/* GStreamer * Copyright (C) <1999> Erik Walthinsen * * 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., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. */ /*#define GST_DEBUG_ENABLED */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include /* elementfactory information */ static GstElementDetails mp3parse_details = { "MPEG1 Audio Parser", "Codec/Parser/Audio", "Parses and frames mpeg1 audio streams (levels 1-3), provides seek", "Erik Walthinsen " }; static GstStaticPadTemplate mp3_src_template = GST_STATIC_PAD_TEMPLATE ( "src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/mpeg, " "mpegversion = (int) 1, " "layer = (int) [ 1, 3 ], " "rate = (int) [ 8000, 48000], " "channels = (int) [ 1, 2 ]") ); static GstStaticPadTemplate mp3_sink_template = GST_STATIC_PAD_TEMPLATE ( "sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/mpeg, " "mpegversion = (int) 1" ) ); /* GstMPEGAudioParse signals and args */ enum { /* FILL ME */ LAST_SIGNAL }; enum { ARG_0, ARG_SKIP, ARG_BIT_RATE, /* FILL ME */ }; static void gst_mp3parse_class_init (GstMPEGAudioParseClass *klass); static void gst_mp3parse_base_init (GstMPEGAudioParseClass *klass); static void gst_mp3parse_init (GstMPEGAudioParse *mp3parse); static void gst_mp3parse_chain (GstPad *pad,GstData *_data); static long bpf_from_header (GstMPEGAudioParse *parse, unsigned long header); static int head_check (unsigned long head); static void gst_mp3parse_set_property (GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec); static void gst_mp3parse_get_property (GObject *object, guint prop_id, GValue *value, GParamSpec *pspec); static GstElementStateReturn gst_mp3parse_change_state (GstElement *element); static GstElementClass *parent_class = NULL; /*static guint gst_mp3parse_signals[LAST_SIGNAL] = { 0 }; */ GType gst_mp3parse_get_type(void) { static GType mp3parse_type = 0; if (!mp3parse_type) { static const GTypeInfo mp3parse_info = { sizeof(GstMPEGAudioParseClass), (GBaseInitFunc)gst_mp3parse_base_init, NULL, (GClassInitFunc)gst_mp3parse_class_init, NULL, NULL, sizeof(GstMPEGAudioParse), 0, (GInstanceInitFunc)gst_mp3parse_init, }; mp3parse_type = g_type_register_static (GST_TYPE_ELEMENT, "GstMPEGAudioParse", &mp3parse_info, 0); } return mp3parse_type; } static guint mp3types_bitrates[2][3][16] = { { {0, 32, 64, 96, 128, 160, 192, 224, 256, 288, 320, 352, 384, 416, 448, }, {0, 32, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384, }, {0, 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, } }, { {0, 32, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224, 256, }, {0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160, }, {0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160, } }, }; static guint mp3types_freqs[3][3] = { {44100, 48000, 32000}, {22050, 24000, 16000}, {11025, 12000, 8000}}; static inline guint mp3_type_frame_length_from_header (guint32 header, guint *put_layer, guint *put_channels, guint *put_bitrate, guint *put_samplerate) { guint length; gulong mode, samplerate, bitrate, layer, channels, padding; gint lsf, mpg25; if (header & (1 << 20)) { lsf = (header & (1 << 19)) ? 0 : 1; mpg25 = 0; } else { lsf = 1; mpg25 = 1; } mode = (header >> 6) & 0x3; channels = (mode == 3) ? 1 : 2; samplerate = (header >> 10) & 0x3; samplerate = mp3types_freqs[lsf + mpg25][samplerate]; layer = 4 - ((header >> 17) & 0x3); bitrate = (header >> 12) & 0xF; bitrate = mp3types_bitrates[lsf][layer - 1][bitrate] * 1000; if (bitrate == 0) return 0; padding = (header >> 9) & 0x1; switch (layer) { case 1: length = (bitrate * 12) / samplerate + 4 * padding; break; case 2: length = (bitrate * 144) / samplerate + padding; break; default: case 3: length = (bitrate * 144) / (samplerate << lsf) + padding; break; } GST_DEBUG ("Calculated mp3 frame length of %u bytes", length); GST_DEBUG ("samplerate = %lu, bitrate = %lu, layer = %lu, channels = %lu", samplerate, bitrate, layer, channels); if (put_layer) *put_layer = layer; if (put_channels) *put_channels = channels; if (put_bitrate) *put_bitrate = bitrate; if (put_samplerate) *put_samplerate = samplerate; return length; } /** * The chance that random data is identified as a valid mp3 header is 63 / 2^18 * (0.024%) per try. This makes the function for calculating false positives * 1 - (1 - ((63 / 2 ^18) ^ GST_MP3_TYPEFIND_MIN_HEADERS)) ^ buffersize) * This has the following probabilities of false positives: * bufsize MIN_HEADERS * (bytes) 1 2 3 4 * 4096 62.6% 0.02% 0% 0% * 16384 98% 0.09% 0% 0% * 1 MiB 100% 5.88% 0% 0% * 1 GiB 100% 100% 1.44% 0% * 1 TiB 100% 100% 100% 0.35% * This means that the current choice (3 headers by most of the time 4096 byte * buffers is pretty safe for now. * * The max. size of each frame is 1440 bytes, which means that for N frames * to be detected, we need 1440 * GST_MP3_TYPEFIND_MIN_HEADERS + 3 of data. * Assuming we step into the stream right after the frame header, this * means we need 1440 * (GST_MP3_TYPEFIND_MIN_HEADERS + 1) - 1 + 3 bytes * of data (5762) to always detect any mp3. */ /* increase this value when this function finds too many false positives */ #define GST_MP3_TYPEFIND_MIN_HEADERS 3 #define GST_MP3_TYPEFIND_MIN_DATA (1440 * (GST_MP3_TYPEFIND_MIN_HEADERS + 1) - 1 + 3) static GstCaps * mp3_caps_create (guint layer, guint channels, guint bitrate, guint samplerate) { GstCaps *new; g_assert (layer); g_assert (samplerate); g_assert (bitrate); g_assert (channels); new = gst_caps_new_simple ("audio/mpeg", "mpegversion", G_TYPE_INT, 1, "layer", G_TYPE_INT, layer, "rate", G_TYPE_INT, samplerate, "channels", G_TYPE_INT, channels, NULL); return new; } static void gst_mp3parse_base_init (GstMPEGAudioParseClass *klass) { GstElementClass *element_class = GST_ELEMENT_CLASS (klass); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&mp3_sink_template)); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&mp3_src_template)); gst_element_class_set_details (element_class, &mp3parse_details); } static void gst_mp3parse_class_init (GstMPEGAudioParseClass *klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; gobject_class = (GObjectClass*)klass; gstelement_class = (GstElementClass*)klass; g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_SKIP, g_param_spec_int("skip","skip","skip", G_MININT,G_MAXINT,0,G_PARAM_READWRITE)); /* CHECKME */ g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_BIT_RATE, g_param_spec_int("bitrate","Bitrate","Bit Rate", G_MININT,G_MAXINT,0,G_PARAM_READABLE)); /* CHECKME */ parent_class = g_type_class_ref(GST_TYPE_ELEMENT); gobject_class->set_property = gst_mp3parse_set_property; gobject_class->get_property = gst_mp3parse_get_property; gstelement_class->change_state = gst_mp3parse_change_state; } static void gst_mp3parse_init (GstMPEGAudioParse *mp3parse) { mp3parse->sinkpad = gst_pad_new_from_template( gst_static_pad_template_get (&mp3_sink_template), "sink"); gst_element_add_pad(GST_ELEMENT(mp3parse),mp3parse->sinkpad); gst_pad_set_chain_function(mp3parse->sinkpad,gst_mp3parse_chain); gst_element_set_loop_function (GST_ELEMENT(mp3parse),NULL); mp3parse->srcpad = gst_pad_new_from_template( gst_static_pad_template_get (&mp3_src_template), "src"); gst_element_add_pad(GST_ELEMENT(mp3parse),mp3parse->srcpad); gst_pad_use_explicit_caps (mp3parse->srcpad); /*gst_pad_set_type_id(mp3parse->srcpad, mp3frametype); */ mp3parse->partialbuf = NULL; mp3parse->skip = 0; mp3parse->in_flush = FALSE; mp3parse->rate = mp3parse->channels = mp3parse->layer = -1; } static void gst_mp3parse_chain (GstPad *pad, GstData *_data) { GstBuffer *buf = GST_BUFFER (_data); GstMPEGAudioParse *mp3parse; guchar *data; glong size,offset = 0; guint32 header; int bpf; GstBuffer *outbuf; guint64 last_ts; g_return_if_fail(pad != NULL); g_return_if_fail(GST_IS_PAD(pad)); g_return_if_fail(buf != NULL); /* g_return_if_fail(GST_IS_BUFFER(buf)); */ mp3parse = GST_MP3PARSE (gst_pad_get_parent (pad)); GST_DEBUG ("mp3parse: received buffer of %d bytes",GST_BUFFER_SIZE(buf)); last_ts = GST_BUFFER_TIMESTAMP(buf); /* FIXME, do flush */ /* if (mp3parse->partialbuf) { gst_buffer_unref(mp3parse->partialbuf); mp3parse->partialbuf = NULL; } mp3parse->in_flush = TRUE; */ /* if we have something left from the previous frame */ if (mp3parse->partialbuf) { GstBuffer *newbuf; newbuf = gst_buffer_merge(mp3parse->partialbuf, buf); /* and the one we received.. */ gst_buffer_unref(buf); gst_buffer_unref(mp3parse->partialbuf); mp3parse->partialbuf = newbuf; } else { mp3parse->partialbuf = buf; } size = GST_BUFFER_SIZE(mp3parse->partialbuf); data = GST_BUFFER_DATA(mp3parse->partialbuf); /* while we still have bytes left -4 for the header */ while (offset < size-4) { int skipped = 0; GST_DEBUG ("mp3parse: offset %ld, size %ld ",offset, size); /* search for a possible start byte */ for (;((data[offset] != 0xff) && (offset < size));offset++) skipped++; if (skipped && !mp3parse->in_flush) { GST_DEBUG ("mp3parse: **** now at %ld skipped %d bytes",offset,skipped); } /* construct the header word */ header = GUINT32_FROM_BE(*((guint32 *)(data+offset))); /* if it's a valid header, go ahead and send off the frame */ if (head_check(header)) { /* calculate the bpf of the frame */ bpf = bpf_from_header(mp3parse, header); /******************************************************************************** * robust seek support * - This performs additional frame validation if the in_flush flag is set * (indicating a discontinuous stream). * - The current frame header is not accepted as valid unless the NEXT frame * header has the same values for most fields. This significantly increases * the probability that we aren't processing random data. * - It is not clear if this is sufficient for robust seeking of Layer III * streams which utilize the concept of a "bit reservoir" by borrow bitrate * from previous frames. In this case, seeking may be more complicated because * the frames are not independently coded. ********************************************************************************/ if ( mp3parse->in_flush ) { guint32 header2; if ((size-offset)<(bpf+4)) { if (mp3parse->in_flush) break; } /* wait until we have the the entire current frame as well as the next frame header */ header2 = GUINT32_FROM_BE(*((guint32 *)(data+offset+bpf))); GST_DEBUG ("mp3parse: header=%08X, header2=%08X, bpf=%d", (unsigned int)header, (unsigned int)header2, bpf ); /* mask the bits which are allowed to differ between frames */ #define HDRMASK ~((0xF << 12) /* bitrate */ | \ (0x1 << 9) /* padding */ | \ (0x3 << 4)) /*mode extension*/ if ( (header2&HDRMASK) != (header&HDRMASK) ) { /* require 2 matching headers in a row */ GST_DEBUG ("mp3parse: next header doesn't match (header=%08X, header2=%08X, bpf=%d)", (unsigned int)header, (unsigned int)header2, bpf ); offset++; /* This frame is invalid. Start looking for a valid frame at the next position in the stream */ continue; } } /* if we don't have the whole frame... */ if ((size - offset) < bpf) { GST_DEBUG ("mp3parse: partial buffer needed %ld < %d ",(size-offset), bpf); break; } else { guint bitrate, layer, rate, channels; if (!mp3_type_frame_length_from_header (header, &layer, &channels, &bitrate, &rate)) { g_error("Header failed internal error"); } if (channels != mp3parse->channels || rate != mp3parse->rate || layer != mp3parse->layer || bitrate != mp3parse->bit_rate) { GstCaps *caps = mp3_caps_create (layer, channels, bitrate, rate); gst_pad_set_explicit_caps(mp3parse->srcpad, caps); mp3parse->channels = channels; mp3parse->layer = layer; mp3parse->rate = rate; mp3parse->bit_rate = bitrate; } outbuf = gst_buffer_create_sub(mp3parse->partialbuf,offset,bpf); offset += bpf; if (mp3parse->skip == 0) { GST_DEBUG ("mp3parse: pushing buffer of %d bytes",GST_BUFFER_SIZE(outbuf)); if (mp3parse->in_flush) { /* FIXME do some sort of flush event */ mp3parse->in_flush = FALSE; } GST_BUFFER_TIMESTAMP(outbuf) = last_ts; GST_BUFFER_DURATION(outbuf) = 8 * (GST_SECOND/1000) * GST_BUFFER_SIZE(outbuf) / mp3parse->bit_rate; if (GST_PAD_CAPS (mp3parse->srcpad) != NULL) { gst_pad_push(mp3parse->srcpad,GST_DATA (outbuf)); } else { GST_DEBUG ("No capsnego yet, delaying buffer push"); gst_buffer_unref (outbuf); } } else { GST_DEBUG ("mp3parse: skipping buffer of %d bytes",GST_BUFFER_SIZE(outbuf)); gst_buffer_unref(outbuf); mp3parse->skip--; } } } else { offset++; if (!mp3parse->in_flush) GST_DEBUG ("mp3parse: *** wrong header, skipping byte (FIXME?)"); } } /* if we have processed this block and there are still */ /* bytes left not in a partial block, copy them over. */ if (size-offset > 0) { glong remainder = (size - offset); GST_DEBUG ("mp3parse: partial buffer needed %ld for trailing bytes",remainder); outbuf = gst_buffer_create_sub(mp3parse->partialbuf,offset,remainder); gst_buffer_unref(mp3parse->partialbuf); mp3parse->partialbuf = outbuf; } else { gst_buffer_unref(mp3parse->partialbuf); mp3parse->partialbuf = NULL; } } static long bpf_from_header (GstMPEGAudioParse *parse, unsigned long header) { guint bitrate, layer, rate, channels, length; if (!(length = mp3_type_frame_length_from_header (header, &layer, &channels, &bitrate, &rate))) { return 0; } return length; } static gboolean head_check (unsigned long head) { GST_DEBUG ("checking mp3 header 0x%08lx",head); /* if it's not a valid sync */ if ((head & 0xffe00000) != 0xffe00000) { GST_DEBUG ("invalid sync");return FALSE; } /* if it's an invalid MPEG version */ if (((head >> 19) & 3) == 0x1) { GST_DEBUG ("invalid MPEG version");return FALSE; } /* if it's an invalid layer */ if (!((head >> 17) & 3)) { GST_DEBUG ("invalid layer");return FALSE; } /* if it's an invalid bitrate */ if (((head >> 12) & 0xf) == 0x0) { GST_DEBUG ("invalid bitrate");return FALSE; } if (((head >> 12) & 0xf) == 0xf) { GST_DEBUG ("invalid bitrate");return FALSE; } /* if it's an invalid samplerate */ if (((head >> 10) & 0x3) == 0x3) { GST_DEBUG ("invalid samplerate");return FALSE; } if ((head & 0xffff0000) == 0xfffe0000) { GST_DEBUG ("invalid sync");return FALSE; } if (head & 0x00000002) { GST_DEBUG ("invalid emphasis");return FALSE; } return TRUE; } static void gst_mp3parse_set_property (GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec) { GstMPEGAudioParse *src; /* it's not null if we got it, but it might not be ours */ g_return_if_fail(GST_IS_MP3PARSE(object)); src = GST_MP3PARSE(object); switch (prop_id) { case ARG_SKIP: src->skip = g_value_get_int (value); break; default: break; } } static void gst_mp3parse_get_property (GObject *object, guint prop_id, GValue *value, GParamSpec *pspec) { GstMPEGAudioParse *src; /* it's not null if we got it, but it might not be ours */ g_return_if_fail(GST_IS_MP3PARSE(object)); src = GST_MP3PARSE(object); switch (prop_id) { case ARG_SKIP: g_value_set_int (value, src->skip); break; case ARG_BIT_RATE: g_value_set_int (value, src->bit_rate * 1000); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static GstElementStateReturn gst_mp3parse_change_state (GstElement *element) { GstMPEGAudioParse *src; g_return_val_if_fail(GST_IS_MP3PARSE(element), GST_STATE_FAILURE); src = GST_MP3PARSE(element); switch (GST_STATE_TRANSITION (element)) { case GST_STATE_PAUSED_TO_READY: src->channels = -1; src->rate = -1; src->layer = -1; break; default: break; } if (GST_ELEMENT_CLASS(parent_class)->change_state) return GST_ELEMENT_CLASS(parent_class)->change_state(element); return GST_STATE_SUCCESS; } static gboolean plugin_init (GstPlugin *plugin) { return gst_element_register (plugin, "mp3parse", GST_RANK_NONE, GST_TYPE_MP3PARSE); } GST_PLUGIN_DEFINE ( GST_VERSION_MAJOR, GST_VERSION_MINOR, "mpegaudioparse", "MPEG-1 layer 1/2/3 audio parser", plugin_init, VERSION, "LGPL", GST_PACKAGE, GST_ORIGIN )