--- /dev/null
+/* GStreamer MPEG audio parser
+ * Copyright (C) 2006-2007 Jan Schmidt <thaytan@mad.scientist.com>
+ * Copyright (C) 2010 Mark Nauwelaerts <mnauw users sf net>
+ * Copyright (C) 2010 Nokia Corporation. All rights reserved.
+ * Contact: Stefan Kost <stefan.kost@nokia.com>
+ *
+ * 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.
+ */
+/**
+ * SECTION:element-mpegaudioparse
+ * @short_description: MPEG audio parser
+ * @see_also: #GstAmrParse, #GstAACParse
+ *
+ * <refsect2>
+ * <para>
+ * Parses and frames mpeg1 audio streams. Provides seeking.
+ * </para>
+ * <title>Example launch line</title>
+ * <para>
+ * <programlisting>
+ * gst-launch filesrc location=test.mp3 ! mp3parse ! mad ! autoaudiosink
+ * </programlisting>
+ * </para>
+ * </refsect2>
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <string.h>
+
+#include "gstmpegaudioparse.h"
+#include <gst/base/gstbytereader.h>
+
+GST_DEBUG_CATEGORY_STATIC (mpeg_audio_parse_debug);
+#define GST_CAT_DEFAULT mpeg_audio_parse_debug
+
+#define MPEG_AUDIO_CHANNEL_MODE_UNKNOWN -1
+#define MPEG_AUDIO_CHANNEL_MODE_STEREO 0
+#define MPEG_AUDIO_CHANNEL_MODE_JOINT_STEREO 1
+#define MPEG_AUDIO_CHANNEL_MODE_DUAL_CHANNEL 2
+#define MPEG_AUDIO_CHANNEL_MODE_MONO 3
+
+#define CRC_UNKNOWN -1
+#define CRC_PROTECTED 0
+#define CRC_NOT_PROTECTED 1
+
+#define XING_FRAMES_FLAG 0x0001
+#define XING_BYTES_FLAG 0x0002
+#define XING_TOC_FLAG 0x0004
+#define XING_VBR_SCALE_FLAG 0x0008
+
+static GstStaticPadTemplate 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 ],"
+ "parsed=(boolean) true")
+ );
+
+static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/mpeg, mpegversion = (int) 1, parsed=(boolean)false")
+ );
+
+static void gst_mpeg_audio_parse_finalize (GObject * object);
+
+static gboolean gst_mpeg_audio_parse_start (GstBaseParse * parse);
+static gboolean gst_mpeg_audio_parse_stop (GstBaseParse * parse);
+static gboolean gst_mpeg_audio_parse_check_valid_frame (GstBaseParse * parse,
+ GstBuffer * buffer, guint * size, gint * skipsize);
+static GstFlowReturn gst_mpeg_audio_parse_parse_frame (GstBaseParse * parse,
+ GstBuffer * buf);
+static GstFlowReturn gst_mpeg_audio_parse_pre_push_buffer (GstBaseParse * parse,
+ GstBuffer * buf);
+static gboolean gst_mpeg_audio_parse_convert (GstBaseParse * parse,
+ GstFormat src_format, gint64 src_value,
+ GstFormat dest_format, gint64 * dest_value);
+
+GST_BOILERPLATE (GstMpegAudioParse, gst_mpeg_audio_parse, GstBaseParse,
+ GST_TYPE_BASE_PARSE);
+
+#define GST_TYPE_MPEG_AUDIO_CHANNEL_MODE \
+ (gst_mpeg_audio_channel_mode_get_type())
+
+static const GEnumValue mpeg_audio_channel_mode[] = {
+ {MPEG_AUDIO_CHANNEL_MODE_UNKNOWN, "Unknown", "unknown"},
+ {MPEG_AUDIO_CHANNEL_MODE_MONO, "Mono", "mono"},
+ {MPEG_AUDIO_CHANNEL_MODE_DUAL_CHANNEL, "Dual Channel", "dual-channel"},
+ {MPEG_AUDIO_CHANNEL_MODE_JOINT_STEREO, "Joint Stereo", "joint-stereo"},
+ {MPEG_AUDIO_CHANNEL_MODE_STEREO, "Stereo", "stereo"},
+ {0, NULL, NULL},
+};
+
+static GType
+gst_mpeg_audio_channel_mode_get_type (void)
+{
+ static GType mpeg_audio_channel_mode_type = 0;
+
+ if (!mpeg_audio_channel_mode_type) {
+ mpeg_audio_channel_mode_type =
+ g_enum_register_static ("GstMpegAudioChannelMode",
+ mpeg_audio_channel_mode);
+ }
+ return mpeg_audio_channel_mode_type;
+}
+
+static const gchar *
+gst_mpeg_audio_channel_mode_get_nick (gint mode)
+{
+ guint i;
+ for (i = 0; i < G_N_ELEMENTS (mpeg_audio_channel_mode); i++) {
+ if (mpeg_audio_channel_mode[i].value == mode)
+ return mpeg_audio_channel_mode[i].value_nick;
+ }
+ return NULL;
+}
+
+static void
+gst_mpeg_audio_parse_base_init (gpointer klass)
+{
+ GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&sink_template));
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&src_template));
+
+ gst_element_class_set_details_simple (element_class, "MPEG1 Audio Parser",
+ "Codec/Parser/Audio",
+ "Parses and frames mpeg1 audio streams (levels 1-3), provides seek",
+ "Jan Schmidt <thaytan@mad.scientist.com>,"
+ "Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>");
+}
+
+static void
+gst_mpeg_audio_parse_class_init (GstMpegAudioParseClass * klass)
+{
+ GstBaseParseClass *parse_class = GST_BASE_PARSE_CLASS (klass);
+ GObjectClass *object_class = G_OBJECT_CLASS (klass);
+
+ GST_DEBUG_CATEGORY_INIT (mpeg_audio_parse_debug, "mpegaudioparse", 0,
+ "MPEG1 audio stream parser");
+
+ object_class->finalize = gst_mpeg_audio_parse_finalize;
+
+ parse_class->start = GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_start);
+ parse_class->stop = GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_stop);
+ parse_class->check_valid_frame =
+ GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_check_valid_frame);
+ parse_class->parse_frame =
+ GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_parse_frame);
+ parse_class->pre_push_buffer =
+ GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_pre_push_buffer);
+ parse_class->convert = GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_convert);
+
+ /* register tags */
+#define GST_TAG_CRC "has-crc"
+#define GST_TAG_MODE "channel-mode"
+
+ gst_tag_register (GST_TAG_CRC, GST_TAG_FLAG_META, G_TYPE_BOOLEAN,
+ "has crc", "Using CRC", NULL);
+ gst_tag_register (GST_TAG_MODE, GST_TAG_FLAG_ENCODED, G_TYPE_STRING,
+ "channel mode", "MPEG audio channel mode", NULL);
+
+ g_type_class_ref (GST_TYPE_MPEG_AUDIO_CHANNEL_MODE);
+}
+
+static void
+gst_mpeg_audio_parse_reset (GstMpegAudioParse * mp3parse)
+{
+ mp3parse->channels = -1;
+ mp3parse->rate = -1;
+ mp3parse->sent_codec_tag = FALSE;
+ mp3parse->last_posted_crc = CRC_UNKNOWN;
+ mp3parse->last_posted_channel_mode = MPEG_AUDIO_CHANNEL_MODE_UNKNOWN;
+
+ mp3parse->xing_flags = 0;
+ mp3parse->xing_bitrate = 0;
+ mp3parse->xing_frames = 0;
+ mp3parse->xing_total_time = 0;
+ mp3parse->xing_bytes = 0;
+ mp3parse->xing_vbr_scale = 0;
+ memset (mp3parse->xing_seek_table, 0, 100);
+ memset (mp3parse->xing_seek_table_inverse, 0, 256);
+
+ mp3parse->vbri_bitrate = 0;
+ mp3parse->vbri_frames = 0;
+ mp3parse->vbri_total_time = 0;
+ mp3parse->vbri_bytes = 0;
+ mp3parse->vbri_seek_points = 0;
+ mp3parse->vbri_seek_table = NULL;
+ g_free (mp3parse->vbri_seek_table);
+}
+
+static void
+gst_mpeg_audio_parse_init (GstMpegAudioParse * mp3parse,
+ GstMpegAudioParseClass * klass)
+{
+ gst_mpeg_audio_parse_reset (mp3parse);
+}
+
+static void
+gst_mpeg_audio_parse_finalize (GObject * object)
+{
+ G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+static gboolean
+gst_mpeg_audio_parse_start (GstBaseParse * parse)
+{
+ GstMpegAudioParse *mp3parse = GST_MPEG_AUDIO_PARSE (parse);
+
+ gst_base_parse_set_min_frame_size (GST_BASE_PARSE (mp3parse), 1024);
+ GST_DEBUG_OBJECT (parse, "starting");
+
+ gst_mpeg_audio_parse_reset (mp3parse);
+
+ return TRUE;
+}
+
+static gboolean
+gst_mpeg_audio_parse_stop (GstBaseParse * parse)
+{
+ GstMpegAudioParse *mp3parse = GST_MPEG_AUDIO_PARSE (parse);
+
+ GST_DEBUG_OBJECT (parse, "stopping");
+
+ gst_mpeg_audio_parse_reset (mp3parse);
+
+ return TRUE;
+}
+
+static const 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 const guint mp3types_freqs[3][3] = { {44100, 48000, 32000},
+{22050, 24000, 16000},
+{11025, 12000, 8000}
+};
+
+static inline guint
+mp3_type_frame_length_from_header (GstMpegAudioParse * mp3parse, guint32 header,
+ guint * put_version, guint * put_layer, guint * put_channels,
+ guint * put_bitrate, guint * put_samplerate, guint * put_mode,
+ guint * put_crc)
+{
+ guint length;
+ gulong mode, samplerate, bitrate, layer, channels, padding, crc;
+ gulong version;
+ gint lsf, mpg25;
+
+ if (header & (1 << 20)) {
+ lsf = (header & (1 << 19)) ? 0 : 1;
+ mpg25 = 0;
+ } else {
+ lsf = 1;
+ mpg25 = 1;
+ }
+
+ version = 1 + lsf + mpg25;
+
+ layer = 4 - ((header >> 17) & 0x3);
+
+ crc = (header >> 16) & 0x1;
+
+ bitrate = (header >> 12) & 0xF;
+ bitrate = mp3types_bitrates[lsf][layer - 1][bitrate] * 1000;
+ /* The caller has ensured we have a valid header, so bitrate can't be
+ zero here. */
+ g_assert (bitrate != 0);
+
+ samplerate = (header >> 10) & 0x3;
+ samplerate = mp3types_freqs[lsf + mpg25][samplerate];
+
+ padding = (header >> 9) & 0x1;
+
+ mode = (header >> 6) & 0x3;
+ channels = (mode == 3) ? 1 : 2;
+
+ switch (layer) {
+ case 1:
+ length = 4 * ((bitrate * 12) / samplerate + padding);
+ break;
+ case 2:
+ length = (bitrate * 144) / samplerate + padding;
+ break;
+ default:
+ case 3:
+ length = (bitrate * 144) / (samplerate << lsf) + padding;
+ break;
+ }
+
+ GST_DEBUG_OBJECT (mp3parse, "Calculated mp3 frame length of %u bytes",
+ length);
+ GST_DEBUG_OBJECT (mp3parse, "samplerate = %lu, bitrate = %lu, version = %lu, "
+ "layer = %lu, channels = %lu, mode = %s", samplerate, bitrate, version,
+ layer, channels, gst_mpeg_audio_channel_mode_get_nick (mode));
+
+ if (put_version)
+ *put_version = version;
+ if (put_layer)
+ *put_layer = layer;
+ if (put_channels)
+ *put_channels = channels;
+ if (put_bitrate)
+ *put_bitrate = bitrate;
+ if (put_samplerate)
+ *put_samplerate = samplerate;
+ if (put_mode)
+ *put_mode = mode;
+ if (put_crc)
+ *put_crc = crc;
+
+ return length;
+}
+
+/* Minimum number of consecutive, valid-looking frames to consider
+ * for resyncing */
+#define MIN_RESYNC_FRAMES 3
+
+/* Perform extended validation to check that subsequent headers match
+ * the first header given here in important characteristics, to avoid
+ * false sync. We look for a minimum of MIN_RESYNC_FRAMES consecutive
+ * frames to match their major characteristics.
+ *
+ * If at_eos is set to TRUE, we just check that we don't find any invalid
+ * frames in whatever data is available, rather than requiring a full
+ * MIN_RESYNC_FRAMES of data.
+ *
+ * Returns TRUE if we've seen enough data to validate or reject the frame.
+ * If TRUE is returned, then *valid contains TRUE if it validated, or false
+ * if we decided it was false sync.
+ * If FALSE is returned, then *valid contains minimum needed data.
+ */
+static gboolean
+gst_mp3parse_validate_extended (GstMpegAudioParse * mp3parse, GstBuffer * buf,
+ guint32 header, int bpf, gboolean at_eos, gint * valid)
+{
+ guint32 next_header;
+ const guint8 *data;
+ guint available;
+ int frames_found = 1;
+ int offset = bpf;
+
+ available = GST_BUFFER_SIZE (buf);
+ data = GST_BUFFER_DATA (buf);
+
+ while (frames_found < MIN_RESYNC_FRAMES) {
+ /* Check if we have enough data for all these frames, plus the next
+ frame header. */
+ if (available < offset + 4) {
+ if (at_eos) {
+ /* Running out of data at EOS is fine; just accept it */
+ *valid = TRUE;
+ return TRUE;
+ } else {
+ *valid = offset + 4;
+ return FALSE;
+ }
+ }
+
+ next_header = GST_READ_UINT32_BE (data + offset);
+ GST_DEBUG_OBJECT (mp3parse, "At %d: header=%08X, header2=%08X, bpf=%d",
+ offset, (unsigned int) header, (unsigned int) next_header, bpf);
+
+/* mask the bits which are allowed to differ between frames */
+#define HDRMASK ~((0xF << 12) /* bitrate */ | \
+ (0x1 << 9) /* padding */ | \
+ (0xf << 4) /* mode|mode extension */ | \
+ (0xf)) /* copyright|emphasis */
+
+ if ((next_header & HDRMASK) != (header & HDRMASK)) {
+ /* If any of the unmasked bits don't match, then it's not valid */
+ GST_DEBUG_OBJECT (mp3parse, "next header doesn't match "
+ "(header=%08X (%08X), header2=%08X (%08X), bpf=%d)",
+ (guint) header, (guint) header & HDRMASK, (guint) next_header,
+ (guint) next_header & HDRMASK, bpf);
+ *valid = FALSE;
+ return TRUE;
+ } else if ((((next_header >> 12) & 0xf) == 0) ||
+ (((next_header >> 12) & 0xf) == 0xf)) {
+ /* The essential parts were the same, but the bitrate held an
+ invalid value - also reject */
+ GST_DEBUG_OBJECT (mp3parse, "next header invalid (bitrate)");
+ *valid = FALSE;
+ return TRUE;
+ }
+
+ bpf = mp3_type_frame_length_from_header (mp3parse, next_header,
+ NULL, NULL, NULL, NULL, NULL, NULL, NULL);
+
+ offset += bpf;
+ frames_found++;
+ }
+
+ *valid = TRUE;
+ return TRUE;
+}
+
+static gboolean
+gst_mpeg_audio_parse_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 gboolean
+gst_mpeg_audio_parse_check_valid_frame (GstBaseParse * parse, GstBuffer * buf,
+ guint * framesize, gint * skipsize)
+{
+ GstByteReader reader = GST_BYTE_READER_INIT_FROM_BUFFER (buf);
+ GstMpegAudioParse *mp3parse = GST_MPEG_AUDIO_PARSE (parse);
+ gint off, bpf;
+ gboolean sync, drain, valid, caps_change;
+ guint32 header;
+ guint bitrate, layer, rate, channels, version, mode, crc;
+
+ if (G_UNLIKELY (GST_BUFFER_SIZE (buf) < 6))
+ return FALSE;
+
+ off = gst_byte_reader_masked_scan_uint32 (&reader, 0xffe00000, 0xffe00000,
+ 0, GST_BUFFER_SIZE (buf));
+
+ GST_LOG_OBJECT (parse, "possible sync at buffer offset %d", off);
+
+ /* didn't find anything that looks like a sync word, skip */
+ if (off < 0) {
+ *skipsize = GST_BUFFER_SIZE (buf) - 3;
+ return FALSE;
+ }
+
+ /* possible frame header, but not at offset 0? skip bytes before sync */
+ if (off > 0) {
+ *skipsize = off;
+ return FALSE;
+ }
+
+ /* make sure the values in the frame header look sane */
+ header = GST_READ_UINT32_BE (GST_BUFFER_DATA (buf));
+ if (!gst_mpeg_audio_parse_head_check (mp3parse, header)) {
+ *skipsize = 1;
+ return FALSE;
+ }
+
+ GST_LOG_OBJECT (parse, "got frame");
+
+ bpf = mp3_type_frame_length_from_header (mp3parse, header,
+ &version, &layer, &channels, &bitrate, &rate, &mode, &crc);
+ g_assert (bpf != 0);
+
+ if (channels != mp3parse->channels || rate != mp3parse->rate ||
+ layer != mp3parse->layer || version != mp3parse->version)
+ caps_change = TRUE;
+ else
+ caps_change = FALSE;
+
+ sync = gst_base_parse_get_sync (parse);
+ drain = gst_base_parse_get_drain (parse);
+
+ if (!drain && (!sync || caps_change)) {
+ if (!gst_mp3parse_validate_extended (mp3parse, buf, header, bpf, drain,
+ &valid)) {
+ /* not enough data */
+ gst_base_parse_set_min_frame_size (parse, valid);
+ *skipsize = 0;
+ return FALSE;
+ } else {
+ if (!valid) {
+ *skipsize = off + 2;
+ return FALSE;
+ }
+ }
+ }
+
+ *framesize = bpf;
+ return TRUE;
+}
+
+static void
+gst_mpeg_audio_parse_handle_first_frame (GstMpegAudioParse * mp3parse,
+ GstBuffer * buf)
+{
+ 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;
+ gint64 upstream_total_bytes = 0;
+ GstFormat fmt = GST_FORMAT_BYTES;
+ guint32 read_id;
+ const guint8 *data;
+ GstBaseParseSeekable seekable;
+ guint bitrate;
+
+ if (mp3parse->sent_codec_tag)
+ return;
+
+ /* 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_BUFFER_SIZE (buf);
+ data = GST_BUFFER_DATA (buf);
+ if (avail < offset + 8)
+ return;
+
+ /* The header starts at the provided offset */
+ data += offset;
+
+ /* obtain real upstream total bytes */
+ fmt = GST_FORMAT_BYTES;
+ if (!gst_pad_query_peer_duration (GST_BASE_PARSE_SINK_PAD (GST_BASE_PARSE
+ (mp3parse)), &fmt, &upstream_total_bytes))
+ upstream_total_bytes = 0;
+
+ 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_BUFFER_DATA (buf);
+ 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 ((total_time = mp3parse->xing_total_time) &&
+ (total_bytes = mp3parse->xing_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, new;
+ guint first;
+
+ first = data[0];
+ GST_DEBUG_OBJECT (mp3parse,
+ "Subtracting initial offset of %d bytes from Xing TOC", first);
+
+ /* xing seek table: percent time -> 1/256 bytepos */
+ for (i = 0; i < 100; i++) {
+ new = data[i] - first;
+ if (old > new) {
+ GST_WARNING_OBJECT (mp3parse, "Skipping broken Xing TOC");
+ mp3parse->xing_flags &= ~XING_TOC_FLAG;
+ goto skip_toc;
+ }
+ mp3parse->xing_seek_table[i] = old = new;
+ }
+
+ /* 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 == 99) {
+ gdouble fa, fb, fx;
+ gint a = percent, 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);
+ } 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);
+
+ /* check for truncated file */
+ if (upstream_total_bytes && mp3parse->xing_bytes &&
+ mp3parse->xing_bytes * 0.8 > upstream_total_bytes) {
+ GST_WARNING_OBJECT (mp3parse, "File appears to have been truncated; "
+ "invalidating Xing header duration and size");
+ mp3parse->xing_flags &= ~XING_BYTES_FLAG;
+ mp3parse->xing_flags &= ~XING_FRAMES_FLAG;
+ }
+ } 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_BUFFER_DATA (buf);
+ 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 ((total_time = mp3parse->vbri_total_time) &&
+ (total_bytes = mp3parse->vbri_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);
+
+ 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;
+ }
+
+ if (avail < offset + 26) {
+ GST_DEBUG_OBJECT (mp3parse,
+ "Not enough data to read VBRI header (need %d)",
+ offset + 26 + nseek_points * seek_bytes);
+ return;
+ }
+
+ data = GST_BUFFER_DATA (buf);
+ 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);
+
+ /* check for truncated file */
+ if (upstream_total_bytes && mp3parse->vbri_bytes &&
+ mp3parse->vbri_bytes * 0.8 > upstream_total_bytes) {
+ GST_WARNING_OBJECT (mp3parse, "File appears to have been truncated; "
+ "invalidating VBRI header duration and size");
+ mp3parse->vbri_valid = FALSE;
+ } else {
+ mp3parse->vbri_valid = TRUE;
+ }
+ } else {
+ GST_DEBUG_OBJECT (mp3parse,
+ "Xing, LAME or VBRI header not found in first frame");
+ }
+
+ /* set duration if tables provided a valid one */
+ if (mp3parse->xing_flags & XING_FRAMES_FLAG) {
+ gst_base_parse_set_duration (GST_BASE_PARSE (mp3parse), GST_FORMAT_TIME,
+ mp3parse->xing_total_time, 0);
+ }
+ if (mp3parse->vbri_total_time != 0 && mp3parse->vbri_valid) {
+ gst_base_parse_set_duration (GST_BASE_PARSE (mp3parse), GST_FORMAT_TIME,
+ mp3parse->vbri_total_time, 0);
+ }
+
+ /* tell baseclass how nicely we can seek, and a bitrate if one found */
+ seekable = GST_BASE_PARSE_SEEK_DEFAULT;
+ if ((mp3parse->xing_flags & XING_TOC_FLAG) && mp3parse->xing_bytes &&
+ mp3parse->xing_total_time)
+ seekable = GST_BASE_PARSE_SEEK_TABLE;
+
+ if (mp3parse->vbri_seek_table && mp3parse->vbri_bytes &&
+ mp3parse->vbri_total_time)
+ seekable = GST_BASE_PARSE_SEEK_TABLE;
+
+ if (mp3parse->xing_bitrate)
+ bitrate = mp3parse->xing_bitrate;
+ else if (mp3parse->vbri_bitrate)
+ bitrate = mp3parse->vbri_bitrate;
+ else
+ bitrate = 0;
+
+ gst_base_parse_set_seek (GST_BASE_PARSE (mp3parse), seekable, bitrate);
+}
+
+static GstFlowReturn
+gst_mpeg_audio_parse_parse_frame (GstBaseParse * parse, GstBuffer * buf)
+{
+ GstMpegAudioParse *mp3parse = GST_MPEG_AUDIO_PARSE (parse);
+ guint bitrate, layer, rate, channels, version, mode, crc;
+
+ g_return_val_if_fail (GST_BUFFER_SIZE (buf) >= 4, GST_FLOW_ERROR);
+
+ if (!mp3_type_frame_length_from_header (mp3parse,
+ GST_READ_UINT32_BE (GST_BUFFER_DATA (buf)),
+ &version, &layer, &channels, &bitrate, &rate, &mode, &crc))
+ goto broken_header;
+
+ if (G_UNLIKELY (channels != mp3parse->channels || rate != mp3parse->rate ||
+ layer != mp3parse->layer || version != mp3parse->version)) {
+ GstCaps *caps = gst_caps_new_simple ("audio/mpeg",
+ "mpegversion", G_TYPE_INT, 1,
+ "mpegaudioversion", G_TYPE_INT, version,
+ "layer", G_TYPE_INT, layer,
+ "rate", G_TYPE_INT, rate,
+ "channels", G_TYPE_INT, channels, "parsed", G_TYPE_BOOLEAN, TRUE, NULL);
+ gst_buffer_set_caps (buf, caps);
+ gst_pad_set_caps (GST_BASE_PARSE_SRC_PAD (parse), caps);
+ gst_caps_unref (caps);
+
+ mp3parse->rate = rate;
+ mp3parse->channels = channels;
+ 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 == 1) {
+ mp3parse->spf = 1152;
+ } else {
+ /* MPEG-2 or "2.5" */
+ mp3parse->spf = 576;
+ }
+
+ /* lead_in:
+ * 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).
+ *
+ * lead_out:
+ * Some mp3 streams have an offset in the timestamps, for which we have to
+ * push the frame *after* the end position in order for the decoder to be
+ * able to decode everything up until the segment.stop position. */
+ gst_base_parse_set_frame_props (parse, mp3parse->rate, mp3parse->spf,
+ (version == 1) ? 10 : 30, 2);
+ }
+
+ /* For first frame; check for seek tables and output a codec tag */
+ gst_mpeg_audio_parse_handle_first_frame (mp3parse, buf);
+
+ /* store some frame info for later processing */
+ mp3parse->last_crc = crc;
+ mp3parse->last_mode = mode;
+
+ return GST_FLOW_OK;
+
+/* ERRORS */
+broken_header:
+ {
+ /* this really shouldn't ever happen */
+ GST_ELEMENT_ERROR (parse, STREAM, DECODE, (NULL), (NULL));
+ return GST_FLOW_ERROR;
+ }
+}
+
+static gboolean
+gst_mpeg_audio_parse_time_to_bytepos (GstMpegAudioParse * mp3parse,
+ GstClockTime ts, gint64 * bytepos)
+{
+ gint64 total_bytes;
+ GstClockTime total_time;
+
+ /* If XING seek table exists use this for time->byte conversion */
+ if ((mp3parse->xing_flags & XING_TOC_FLAG) &&
+ (total_bytes = mp3parse->xing_bytes) &&
+ (total_time = mp3parse->xing_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 && (total_bytes = mp3parse->vbri_bytes) &&
+ (total_time = mp3parse->vbri_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;
+ }
+
+ return FALSE;
+}
+
+static gboolean
+gst_mpeg_audio_parse_bytepos_to_time (GstMpegAudioParse * mp3parse,
+ gint64 bytepos, GstClockTime * ts)
+{
+ gint64 total_bytes;
+ GstClockTime total_time;
+
+ /* If XING seek table exists use this for byte->time conversion */
+ if ((mp3parse->xing_flags & XING_TOC_FLAG) &&
+ (total_bytes = mp3parse->xing_bytes) &&
+ (total_time = mp3parse->xing_total_time)) {
+ gdouble fa, fb, fx;
+ gdouble pos;
+ gint index;
+
+ pos = CLAMP ((bytepos * 256.0) / total_bytes, 0.0, 256.0);
+ 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 (mp3parse->vbri_seek_table &&
+ (total_bytes = mp3parse->vbri_bytes) &&
+ (total_time = mp3parse->vbri_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;
+ }
+
+ return FALSE;
+}
+
+static gboolean
+gst_mpeg_audio_parse_convert (GstBaseParse * parse, GstFormat src_format,
+ gint64 src_value, GstFormat dest_format, gint64 * dest_value)
+{
+ GstMpegAudioParse *mp3parse = GST_MPEG_AUDIO_PARSE (parse);
+ gboolean res = FALSE;
+
+ if (src_format == GST_FORMAT_TIME && dest_format == GST_FORMAT_BYTES)
+ res =
+ gst_mpeg_audio_parse_time_to_bytepos (mp3parse, src_value, dest_value);
+ else if (src_format == GST_FORMAT_BYTES && dest_format == GST_FORMAT_TIME)
+ res = gst_mpeg_audio_parse_bytepos_to_time (mp3parse, src_value,
+ (GstClockTime *) dest_value);
+
+ /* if no tables, fall back to default estimated rate based conversion */
+ if (!res)
+ return gst_base_parse_convert_default (parse, src_format, src_value,
+ dest_format, dest_value);
+
+ return res;
+}
+
+static GstFlowReturn
+gst_mpeg_audio_parse_pre_push_buffer (GstBaseParse * parse, GstBuffer * buf)
+{
+ GstMpegAudioParse *mp3parse = GST_MPEG_AUDIO_PARSE (parse);
+ GstTagList *taglist;
+
+ /* tag sending done late enough in hook to ensure pending events
+ * have already been sent */
+
+ if (!mp3parse->sent_codec_tag) {
+ gchar *codec;
+
+ /* 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),
+ GST_BASE_PARSE_SRC_PAD (mp3parse), taglist);
+ g_free (codec);
+
+ /* also signals the end of first-frame processing */
+ mp3parse->sent_codec_tag = TRUE;
+ }
+
+ /* we will create a taglist (if any of the parameters has changed)
+ * to add the tags that changed */
+ taglist = NULL;
+ if (mp3parse->last_posted_crc != mp3parse->last_crc) {
+ gboolean using_crc;
+
+ if (!taglist) {
+ taglist = gst_tag_list_new ();
+ }
+ mp3parse->last_posted_crc = mp3parse->last_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 != mp3parse->last_mode) {
+ if (!taglist) {
+ taglist = gst_tag_list_new ();
+ }
+ mp3parse->last_posted_channel_mode = mp3parse->last_mode;
+
+ gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE, GST_TAG_MODE,
+ gst_mpeg_audio_channel_mode_get_nick (mp3parse->last_mode), NULL);
+ }
+
+ /* if the taglist exists, we need to send it */
+ if (taglist) {
+ gst_element_found_tags_for_pad (GST_ELEMENT (mp3parse),
+ GST_BASE_PARSE_SRC_PAD (mp3parse), taglist);
+ }
+
+ /* usual clipping applies */
+ return GST_BASE_PARSE_FLOW_CLIP;
+}
projects / platform / upstream / gst-plugins-good.git / commitdiff