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567 lines
17 KiB
C
567 lines
17 KiB
C
/* GStreamer
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* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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/*#define GST_DEBUG_ENABLED */
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include "gstmpegaudioparse.h"
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/* elementfactory information */
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static GstElementDetails mp3parse_details = {
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"MPEG1 Audio Parser",
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"Codec/Parser/Audio",
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"Parses and frames mpeg1 audio streams (levels 1-3), provides seek",
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"Erik Walthinsen <omega@cse.ogi.edu>"
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};
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static GstStaticPadTemplate mp3_src_template = GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/mpeg, "
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"mpegversion = (int) 1, "
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"layer = (int) [ 1, 3 ], "
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"rate = (int) [ 8000, 48000], " "channels = (int) [ 1, 2 ]")
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);
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static GstStaticPadTemplate mp3_sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/mpeg, " "mpegversion = (int) 1")
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);
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/* GstMPEGAudioParse signals and args */
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enum
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{
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/* FILL ME */
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LAST_SIGNAL
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};
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enum
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{
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ARG_0,
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ARG_SKIP,
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ARG_BIT_RATE
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/* FILL ME */
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};
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static void gst_mp3parse_class_init (GstMPEGAudioParseClass * klass);
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static void gst_mp3parse_base_init (GstMPEGAudioParseClass * klass);
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static void gst_mp3parse_init (GstMPEGAudioParse * mp3parse);
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static GstFlowReturn gst_mp3parse_chain (GstPad * pad, GstBuffer * buffer);
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static int head_check (unsigned long head);
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static void gst_mp3parse_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_mp3parse_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static GstStateChangeReturn gst_mp3parse_change_state (GstElement * element,
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GstStateChange transition);
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static GstElementClass *parent_class = NULL;
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/*static guint gst_mp3parse_signals[LAST_SIGNAL] = { 0 }; */
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GType
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gst_mp3parse_get_type (void)
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{
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static GType mp3parse_type = 0;
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if (!mp3parse_type) {
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static const GTypeInfo mp3parse_info = {
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sizeof (GstMPEGAudioParseClass),
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(GBaseInitFunc) gst_mp3parse_base_init,
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NULL,
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(GClassInitFunc) gst_mp3parse_class_init,
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NULL,
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NULL,
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sizeof (GstMPEGAudioParse),
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0,
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(GInstanceInitFunc) gst_mp3parse_init,
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};
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mp3parse_type = g_type_register_static (GST_TYPE_ELEMENT,
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"GstMPEGAudioParse", &mp3parse_info, 0);
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}
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return mp3parse_type;
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}
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static guint mp3types_bitrates[2][3][16] =
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{ {{0, 32, 64, 96, 128, 160, 192, 224, 256, 288, 320, 352, 384, 416, 448,},
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{0, 32, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384,},
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{0, 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320,}},
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{{0, 32, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224, 256,},
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{0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160,},
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{0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160,}},
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};
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static guint mp3types_freqs[3][3] = { {44100, 48000, 32000},
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{22050, 24000, 16000},
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{11025, 12000, 8000}
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};
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static inline guint
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mp3_type_frame_length_from_header (guint32 header, guint * put_layer,
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guint * put_channels, guint * put_bitrate, guint * put_samplerate)
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{
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guint length;
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gulong mode, samplerate, bitrate, layer, channels, padding;
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gint lsf, mpg25;
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if (header & (1 << 20)) {
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lsf = (header & (1 << 19)) ? 0 : 1;
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mpg25 = 0;
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} else {
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lsf = 1;
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mpg25 = 1;
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}
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mode = (header >> 6) & 0x3;
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channels = (mode == 3) ? 1 : 2;
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samplerate = (header >> 10) & 0x3;
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samplerate = mp3types_freqs[lsf + mpg25][samplerate];
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layer = 4 - ((header >> 17) & 0x3);
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bitrate = (header >> 12) & 0xF;
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bitrate = mp3types_bitrates[lsf][layer - 1][bitrate] * 1000;
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if (bitrate == 0)
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return 0;
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padding = (header >> 9) & 0x1;
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switch (layer) {
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case 1:
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length = (bitrate * 12) / samplerate + 4 * padding;
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break;
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case 2:
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length = (bitrate * 144) / samplerate + padding;
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break;
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default:
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case 3:
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length = (bitrate * 144) / (samplerate << lsf) + padding;
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break;
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}
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GST_DEBUG ("Calculated mp3 frame length of %u bytes", length);
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GST_DEBUG ("samplerate = %lu, bitrate = %lu, layer = %lu, channels = %lu",
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samplerate, bitrate, layer, channels);
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if (put_layer)
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*put_layer = layer;
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if (put_channels)
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*put_channels = channels;
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if (put_bitrate)
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*put_bitrate = bitrate;
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if (put_samplerate)
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*put_samplerate = samplerate;
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return length;
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}
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/*
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* The chance that random data is identified as a valid mp3 header is 63 / 2^18
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* (0.024%) per try. This makes the function for calculating false positives
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* 1 - (1 - ((63 / 2 ^18) ^ GST_MP3_TYPEFIND_MIN_HEADERS)) ^ buffersize)
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* This has the following probabilities of false positives:
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* bufsize MIN_HEADERS
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* (bytes) 1 2 3 4
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* 4096 62.6% 0.02% 0% 0%
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* 16384 98% 0.09% 0% 0%
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* 1 MiB 100% 5.88% 0% 0%
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* 1 GiB 100% 100% 1.44% 0%
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* 1 TiB 100% 100% 100% 0.35%
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* This means that the current choice (3 headers by most of the time 4096 byte
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* buffers is pretty safe for now.
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*
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* The max. size of each frame is 1440 bytes, which means that for N frames
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* to be detected, we need 1440 * GST_MP3_TYPEFIND_MIN_HEADERS + 3 of data.
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* Assuming we step into the stream right after the frame header, this
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* means we need 1440 * (GST_MP3_TYPEFIND_MIN_HEADERS + 1) - 1 + 3 bytes
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* of data (5762) to always detect any mp3.
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*/
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/* increase this value when this function finds too many false positives */
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#define GST_MP3_TYPEFIND_MIN_HEADERS 3
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#define GST_MP3_TYPEFIND_MIN_DATA (1440 * (GST_MP3_TYPEFIND_MIN_HEADERS + 1) - 1 + 3)
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static GstCaps *
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mp3_caps_create (guint layer, guint channels, guint bitrate, guint samplerate)
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{
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GstCaps *new;
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g_assert (layer);
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g_assert (samplerate);
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g_assert (bitrate);
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g_assert (channels);
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new = gst_caps_new_simple ("audio/mpeg",
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"mpegversion", G_TYPE_INT, 1,
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"layer", G_TYPE_INT, layer,
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"rate", G_TYPE_INT, samplerate, "channels", G_TYPE_INT, channels, NULL);
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return new;
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}
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static void
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gst_mp3parse_base_init (GstMPEGAudioParseClass * klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&mp3_sink_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&mp3_src_template));
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gst_element_class_set_details (element_class, &mp3parse_details);
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}
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static void
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gst_mp3parse_class_init (GstMPEGAudioParseClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
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gobject_class->set_property = gst_mp3parse_set_property;
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gobject_class->get_property = gst_mp3parse_get_property;
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g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_SKIP,
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g_param_spec_int ("skip", "skip", "skip",
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G_MININT, G_MAXINT, 0, G_PARAM_READWRITE));
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g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_BIT_RATE,
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g_param_spec_int ("bitrate", "Bitrate", "Bit Rate",
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G_MININT, G_MAXINT, 0, G_PARAM_READABLE));
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gstelement_class->change_state = gst_mp3parse_change_state;
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}
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static void
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gst_mp3parse_init (GstMPEGAudioParse * mp3parse)
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{
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mp3parse->sinkpad =
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gst_pad_new_from_template (gst_static_pad_template_get
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(&mp3_sink_template), "sink");
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gst_pad_set_chain_function (mp3parse->sinkpad, gst_mp3parse_chain);
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gst_element_add_pad (GST_ELEMENT (mp3parse), mp3parse->sinkpad);
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mp3parse->srcpad =
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gst_pad_new_from_template (gst_static_pad_template_get
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(&mp3_src_template), "src");
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gst_pad_use_fixed_caps (mp3parse->srcpad);
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gst_element_add_pad (GST_ELEMENT (mp3parse), mp3parse->srcpad);
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/*gst_pad_set_type_id(mp3parse->srcpad, mp3frametype); */
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mp3parse->partialbuf = NULL;
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mp3parse->skip = 0;
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mp3parse->in_flush = FALSE;
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mp3parse->rate = mp3parse->channels = mp3parse->layer = -1;
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}
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/* FIXME, use adapter */
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static GstFlowReturn
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gst_mp3parse_chain (GstPad * pad, GstBuffer * buf)
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{
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GstMPEGAudioParse *mp3parse;
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guchar *data;
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glong size, offset = 0;
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guint32 header;
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int bpf;
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GstBuffer *outbuf;
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guint64 last_ts;
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mp3parse = GST_MP3PARSE (gst_pad_get_parent (pad));
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GST_DEBUG ("mp3parse: received buffer of %d bytes", GST_BUFFER_SIZE (buf));
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last_ts = GST_BUFFER_TIMESTAMP (buf);
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/* if we have something left from the previous frame */
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if (mp3parse->partialbuf) {
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GstBuffer *newbuf;
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newbuf = gst_buffer_merge (mp3parse->partialbuf, buf);
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/* and the one we received.. */
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gst_buffer_unref (buf);
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gst_buffer_unref (mp3parse->partialbuf);
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mp3parse->partialbuf = newbuf;
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} else {
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mp3parse->partialbuf = buf;
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}
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size = GST_BUFFER_SIZE (mp3parse->partialbuf);
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data = GST_BUFFER_DATA (mp3parse->partialbuf);
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/* while we still have bytes left -4 for the header */
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while (offset < size - 4) {
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int skipped = 0;
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GST_DEBUG ("mp3parse: offset %ld, size %ld ", offset, size);
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/* search for a possible start byte */
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for (; ((offset < size - 4) && (data[offset] != 0xff)); offset++)
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skipped++;
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if (skipped) {
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GST_DEBUG ("mp3parse: **** now at %ld skipped %d bytes", offset, skipped);
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}
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/* construct the header word */
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header = GST_READ_UINT32_BE (data + offset);
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/* if it's a valid header, go ahead and send off the frame */
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if (head_check (header)) {
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guint bitrate = 0, layer = 0, rate = 0, channels = 0;
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if (!(bpf = mp3_type_frame_length_from_header (header, &layer,
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&channels, &bitrate, &rate))) {
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g_error ("Header failed internal error");
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}
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/********************************************************************************
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* robust seek support
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* - This performs additional frame validation if the in_flush flag is set
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* (indicating a discontinuous stream).
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* - The current frame header is not accepted as valid unless the NEXT frame
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* header has the same values for most fields. This significantly increases
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* the probability that we aren't processing random data.
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* - It is not clear if this is sufficient for robust seeking of Layer III
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* streams which utilize the concept of a "bit reservoir" by borrow bitrate
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* from previous frames. In this case, seeking may be more complicated because
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* the frames are not independently coded.
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********************************************************************************/
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if (mp3parse->in_flush) {
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guint32 header2;
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if ((size - offset) < (bpf + 4)) {
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if (mp3parse->in_flush)
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break;
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}
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/* wait until we have the the entire current frame as well as the next frame header */
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header2 = GST_READ_UINT32_BE (data + offset + bpf);
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GST_DEBUG ("mp3parse: header=%08X, header2=%08X, bpf=%d",
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(unsigned int) header, (unsigned int) header2, bpf);
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/* mask the bits which are allowed to differ between frames */
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#define HDRMASK ~((0xF << 12) /* bitrate */ | \
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(0x1 << 9) /* padding */ | \
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(0x3 << 4)) /*mode extension */
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if ((header2 & HDRMASK) != (header & HDRMASK)) { /* require 2 matching headers in a row */
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GST_DEBUG
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("mp3parse: next header doesn't match (header=%08X, header2=%08X, bpf=%d)",
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(unsigned int) header, (unsigned int) header2, bpf);
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offset++; /* This frame is invalid. Start looking for a valid frame at the next position in the stream */
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continue;
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}
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}
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/* if we don't have the whole frame... */
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if ((size - offset) < bpf) {
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GST_DEBUG ("mp3parse: partial buffer needed %ld < %d ", (size - offset),
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bpf);
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break;
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} else {
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if (channels != mp3parse->channels ||
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rate != mp3parse->rate ||
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layer != mp3parse->layer || bitrate != mp3parse->bit_rate) {
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GstCaps *caps = mp3_caps_create (layer, channels, bitrate, rate);
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gst_pad_set_caps (mp3parse->srcpad, caps);
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gst_caps_unref (caps);
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mp3parse->channels = channels;
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mp3parse->layer = layer;
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mp3parse->rate = rate;
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mp3parse->bit_rate = bitrate;
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}
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outbuf = gst_buffer_create_sub (mp3parse->partialbuf, offset, bpf);
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offset += bpf;
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if (mp3parse->skip == 0) {
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GST_DEBUG ("mp3parse: pushing buffer of %d bytes",
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GST_BUFFER_SIZE (outbuf));
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GST_BUFFER_TIMESTAMP (outbuf) = last_ts;
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if (mp3parse->layer == 1) {
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GST_BUFFER_DURATION (outbuf) = 384 * GST_SECOND / mp3parse->rate;
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} else {
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GST_BUFFER_DURATION (outbuf) = 1152 * GST_SECOND / mp3parse->rate;
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}
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gst_buffer_set_caps (outbuf, GST_PAD_CAPS (pad));
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gst_pad_push (mp3parse->srcpad, outbuf);
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} else {
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GST_DEBUG ("mp3parse: skipping buffer of %d bytes",
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GST_BUFFER_SIZE (outbuf));
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gst_buffer_unref (outbuf);
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mp3parse->skip--;
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}
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}
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} else {
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offset++;
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GST_DEBUG ("mp3parse: *** wrong header, skipping byte (FIXME?)");
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}
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}
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/* if we have processed this block and there are still */
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/* bytes left not in a partial block, copy them over. */
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if (size - offset > 0) {
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glong remainder = (size - offset);
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GST_DEBUG ("mp3parse: partial buffer needed %ld for trailing bytes",
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remainder);
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outbuf = gst_buffer_create_sub (mp3parse->partialbuf, offset, remainder);
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gst_buffer_unref (mp3parse->partialbuf);
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mp3parse->partialbuf = outbuf;
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} else {
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gst_buffer_unref (mp3parse->partialbuf);
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mp3parse->partialbuf = NULL;
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}
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gst_object_unref (mp3parse);
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return GST_FLOW_OK;
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}
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static gboolean
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head_check (unsigned long head)
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{
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GST_DEBUG ("checking mp3 header 0x%08lx", head);
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/* if it's not a valid sync */
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if ((head & 0xffe00000) != 0xffe00000) {
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GST_DEBUG ("invalid sync");
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return FALSE;
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}
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/* if it's an invalid MPEG version */
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if (((head >> 19) & 3) == 0x1) {
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GST_DEBUG ("invalid MPEG version");
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return FALSE;
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}
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/* if it's an invalid layer */
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if (!((head >> 17) & 3)) {
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GST_DEBUG ("invalid layer");
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return FALSE;
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}
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/* 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)
|