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gstreamer/gst/audioparsers/gstmpegaudioparse.c
Zebediah Figura 71bb53a648 mpegaudioparse: Use a constant bit rate to convert between time and bytes if possible. 
This should result in no worse accuracy than the base parse element, and may
result in better accuracy. In particular, the number of bytes processed at any
given point, as accumulated by baseparse, can be only accurate to
(1 / # of frames) bytes per second, and if we try to seek immediately after
pausing the pipeline to a large offset, this small inaccuracy can propagate to
something noticeable.

The use case that prompted this patch is a 45-minute MPEG-1 layer 3 file, which
has a constant bit rate but no seek tables. Trying to seek the pipeline
immediately after pauisng it, without the ACCURATE flag, to a location 41
minutes in, yields a location that is, even with <https://gitlab.freedesktop.org/gstreamer/gstreamer/merge_requests/374>,
still audibly incorrect. This patch yields a much closer position, no longer
audibly incorrect, and likely within a frame of the most correct position.
2020-03-19 14:02:44 +00:00

1475 lines
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/* 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-mpegaudioparse
* @title: mpegaudioparse
* @short_description: MPEG audio parser
* @see_also: #GstAmrParse, #GstAACParse
*
* Parses and frames mpeg1 audio streams. Provides seeking.
*
* ## Example launch line
* |[
* gst-launch-1.0 filesrc location=test.mp3 ! mpegaudioparse ! mpg123audiodec
* ! audioconvert ! audioresample ! autoaudiosink
* ]|
*
*/
/* FIXME: we should make the base class (GstBaseParse) aware of the
* XING seek table somehow, so it can use it properly for things like
* accurate seeks. Currently it can only do a lookup via the convert function,
* but then doesn't know what the result represents exactly. One could either
* add a vfunc for index lookup, or just make mpegaudioparse populate the
* base class's index via the API provided.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include "gstmpegaudioparse.h"
#include <gst/base/gstbytereader.h>
#include <gst/pbutils/pbutils.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
#define MIN_FRAME_SIZE 6
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 ], "
"mpegaudioversion = (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")
);
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 GstFlowReturn gst_mpeg_audio_parse_handle_frame (GstBaseParse * parse,
GstBaseParseFrame * frame, gint * skipsize);
static GstFlowReturn gst_mpeg_audio_parse_pre_push_frame (GstBaseParse * parse,
GstBaseParseFrame * frame);
static gboolean gst_mpeg_audio_parse_convert (GstBaseParse * parse,
GstFormat src_format, gint64 src_value,
GstFormat dest_format, gint64 * dest_value);
static GstCaps *gst_mpeg_audio_parse_get_sink_caps (GstBaseParse * parse,
GstCaps * filter);
static void gst_mpeg_audio_parse_handle_first_frame (GstMpegAudioParse *
mp3parse, GstBuffer * buf);
#define gst_mpeg_audio_parse_parent_class parent_class
G_DEFINE_TYPE (GstMpegAudioParse, gst_mpeg_audio_parse, 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_class_init (GstMpegAudioParseClass * klass)
{
GstBaseParseClass *parse_class = GST_BASE_PARSE_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_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->handle_frame =
GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_handle_frame);
parse_class->pre_push_frame =
GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_pre_push_frame);
parse_class->convert = GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_convert);
parse_class->get_sink_caps =
GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_get_sink_caps);
/* 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);
gst_element_class_add_static_pad_template (element_class, &sink_template);
gst_element_class_add_static_pad_template (element_class, &src_template);
gst_element_class_set_static_metadata (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_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->freerate = 0;
mp3parse->hdr_bitrate = 0;
mp3parse->bitrate_is_constant = TRUE;
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, sizeof (mp3parse->xing_seek_table));
memset (mp3parse->xing_seek_table_inverse, 0,
sizeof (mp3parse->xing_seek_table_inverse));
mp3parse->vbri_bitrate = 0;
mp3parse->vbri_frames = 0;
mp3parse->vbri_total_time = 0;
mp3parse->vbri_bytes = 0;
mp3parse->vbri_seek_points = 0;
g_free (mp3parse->vbri_seek_table);
mp3parse->vbri_seek_table = NULL;
mp3parse->encoder_delay = 0;
mp3parse->encoder_padding = 0;
}
static void
gst_mpeg_audio_parse_init (GstMpegAudioParse * mp3parse)
{
gst_mpeg_audio_parse_reset (mp3parse);
GST_PAD_SET_ACCEPT_INTERSECT (GST_BASE_PARSE_SINK_PAD (mp3parse));
GST_PAD_SET_ACCEPT_TEMPLATE (GST_BASE_PARSE_SINK_PAD (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), MIN_FRAME_SIZE);
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;
if (!bitrate) {
GST_LOG_OBJECT (mp3parse, "using freeform bitrate");
bitrate = mp3parse->freerate;
}
samplerate = (header >> 10) & 0x3;
samplerate = mp3types_freqs[lsf + mpg25][samplerate];
/* force 0 length if 0 bitrate */
padding = (bitrate > 0) ? (header >> 9) & 0x1 : 0;
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;
GstMapInfo map;
gboolean res = TRUE;
int frames_found = 1;
int offset = bpf;
gst_buffer_map (buf, &map, GST_MAP_READ);
while (frames_found < MIN_RESYNC_FRAMES) {
/* Check if we have enough data for all these frames, plus the next
frame header. */
if (map.size < offset + 4) {
if (at_eos) {
/* Running out of data at EOS is fine; just accept it */
*valid = TRUE;
goto cleanup;
} else {
*valid = offset + 4;
res = FALSE;
goto cleanup;
}
}
next_header = GST_READ_UINT32_BE (map.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;
goto cleanup;
} else if (((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;
goto cleanup;
}
bpf = mp3_type_frame_length_from_header (mp3parse, next_header,
NULL, NULL, NULL, NULL, NULL, NULL, NULL);
/* if no bitrate, and no freeform rate known, then fail */
if (G_UNLIKELY (!bpf)) {
GST_DEBUG_OBJECT (mp3parse, "next header invalid (bitrate 0)");
*valid = FALSE;
goto cleanup;
}
offset += bpf;
frames_found++;
}
*valid = TRUE;
cleanup:
gst_buffer_unmap (buf, &map);
return res;
}
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) == 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;
}
/* Determines possible freeform frame rate/size by looking for next
* header with valid bitrate (0 or otherwise valid) (and sufficiently
* matching current header).
*
* Returns TRUE if we've found such one, and *rate then contains rate
* (or *rate contains 0 if decided no freeframe size could be determined).
* If not enough data, returns FALSE.
*/
static gboolean
gst_mp3parse_find_freerate (GstMpegAudioParse * mp3parse, GstMapInfo * map,
guint32 header, gboolean at_eos, gint * _rate)
{
guint32 next_header;
const guint8 *data;
guint available;
int offset = 4;
gulong samplerate, rate, layer, padding;
gboolean valid;
gint lsf, mpg25;
available = map->size;
data = map->data;
*_rate = 0;
/* pick apart header again partially */
if (header & (1 << 20)) {
lsf = (header & (1 << 19)) ? 0 : 1;
mpg25 = 0;
} else {
lsf = 1;
mpg25 = 1;
}
layer = 4 - ((header >> 17) & 0x3);
samplerate = (header >> 10) & 0x3;
samplerate = mp3types_freqs[lsf + mpg25][samplerate];
padding = (header >> 9) & 0x1;
for (; offset < available; ++offset) {
/* 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; failed to determine size */
return TRUE;
} else {
return FALSE;
}
}
valid = FALSE;
next_header = GST_READ_UINT32_BE (data + offset);
if ((next_header & 0xFFE00000) != 0xFFE00000)
goto next;
GST_DEBUG_OBJECT (mp3parse, "At %d: header=%08X, header2=%08X",
offset, (unsigned int) header, (unsigned int) next_header);
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))",
(guint) header, (guint) header & HDRMASK, (guint) next_header,
(guint) next_header & HDRMASK);
goto next;
} else if (((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)");
goto next;
}
valid = TRUE;
next:
/* almost accept as free frame */
if (layer == 1) {
rate = samplerate * (offset - 4 * padding + 4) / 48000;
} else {
rate = samplerate * (offset - padding + 1) / (144 >> lsf) / 1000;
}
if (valid) {
GST_LOG_OBJECT (mp3parse, "calculated rate %lu", rate * 1000);
if (rate < 8 || (layer == 3 && rate > 640)) {
GST_DEBUG_OBJECT (mp3parse, "rate invalid");
if (rate < 8) {
/* maybe some hope */
continue;
} else {
GST_DEBUG_OBJECT (mp3parse, "aborting");
/* give up */
break;
}
}
*_rate = rate * 1000;
break;
} else {
/* avoid indefinite searching */
if (rate > 1000) {
GST_DEBUG_OBJECT (mp3parse, "exceeded sanity rate; aborting");
break;
}
}
}
return TRUE;
}
static GstFlowReturn
gst_mpeg_audio_parse_handle_frame (GstBaseParse * parse,
GstBaseParseFrame * frame, gint * skipsize)
{
GstMpegAudioParse *mp3parse = GST_MPEG_AUDIO_PARSE (parse);
GstBuffer *buf = frame->buffer;
GstByteReader reader;
gint off, bpf = 0;
gboolean lost_sync, draining, valid, caps_change;
guint32 header;
guint bitrate, layer, rate, channels, version, mode, crc;
GstMapInfo map;
gboolean res = FALSE;
gst_buffer_map (buf, &map, GST_MAP_READ);
if (G_UNLIKELY (map.size < 6)) {
*skipsize = 1;
goto cleanup;
}
gst_byte_reader_init (&reader, map.data, map.size);
off = gst_byte_reader_masked_scan_uint32 (&reader, 0xffe00000, 0xffe00000,
0, map.size);
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 = map.size - 3;
goto cleanup;
}
/* possible frame header, but not at offset 0? skip bytes before sync */
if (off > 0) {
*skipsize = off;
goto cleanup;
}
/* make sure the values in the frame header look sane */
header = GST_READ_UINT32_BE (map.data);
if (!gst_mpeg_audio_parse_head_check (mp3parse, header)) {
*skipsize = 1;
goto cleanup;
}
GST_LOG_OBJECT (parse, "got frame");
lost_sync = GST_BASE_PARSE_LOST_SYNC (parse);
draining = GST_BASE_PARSE_DRAINING (parse);
if (G_UNLIKELY (lost_sync))
mp3parse->freerate = 0;
bpf = mp3_type_frame_length_from_header (mp3parse, header,
&version, &layer, &channels, &bitrate, &rate, &mode, &crc);
if (channels != mp3parse->channels || rate != mp3parse->rate ||
layer != mp3parse->layer || version != mp3parse->version)
caps_change = TRUE;
else
caps_change = FALSE;
/* maybe free format */
if (bpf == 0) {
GST_LOG_OBJECT (mp3parse, "possibly free format");
if (lost_sync || mp3parse->freerate == 0) {
GST_DEBUG_OBJECT (mp3parse, "finding free format rate");
if (!gst_mp3parse_find_freerate (mp3parse, &map, header, draining,
&valid)) {
/* not enough data */
gst_base_parse_set_min_frame_size (parse, valid);
*skipsize = 0;
goto cleanup;
} else {
GST_DEBUG_OBJECT (parse, "determined freeform size %d", valid);
mp3parse->freerate = valid;
}
}
/* try again */
bpf = mp3_type_frame_length_from_header (mp3parse, header,
&version, &layer, &channels, &bitrate, &rate, &mode, &crc);
if (!bpf) {
/* did not come up with valid freeform length, reject after all */
*skipsize = 1;
goto cleanup;
}
}
if (!draining && (lost_sync || caps_change)) {
if (!gst_mp3parse_validate_extended (mp3parse, buf, header, bpf, draining,
&valid)) {
/* not enough data */
gst_base_parse_set_min_frame_size (parse, valid);
*skipsize = 0;
goto cleanup;
} else {
if (!valid) {
*skipsize = off + 2;
goto cleanup;
}
}
} else if (draining && lost_sync && caps_change && mp3parse->rate > 0) {
/* avoid caps jitter that we can't be sure of */
*skipsize = off + 2;
goto cleanup;
}
/* restore default minimum */
gst_base_parse_set_min_frame_size (parse, MIN_FRAME_SIZE);
res = TRUE;
/* metadata handling */
if (G_UNLIKELY (caps_change)) {
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_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_rate (parse, mp3parse->rate, mp3parse->spf,
(version == 1) ? 10 : 30, 2);
}
if (mp3parse->hdr_bitrate && mp3parse->hdr_bitrate != bitrate) {
mp3parse->bitrate_is_constant = FALSE;
}
mp3parse->hdr_bitrate = bitrate;
/* 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;
cleanup:
gst_buffer_unmap (buf, &map);
if (res && bpf <= map.size) {
return gst_base_parse_finish_frame (parse, frame, bpf);
}
return GST_FLOW_OK;
}
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 */
const guint32 lame_id = 0x4c414d45; /* 'LAME' in hex */
gint offset_xing, offset_vbri;
guint64 avail;
gint64 upstream_total_bytes = 0;
guint32 read_id_xing = 0, read_id_vbri = 0;
GstMapInfo map;
guint8 *data;
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_xing = 0x11;
else
offset_xing = 0x20;
} else { /* MPEG-2 header */
if (mp3parse->channels == 1)
offset_xing = 0x09;
else
offset_xing = 0x11;
}
/* The VBRI tag is always at offset 0x20 */
offset_vbri = 0x20;
/* Skip the 4 bytes of the MP3 header too */
offset_xing += 4;
offset_vbri += 4;
/* Check if we have enough data to read the Xing header */
gst_buffer_map (buf, &map, GST_MAP_READ);
data = map.data;
avail = map.size;
if (avail >= offset_xing + 4) {
read_id_xing = GST_READ_UINT32_BE (data + offset_xing);
}
if (avail >= offset_vbri + 4) {
read_id_vbri = GST_READ_UINT32_BE (data + offset_vbri);
}
/* obtain real upstream total bytes */
if (!gst_pad_peer_query_duration (GST_BASE_PARSE_SINK_PAD (mp3parse),
GST_FORMAT_BYTES, &upstream_total_bytes))
upstream_total_bytes = 0;
if (read_id_xing == xing_id || read_id_xing == info_id) {
guint32 xing_flags;
guint bytes_needed = offset_xing + 8;
gint64 total_bytes;
GstClockTime total_time;
GST_DEBUG_OBJECT (mp3parse, "Found Xing header marker 0x%x", xing_id);
/* Move data after Xing header */
data += offset_xing + 4;
/* Read 4 base bytes of flags, big-endian */
xing_flags = GST_READ_UINT32_BE (data);
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);
goto cleanup;
}
GST_DEBUG_OBJECT (mp3parse, "Reading Xing header");
mp3parse->xing_flags = xing_flags;
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 (percent < 99 && table[percent]) {
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, sizeof (mp3parse->xing_seek_table));
memset (mp3parse->xing_seek_table_inverse, 0,
sizeof (mp3parse->xing_seek_table_inverse));
}
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);
/* 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;
}
/* Optional LAME tag? */
if (avail - bytes_needed >= 36 && GST_READ_UINT32_BE (data) == lame_id) {
gchar lame_version[10] = { 0, };
guint tag_rev;
guint32 encoder_delay, encoder_padding;
memcpy (lame_version, data, 9);
data += 9;
tag_rev = data[0] >> 4;
GST_DEBUG_OBJECT (mp3parse, "Found LAME tag revision %d created by '%s'",
tag_rev, lame_version);
/* Skip all the information we're not interested in */
data += 12;
/* Encoder delay and end padding */
encoder_delay = GST_READ_UINT24_BE (data);
encoder_delay >>= 12;
encoder_padding = GST_READ_UINT24_BE (data);
encoder_padding &= 0x000fff;
mp3parse->encoder_delay = encoder_delay;
mp3parse->encoder_padding = encoder_padding;
GST_DEBUG_OBJECT (mp3parse, "Encoder delay %u, encoder padding %u",
encoder_delay, encoder_padding);
}
} else if (read_id_vbri == 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_vbri + 26) {
GST_DEBUG_OBJECT (mp3parse,
"Not enough data to read VBRI header (need %d)", offset_vbri + 26);
goto cleanup;
}
GST_DEBUG_OBJECT (mp3parse, "Reading VBRI header");
/* Move data after VBRI header */
data += offset_vbri + 4;
if (GST_READ_UINT16_BE (data) != 0x0001) {
GST_WARNING_OBJECT (mp3parse,
"Unsupported VBRI version 0x%x", GST_READ_UINT16_BE (data));
goto cleanup;
}
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_vbri + 26 + nseek_points * seek_bytes) {
GST_WARNING_OBJECT (mp3parse,
"Not enough data to read VBRI seek table (need %d)",
offset_vbri + 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 = map.data;
data += offset_vbri + 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 */
/* FIXME: fill index with seek table */
#if 0
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;
#endif
if (mp3parse->xing_bitrate)
bitrate = mp3parse->xing_bitrate;
else if (mp3parse->vbri_bitrate)
bitrate = mp3parse->vbri_bitrate;
else
bitrate = 0;
gst_base_parse_set_average_bitrate (GST_BASE_PARSE (mp3parse), bitrate);
cleanup:
gst_buffer_unmap (buf, &map);
}
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;
}
/* If we have had a constant bit rate (so far), use it directly, as it
* may give slightly more accurate results than the base class. */
if (mp3parse->bitrate_is_constant && mp3parse->hdr_bitrate) {
*bytepos = gst_util_uint64_scale (ts, mp3parse->hdr_bitrate,
8 * GST_SECOND);
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;
}
/* If we have had a constant bit rate (so far), use it directly, as it
* may give slightly more accurate results than the base class. */
if (mp3parse->bitrate_is_constant && mp3parse->hdr_bitrate) {
*ts = gst_util_uint64_scale (bytepos, 8 * GST_SECOND,
mp3parse->hdr_bitrate);
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_frame (GstBaseParse * parse,
GstBaseParseFrame * frame)
{
GstMpegAudioParse *mp3parse = GST_MPEG_AUDIO_PARSE (parse);
GstTagList *taglist = NULL;
/* we will create a taglist (if any of the parameters has changed)
* to add the tags that changed */
if (mp3parse->last_posted_crc != mp3parse->last_crc) {
gboolean using_crc;
if (!taglist)
taglist = gst_tag_list_new_empty ();
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_empty ();
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);
}
/* tag sending done late enough in hook to ensure pending events
* have already been sent */
if (taglist != NULL || !mp3parse->sent_codec_tag) {
GstCaps *caps;
if (taglist == NULL)
taglist = gst_tag_list_new_empty ();
/* codec tag */
caps = gst_pad_get_current_caps (GST_BASE_PARSE_SRC_PAD (parse));
if (G_UNLIKELY (caps == NULL)) {
gst_tag_list_unref (taglist);
if (GST_PAD_IS_FLUSHING (GST_BASE_PARSE_SRC_PAD (parse))) {
GST_INFO_OBJECT (parse, "Src pad is flushing");
return GST_FLOW_FLUSHING;
} else {
GST_INFO_OBJECT (parse, "Src pad is not negotiated!");
return GST_FLOW_NOT_NEGOTIATED;
}
}
gst_pb_utils_add_codec_description_to_tag_list (taglist,
GST_TAG_AUDIO_CODEC, caps);
gst_caps_unref (caps);
if (mp3parse->hdr_bitrate > 0 && mp3parse->xing_bitrate == 0 &&
mp3parse->vbri_bitrate == 0) {
/* We don't have a VBR bitrate, so post the available bitrate as
* nominal and let baseparse calculate the real bitrate */
gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE,
GST_TAG_NOMINAL_BITRATE, mp3parse->hdr_bitrate, NULL);
}
/* also signals the end of first-frame processing */
mp3parse->sent_codec_tag = TRUE;
}
/* if the taglist exists, we need to update it so it gets sent out */
if (taglist) {
gst_base_parse_merge_tags (parse, taglist, GST_TAG_MERGE_REPLACE);
gst_tag_list_unref (taglist);
}
/* usual clipping applies */
frame->flags |= GST_BASE_PARSE_FRAME_FLAG_CLIP;
return GST_FLOW_OK;
}
static void
remove_fields (GstCaps * caps)
{
guint i, n;
n = gst_caps_get_size (caps);
for (i = 0; i < n; i++) {
GstStructure *s = gst_caps_get_structure (caps, i);
gst_structure_remove_field (s, "parsed");
}
}
static GstCaps *
gst_mpeg_audio_parse_get_sink_caps (GstBaseParse * parse, GstCaps * filter)
{
GstCaps *peercaps, *templ;
GstCaps *res;
templ = gst_pad_get_pad_template_caps (GST_BASE_PARSE_SINK_PAD (parse));
if (filter) {
GstCaps *fcopy = gst_caps_copy (filter);
/* Remove the fields we convert */
remove_fields (fcopy);
peercaps = gst_pad_peer_query_caps (GST_BASE_PARSE_SRC_PAD (parse), fcopy);
gst_caps_unref (fcopy);
} else
peercaps = gst_pad_peer_query_caps (GST_BASE_PARSE_SRC_PAD (parse), NULL);
if (peercaps) {
/* Remove the parsed field */
peercaps = gst_caps_make_writable (peercaps);
remove_fields (peercaps);
res = gst_caps_intersect_full (peercaps, templ, GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (peercaps);
gst_caps_unref (templ);
} else {
res = templ;
}
if (filter) {
GstCaps *intersection;
intersection =
gst_caps_intersect_full (filter, res, GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (res);
res = intersection;
}
return res;
}
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