mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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6d5dba30d2
Original commit message from CVS: * gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_sink_event), (gst_mp3parse_emit_frame), (mp3parse_total_time), (mp3parse_bytepos_to_time): Don't recurse from mp3parse_bytepos_to_time() to mp3parse_total_time() if we're called from there already. Otherwise we end up in a endless recursion and crash with a stack overflow. This can happen when a Xing or VBRI header with TOC exists but it doesn't contain the total time. Fixes bug #545370.
1970 lines
59 KiB
C
1970 lines
59 KiB
C
/* GStreamer
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* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
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* Copyright (C) <2006-2007> Jan Schmidt <thaytan@mad.scientist.com>
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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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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <string.h>
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#include "gstmpegaudioparse.h"
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GST_DEBUG_CATEGORY_STATIC (mp3parse_debug);
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#define GST_CAT_DEFAULT mp3parse_debug
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#define MP3_CHANNEL_MODE_UNKNOWN -1
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#define MP3_CHANNEL_MODE_STEREO 0
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#define MP3_CHANNEL_MODE_JOINT_STEREO 1
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#define MP3_CHANNEL_MODE_DUAL_CHANNEL 2
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#define MP3_CHANNEL_MODE_MONO 3
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#define CRC_UNKNOWN -1
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#define CRC_PROTECTED 0
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#define CRC_NOT_PROTECTED 1
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#define XING_FRAMES_FLAG 0x0001
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#define XING_BYTES_FLAG 0x0002
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#define XING_TOC_FLAG 0x0004
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#define XING_VBR_SCALE_FLAG 0x0008
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#define GST_READ_UINT24_BE(p) (p[2] | (p[1] << 8) | (p[0] << 16))
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static inline MPEGAudioSeekEntry *
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mpeg_audio_seek_entry_new ()
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{
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return g_slice_new (MPEGAudioSeekEntry);
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}
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static inline void
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mpeg_audio_seek_entry_free (MPEGAudioSeekEntry * entry)
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{
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g_slice_free (MPEGAudioSeekEntry, entry);
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}
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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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"Jan Schmidt <thaytan@mad.scientist.com>\n"
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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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"parsed=(boolean) true")
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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, parsed=(boolean)false")
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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 (gpointer klass);
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static void gst_mp3parse_init (GstMPEGAudioParse * mp3parse,
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GstMPEGAudioParseClass * klass);
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static gboolean gst_mp3parse_sink_event (GstPad * pad, GstEvent * event);
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static GstFlowReturn gst_mp3parse_chain (GstPad * pad, GstBuffer * buffer);
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static gboolean mp3parse_src_query (GstPad * pad, GstQuery * query);
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static const GstQueryType *mp3parse_get_query_types (GstPad * pad);
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static gboolean mp3parse_src_event (GstPad * pad, GstEvent * event);
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static int head_check (GstMPEGAudioParse * mp3parse, unsigned long head);
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static void gst_mp3parse_dispose (GObject * object);
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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 gboolean mp3parse_bytepos_to_time (GstMPEGAudioParse * mp3parse,
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gint64 bytepos, GstClockTime * ts, gboolean from_total_time);
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static gboolean
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mp3parse_total_bytes (GstMPEGAudioParse * mp3parse, gint64 * total);
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static gboolean
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mp3parse_total_time (GstMPEGAudioParse * mp3parse, GstClockTime * total);
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/* static guint gst_mp3parse_signals[LAST_SIGNAL] = { 0 }; */
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GST_BOILERPLATE (GstMPEGAudioParse, gst_mp3parse, GstElement, GST_TYPE_ELEMENT);
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#define GST_TYPE_MP3_CHANNEL_MODE (gst_mp3_channel_mode_get_type())
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G_GNUC_UNUSED static GType
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gst_mp3_channel_mode_get_type (void)
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{
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static GType mp3_channel_mode_type = 0;
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static GEnumValue mp3_channel_mode[] = {
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{MP3_CHANNEL_MODE_UNKNOWN, "Unknown", "unknown"},
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{MP3_CHANNEL_MODE_MONO, "Mono", "mono"},
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{MP3_CHANNEL_MODE_DUAL_CHANNEL, "Dual Channel", "dual-channel"},
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{MP3_CHANNEL_MODE_JOINT_STEREO, "Joint Stereo", "joint-stereo"},
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{MP3_CHANNEL_MODE_STEREO, "Stereo", "stereo"},
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{0, NULL, NULL},
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};
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if (!mp3_channel_mode_type) {
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mp3_channel_mode_type =
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g_enum_register_static ("GstMp3ChannelMode", mp3_channel_mode);
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}
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return mp3_channel_mode_type;
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}
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static const guint mp3types_bitrates[2][3][16] = {
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{
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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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},
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{
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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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};
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static const 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 (GstMPEGAudioParse * mp3parse, guint32 header,
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guint * put_version, guint * put_layer, guint * put_channels,
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guint * put_bitrate, guint * put_samplerate, guint * put_mode,
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guint * put_crc)
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{
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guint length;
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gulong mode, samplerate, bitrate, layer, channels, padding, crc;
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gulong version;
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gint lsf, mpg25;
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GEnumValue *mode_enum;
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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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version = 1 + lsf + mpg25;
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layer = 4 - ((header >> 17) & 0x3);
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crc = (header >> 16) & 0x1;
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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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samplerate = (header >> 10) & 0x3;
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samplerate = mp3types_freqs[lsf + mpg25][samplerate];
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padding = (header >> 9) & 0x1;
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mode = (header >> 6) & 0x3;
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channels = (mode == 3) ? 1 : 2;
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switch (layer) {
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case 1:
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length = 4 * ((bitrate * 12) / samplerate + 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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mode_enum =
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g_enum_get_value (g_type_class_ref (GST_TYPE_MP3_CHANNEL_MODE), mode);
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GST_DEBUG_OBJECT (mp3parse, "Calculated mp3 frame length of %u bytes",
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length);
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GST_DEBUG_OBJECT (mp3parse, "samplerate = %lu, bitrate = %lu, version = %lu, "
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"layer = %lu, channels = %lu, mode = %s", samplerate, bitrate, version,
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layer, channels, mode_enum->value_nick);
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if (put_version)
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*put_version = version;
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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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if (put_mode)
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*put_mode = mode;
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if (put_crc)
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*put_crc = crc;
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return length;
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}
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static GstCaps *
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mp3_caps_create (guint version, guint layer, guint channels, guint samplerate)
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{
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GstCaps *new;
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g_assert (version);
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g_assert (layer);
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g_assert (samplerate);
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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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"mpegaudioversion", G_TYPE_INT, version,
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"layer", G_TYPE_INT, layer,
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"rate", G_TYPE_INT, samplerate,
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"channels", G_TYPE_INT, channels, "parsed", G_TYPE_BOOLEAN, TRUE, 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 (gpointer 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_DEBUG_CATEGORY_INIT (mp3parse_debug, "mp3parse", 0, "MPEG Audio Parser");
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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_peek_parent (klass);
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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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gobject_class->dispose = gst_mp3parse_dispose;
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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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/* register tags */
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#define GST_TAG_CRC "has-crc"
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#define GST_TAG_MODE "channel-mode"
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gst_tag_register (GST_TAG_CRC, GST_TAG_FLAG_META, G_TYPE_BOOLEAN,
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"has crc", "Using CRC", NULL);
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gst_tag_register (GST_TAG_MODE, GST_TAG_FLAG_ENCODED, G_TYPE_STRING,
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"channel mode", "MPEG audio channel mode", NULL);
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}
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static void
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gst_mp3parse_reset (GstMPEGAudioParse * mp3parse)
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{
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mp3parse->skip = 0;
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mp3parse->resyncing = TRUE;
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mp3parse->next_ts = GST_CLOCK_TIME_NONE;
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mp3parse->cur_offset = -1;
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mp3parse->tracked_offset = 0;
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mp3parse->pending_ts = GST_CLOCK_TIME_NONE;
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mp3parse->pending_offset = -1;
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gst_adapter_clear (mp3parse->adapter);
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mp3parse->rate = mp3parse->channels = mp3parse->layer = -1;
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mp3parse->version = 1;
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mp3parse->max_bitreservoir = GST_CLOCK_TIME_NONE;
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mp3parse->avg_bitrate = 0;
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mp3parse->bitrate_sum = 0;
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mp3parse->last_posted_bitrate = 0;
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mp3parse->frame_count = 0;
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mp3parse->sent_codec_tag = FALSE;
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mp3parse->last_posted_crc = CRC_UNKNOWN;
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mp3parse->last_posted_channel_mode = MP3_CHANNEL_MODE_UNKNOWN;
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mp3parse->xing_flags = 0;
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mp3parse->xing_bitrate = 0;
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mp3parse->xing_frames = 0;
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mp3parse->xing_total_time = 0;
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mp3parse->xing_bytes = 0;
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mp3parse->xing_vbr_scale = 0;
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memset (mp3parse->xing_seek_table, 0, 100);
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memset (mp3parse->xing_seek_table_inverse, 0, 256);
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mp3parse->vbri_bitrate = 0;
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mp3parse->vbri_frames = 0;
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mp3parse->vbri_total_time = 0;
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mp3parse->vbri_bytes = 0;
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mp3parse->vbri_seek_points = 0;
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g_free (mp3parse->vbri_seek_table);
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mp3parse->vbri_seek_table = NULL;
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if (mp3parse->seek_table) {
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g_list_foreach (mp3parse->seek_table, (GFunc) mpeg_audio_seek_entry_free,
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NULL);
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g_list_free (mp3parse->seek_table);
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mp3parse->seek_table = NULL;
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}
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g_mutex_lock (mp3parse->pending_accurate_seeks_lock);
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if (mp3parse->pending_accurate_seeks) {
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g_slist_foreach (mp3parse->pending_accurate_seeks, (GFunc) g_free, NULL);
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g_slist_free (mp3parse->pending_accurate_seeks);
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mp3parse->pending_accurate_seeks = NULL;
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}
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g_mutex_unlock (mp3parse->pending_accurate_seeks_lock);
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if (mp3parse->pending_segment) {
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GstEvent **eventp = &mp3parse->pending_segment;
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gst_event_replace (eventp, NULL);
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}
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mp3parse->exact_position = FALSE;
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gst_segment_init (&mp3parse->segment, GST_FORMAT_TIME);
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}
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static void
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gst_mp3parse_init (GstMPEGAudioParse * mp3parse, GstMPEGAudioParseClass * klass)
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{
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mp3parse->sinkpad =
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gst_pad_new_from_static_template (&mp3_sink_template, "sink");
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gst_pad_set_event_function (mp3parse->sinkpad, gst_mp3parse_sink_event);
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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_static_template (&mp3_src_template, "src");
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gst_pad_use_fixed_caps (mp3parse->srcpad);
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gst_pad_set_event_function (mp3parse->srcpad, mp3parse_src_event);
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gst_pad_set_query_function (mp3parse->srcpad, mp3parse_src_query);
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gst_pad_set_query_type_function (mp3parse->srcpad, mp3parse_get_query_types);
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gst_element_add_pad (GST_ELEMENT (mp3parse), mp3parse->srcpad);
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mp3parse->adapter = gst_adapter_new ();
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mp3parse->pending_accurate_seeks_lock = g_mutex_new ();
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gst_mp3parse_reset (mp3parse);
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}
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static void
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gst_mp3parse_dispose (GObject * object)
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{
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GstMPEGAudioParse *mp3parse = GST_MP3PARSE (object);
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gst_mp3parse_reset (mp3parse);
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if (mp3parse->adapter) {
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g_object_unref (mp3parse->adapter);
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mp3parse->adapter = NULL;
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}
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g_mutex_free (mp3parse->pending_accurate_seeks_lock);
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mp3parse->pending_accurate_seeks_lock = NULL;
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g_list_foreach (mp3parse->pending_events, (GFunc) gst_mini_object_unref,
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NULL);
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g_list_free (mp3parse->pending_events);
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mp3parse->pending_events = NULL;
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G_OBJECT_CLASS (parent_class)->dispose (object);
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}
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static gboolean
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gst_mp3parse_sink_event (GstPad * pad, GstEvent * event)
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|
{
|
|
gboolean res = TRUE;
|
|
|
|
GstMPEGAudioParse *mp3parse;
|
|
|
|
GstEvent **eventp;
|
|
|
|
mp3parse = GST_MP3PARSE (gst_pad_get_parent (pad));
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_NEWSEGMENT:
|
|
{
|
|
gdouble rate, applied_rate;
|
|
|
|
GstFormat format;
|
|
|
|
gint64 start, stop, pos;
|
|
|
|
gboolean update;
|
|
|
|
gst_event_parse_new_segment_full (event, &update, &rate, &applied_rate,
|
|
&format, &start, &stop, &pos);
|
|
|
|
g_mutex_lock (mp3parse->pending_accurate_seeks_lock);
|
|
if (format == GST_FORMAT_BYTES && mp3parse->pending_accurate_seeks) {
|
|
MPEGAudioPendingAccurateSeek *seek = NULL;
|
|
|
|
GSList *node;
|
|
|
|
for (node = mp3parse->pending_accurate_seeks; node; node = node->next) {
|
|
MPEGAudioPendingAccurateSeek *tmp = node->data;
|
|
|
|
if (tmp->upstream_start == pos) {
|
|
seek = tmp;
|
|
break;
|
|
}
|
|
}
|
|
if (seek) {
|
|
GstSegment *s = &seek->segment;
|
|
|
|
event =
|
|
gst_event_new_new_segment_full (FALSE, s->rate, s->applied_rate,
|
|
GST_FORMAT_TIME, s->start, s->stop, s->last_stop);
|
|
|
|
mp3parse->segment = seek->segment;
|
|
|
|
mp3parse->resyncing = FALSE;
|
|
mp3parse->cur_offset = pos;
|
|
mp3parse->next_ts = seek->timestamp_start;
|
|
mp3parse->pending_ts = GST_CLOCK_TIME_NONE;
|
|
mp3parse->tracked_offset = 0;
|
|
|
|
gst_event_parse_new_segment_full (event, &update, &rate,
|
|
&applied_rate, &format, &start, &stop, &pos);
|
|
|
|
GST_DEBUG_OBJECT (mp3parse,
|
|
"Pushing accurate newseg rate %g, applied rate %g, "
|
|
"format %d, start %lld, stop %lld, pos %lld", rate,
|
|
applied_rate, format, start, stop, pos);
|
|
|
|
g_free (seek);
|
|
mp3parse->pending_accurate_seeks =
|
|
g_slist_delete_link (mp3parse->pending_accurate_seeks, node);
|
|
|
|
g_mutex_unlock (mp3parse->pending_accurate_seeks_lock);
|
|
res = gst_pad_push_event (mp3parse->srcpad, event);
|
|
|
|
return res;
|
|
} else {
|
|
GST_WARNING_OBJECT (mp3parse,
|
|
"Accurate seek not possible, didn't get an appropiate upstream segment");
|
|
}
|
|
}
|
|
g_mutex_unlock (mp3parse->pending_accurate_seeks_lock);
|
|
|
|
mp3parse->exact_position = FALSE;
|
|
|
|
if (format == GST_FORMAT_BYTES) {
|
|
GstClockTime seg_start, seg_stop, seg_pos;
|
|
|
|
/* stop time is allowed to be open-ended, but not start & pos */
|
|
if (!mp3parse_bytepos_to_time (mp3parse, stop, &seg_stop, FALSE))
|
|
seg_stop = GST_CLOCK_TIME_NONE;
|
|
if (mp3parse_bytepos_to_time (mp3parse, start, &seg_start, FALSE) &&
|
|
mp3parse_bytepos_to_time (mp3parse, pos, &seg_pos, FALSE)) {
|
|
gst_event_unref (event);
|
|
event = gst_event_new_new_segment_full (update, rate, applied_rate,
|
|
GST_FORMAT_TIME, seg_start, seg_stop, seg_pos);
|
|
format = GST_FORMAT_TIME;
|
|
GST_DEBUG_OBJECT (mp3parse, "Converted incoming segment to TIME. "
|
|
"start = %" GST_TIME_FORMAT ", stop = %" GST_TIME_FORMAT
|
|
", pos = %" GST_TIME_FORMAT, GST_TIME_ARGS (seg_start),
|
|
GST_TIME_ARGS (seg_stop), GST_TIME_ARGS (seg_pos));
|
|
}
|
|
}
|
|
|
|
if (format != GST_FORMAT_TIME) {
|
|
/* Unknown incoming segment format. Output a default open-ended
|
|
* TIME segment */
|
|
gst_event_unref (event);
|
|
event = gst_event_new_new_segment_full (update, rate, applied_rate,
|
|
GST_FORMAT_TIME, 0, GST_CLOCK_TIME_NONE, 0);
|
|
}
|
|
|
|
mp3parse->resyncing = TRUE;
|
|
mp3parse->cur_offset = -1;
|
|
mp3parse->next_ts = GST_CLOCK_TIME_NONE;
|
|
mp3parse->pending_ts = GST_CLOCK_TIME_NONE;
|
|
mp3parse->tracked_offset = 0;
|
|
|
|
gst_event_parse_new_segment_full (event, &update, &rate, &applied_rate,
|
|
&format, &start, &stop, &pos);
|
|
GST_DEBUG_OBJECT (mp3parse, "Pushing newseg rate %g, applied rate %g, "
|
|
"format %d, start %lld, stop %lld, pos %lld",
|
|
rate, applied_rate, format, start, stop, pos);
|
|
|
|
gst_segment_set_newsegment_full (&mp3parse->segment, update, rate,
|
|
applied_rate, format, start, stop, pos);
|
|
|
|
/* save the segment for later, right before we push a new buffer so that
|
|
* the caps are fixed and the next linked element can receive the segment. */
|
|
eventp = &mp3parse->pending_segment;
|
|
gst_event_replace (eventp, event);
|
|
gst_event_unref (event);
|
|
res = TRUE;
|
|
break;
|
|
}
|
|
case GST_EVENT_FLUSH_STOP:
|
|
/* Clear our adapter and set up for a new position */
|
|
gst_adapter_clear (mp3parse->adapter);
|
|
eventp = &mp3parse->pending_segment;
|
|
gst_event_replace (eventp, NULL);
|
|
res = gst_pad_push_event (mp3parse->srcpad, event);
|
|
break;
|
|
default:
|
|
if (mp3parse->pending_segment &&
|
|
(GST_EVENT_TYPE (event) != GST_EVENT_EOS) &&
|
|
(GST_EVENT_TYPE (event) != GST_EVENT_FLUSH_START)) {
|
|
/* Cache all events except EOS and the ones above if we have
|
|
* a pending segment */
|
|
mp3parse->pending_events =
|
|
g_list_append (mp3parse->pending_events, event);
|
|
} else {
|
|
res = gst_pad_push_event (mp3parse->srcpad, event);
|
|
}
|
|
break;
|
|
}
|
|
|
|
gst_object_unref (mp3parse);
|
|
|
|
return res;
|
|
}
|
|
|
|
static MPEGAudioSeekEntry *
|
|
mp3parse_seek_table_last_entry (GstMPEGAudioParse * mp3parse)
|
|
{
|
|
MPEGAudioSeekEntry *ret = NULL;
|
|
|
|
if (mp3parse->seek_table) {
|
|
ret = mp3parse->seek_table->data;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
/* Prepare a buffer of the indicated size, timestamp it and output */
|
|
static GstFlowReturn
|
|
gst_mp3parse_emit_frame (GstMPEGAudioParse * mp3parse, guint size,
|
|
guint mode, guint crc)
|
|
{
|
|
GstBuffer *outbuf;
|
|
|
|
guint bitrate;
|
|
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
|
|
GstClockTime push_start;
|
|
|
|
GstTagList *taglist;
|
|
|
|
outbuf = gst_adapter_take_buffer (mp3parse->adapter, size);
|
|
|
|
GST_BUFFER_DURATION (outbuf) =
|
|
gst_util_uint64_scale (GST_SECOND, mp3parse->spf, mp3parse->rate);
|
|
|
|
GST_BUFFER_OFFSET (outbuf) = mp3parse->cur_offset;
|
|
|
|
/* Check if we have a pending timestamp from an incoming buffer to apply
|
|
* here */
|
|
if (GST_CLOCK_TIME_IS_VALID (mp3parse->pending_ts)) {
|
|
if (mp3parse->tracked_offset >= mp3parse->pending_offset) {
|
|
/* If the incoming timestamp differs from our expected by more than 2
|
|
* 90khz MPEG ticks, then take it and, if needed, set the discont flag.
|
|
* This avoids creating imperfect streams just because of
|
|
* quantization in the MPEG clock sampling */
|
|
GstClockTimeDiff diff = mp3parse->next_ts - mp3parse->pending_ts;
|
|
|
|
if (diff < -2 * (GST_SECOND / 90000) || diff > 2 * (GST_SECOND / 90000)) {
|
|
GST_DEBUG_OBJECT (mp3parse, "Updating next_ts from %" GST_TIME_FORMAT
|
|
" to pending ts %" GST_TIME_FORMAT
|
|
" at offset %lld (pending offset was %lld)",
|
|
GST_TIME_ARGS (mp3parse->next_ts),
|
|
GST_TIME_ARGS (mp3parse->pending_ts), mp3parse->tracked_offset,
|
|
mp3parse->pending_offset);
|
|
|
|
/* Only set discont if we sent out some timestamps already and we're
|
|
* adjusting */
|
|
if (GST_CLOCK_TIME_IS_VALID (mp3parse->next_ts))
|
|
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
|
|
mp3parse->next_ts = mp3parse->pending_ts;
|
|
}
|
|
mp3parse->pending_ts = GST_CLOCK_TIME_NONE;
|
|
}
|
|
}
|
|
|
|
/* Decide what timestamp we're going to apply */
|
|
if (GST_CLOCK_TIME_IS_VALID (mp3parse->next_ts)) {
|
|
GST_BUFFER_TIMESTAMP (outbuf) = mp3parse->next_ts;
|
|
} else {
|
|
GstClockTime ts;
|
|
|
|
/* No timestamp yet, convert our offset to a timestamp if we can, or
|
|
* start at 0 */
|
|
if (mp3parse_bytepos_to_time (mp3parse, mp3parse->cur_offset, &ts, FALSE) &&
|
|
GST_CLOCK_TIME_IS_VALID (ts))
|
|
GST_BUFFER_TIMESTAMP (outbuf) = ts;
|
|
else {
|
|
GST_BUFFER_TIMESTAMP (outbuf) = 0;
|
|
}
|
|
}
|
|
|
|
if (GST_BUFFER_TIMESTAMP (outbuf) == 0)
|
|
mp3parse->exact_position = TRUE;
|
|
|
|
if (mp3parse->exact_position && GST_BUFFER_TIMESTAMP_IS_VALID (outbuf) &&
|
|
mp3parse->cur_offset != GST_BUFFER_OFFSET_NONE &&
|
|
(!mp3parse->seek_table ||
|
|
(mp3parse_seek_table_last_entry (mp3parse))->byte <
|
|
GST_BUFFER_OFFSET (outbuf))) {
|
|
MPEGAudioSeekEntry *entry = mpeg_audio_seek_entry_new ();
|
|
|
|
entry->byte = mp3parse->cur_offset;
|
|
entry->timestamp = GST_BUFFER_TIMESTAMP (outbuf);
|
|
mp3parse->seek_table = g_list_prepend (mp3parse->seek_table, entry);
|
|
GST_DEBUG_OBJECT (mp3parse, "Adding index entry %" GST_TIME_FORMAT
|
|
" @ offset 0x%08" G_GINT64_MODIFIER "x",
|
|
GST_TIME_ARGS (entry->timestamp), entry->byte);
|
|
}
|
|
|
|
/* Update our byte offset tracking */
|
|
if (mp3parse->cur_offset != -1) {
|
|
mp3parse->cur_offset += size;
|
|
}
|
|
mp3parse->tracked_offset += size;
|
|
|
|
if (GST_BUFFER_TIMESTAMP_IS_VALID (outbuf)
|
|
&& GST_BUFFER_DURATION_IS_VALID (outbuf))
|
|
mp3parse->next_ts =
|
|
GST_BUFFER_TIMESTAMP (outbuf) + GST_BUFFER_DURATION (outbuf);
|
|
|
|
gst_buffer_set_caps (outbuf, GST_PAD_CAPS (mp3parse->srcpad));
|
|
|
|
/* Post a bitrate tag if we need to before pushing the buffer */
|
|
if (mp3parse->xing_bitrate != 0)
|
|
bitrate = mp3parse->xing_bitrate;
|
|
else if (mp3parse->vbri_bitrate != 0)
|
|
bitrate = mp3parse->vbri_bitrate;
|
|
else
|
|
bitrate = mp3parse->avg_bitrate;
|
|
|
|
/* we will create a taglist (if any of the parameters has changed)
|
|
* to add the tags that changed */
|
|
taglist = NULL;
|
|
if ((mp3parse->last_posted_bitrate / 10000) != (bitrate / 10000)) {
|
|
taglist = gst_tag_list_new ();
|
|
mp3parse->last_posted_bitrate = bitrate;
|
|
gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE, GST_TAG_BITRATE,
|
|
mp3parse->last_posted_bitrate, NULL);
|
|
|
|
/* Post a new duration message if the average bitrate changes that much
|
|
* so applications can update their cached values
|
|
*/
|
|
if ((mp3parse->xing_flags & XING_TOC_FLAG) == 0
|
|
&& mp3parse->vbri_total_time == 0) {
|
|
gst_element_post_message (GST_ELEMENT (mp3parse),
|
|
gst_message_new_duration (GST_OBJECT (mp3parse), GST_FORMAT_TIME,
|
|
-1));
|
|
}
|
|
}
|
|
|
|
if (mp3parse->last_posted_crc != crc) {
|
|
gboolean using_crc;
|
|
|
|
if (!taglist) {
|
|
taglist = gst_tag_list_new ();
|
|
}
|
|
mp3parse->last_posted_crc = crc;
|
|
if (mp3parse->last_posted_crc == CRC_PROTECTED) {
|
|
using_crc = TRUE;
|
|
} else {
|
|
using_crc = FALSE;
|
|
}
|
|
gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE, GST_TAG_CRC,
|
|
using_crc, NULL);
|
|
}
|
|
|
|
if (mp3parse->last_posted_channel_mode != mode) {
|
|
GEnumValue *mode_enum;
|
|
|
|
if (!taglist) {
|
|
taglist = gst_tag_list_new ();
|
|
}
|
|
mp3parse->last_posted_channel_mode = mode;
|
|
|
|
mode_enum = g_enum_get_value (g_type_class_ref (GST_TYPE_MP3_CHANNEL_MODE),
|
|
mp3parse->last_posted_channel_mode);
|
|
|
|
gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE, GST_TAG_MODE,
|
|
mode_enum->value_nick, NULL);
|
|
}
|
|
|
|
/* if the taglist exists, we need to send it */
|
|
if (taglist) {
|
|
gst_element_found_tags_for_pad (GST_ELEMENT (mp3parse),
|
|
mp3parse->srcpad, taglist);
|
|
}
|
|
|
|
/* We start pushing 9 frames earlier (29 frames for MPEG2) than
|
|
* segment start to be able to decode the first frame we want.
|
|
* 9 (29) frames are the theoretical maximum of frames that contain
|
|
* data for the current frame (bit reservoir).
|
|
*/
|
|
if (mp3parse->segment.start == 0) {
|
|
push_start = 0;
|
|
} else if (GST_CLOCK_TIME_IS_VALID (mp3parse->max_bitreservoir)) {
|
|
if (GST_CLOCK_TIME_IS_VALID (mp3parse->segment.start) &&
|
|
mp3parse->segment.start > mp3parse->max_bitreservoir)
|
|
push_start = mp3parse->segment.start - mp3parse->max_bitreservoir;
|
|
else
|
|
push_start = 0;
|
|
} else {
|
|
push_start = mp3parse->segment.start;
|
|
}
|
|
|
|
if (G_UNLIKELY ((GST_CLOCK_TIME_IS_VALID (push_start) &&
|
|
GST_BUFFER_TIMESTAMP_IS_VALID (outbuf) &&
|
|
GST_BUFFER_DURATION_IS_VALID (outbuf) &&
|
|
GST_BUFFER_TIMESTAMP (outbuf) + GST_BUFFER_DURATION (outbuf)
|
|
< push_start))) {
|
|
GST_DEBUG_OBJECT (mp3parse,
|
|
"Buffer before configured segment range %" GST_TIME_FORMAT
|
|
" to %" GST_TIME_FORMAT ", dropping, timestamp %"
|
|
GST_TIME_FORMAT " duration %" GST_TIME_FORMAT
|
|
", offset 0x%08" G_GINT64_MODIFIER "x", GST_TIME_ARGS (push_start),
|
|
GST_TIME_ARGS (mp3parse->segment.stop),
|
|
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
|
|
GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)),
|
|
GST_BUFFER_OFFSET (outbuf));
|
|
|
|
gst_buffer_unref (outbuf);
|
|
ret = GST_FLOW_OK;
|
|
} else if (G_UNLIKELY (GST_BUFFER_TIMESTAMP_IS_VALID (outbuf) &&
|
|
GST_CLOCK_TIME_IS_VALID (mp3parse->segment.stop) &&
|
|
GST_BUFFER_TIMESTAMP (outbuf) >= mp3parse->segment.stop)) {
|
|
GST_DEBUG_OBJECT (mp3parse,
|
|
"Buffer after configured segment range %" GST_TIME_FORMAT
|
|
" to %" GST_TIME_FORMAT ", returning GST_FLOW_UNEXPECTED, timestamp %"
|
|
GST_TIME_FORMAT " duration %" GST_TIME_FORMAT ", offset 0x%08"
|
|
G_GINT64_MODIFIER "x", GST_TIME_ARGS (push_start),
|
|
GST_TIME_ARGS (mp3parse->segment.stop),
|
|
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
|
|
GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)),
|
|
GST_BUFFER_OFFSET (outbuf));
|
|
|
|
gst_buffer_unref (outbuf);
|
|
ret = GST_FLOW_UNEXPECTED;
|
|
} else {
|
|
GST_DEBUG_OBJECT (mp3parse,
|
|
"pushing buffer of %d bytes, timestamp %" GST_TIME_FORMAT
|
|
", offset 0x%08" G_GINT64_MODIFIER "x", size,
|
|
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
|
|
GST_BUFFER_OFFSET (outbuf));
|
|
mp3parse->segment.last_stop = GST_BUFFER_TIMESTAMP (outbuf);
|
|
/* push any pending segment now */
|
|
if (mp3parse->pending_segment) {
|
|
gst_pad_push_event (mp3parse->srcpad, mp3parse->pending_segment);
|
|
mp3parse->pending_segment = NULL;
|
|
}
|
|
if (mp3parse->pending_events) {
|
|
GList *l;
|
|
|
|
for (l = mp3parse->pending_events; l != NULL; l = l->next) {
|
|
gst_pad_push_event (mp3parse->srcpad, GST_EVENT (l->data));
|
|
}
|
|
g_list_free (mp3parse->pending_events);
|
|
mp3parse->pending_events = NULL;
|
|
}
|
|
ret = gst_pad_push (mp3parse->srcpad, outbuf);
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static void
|
|
gst_mp3parse_handle_first_frame (GstMPEGAudioParse * mp3parse)
|
|
{
|
|
GstTagList *taglist;
|
|
|
|
gchar *codec;
|
|
|
|
const guint32 xing_id = 0x58696e67; /* 'Xing' in hex */
|
|
|
|
const guint32 info_id = 0x496e666f; /* 'Info' in hex - found in LAME CBR files */
|
|
|
|
const guint32 vbri_id = 0x56425249; /* 'VBRI' in hex */
|
|
|
|
gint offset;
|
|
|
|
guint64 avail;
|
|
|
|
guint32 read_id;
|
|
|
|
const guint8 *data;
|
|
|
|
/* Output codec tag */
|
|
if (!mp3parse->sent_codec_tag) {
|
|
if (mp3parse->layer == 3) {
|
|
codec = g_strdup_printf ("MPEG %d Audio, Layer %d (MP3)",
|
|
mp3parse->version, mp3parse->layer);
|
|
} else {
|
|
codec = g_strdup_printf ("MPEG %d Audio, Layer %d",
|
|
mp3parse->version, mp3parse->layer);
|
|
}
|
|
|
|
taglist = gst_tag_list_new ();
|
|
gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE,
|
|
GST_TAG_AUDIO_CODEC, codec, NULL);
|
|
gst_element_found_tags_for_pad (GST_ELEMENT (mp3parse),
|
|
mp3parse->srcpad, taglist);
|
|
g_free (codec);
|
|
|
|
mp3parse->sent_codec_tag = TRUE;
|
|
}
|
|
/* end setting the tag */
|
|
|
|
/* Check first frame for Xing info */
|
|
if (mp3parse->version == 1) { /* MPEG-1 file */
|
|
if (mp3parse->channels == 1)
|
|
offset = 0x11;
|
|
else
|
|
offset = 0x20;
|
|
} else { /* MPEG-2 header */
|
|
if (mp3parse->channels == 1)
|
|
offset = 0x09;
|
|
else
|
|
offset = 0x11;
|
|
}
|
|
/* Skip the 4 bytes of the MP3 header too */
|
|
offset += 4;
|
|
|
|
/* Check if we have enough data to read the Xing header */
|
|
avail = gst_adapter_available (mp3parse->adapter);
|
|
|
|
if (avail < offset + 8)
|
|
return;
|
|
|
|
data = gst_adapter_peek (mp3parse->adapter, offset + 8);
|
|
if (data == NULL)
|
|
return;
|
|
/* The header starts at the provided offset */
|
|
data += offset;
|
|
|
|
read_id = GST_READ_UINT32_BE (data);
|
|
if (read_id == xing_id || read_id == info_id) {
|
|
guint32 xing_flags;
|
|
|
|
guint bytes_needed = offset + 8;
|
|
|
|
gint64 total_bytes;
|
|
|
|
GstClockTime total_time;
|
|
|
|
GST_DEBUG_OBJECT (mp3parse, "Found Xing header marker 0x%x", xing_id);
|
|
|
|
/* Read 4 base bytes of flags, big-endian */
|
|
xing_flags = GST_READ_UINT32_BE (data + 4);
|
|
if (xing_flags & XING_FRAMES_FLAG)
|
|
bytes_needed += 4;
|
|
if (xing_flags & XING_BYTES_FLAG)
|
|
bytes_needed += 4;
|
|
if (xing_flags & XING_TOC_FLAG)
|
|
bytes_needed += 100;
|
|
if (xing_flags & XING_VBR_SCALE_FLAG)
|
|
bytes_needed += 4;
|
|
if (avail < bytes_needed) {
|
|
GST_DEBUG_OBJECT (mp3parse,
|
|
"Not enough data to read Xing header (need %d)", bytes_needed);
|
|
return;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (mp3parse, "Reading Xing header");
|
|
mp3parse->xing_flags = xing_flags;
|
|
data = gst_adapter_peek (mp3parse->adapter, bytes_needed);
|
|
data += offset + 8;
|
|
|
|
if (xing_flags & XING_FRAMES_FLAG) {
|
|
mp3parse->xing_frames = GST_READ_UINT32_BE (data);
|
|
if (mp3parse->xing_frames == 0) {
|
|
GST_WARNING_OBJECT (mp3parse,
|
|
"Invalid number of frames in Xing header");
|
|
mp3parse->xing_flags &= ~XING_FRAMES_FLAG;
|
|
} else {
|
|
mp3parse->xing_total_time = gst_util_uint64_scale (GST_SECOND,
|
|
(guint64) (mp3parse->xing_frames) * (mp3parse->spf),
|
|
mp3parse->rate);
|
|
}
|
|
|
|
data += 4;
|
|
} else {
|
|
mp3parse->xing_frames = 0;
|
|
mp3parse->xing_total_time = 0;
|
|
}
|
|
|
|
if (xing_flags & XING_BYTES_FLAG) {
|
|
mp3parse->xing_bytes = GST_READ_UINT32_BE (data);
|
|
if (mp3parse->xing_bytes == 0) {
|
|
GST_WARNING_OBJECT (mp3parse, "Invalid number of bytes in Xing header");
|
|
mp3parse->xing_flags &= ~XING_BYTES_FLAG;
|
|
}
|
|
|
|
data += 4;
|
|
} else {
|
|
mp3parse->xing_bytes = 0;
|
|
}
|
|
|
|
/* If we know the upstream size and duration, compute the
|
|
* total bitrate, rounded up to the nearest kbit/sec */
|
|
if (mp3parse_total_time (mp3parse, &total_time) &&
|
|
mp3parse_total_bytes (mp3parse, &total_bytes)) {
|
|
mp3parse->xing_bitrate = gst_util_uint64_scale (total_bytes,
|
|
8 * GST_SECOND, total_time);
|
|
mp3parse->xing_bitrate += 500;
|
|
mp3parse->xing_bitrate -= mp3parse->xing_bitrate % 1000;
|
|
}
|
|
|
|
if (xing_flags & XING_TOC_FLAG) {
|
|
int i, percent = 0;
|
|
|
|
guchar *table = mp3parse->xing_seek_table;
|
|
|
|
guchar old = 0;
|
|
|
|
if (data[0] != 0) {
|
|
GST_WARNING_OBJECT (mp3parse, "Skipping broken Xing TOC");
|
|
mp3parse->xing_flags &= ~XING_TOC_FLAG;
|
|
goto skip_toc;
|
|
}
|
|
|
|
/* xing seek table: percent time -> 1/256 bytepos */
|
|
for (i = 0; i < 100; i++) {
|
|
mp3parse->xing_seek_table[i] = data[i];
|
|
if (old > data[i]) {
|
|
GST_WARNING_OBJECT (mp3parse, "Skipping broken Xing TOC");
|
|
mp3parse->xing_flags &= ~XING_TOC_FLAG;
|
|
goto skip_toc;
|
|
}
|
|
}
|
|
|
|
/* build inverse table: 1/256 bytepos -> 1/100 percent time */
|
|
for (i = 0; i < 256; i++) {
|
|
while (percent < 99 && table[percent + 1] <= i)
|
|
percent++;
|
|
|
|
if (table[percent] == i) {
|
|
mp3parse->xing_seek_table_inverse[i] = percent * 100;
|
|
} else if (table[percent] < i && percent < 99) {
|
|
gdouble fa, fb, fx;
|
|
|
|
gint a = percent, b = percent + 1;
|
|
|
|
fa = table[a];
|
|
fb = table[b];
|
|
fx = (b - a) / (fb - fa) * (i - fa) + a;
|
|
mp3parse->xing_seek_table_inverse[i] = (guint16) (fx * 100);
|
|
} else if (percent == 98 && table[percent + 1] <= i) {
|
|
gdouble fa, fb, fx;
|
|
|
|
gint a = percent + 1, b = 100;
|
|
|
|
fa = table[a];
|
|
fb = 256.0;
|
|
fx = (b - a) / (fb - fa) * (i - fa) + a;
|
|
mp3parse->xing_seek_table_inverse[i] = (guint16) (fx * 100);
|
|
}
|
|
}
|
|
skip_toc:
|
|
data += 100;
|
|
} else {
|
|
memset (mp3parse->xing_seek_table, 0, 100);
|
|
memset (mp3parse->xing_seek_table_inverse, 0, 256);
|
|
}
|
|
|
|
if (xing_flags & XING_VBR_SCALE_FLAG) {
|
|
mp3parse->xing_vbr_scale = GST_READ_UINT32_BE (data);
|
|
data += 4;
|
|
} else
|
|
mp3parse->xing_vbr_scale = 0;
|
|
|
|
GST_DEBUG_OBJECT (mp3parse, "Xing header reported %u frames, time %"
|
|
GST_TIME_FORMAT ", %u bytes, vbr scale %u", mp3parse->xing_frames,
|
|
GST_TIME_ARGS (mp3parse->xing_total_time), mp3parse->xing_bytes,
|
|
mp3parse->xing_vbr_scale);
|
|
} else if (read_id == vbri_id) {
|
|
gint64 total_bytes, total_frames;
|
|
|
|
GstClockTime total_time;
|
|
|
|
guint16 nseek_points;
|
|
|
|
GST_DEBUG_OBJECT (mp3parse, "Found VBRI header marker 0x%x", vbri_id);
|
|
if (avail < offset + 26) {
|
|
GST_DEBUG_OBJECT (mp3parse,
|
|
"Not enough data to read VBRI header (need %d)", offset + 26);
|
|
return;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (mp3parse, "Reading VBRI header");
|
|
data = gst_adapter_peek (mp3parse->adapter, offset + 26);
|
|
data += offset + 4;
|
|
|
|
if (GST_READ_UINT16_BE (data) != 0x0001) {
|
|
GST_WARNING_OBJECT (mp3parse,
|
|
"Unsupported VBRI version 0x%x", GST_READ_UINT16_BE (data));
|
|
return;
|
|
}
|
|
data += 2;
|
|
|
|
/* Skip encoder delay */
|
|
data += 2;
|
|
|
|
/* Skip quality */
|
|
data += 2;
|
|
|
|
total_bytes = GST_READ_UINT32_BE (data);
|
|
if (total_bytes != 0)
|
|
mp3parse->vbri_bytes = total_bytes;
|
|
data += 4;
|
|
|
|
total_frames = GST_READ_UINT32_BE (data);
|
|
if (total_frames != 0) {
|
|
mp3parse->vbri_frames = total_frames;
|
|
mp3parse->vbri_total_time = gst_util_uint64_scale (GST_SECOND,
|
|
(guint64) (mp3parse->vbri_frames) * (mp3parse->spf), mp3parse->rate);
|
|
}
|
|
data += 4;
|
|
|
|
/* If we know the upstream size and duration, compute the
|
|
* total bitrate, rounded up to the nearest kbit/sec */
|
|
if (mp3parse_total_time (mp3parse, &total_time) &&
|
|
mp3parse_total_bytes (mp3parse, &total_bytes)) {
|
|
mp3parse->vbri_bitrate = gst_util_uint64_scale (total_bytes,
|
|
8 * GST_SECOND, total_time);
|
|
mp3parse->vbri_bitrate += 500;
|
|
mp3parse->vbri_bitrate -= mp3parse->vbri_bitrate % 1000;
|
|
}
|
|
|
|
nseek_points = GST_READ_UINT16_BE (data);
|
|
data += 2;
|
|
|
|
if (nseek_points > 0) {
|
|
guint scale, seek_bytes, seek_frames;
|
|
|
|
gint i;
|
|
|
|
mp3parse->vbri_seek_points = nseek_points;
|
|
|
|
scale = GST_READ_UINT16_BE (data);
|
|
data += 2;
|
|
|
|
seek_bytes = GST_READ_UINT16_BE (data);
|
|
data += 2;
|
|
|
|
seek_frames = GST_READ_UINT16_BE (data);
|
|
data += 2;
|
|
|
|
if (scale == 0 || seek_bytes == 0 || seek_bytes > 4 || seek_frames == 0) {
|
|
GST_WARNING_OBJECT (mp3parse, "Unsupported VBRI seek table");
|
|
goto out_vbri;
|
|
}
|
|
|
|
if (avail < offset + 26 + nseek_points * seek_bytes) {
|
|
GST_WARNING_OBJECT (mp3parse,
|
|
"Not enough data to read VBRI seek table (need %d)",
|
|
offset + 26 + nseek_points * seek_bytes);
|
|
goto out_vbri;
|
|
}
|
|
|
|
if (seek_frames * nseek_points < total_frames - seek_frames ||
|
|
seek_frames * nseek_points > total_frames + seek_frames) {
|
|
GST_WARNING_OBJECT (mp3parse,
|
|
"VBRI seek table doesn't cover the complete file");
|
|
goto out_vbri;
|
|
}
|
|
|
|
data =
|
|
gst_adapter_peek (mp3parse->adapter,
|
|
offset + 26 + nseek_points * seek_bytes);
|
|
data += offset + 26;
|
|
|
|
|
|
/* VBRI seek table: frame/seek_frames -> byte */
|
|
mp3parse->vbri_seek_table = g_new (guint32, nseek_points);
|
|
if (seek_bytes == 4)
|
|
for (i = 0; i < nseek_points; i++) {
|
|
mp3parse->vbri_seek_table[i] = GST_READ_UINT32_BE (data) * scale;
|
|
data += 4;
|
|
} else if (seek_bytes == 3)
|
|
for (i = 0; i < nseek_points; i++) {
|
|
mp3parse->vbri_seek_table[i] = GST_READ_UINT24_BE (data) * scale;
|
|
data += 3;
|
|
} else if (seek_bytes == 2)
|
|
for (i = 0; i < nseek_points; i++) {
|
|
mp3parse->vbri_seek_table[i] = GST_READ_UINT16_BE (data) * scale;
|
|
data += 2;
|
|
} else /* seek_bytes == 1 */
|
|
for (i = 0; i < nseek_points; i++) {
|
|
mp3parse->vbri_seek_table[i] = GST_READ_UINT8 (data) * scale;
|
|
data += 1;
|
|
}
|
|
}
|
|
out_vbri:
|
|
|
|
GST_DEBUG_OBJECT (mp3parse, "VBRI header reported %u frames, time %"
|
|
GST_TIME_FORMAT ", bytes %u", mp3parse->vbri_frames,
|
|
GST_TIME_ARGS (mp3parse->vbri_total_time), mp3parse->vbri_bytes);
|
|
} else {
|
|
GST_DEBUG_OBJECT (mp3parse,
|
|
"Xing, LAME or VBRI header not found in first frame");
|
|
}
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_mp3parse_chain (GstPad * pad, GstBuffer * buf)
|
|
{
|
|
GstFlowReturn flow = GST_FLOW_OK;
|
|
|
|
GstMPEGAudioParse *mp3parse;
|
|
|
|
const guchar *data;
|
|
|
|
guint32 header;
|
|
|
|
int bpf;
|
|
|
|
guint available;
|
|
|
|
GstClockTime timestamp;
|
|
|
|
mp3parse = GST_MP3PARSE (GST_PAD_PARENT (pad));
|
|
|
|
GST_LOG_OBJECT (mp3parse, "buffer of %d bytes", GST_BUFFER_SIZE (buf));
|
|
|
|
timestamp = GST_BUFFER_TIMESTAMP (buf);
|
|
|
|
/* If we don't yet have a next timestamp, save it and the incoming offset
|
|
* so we can apply it to the right outgoing buffer */
|
|
if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
|
|
gint64 avail = gst_adapter_available (mp3parse->adapter);
|
|
|
|
mp3parse->pending_ts = timestamp;
|
|
mp3parse->pending_offset = mp3parse->tracked_offset + avail;
|
|
|
|
/* If we have no data pending and the next timestamp is
|
|
* invalid we can use the upstream timestamp for the next frame.
|
|
*
|
|
* This will give us a timestamp if we're resyncing and upstream
|
|
* gave us -1 as offset. */
|
|
if (avail == 0 && !GST_CLOCK_TIME_IS_VALID (mp3parse->next_ts))
|
|
mp3parse->next_ts = timestamp;
|
|
|
|
GST_LOG_OBJECT (mp3parse, "Have pending ts %" GST_TIME_FORMAT
|
|
" to apply in %lld bytes (@ off %lld)",
|
|
GST_TIME_ARGS (mp3parse->pending_ts), avail, mp3parse->pending_offset);
|
|
}
|
|
|
|
/* Update the cur_offset we'll apply to outgoing buffers */
|
|
if (mp3parse->cur_offset == -1 && GST_BUFFER_OFFSET (buf) != -1)
|
|
mp3parse->cur_offset = GST_BUFFER_OFFSET (buf);
|
|
|
|
/* And add the data to the pool */
|
|
gst_adapter_push (mp3parse->adapter, buf);
|
|
|
|
/* while we still have at least 4 bytes (for the header) available */
|
|
while (gst_adapter_available (mp3parse->adapter) >= 4) {
|
|
/* search for a possible start byte */
|
|
data = gst_adapter_peek (mp3parse->adapter, 4);
|
|
if (*data != 0xff) {
|
|
/* It'd be nice to make this efficient, but it's ok for now; this is only
|
|
* when resyncing */
|
|
mp3parse->resyncing = TRUE;
|
|
gst_adapter_flush (mp3parse->adapter, 1);
|
|
if (mp3parse->cur_offset != -1)
|
|
mp3parse->cur_offset++;
|
|
mp3parse->tracked_offset++;
|
|
continue;
|
|
}
|
|
|
|
available = gst_adapter_available (mp3parse->adapter);
|
|
|
|
/* construct the header word */
|
|
header = GST_READ_UINT32_BE (data);
|
|
/* if it's a valid header, go ahead and send off the frame */
|
|
if (head_check (mp3parse, header)) {
|
|
guint bitrate = 0, layer = 0, rate = 0, channels = 0, version = 0, mode =
|
|
0, crc = 0;
|
|
|
|
if (!(bpf = mp3_type_frame_length_from_header (mp3parse, header,
|
|
&version, &layer, &channels, &bitrate, &rate, &mode, &crc)))
|
|
goto header_error;
|
|
|
|
/*************************************************************************
|
|
* robust seek support
|
|
* - This performs additional frame validation if the resyncing 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 borrowing
|
|
* bitrate from previous frames. In this case, seeking may be more
|
|
* complicated because the frames are not independently coded.
|
|
*************************************************************************/
|
|
if (mp3parse->resyncing) {
|
|
guint32 header2;
|
|
|
|
const guint8 *data2;
|
|
|
|
/* wait until we have the the entire current frame as well as the next
|
|
* frame header */
|
|
if (available < bpf + 4)
|
|
break;
|
|
|
|
data2 = gst_adapter_peek (mp3parse->adapter, bpf + 4);
|
|
header2 = GST_READ_UINT32_BE (data2 + bpf);
|
|
GST_DEBUG_OBJECT (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 */
|
|
|
|
/* require 2 matching headers in a row */
|
|
if ((header2 & HDRMASK) != (header & HDRMASK)) {
|
|
GST_DEBUG_OBJECT (mp3parse, "next header doesn't match "
|
|
"(header=%08X, header2=%08X, bpf=%d)",
|
|
(unsigned int) header, (unsigned int) header2, bpf);
|
|
/* This frame is invalid. Start looking for a valid frame at the
|
|
* next position in the stream */
|
|
mp3parse->resyncing = TRUE;
|
|
gst_adapter_flush (mp3parse->adapter, 1);
|
|
if (mp3parse->cur_offset != -1)
|
|
mp3parse->cur_offset++;
|
|
mp3parse->tracked_offset++;
|
|
continue;
|
|
}
|
|
}
|
|
|
|
/* if we don't have the whole frame... */
|
|
if (available < bpf) {
|
|
GST_DEBUG_OBJECT (mp3parse, "insufficient data available, need "
|
|
"%d bytes, have %d", bpf, available);
|
|
break;
|
|
}
|
|
|
|
if (channels != mp3parse->channels ||
|
|
rate != mp3parse->rate || layer != mp3parse->layer ||
|
|
version != mp3parse->version) {
|
|
GstCaps *caps;
|
|
|
|
caps = mp3_caps_create (version, layer, channels, rate);
|
|
gst_pad_set_caps (mp3parse->srcpad, caps);
|
|
gst_caps_unref (caps);
|
|
|
|
mp3parse->channels = channels;
|
|
mp3parse->rate = rate;
|
|
}
|
|
|
|
if (layer != mp3parse->layer || version != mp3parse->version) {
|
|
mp3parse->layer = layer;
|
|
mp3parse->version = version;
|
|
|
|
/* see http://www.codeproject.com/audio/MPEGAudioInfo.asp */
|
|
if (mp3parse->layer == 1)
|
|
mp3parse->spf = 384;
|
|
else if (mp3parse->layer == 2)
|
|
mp3parse->spf = 1152;
|
|
else if (mp3parse->version == 2) {
|
|
mp3parse->spf = 576;
|
|
} else
|
|
mp3parse->spf = 1152;
|
|
}
|
|
|
|
mp3parse->bit_rate = bitrate;
|
|
|
|
mp3parse->max_bitreservoir = gst_util_uint64_scale (GST_SECOND,
|
|
((version == 1) ? 10 : 30) * mp3parse->spf, mp3parse->rate);
|
|
|
|
/* Check the first frame for a Xing header to get our total length */
|
|
if (mp3parse->frame_count == 0) {
|
|
/* For the first frame in the file, look for a Xing frame after
|
|
* the header, and output a codec tag */
|
|
gst_mp3parse_handle_first_frame (mp3parse);
|
|
}
|
|
|
|
/* Update VBR stats */
|
|
mp3parse->bitrate_sum += mp3parse->bit_rate;
|
|
mp3parse->frame_count++;
|
|
/* Compute the average bitrate, rounded up to the nearest 1000 bits */
|
|
mp3parse->avg_bitrate =
|
|
(mp3parse->bitrate_sum / mp3parse->frame_count + 500);
|
|
mp3parse->avg_bitrate -= mp3parse->avg_bitrate % 1000;
|
|
|
|
if (!mp3parse->skip) {
|
|
mp3parse->resyncing = FALSE;
|
|
flow = gst_mp3parse_emit_frame (mp3parse, bpf, mode, crc);
|
|
} else {
|
|
GST_DEBUG_OBJECT (mp3parse, "skipping buffer of %d bytes", bpf);
|
|
gst_adapter_flush (mp3parse->adapter, bpf);
|
|
if (mp3parse->cur_offset != -1)
|
|
mp3parse->cur_offset += bpf;
|
|
mp3parse->tracked_offset += bpf;
|
|
mp3parse->skip--;
|
|
}
|
|
} else {
|
|
mp3parse->resyncing = TRUE;
|
|
gst_adapter_flush (mp3parse->adapter, 1);
|
|
if (mp3parse->cur_offset != -1)
|
|
mp3parse->cur_offset++;
|
|
mp3parse->tracked_offset++;
|
|
GST_DEBUG_OBJECT (mp3parse, "wrong header, skipping byte");
|
|
}
|
|
|
|
if (GST_FLOW_IS_FATAL (flow))
|
|
break;
|
|
}
|
|
|
|
return flow;
|
|
|
|
header_error:
|
|
GST_ELEMENT_ERROR (mp3parse, STREAM, DECODE,
|
|
("Invalid MP3 header found"), (NULL));
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
|
|
static gboolean
|
|
head_check (GstMPEGAudioParse * mp3parse, unsigned long head)
|
|
{
|
|
GST_DEBUG_OBJECT (mp3parse, "checking mp3 header 0x%08lx", head);
|
|
/* if it's not a valid sync */
|
|
if ((head & 0xffe00000) != 0xffe00000) {
|
|
GST_WARNING_OBJECT (mp3parse, "invalid sync");
|
|
return FALSE;
|
|
}
|
|
/* if it's an invalid MPEG version */
|
|
if (((head >> 19) & 3) == 0x1) {
|
|
GST_WARNING_OBJECT (mp3parse, "invalid MPEG version: 0x%lx",
|
|
(head >> 19) & 3);
|
|
return FALSE;
|
|
}
|
|
/* if it's an invalid layer */
|
|
if (!((head >> 17) & 3)) {
|
|
GST_WARNING_OBJECT (mp3parse, "invalid layer: 0x%lx", (head >> 17) & 3);
|
|
return FALSE;
|
|
}
|
|
/* if it's an invalid bitrate */
|
|
if (((head >> 12) & 0xf) == 0x0) {
|
|
GST_WARNING_OBJECT (mp3parse, "invalid bitrate: 0x%lx."
|
|
"Free format files are not supported yet", (head >> 12) & 0xf);
|
|
return FALSE;
|
|
}
|
|
if (((head >> 12) & 0xf) == 0xf) {
|
|
GST_WARNING_OBJECT (mp3parse, "invalid bitrate: 0x%lx", (head >> 12) & 0xf);
|
|
return FALSE;
|
|
}
|
|
/* if it's an invalid samplerate */
|
|
if (((head >> 10) & 0x3) == 0x3) {
|
|
GST_WARNING_OBJECT (mp3parse, "invalid samplerate: 0x%lx",
|
|
(head >> 10) & 0x3);
|
|
return FALSE;
|
|
}
|
|
|
|
if ((head & 0x3) == 0x2) {
|
|
/* Ignore this as there are some files with emphasis 0x2 that can
|
|
* be played fine. See BGO #537235 */
|
|
GST_WARNING_OBJECT (mp3parse, "invalid emphasis: 0x%lx", head & 0x3);
|
|
}
|
|
|
|
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 *mp3parse;
|
|
|
|
GstStateChangeReturn result;
|
|
|
|
mp3parse = GST_MP3PARSE (element);
|
|
|
|
result = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
gst_mp3parse_reset (mp3parse);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return result;
|
|
}
|
|
|
|
static gboolean
|
|
mp3parse_total_bytes (GstMPEGAudioParse * mp3parse, gint64 * total)
|
|
{
|
|
GstFormat fmt = GST_FORMAT_BYTES;
|
|
|
|
if (gst_pad_query_peer_duration (mp3parse->sinkpad, &fmt, total))
|
|
return TRUE;
|
|
|
|
if (mp3parse->xing_flags & XING_BYTES_FLAG) {
|
|
*total = mp3parse->xing_bytes;
|
|
return TRUE;
|
|
}
|
|
|
|
if (mp3parse->vbri_bytes != 0) {
|
|
*total = mp3parse->vbri_bytes;
|
|
return TRUE;
|
|
}
|
|
|
|
return FALSE;
|
|
}
|
|
|
|
static gboolean
|
|
mp3parse_total_time (GstMPEGAudioParse * mp3parse, GstClockTime * total)
|
|
{
|
|
gint64 total_bytes;
|
|
|
|
*total = GST_CLOCK_TIME_NONE;
|
|
|
|
if (mp3parse->xing_flags & XING_FRAMES_FLAG) {
|
|
*total = mp3parse->xing_total_time;
|
|
return TRUE;
|
|
}
|
|
|
|
if (mp3parse->vbri_total_time != 0) {
|
|
*total = mp3parse->vbri_total_time;
|
|
return TRUE;
|
|
}
|
|
|
|
/* Calculate time from the measured bitrate */
|
|
if (!mp3parse_total_bytes (mp3parse, &total_bytes))
|
|
return FALSE;
|
|
|
|
if (total_bytes != -1
|
|
&& !mp3parse_bytepos_to_time (mp3parse, total_bytes, total, TRUE))
|
|
return FALSE;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/* Convert a timestamp to the file position required to start decoding that
|
|
* timestamp. For now, this just uses the avg bitrate. Later, use an
|
|
* incrementally accumulated seek table */
|
|
static gboolean
|
|
mp3parse_time_to_bytepos (GstMPEGAudioParse * mp3parse, GstClockTime ts,
|
|
gint64 * bytepos)
|
|
{
|
|
gint64 total_bytes;
|
|
|
|
GstClockTime total_time;
|
|
|
|
/* -1 always maps to -1 */
|
|
if (ts == -1) {
|
|
*bytepos = -1;
|
|
return TRUE;
|
|
}
|
|
|
|
/* If XING seek table exists use this for time->byte conversion */
|
|
if ((mp3parse->xing_flags & XING_TOC_FLAG) &&
|
|
mp3parse_total_bytes (mp3parse, &total_bytes) &&
|
|
mp3parse_total_time (mp3parse, &total_time)) {
|
|
gdouble fa, fb, fx;
|
|
|
|
gdouble percent =
|
|
CLAMP ((100.0 * gst_util_guint64_to_gdouble (ts)) /
|
|
gst_util_guint64_to_gdouble (total_time), 0.0, 100.0);
|
|
gint index = CLAMP (percent, 0, 99);
|
|
|
|
fa = mp3parse->xing_seek_table[index];
|
|
if (index < 99)
|
|
fb = mp3parse->xing_seek_table[index + 1];
|
|
else
|
|
fb = 256.0;
|
|
|
|
fx = fa + (fb - fa) * (percent - index);
|
|
|
|
*bytepos = (1.0 / 256.0) * fx * total_bytes;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
if (mp3parse->vbri_seek_table &&
|
|
mp3parse_total_bytes (mp3parse, &total_bytes) &&
|
|
mp3parse_total_time (mp3parse, &total_time)) {
|
|
gint i, j;
|
|
|
|
gdouble a, b, fa, fb;
|
|
|
|
i = gst_util_uint64_scale (ts, mp3parse->vbri_seek_points - 1, total_time);
|
|
i = CLAMP (i, 0, mp3parse->vbri_seek_points - 1);
|
|
|
|
a = gst_guint64_to_gdouble (gst_util_uint64_scale (i, total_time,
|
|
mp3parse->vbri_seek_points));
|
|
fa = 0.0;
|
|
for (j = i; j >= 0; j--)
|
|
fa += mp3parse->vbri_seek_table[j];
|
|
|
|
if (i + 1 < mp3parse->vbri_seek_points) {
|
|
b = gst_guint64_to_gdouble (gst_util_uint64_scale (i + 1, total_time,
|
|
mp3parse->vbri_seek_points));
|
|
fb = fa + mp3parse->vbri_seek_table[i + 1];
|
|
} else {
|
|
b = gst_guint64_to_gdouble (total_time);
|
|
fb = total_bytes;
|
|
}
|
|
|
|
*bytepos = fa + ((fb - fa) / (b - a)) * (gst_guint64_to_gdouble (ts) - a);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
if (mp3parse->avg_bitrate == 0)
|
|
goto no_bitrate;
|
|
|
|
*bytepos =
|
|
gst_util_uint64_scale (ts, mp3parse->avg_bitrate, (8 * GST_SECOND));
|
|
return TRUE;
|
|
no_bitrate:
|
|
GST_DEBUG_OBJECT (mp3parse, "Cannot seek yet - no average bitrate");
|
|
return FALSE;
|
|
}
|
|
|
|
static gboolean
|
|
mp3parse_bytepos_to_time (GstMPEGAudioParse * mp3parse,
|
|
gint64 bytepos, GstClockTime * ts, gboolean from_total_time)
|
|
{
|
|
gint64 total_bytes;
|
|
|
|
GstClockTime total_time;
|
|
|
|
if (bytepos == -1) {
|
|
*ts = GST_CLOCK_TIME_NONE;
|
|
return TRUE;
|
|
}
|
|
|
|
if (bytepos == 0) {
|
|
*ts = 0;
|
|
return TRUE;
|
|
}
|
|
|
|
/* If XING seek table exists use this for byte->time conversion */
|
|
if (!from_total_time && (mp3parse->xing_flags & XING_TOC_FLAG) &&
|
|
mp3parse_total_bytes (mp3parse, &total_bytes) &&
|
|
mp3parse_total_time (mp3parse, &total_time)) {
|
|
gdouble fa, fb, fx;
|
|
|
|
gdouble pos = CLAMP ((bytepos * 256.0) / total_bytes, 0.0, 256.0);
|
|
|
|
gint index = CLAMP (pos, 0, 255);
|
|
|
|
fa = mp3parse->xing_seek_table_inverse[index];
|
|
if (index < 255)
|
|
fb = mp3parse->xing_seek_table_inverse[index + 1];
|
|
else
|
|
fb = 10000.0;
|
|
|
|
fx = fa + (fb - fa) * (pos - index);
|
|
|
|
*ts = (1.0 / 10000.0) * fx * gst_util_guint64_to_gdouble (total_time);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
if (!from_total_time && mp3parse->vbri_seek_table &&
|
|
mp3parse_total_bytes (mp3parse, &total_bytes) &&
|
|
mp3parse_total_time (mp3parse, &total_time)) {
|
|
gint i = 0;
|
|
|
|
guint64 sum = 0;
|
|
|
|
gdouble a, b, fa, fb;
|
|
|
|
|
|
do {
|
|
sum += mp3parse->vbri_seek_table[i];
|
|
i++;
|
|
} while (i + 1 < mp3parse->vbri_seek_points
|
|
&& sum + mp3parse->vbri_seek_table[i] < bytepos);
|
|
i--;
|
|
|
|
a = gst_guint64_to_gdouble (sum);
|
|
fa = gst_guint64_to_gdouble (gst_util_uint64_scale (i, total_time,
|
|
mp3parse->vbri_seek_points));
|
|
|
|
if (i + 1 < mp3parse->vbri_seek_points) {
|
|
b = a + mp3parse->vbri_seek_table[i + 1];
|
|
fb = gst_guint64_to_gdouble (gst_util_uint64_scale (i + 1, total_time,
|
|
mp3parse->vbri_seek_points));
|
|
} else {
|
|
b = total_bytes;
|
|
fb = gst_guint64_to_gdouble (total_time);
|
|
}
|
|
|
|
*ts = gst_gdouble_to_guint64 (fa + ((fb - fa) / (b - a)) * (bytepos - a));
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/* Cannot convert anything except 0 if we don't have a bitrate yet */
|
|
if (mp3parse->avg_bitrate == 0)
|
|
return FALSE;
|
|
|
|
*ts = (GstClockTime) gst_util_uint64_scale (GST_SECOND, bytepos * 8,
|
|
mp3parse->avg_bitrate);
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
mp3parse_handle_seek (GstMPEGAudioParse * mp3parse, GstEvent * event)
|
|
{
|
|
GstFormat format;
|
|
|
|
gdouble rate;
|
|
|
|
GstSeekFlags flags;
|
|
|
|
GstSeekType cur_type, stop_type;
|
|
|
|
gint64 cur, stop;
|
|
|
|
gint64 byte_cur, byte_stop;
|
|
|
|
gst_event_parse_seek (event, &rate, &format, &flags, &cur_type, &cur,
|
|
&stop_type, &stop);
|
|
|
|
GST_DEBUG_OBJECT (mp3parse, "Performing seek to %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (cur));
|
|
|
|
/* For any format other than TIME, see if upstream handles
|
|
* it directly or fail. For TIME, try upstream, but do it ourselves if
|
|
* it fails upstream */
|
|
if (format != GST_FORMAT_TIME) {
|
|
gst_event_ref (event);
|
|
return gst_pad_push_event (mp3parse->sinkpad, event);
|
|
} else {
|
|
gst_event_ref (event);
|
|
if (gst_pad_push_event (mp3parse->sinkpad, event))
|
|
return TRUE;
|
|
}
|
|
|
|
/* Handle TIME based seeks by converting to a BYTE position */
|
|
|
|
/* For accurate seeking get the frame 9 (MPEG1) or 29 (MPEG2) frames
|
|
* before the one we want to seek to and push them all to the decoder.
|
|
*
|
|
* This is necessary because of the bit reservoir. See
|
|
* http://www.mars.org/mailman/public/mad-dev/2002-May/000634.html
|
|
*
|
|
*/
|
|
|
|
if (flags & GST_SEEK_FLAG_ACCURATE) {
|
|
MPEGAudioPendingAccurateSeek *seek =
|
|
g_new0 (MPEGAudioPendingAccurateSeek, 1);
|
|
GstClockTime start;
|
|
|
|
seek->segment = mp3parse->segment;
|
|
|
|
gst_segment_set_seek (&seek->segment, rate, GST_FORMAT_TIME,
|
|
flags, cur_type, cur, stop_type, stop, NULL);
|
|
|
|
if (!mp3parse->seek_table) {
|
|
byte_cur = 0;
|
|
byte_stop = -1;
|
|
start = 0;
|
|
} else {
|
|
MPEGAudioSeekEntry *entry = NULL, *start_entry = NULL, *stop_entry = NULL;
|
|
|
|
GList *start_node, *stop_node;
|
|
|
|
for (start_node = mp3parse->seek_table; start_node;
|
|
start_node = start_node->next) {
|
|
entry = start_node->data;
|
|
|
|
if (cur - mp3parse->max_bitreservoir >= entry->timestamp) {
|
|
start_entry = entry;
|
|
break;
|
|
}
|
|
}
|
|
|
|
if (!start_entry) {
|
|
start_entry = mp3parse->seek_table->data;
|
|
start = start_entry->timestamp;
|
|
byte_cur = start_entry->byte;
|
|
} else {
|
|
start = start_entry->timestamp;
|
|
byte_cur = start_entry->byte;
|
|
}
|
|
|
|
for (stop_node = mp3parse->seek_table; stop_node;
|
|
stop_node = stop_node->next) {
|
|
entry = stop_node->data;
|
|
|
|
if (stop >= entry->timestamp) {
|
|
stop_node = stop_node->prev;
|
|
stop_entry = (stop_node) ? stop_node->data : NULL;
|
|
break;
|
|
}
|
|
}
|
|
|
|
if (!stop_entry) {
|
|
byte_stop = -1;
|
|
} else {
|
|
byte_stop = stop_entry->byte;
|
|
}
|
|
|
|
}
|
|
g_mutex_lock (mp3parse->pending_accurate_seeks_lock);
|
|
event = gst_event_new_seek (rate, GST_FORMAT_BYTES, flags, cur_type,
|
|
byte_cur, stop_type, byte_stop);
|
|
if (gst_pad_push_event (mp3parse->sinkpad, event)) {
|
|
mp3parse->exact_position = TRUE;
|
|
seek->upstream_start = byte_cur;
|
|
seek->timestamp_start = start;
|
|
mp3parse->pending_accurate_seeks =
|
|
g_slist_prepend (mp3parse->pending_accurate_seeks, seek);
|
|
g_mutex_unlock (mp3parse->pending_accurate_seeks_lock);
|
|
return TRUE;
|
|
} else {
|
|
g_mutex_unlock (mp3parse->pending_accurate_seeks_lock);
|
|
mp3parse->exact_position = TRUE;
|
|
g_free (seek);
|
|
return TRUE;
|
|
}
|
|
}
|
|
|
|
mp3parse->exact_position = FALSE;
|
|
|
|
/* Convert the TIME to the appropriate BYTE position at which to resume
|
|
* decoding. */
|
|
if (!mp3parse_time_to_bytepos (mp3parse, (GstClockTime) cur, &byte_cur))
|
|
goto no_pos;
|
|
if (!mp3parse_time_to_bytepos (mp3parse, (GstClockTime) stop, &byte_stop))
|
|
goto no_pos;
|
|
|
|
GST_DEBUG_OBJECT (mp3parse, "Seeking to byte range %" G_GINT64_FORMAT
|
|
" to %" G_GINT64_FORMAT, byte_cur, byte_stop);
|
|
|
|
/* Send BYTE based seek upstream */
|
|
event = gst_event_new_seek (rate, GST_FORMAT_BYTES, flags, cur_type,
|
|
byte_cur, stop_type, byte_stop);
|
|
|
|
return gst_pad_push_event (mp3parse->sinkpad, event);
|
|
no_pos:
|
|
GST_DEBUG_OBJECT (mp3parse,
|
|
"Could not determine byte position for desired time");
|
|
return FALSE;
|
|
}
|
|
|
|
static gboolean
|
|
mp3parse_src_event (GstPad * pad, GstEvent * event)
|
|
{
|
|
GstMPEGAudioParse *mp3parse = GST_MP3PARSE (gst_pad_get_parent (pad));
|
|
|
|
gboolean res = FALSE;
|
|
|
|
g_return_val_if_fail (mp3parse != NULL, FALSE);
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_SEEK:
|
|
res = mp3parse_handle_seek (mp3parse, event);
|
|
gst_event_unref (event);
|
|
break;
|
|
default:
|
|
res = gst_pad_event_default (pad, event);
|
|
break;
|
|
}
|
|
|
|
gst_object_unref (mp3parse);
|
|
return res;
|
|
}
|
|
|
|
static gboolean
|
|
mp3parse_src_query (GstPad * pad, GstQuery * query)
|
|
{
|
|
GstFormat format;
|
|
|
|
GstClockTime total;
|
|
|
|
GstMPEGAudioParse *mp3parse = GST_MP3PARSE (gst_pad_get_parent (pad));
|
|
|
|
gboolean res = FALSE;
|
|
|
|
GstPad *peer;
|
|
|
|
g_return_val_if_fail (mp3parse != NULL, FALSE);
|
|
|
|
switch (GST_QUERY_TYPE (query)) {
|
|
case GST_QUERY_POSITION:
|
|
gst_query_parse_position (query, &format, NULL);
|
|
|
|
if (format == GST_FORMAT_BYTES || format == GST_FORMAT_DEFAULT) {
|
|
if (mp3parse->cur_offset != -1) {
|
|
gst_query_set_position (query, GST_FORMAT_BYTES,
|
|
mp3parse->cur_offset);
|
|
res = TRUE;
|
|
}
|
|
} else if (format == GST_FORMAT_TIME) {
|
|
if (mp3parse->next_ts == GST_CLOCK_TIME_NONE)
|
|
goto out;
|
|
gst_query_set_position (query, GST_FORMAT_TIME, mp3parse->next_ts);
|
|
res = TRUE;
|
|
}
|
|
|
|
/* If no answer above, see if upstream knows */
|
|
if (!res) {
|
|
if ((peer = gst_pad_get_peer (mp3parse->sinkpad)) != NULL) {
|
|
res = gst_pad_query (peer, query);
|
|
gst_object_unref (peer);
|
|
if (res)
|
|
goto out;
|
|
}
|
|
}
|
|
break;
|
|
case GST_QUERY_DURATION:
|
|
gst_query_parse_duration (query, &format, NULL);
|
|
|
|
/* First, see if upstream knows */
|
|
if ((peer = gst_pad_get_peer (mp3parse->sinkpad)) != NULL) {
|
|
res = gst_pad_query (peer, query);
|
|
gst_object_unref (peer);
|
|
if (res)
|
|
goto out;
|
|
}
|
|
|
|
if (format == GST_FORMAT_TIME) {
|
|
if (!mp3parse_total_time (mp3parse, &total) || total == -1)
|
|
goto out;
|
|
gst_query_set_duration (query, format, total);
|
|
res = TRUE;
|
|
}
|
|
break;
|
|
default:
|
|
res = gst_pad_query_default (pad, query);
|
|
break;
|
|
}
|
|
|
|
out:
|
|
gst_object_unref (mp3parse);
|
|
return res;
|
|
}
|
|
|
|
static const GstQueryType *
|
|
mp3parse_get_query_types (GstPad * pad ATTR_UNUSED)
|
|
{
|
|
static const GstQueryType query_types[] = {
|
|
GST_QUERY_POSITION,
|
|
GST_QUERY_DURATION,
|
|
0
|
|
};
|
|
|
|
return query_types;
|
|
}
|