_DEBUG_OBJECT (dcaparse, "didn't find second syncword"); *skipsize = 4; goto cleanup; } /* ok, got sync now, let's assume constant frame size */ gst_base_parse_set_min_frame_size (parse, size); } else { /* wait for some more data */ GST_LOG_OBJECT (dcaparse, "next sync out of reach (%" G_GSIZE_FORMAT " < %u)", map.size, size + 16); goto cleanup; } } #endif /* found frame */ ret = GST_FLOW_OK; /* metadata handling */ block_size = num_blocks * samples_per_block; if (G_UNLIKELY (dcaparse->rate != rate || dcaparse->channels != chans || dcaparse->depth != depth || dcaparse->endianness != endianness || (!terminator && dcaparse->block_size != block_size) || (size != dcaparse->frame_size))) { GstCaps *caps; caps = gst_caps_new_simple ("audio/x-dts", "framed", G_TYPE_BOOLEAN, TRUE, "rate", G_TYPE_INT, rate, "channels", G_TYPE_INT, chans, "endianness", G_TYPE_INT, endianness, "depth", G_TYPE_INT, depth, "block-size", G_TYPE_INT, block_size, "frame-size", G_TYPE_INT, size, NULL); gst_pad_set_caps (GST_BASE_PARSE_SRC_PAD (parse), caps); gst_caps_unref (caps); dcaparse->rate = rate; dcaparse->channels = chans; dcaparse->depth = depth; dcaparse->endianness = endianness; dcaparse->block_size = block_size; dcaparse->frame_size = size; gst_base_parse_set_frame_rate (parse, rate, block_size, 0, 0); } cleanup: /* it is possible that DTS HD substream after DTS core */ if (parse->flags & GST_BASE_PARSE_FLAG_DRAINING || map.size >= size + 9) { extra_size = 0; if (map.size >= size + 9) { const guint8 *next = map.data + size; /* Check for DTS_SYNCWORD_SUBSTREAM */ if (next[0] == 0x64 && next[1] == 0x58 && next[2] == 0x20 && next[3] == 0x25) { /* 7.4.1 Extension Substream Header */ GstBitReader reader; gst_bit_reader_init (&reader, next + 4, 5); gst_bit_reader_skip (&reader, 8 + 2); /* skip UserDefinedBits and nExtSSIndex) */ if (gst_bit_reader_get_bits_uint8_unchecked (&reader, 1) == 0) { gst_bit_reader_skip (&reader, 8); extra_size = gst_bit_reader_get_bits_uint32_unchecked (&reader, 16) + 1; } else { gst_bit_reader_skip (&reader, 12); extra_size = gst_bit_reader_get_bits_uint32_unchecked (&reader, 20) + 1; } } } gst_buffer_unmap (buf, &map); if (ret == GST_FLOW_OK && size + extra_size <= map.size) { ret = gst_base_parse_finish_frame (parse, frame, size + extra_size); } else { ret = GST_FLOW_OK; } } else { gst_buffer_unmap (buf, &map); } return ret; } /* * 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. * 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. * See gstac3parse.c for a more detailed description. * */ static GstFlowReturn gst_dca_parse_chain_priv (GstPad * pad, GstObject * parent, GstBuffer * buffer) { GstDcaParse *dcaparse = GST_DCA_PARSE (parent); GstFlowReturn ret; GstBuffer *newbuf; gsize size; size = gst_buffer_get_size (buffer); if (size >= 2) { newbuf = gst_buffer_copy_region (buffer, GST_BUFFER_COPY_ALL, 2, size - 2); gst_buffer_copy_into (newbuf, buffer, GST_BUFFER_COPY_METADATA, 0, -1); gst_buffer_unref (buffer); ret = dcaparse->baseparse_chainfunc (pad, parent, newbuf); } else { gst_buffer_unref (buffer); ret = GST_FLOW_OK; } return ret; } 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"); } } static GstCaps * gst_dca_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); 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 field */ peercaps = gst_caps_make_writable (peercaps); remove_fields (peercaps); 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_dca_parse_set_sink_caps (GstBaseParse * parse, GstCaps * caps) { GstStructure *s; GstDcaParse *dcaparse = GST_DCA_PARSE (parse); s = gst_caps_get_structure (caps, 0); if (gst_structure_has_name (s, "audio/x-private1-dts")) { gst_pad_set_chain_function (parse->sinkpad, gst_dca_parse_chain_priv); } else { gst_pad_set_chain_function (parse->sinkpad, dcaparse->baseparse_chainfunc); } return TRUE; } static GstFlowReturn gst_dca_parse_pre_push_frame (GstBaseParse * parse, GstBaseParseFrame * frame) { GstDcaParse *dcaparse = GST_DCA_PARSE (parse); if (!dcaparse->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 */ dcaparse->sent_codec_tag = TRUE; } frame->flags |= GST_BASE_PARSE_FRAME_FLAG_CLIP; return GST_FLOW_OK; }