/* GStreamer Wavpack parser * Copyright (C) 2012 Mark Nauwelaerts * Copyright (C) 2012 Nokia Corporation. All rights reserved. * Contact: Stefan Kost * * 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-wavpackparse * @short_description: Wavpack parser * @see_also: #GstAmrParse, #GstAACParse * * This is an Wavpack parser. * * * Example launch line * |[ * gst-launch-1.0 filesrc location=abc.wavpack ! wavpackparse ! wavpackdec ! audioresample ! audioconvert ! autoaudiosink * ]| * */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include #include "gstwavpackparse.h" #include #include #include GST_DEBUG_CATEGORY_STATIC (wavpack_parse_debug); #define GST_CAT_DEFAULT wavpack_parse_debug static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-wavpack, " "depth = (int) [ 1, 32 ], " "channels = (int) [ 1, 8 ], " "rate = (int) [ 6000, 192000 ], " "framed = (boolean) TRUE; " "audio/x-wavpack-correction, " "framed = (boolean) TRUE") ); static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-wavpack")); static void gst_wavpack_parse_finalize (GObject * object); static gboolean gst_wavpack_parse_start (GstBaseParse * parse); static gboolean gst_wavpack_parse_stop (GstBaseParse * parse); static GstFlowReturn gst_wavpack_parse_handle_frame (GstBaseParse * parse, GstBaseParseFrame * frame, gint * skipsize); static GstCaps *gst_wavpack_parse_get_sink_caps (GstBaseParse * parse, GstCaps * filter); static GstFlowReturn gst_wavpack_parse_pre_push_frame (GstBaseParse * parse, GstBaseParseFrame * frame); #define gst_wavpack_parse_parent_class parent_class G_DEFINE_TYPE (GstWavpackParse, gst_wavpack_parse, GST_TYPE_BASE_PARSE); static void gst_wavpack_parse_class_init (GstWavpackParseClass * 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 (wavpack_parse_debug, "wavpackparse", 0, "Wavpack audio stream parser"); object_class->finalize = gst_wavpack_parse_finalize; parse_class->start = GST_DEBUG_FUNCPTR (gst_wavpack_parse_start); parse_class->stop = GST_DEBUG_FUNCPTR (gst_wavpack_parse_stop); parse_class->handle_frame = GST_DEBUG_FUNCPTR (gst_wavpack_parse_handle_frame); parse_class->get_sink_caps = GST_DEBUG_FUNCPTR (gst_wavpack_parse_get_sink_caps); parse_class->pre_push_frame = GST_DEBUG_FUNCPTR (gst_wavpack_parse_pre_push_frame); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&sink_template)); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&src_template)); gst_element_class_set_static_metadata (element_class, "Wavpack audio stream parser", "Codec/Parser/Audio", "Wavpack parser", "Mark Nauwelaerts "); } static void gst_wavpack_parse_reset (GstWavpackParse * wvparse) { wvparse->channels = -1; wvparse->channel_mask = 0; wvparse->sample_rate = -1; wvparse->width = -1; wvparse->total_samples = 0; wvparse->sent_codec_tag = FALSE; } static void gst_wavpack_parse_init (GstWavpackParse * wvparse) { gst_wavpack_parse_reset (wvparse); GST_PAD_SET_ACCEPT_INTERSECT (GST_BASE_PARSE_SINK_PAD (wvparse)); GST_PAD_SET_ACCEPT_TEMPLATE (GST_BASE_PARSE_SINK_PAD (wvparse)); } static void gst_wavpack_parse_finalize (GObject * object) { G_OBJECT_CLASS (parent_class)->finalize (object); } static gboolean gst_wavpack_parse_start (GstBaseParse * parse) { GstWavpackParse *wvparse = GST_WAVPACK_PARSE (parse); GST_DEBUG_OBJECT (parse, "starting"); gst_wavpack_parse_reset (wvparse); /* need header at least */ gst_base_parse_set_min_frame_size (GST_BASE_PARSE (wvparse), sizeof (WavpackHeader)); /* inform baseclass we can come up with ts, based on counters in packets */ gst_base_parse_set_has_timing_info (GST_BASE_PARSE_CAST (wvparse), TRUE); gst_base_parse_set_syncable (GST_BASE_PARSE_CAST (wvparse), TRUE); return TRUE; } static gboolean gst_wavpack_parse_stop (GstBaseParse * parse) { GST_DEBUG_OBJECT (parse, "stopping"); return TRUE; } static gint gst_wavpack_get_default_channel_mask (gint nchannels) { gint channel_mask = 0; /* Set the default channel mask for the given number of channels. * It's the same as for WAVE_FORMAT_EXTENDED: * http://www.microsoft.com/whdc/device/audio/multichaud.mspx */ switch (nchannels) { case 11: channel_mask |= 0x00400; channel_mask |= 0x00200; case 9: channel_mask |= 0x00100; case 8: channel_mask |= 0x00080; channel_mask |= 0x00040; case 6: channel_mask |= 0x00020; channel_mask |= 0x00010; case 4: channel_mask |= 0x00008; case 3: channel_mask |= 0x00004; case 2: channel_mask |= 0x00002; channel_mask |= 0x00001; break; case 1: /* For mono use front center */ channel_mask |= 0x00004; break; } return channel_mask; } static const struct { const guint32 ms_mask; const GstAudioChannelPosition gst_pos; } layout_mapping[] = { { 0x00001, GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT}, { 0x00002, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT}, { 0x00004, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER}, { 0x00008, GST_AUDIO_CHANNEL_POSITION_LFE1}, { 0x00010, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT}, { 0x00020, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT}, { 0x00040, GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER}, { 0x00080, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER}, { 0x00100, GST_AUDIO_CHANNEL_POSITION_REAR_CENTER}, { 0x00200, GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT}, { 0x00400, GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT}, { 0x00800, GST_AUDIO_CHANNEL_POSITION_TOP_CENTER}, { 0x01000, GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_LEFT}, { 0x02000, GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_CENTER}, { 0x04000, GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_RIGHT}, { 0x08000, GST_AUDIO_CHANNEL_POSITION_TOP_REAR_LEFT}, { 0x10000, GST_AUDIO_CHANNEL_POSITION_TOP_REAR_CENTER}, { 0x20000, GST_AUDIO_CHANNEL_POSITION_TOP_REAR_RIGHT} }; #define MAX_CHANNEL_POSITIONS G_N_ELEMENTS (layout_mapping) static gboolean gst_wavpack_get_channel_positions (gint num_channels, gint layout, GstAudioChannelPosition * pos) { gint i, p; if (num_channels == 1 && layout == 0x00004) { pos[0] = GST_AUDIO_CHANNEL_POSITION_MONO; return TRUE; } p = 0; for (i = 0; i < MAX_CHANNEL_POSITIONS; ++i) { if ((layout & layout_mapping[i].ms_mask) != 0) { if (p >= num_channels) { GST_WARNING ("More bits set in the channel layout map than there " "are channels! Broken file"); return FALSE; } if (layout_mapping[i].gst_pos == GST_AUDIO_CHANNEL_POSITION_INVALID) { GST_WARNING ("Unsupported channel position (mask 0x%08x) in channel " "layout map - ignoring those channels", layout_mapping[i].ms_mask); /* what to do? just ignore it and let downstream deal with a channel * layout that has INVALID positions in it for now ... */ } pos[p] = layout_mapping[i].gst_pos; ++p; } } if (p != num_channels) { GST_WARNING ("Only %d bits set in the channel layout map, but there are " "supposed to be %d channels! Broken file", p, num_channels); return FALSE; } return TRUE; } static const guint32 sample_rates[] = { 6000, 8000, 9600, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000, 64000, 88200, 96000, 192000 }; #define CHECK(call) { \ if (!call) \ goto read_failed; \ } /* caller ensures properly sync'ed with enough data */ static gboolean gst_wavpack_parse_frame_metadata (GstWavpackParse * parse, GstBuffer * buf, gint skip, WavpackHeader * wph, WavpackInfo * wpi) { GstByteReader br; gint i; GstMapInfo map; g_return_val_if_fail (wph != NULL || wpi != NULL, FALSE); g_return_val_if_fail (gst_buffer_get_size (buf) >= skip + sizeof (WavpackHeader), FALSE); gst_buffer_map (buf, &map, GST_MAP_READ); gst_byte_reader_init (&br, map.data + skip, wph->ckSize + 8); /* skip past header */ gst_byte_reader_skip_unchecked (&br, sizeof (WavpackHeader)); /* get some basics from header */ i = (wph->flags >> 23) & 0xF; if (!wpi->rate) wpi->rate = (i < G_N_ELEMENTS (sample_rates)) ? sample_rates[i] : 44100; wpi->width = ((wph->flags & 0x3) + 1) * 8; if (!wpi->channels) wpi->channels = (wph->flags & 0x4) ? 1 : 2; if (!wpi->channel_mask) wpi->channel_mask = 5 - wpi->channels; /* need to dig metadata blocks for some more */ while (gst_byte_reader_get_remaining (&br)) { gint size = 0; guint16 size2 = 0; guint8 c, id; const guint8 *data; GstByteReader mbr; CHECK (gst_byte_reader_get_uint8 (&br, &id)); CHECK (gst_byte_reader_get_uint8 (&br, &c)); if (id & ID_LARGE) CHECK (gst_byte_reader_get_uint16_le (&br, &size2)); size = size2; size <<= 8; size += c; size <<= 1; if (id & ID_ODD_SIZE) size--; CHECK (gst_byte_reader_get_data (&br, size + (size & 1), &data)); gst_byte_reader_init (&mbr, data, size); switch (id) { case ID_WVC_BITSTREAM: GST_LOG_OBJECT (parse, "correction bitstream"); wpi->correction = TRUE; break; case ID_WV_BITSTREAM: case ID_WVX_BITSTREAM: break; case ID_SAMPLE_RATE: if (size == 3) { CHECK (gst_byte_reader_get_uint24_le (&mbr, &wpi->rate)); GST_LOG_OBJECT (parse, "updated with custom rate %d", wpi->rate); } else { GST_DEBUG_OBJECT (parse, "unexpected size for SAMPLE_RATE metadata"); } break; case ID_CHANNEL_INFO: { guint16 channels; guint32 mask = 0; if (size == 6) { CHECK (gst_byte_reader_get_uint16_le (&mbr, &channels)); channels = channels & 0xFFF; CHECK (gst_byte_reader_get_uint24_le (&mbr, &mask)); } else if (size) { CHECK (gst_byte_reader_get_uint8 (&mbr, &c)); channels = c; while (gst_byte_reader_get_uint8 (&mbr, &c)) mask |= (((guint32) c) << 8); } else { GST_DEBUG_OBJECT (parse, "unexpected size for CHANNEL_INFO metadata"); break; } wpi->channels = channels; wpi->channel_mask = mask; break; } default: GST_LOG_OBJECT (parse, "unparsed ID 0x%x", id); break; } } gst_buffer_unmap (buf, &map); return TRUE; /* ERRORS */ read_failed: { gst_buffer_unmap (buf, &map); GST_DEBUG_OBJECT (parse, "short read while parsing metadata"); /* let's look the other way anyway */ return TRUE; } } /* caller ensures properly sync'ed with enough data */ static gboolean gst_wavpack_parse_frame_header (GstWavpackParse * parse, GstBuffer * buf, gint skip, WavpackHeader * _wph) { GstByteReader br; WavpackHeader wph = { {0,}, 0, }; GstMapInfo map; gboolean hdl = TRUE; g_return_val_if_fail (gst_buffer_get_size (buf) >= skip + sizeof (WavpackHeader), FALSE); gst_buffer_map (buf, &map, GST_MAP_READ); gst_byte_reader_init (&br, map.data, map.size); /* marker */ gst_byte_reader_skip_unchecked (&br, skip + 4); /* read */ hdl &= gst_byte_reader_get_uint32_le (&br, &wph.ckSize); hdl &= gst_byte_reader_get_uint16_le (&br, &wph.version); hdl &= gst_byte_reader_get_uint8 (&br, &wph.track_no); hdl &= gst_byte_reader_get_uint8 (&br, &wph.index_no); hdl &= gst_byte_reader_get_uint32_le (&br, &wph.total_samples); hdl &= gst_byte_reader_get_uint32_le (&br, &wph.block_index); hdl &= gst_byte_reader_get_uint32_le (&br, &wph.block_samples); hdl &= gst_byte_reader_get_uint32_le (&br, &wph.flags); hdl &= gst_byte_reader_get_uint32_le (&br, &wph.crc); if (!hdl) GST_WARNING_OBJECT (parse, "Error reading header"); /* dump */ GST_LOG_OBJECT (parse, "size %d", wph.ckSize); GST_LOG_OBJECT (parse, "version 0x%x", wph.version); GST_LOG_OBJECT (parse, "total samples %d", wph.total_samples); GST_LOG_OBJECT (parse, "block index %d", wph.block_index); GST_LOG_OBJECT (parse, "block samples %d", wph.block_samples); GST_LOG_OBJECT (parse, "flags 0x%x", wph.flags); GST_LOG_OBJECT (parse, "crc 0x%x", wph.flags); if (!parse->total_samples && wph.block_index == 0 && wph.total_samples != -1) { GST_DEBUG_OBJECT (parse, "determined duration of %u samples", wph.total_samples); parse->total_samples = wph.total_samples; } if (_wph) *_wph = wph; gst_buffer_unmap (buf, &map); return TRUE; } static GstFlowReturn gst_wavpack_parse_handle_frame (GstBaseParse * parse, GstBaseParseFrame * frame, gint * skipsize) { GstWavpackParse *wvparse = GST_WAVPACK_PARSE (parse); GstBuffer *buf = frame->buffer; GstByteReader reader; gint off; guint rate, chans, width, mask; gboolean lost_sync, draining, final; guint frmsize = 0; WavpackHeader wph; WavpackInfo wpi = { 0, }; GstMapInfo map; if (G_UNLIKELY (gst_buffer_get_size (buf) < sizeof (WavpackHeader))) return FALSE; gst_buffer_map (buf, &map, GST_MAP_READ); gst_byte_reader_init (&reader, map.data, map.size); /* scan for 'wvpk' marker */ off = gst_byte_reader_masked_scan_uint32 (&reader, 0xffffffff, 0x7776706b, 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 skip; } /* possible frame header, but not at offset 0? skip bytes before sync */ if (off > 0) { *skipsize = off; goto skip; } /* make sure the values in the frame header look sane */ gst_wavpack_parse_frame_header (wvparse, buf, 0, &wph); frmsize = wph.ckSize + 8; /* need the entire frame for parsing */ if (gst_byte_reader_get_remaining (&reader) < frmsize) goto more; /* got a frame, now we can dig for some more metadata */ GST_LOG_OBJECT (parse, "got frame"); gst_wavpack_parse_frame_metadata (wvparse, buf, 0, &wph, &wpi); lost_sync = GST_BASE_PARSE_LOST_SYNC (parse); draining = GST_BASE_PARSE_DRAINING (parse); while (!(final = (wph.flags & FLAG_FINAL_BLOCK)) || (lost_sync && !draining)) { guint32 word = 0; GST_LOG_OBJECT (wvparse, "checking next frame syncword; " "lost_sync: %d, draining: %d, final: %d", lost_sync, draining, final); if (!gst_byte_reader_skip (&reader, wph.ckSize + 8) || !gst_byte_reader_peek_uint32_be (&reader, &word)) { GST_DEBUG_OBJECT (wvparse, "... but not sufficient data"); frmsize += 4; goto more; } else { if (word != 0x7776706b) { GST_DEBUG_OBJECT (wvparse, "0x%x not OK", word); *skipsize = off + 2; goto skip; } /* need to parse each frame/block for metadata if several ones */ if (!final) { gint av; GST_LOG_OBJECT (wvparse, "checking frame at offset %d (0x%x)", frmsize, frmsize); av = gst_byte_reader_get_remaining (&reader); if (av < sizeof (WavpackHeader)) { frmsize += sizeof (WavpackHeader); goto more; } gst_wavpack_parse_frame_header (wvparse, buf, frmsize, &wph); off = frmsize; frmsize += wph.ckSize + 8; if (av < wph.ckSize + 8) goto more; gst_wavpack_parse_frame_metadata (wvparse, buf, off, &wph, &wpi); /* could also check for matching block_index and block_samples ?? */ } } /* resynced if we make it here */ lost_sync = FALSE; } rate = wpi.rate; width = wpi.width; chans = wpi.channels; mask = wpi.channel_mask; GST_LOG_OBJECT (parse, "rate: %u, width: %u, chans: %u", rate, width, chans); GST_BUFFER_PTS (buf) = gst_util_uint64_scale_int (wph.block_index, GST_SECOND, rate); GST_BUFFER_DTS (buf) = GST_BUFFER_PTS (buf); GST_BUFFER_DURATION (buf) = gst_util_uint64_scale_int (wph.block_index + wph.block_samples, GST_SECOND, rate) - GST_BUFFER_PTS (buf); if (G_UNLIKELY (wvparse->sample_rate != rate || wvparse->channels != chans || wvparse->width != width || wvparse->channel_mask != mask)) { GstCaps *caps; if (wpi.correction) { caps = gst_caps_new_simple ("audio/x-wavpack-correction", "framed", G_TYPE_BOOLEAN, TRUE, NULL); } else { caps = gst_caps_new_simple ("audio/x-wavpack", "channels", G_TYPE_INT, chans, "rate", G_TYPE_INT, rate, "depth", G_TYPE_INT, width, "framed", G_TYPE_BOOLEAN, TRUE, NULL); if (!mask) mask = gst_wavpack_get_default_channel_mask (wvparse->channels); if (mask != 0) { GstAudioChannelPosition pos[64] = { GST_AUDIO_CHANNEL_POSITION_INVALID, }; guint64 gmask; if (!gst_wavpack_get_channel_positions (chans, mask, pos)) { GST_WARNING_OBJECT (wvparse, "Failed to determine channel layout"); } else { gst_audio_channel_positions_to_mask (pos, chans, FALSE, &gmask); if (gmask) gst_caps_set_simple (caps, "channel-mask", GST_TYPE_BITMASK, gmask, NULL); } } } gst_pad_set_caps (GST_BASE_PARSE_SRC_PAD (parse), caps); gst_caps_unref (caps); wvparse->sample_rate = rate; wvparse->channels = chans; wvparse->width = width; wvparse->channel_mask = mask; if (wvparse->total_samples) { GST_DEBUG_OBJECT (wvparse, "setting duration"); gst_base_parse_set_duration (GST_BASE_PARSE (wvparse), GST_FORMAT_TIME, gst_util_uint64_scale_int (wvparse->total_samples, GST_SECOND, wvparse->sample_rate), 0); } } /* return to normal size */ gst_base_parse_set_min_frame_size (parse, sizeof (WavpackHeader)); gst_buffer_unmap (buf, &map); return gst_base_parse_finish_frame (parse, frame, frmsize); skip: gst_buffer_unmap (buf, &map); GST_LOG_OBJECT (wvparse, "skipping %d", *skipsize); return GST_FLOW_OK; more: gst_buffer_unmap (buf, &map); GST_LOG_OBJECT (wvparse, "need at least %u", frmsize); gst_base_parse_set_min_frame_size (parse, frmsize); *skipsize = 0; return GST_FLOW_OK; } 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_wavpack_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 GstFlowReturn gst_wavpack_parse_pre_push_frame (GstBaseParse * parse, GstBaseParseFrame * frame) { GstWavpackParse *wavpackparse = GST_WAVPACK_PARSE (parse); if (!wavpackparse->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 */ wavpackparse->sent_codec_tag = TRUE; } return GST_FLOW_OK; }