+++ /dev/null
-/* GStreamer
- * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
- *
- * 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 <omega@cse.ogi.edu>");
-
-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)
projects / platform / upstream / gstreamer.git / commitdiff