/* 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 "gstmpegaudioparse.h" /* elementfactory information */ static const GstElementDetails mp3parse_details = GST_ELEMENT_DETAILS ("MPEG-1 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 GstFlowReturn gst_mp3parse_chain (GstPad * pad, GstBuffer * buffer); 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 GstStateChangeReturn gst_mp3parse_change_state (GstElement * element, GstStateChange transition); 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; parent_class = g_type_class_peek_parent (klass); gobject_class->set_property = gst_mp3parse_set_property; gobject_class->get_property = gst_mp3parse_get_property; 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)); 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)); 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_pad_set_chain_function (mp3parse->sinkpad, gst_mp3parse_chain); gst_element_add_pad (GST_ELEMENT (mp3parse), mp3parse->sinkpad); mp3parse->srcpad = gst_pad_new_from_template (gst_static_pad_template_get (&mp3_src_template), "src"); gst_pad_use_fixed_caps (mp3parse->srcpad); gst_element_add_pad (GST_ELEMENT (mp3parse), 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; } /* FIXME, use adapter */ static GstFlowReturn gst_mp3parse_chain (GstPad * pad, GstBuffer * buf) { GstMPEGAudioParse *mp3parse; guchar *data; glong size, offset = 0; guint32 header; int bpf; GstBuffer *outbuf; guint64 last_ts; 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); /* 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 (; ((offset < size - 4) && (data[offset] != 0xff)); offset++) skipped++; if (skipped) { GST_DEBUG ("mp3parse: **** now at %ld skipped %d bytes", offset, skipped); } /* construct the header word */ header = GST_READ_UINT32_BE (data + offset); /* if it's a valid header, go ahead and send off the frame */ if (head_check (header)) { guint bitrate = 0, layer = 0, rate = 0, channels = 0; if (!(bpf = mp3_type_frame_length_from_header (header, &layer, &channels, &bitrate, &rate))) { g_error ("Header failed internal error"); } /******************************************************************************** * 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 = GST_READ_UINT32_BE (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 { 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_caps (mp3parse->srcpad, caps); gst_caps_unref (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)); GST_BUFFER_TIMESTAMP (outbuf) = last_ts; if (mp3parse->layer == 1) { GST_BUFFER_DURATION (outbuf) = 384 * GST_SECOND / mp3parse->rate; } else { GST_BUFFER_DURATION (outbuf) = 1152 * GST_SECOND / mp3parse->rate; } gst_buffer_set_caps (outbuf, GST_PAD_CAPS (pad)); gst_pad_push (mp3parse->srcpad, outbuf); } else { GST_DEBUG ("mp3parse: skipping buffer of %d bytes", GST_BUFFER_SIZE (outbuf)); gst_buffer_unref (outbuf); mp3parse->skip--; } } } else { offset++; 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; } gst_object_unref (mp3parse); return GST_FLOW_OK; } 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; 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; 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 GstStateChangeReturn gst_mp3parse_change_state (GstElement * element, GstStateChange transition) { GstMPEGAudioParse *src; GstStateChangeReturn result; src = GST_MP3PARSE (element); switch (transition) { case GST_STATE_CHANGE_PAUSED_TO_READY: src->channels = -1; src->rate = -1; src->layer = -1; break; default: break; } result = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); return result; } 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_NAME, GST_PACKAGE_ORIGIN)