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GST-Tensordecoder-ov_ep/subprojects/gst-plugins-good/gst/audioparsers/gstmpegaudioparse.c
Carlos Rafael Giani 671c89c392 mpg123: Add gapless playback support 
Co-authored-by: Sebastian Dröge <sebastian@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1028>
2022-03-14 10:32:15 +02:00

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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
* ]|
*
*/
/* Notes about gapless playback, "Frankenstein" streams, and the Xing header frame:
*
* Gapless playback is based on the LAME tag, which is located in the Xing
* header frame. The tag contains the encoder delay and encoder padding.
* The encoder delay specifies how many padding nullsamples have been prepended
* by the encoder at the start of the mp3 stream, while the encoder padding
* specifies how many padding nullsamples got added at the end of the stream.
*
* In addition, there is also a "decoder delay". This affects all existing
* mp3 decoders - they themselves introduce a delay into the signal due to
* the way mp3 decoding works. This delay is 529 samples long in all known
* decoders. Unlike the encoder delay, the decoder delay is not specified
* anywhere in the mp3 stream. Players/decoders therefore hardcode the
* decoder delay as 529 samples.
*
* (The LAME tech FAQ mentions 528 samples instead of 529, but LAME seems to
* use 529 samples. Also, decoders like mpg123 use 529 samples instead of 528.
* The situation is a little unclear, but 529 samples seems to be standard.)
*
* For proper gapless playback, both mpegaudioparse and a downstream MPEG
* audio decoder must do their part. mpegaudioparse adjusts buffer PTS/DTS
* and durations, and adds GstAudioClippingMeta to outgoing buffers if
* clipping is necessary. MPEG decoders then clip decoded frames according
* to that meta (if present).
*
* To detect when to add GstAudioClippingMeta and when to adjust PTS/DTS/
* durations, the number of the current frame is retrieved. Based on that, the
* current stream position in samples is calculated. With the sample position,
* it is determined whether or not the current playback position is still
* if the actual playback range (= in the actual playback range of the stream
* that excludes padding samples), or if it is already outside, or partially
* outside.
*
* start_of_actual_samples and end_of_actual_samples define the start/end
* of this actual playback range, in samples. So:
* If sample_pos >= start_of_actual_samples and sample_pos end_of_actual_samples
* -> sample_pos is inside the actual playback range.
*
* (The decoder delay could in theory be left for the decoder to worry
* about. But then, the decoder would also have to adjust PTS/DTS/durations
* of decoded buffers, which is not something a GstAudioDecoder based element
* should have to deal with. So, for convenience, mpegaudioparse also factors
* that delay into its calculations.)
*
*
* "Frankenstein" streams are MPEG streams which have streams beyond
* what the Xing metadata indicates. Such streams typically are the
* result of poorly stitching individual mp3s together, like this:
*
* cat first.mp3 second.mp3 > joined.mp3
*
* The resulting mp3 is not guaranteed to be valid. In particular, this can
* cause confusion when first.mp3 contains a Xing header frame. Its length
* indicator then does not match the actual length (which is bigger). When
* this is detected, a log line about this being a Frankenstein stream is
* generated.
*
*
* Xing header frames are empty dummy MPEG frames. They only exist for
* supplying metadata. They are encoded as valid silent MPEG frames for
* backwards compatibility with older hardware MP3 players, but can be safely
* dropped.
*
* For more about Xing header frames, see:
* https://www.codeproject.com/Articles/8295/MPEG-Audio-Frame-Header#XINGHeader
* https://www.compuphase.com/mp3/mp3loops.htm#PADDING_DELAYS
*
* To facilitate gapless playback and ensure that MPEG audio decoders don't
* actually decode this frame as an empty MPEG frame, it is marked here as
* GST_BUFFER_FLAG_DECODE_ONLY / GST_BUFFER_FLAG_DROPPABLE in mpegaudioparse
* after its metadata got extracted. It is also marked as such if it is
* encountered again after the user for example seeked back to the beginning
* of the mp3 stream. Its duration is also set to zero to make sure that the
* frame does not cause baseparse to increment the timestamp of the frame that
* follows this one.
*
*/
/* 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 "gstaudioparserselements.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_src_query (GstBaseParse * parse,
GstQuery * query);
static gboolean gst_mpeg_audio_parse_sink_event (GstBaseParse * parse,
GstEvent * event);
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 gboolean
gst_mpeg_audio_parse_check_if_is_xing_header_frame (GstMpegAudioParse *
mp3parse, GstBuffer * buf);
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);
GST_ELEMENT_REGISTER_DEFINE (mpegaudioparse, "mpegaudioparse",
GST_RANK_PRIMARY + 2, GST_TYPE_MPEG_AUDIO_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->src_query = GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_src_query);
parse_class->sink_event = GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_sink_event);
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->upstream_format = GST_FORMAT_UNDEFINED;
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->spf = 0;
mp3parse->outgoing_frame_is_xing_header = FALSE;
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;
mp3parse->decoder_delay = 0;
mp3parse->start_of_actual_samples = 0;
mp3parse->end_of_actual_samples = 0;
mp3parse->total_padding_time = GST_CLOCK_TIME_NONE;
mp3parse->start_padding_time = GST_CLOCK_TIME_NONE;
mp3parse->end_padding_time = GST_CLOCK_TIME_NONE;
}
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;
}
/* We need the frame duration for calculating the frame number later
* in gst_mpeg_audio_parse_pre_push_frame (). */
mp3parse->frame_duration = gst_util_uint64_scale (GST_SECOND,
mp3parse->spf, mp3parse->rate);
/* 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;
/* While during normal playback, the Xing header frame is seen only once
* (right at the beginning), we may see it again if the user seeked back
* to the beginning. To make sure it is dropped again and NOT pushed
* downstream, we have to check every frame for Xing IDs.
*
* (sent_codec_tag is TRUE after this Xing frame got parsed.) */
if (G_LIKELY (mp3parse->sent_codec_tag)) {
if (G_UNLIKELY (gst_mpeg_audio_parse_check_if_is_xing_header_frame
(mp3parse, buf))) {
GST_DEBUG_OBJECT (mp3parse, "This is a Xing header frame, which "
"contains no meaningful audio data, and can be safely dropped");
mp3parse->outgoing_frame_is_xing_header = TRUE;
}
}
/* 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);
/* We don't actually drop the frame right here, but rather in
* gst_mpeg_audio_parse_pre_push_frame (), since it is still important
* to let other code bits do their work there even if we want to drop
* the current frame. */
if (G_UNLIKELY (mp3parse->outgoing_frame_is_xing_header)) {
frame->flags |= GST_BASE_PARSE_FRAME_FLAG_NO_FRAME;
/* Set duration to zero to prevent the baseparse class
* from incrementing outgoing timestamps */
GST_BUFFER_DURATION (frame->buffer) = 0;
}
if (res && bpf <= map.size) {
return gst_base_parse_finish_frame (parse, frame, bpf);
}
return GST_FLOW_OK;
}
static gboolean
gst_mpeg_audio_parse_check_if_is_xing_header_frame (GstMpegAudioParse *
mp3parse, GstBuffer * buf)
{
/* TODO: get rid of code duplication
* (see gst_mpeg_audio_parse_handle_first_frame ()) */
const guint32 xing_id = 0x58696e67; /* 'Xing' in hex */
const guint32 info_id = 0x496e666f; /* 'Info' in hex - found in LAME CBR files */
gint offset_xing;
GstMapInfo map;
guint8 *data;
guint64 avail;
guint32 read_id_xing = 0;
gboolean ret = FALSE;
/* 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;
}
/* Skip the 4 bytes of the MP3 header too */
offset_xing += 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);
ret = (read_id_xing == xing_id || read_id_xing == info_id);
}
gst_buffer_unmap (buf, &map);
return ret;
}
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;
guint64 num_xing_samples = 0;
GstClockTime total_time;
GST_DEBUG_OBJECT (mp3parse, "Found Xing header marker 0x%x", xing_id);
GST_DEBUG_OBJECT (mp3parse, "This is a Xing header frame, which contains "
"no meaningful audio data, and can be safely dropped");
mp3parse->outgoing_frame_is_xing_header = TRUE;
/* 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 {
num_xing_samples = (guint64) (mp3parse->xing_frames) * (mp3parse->spf);
mp3parse->xing_total_time = gst_util_uint64_scale (GST_SECOND,
num_xing_samples, mp3parse->rate);
}
data += 4;
} else {
mp3parse->xing_frames = 0;
mp3parse->xing_total_time = 0;
}
/* Store the entire time as actual total time for now. Should there be
* any padding present, this value will get adjusted accordingly. */
mp3parse->xing_actual_total_time = mp3parse->xing_total_time;
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, %"
G_GUINT64_FORMAT " samples, time %" GST_TIME_FORMAT
" (this includes potentially present padding data), %u bytes,"
" vbr scale %u", mp3parse->xing_frames, num_xing_samples,
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;
guint64 total_padding_samples;
guint64 actual_num_xing_samples;
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;
total_padding_samples = encoder_delay + encoder_padding;
mp3parse->encoder_delay = encoder_delay;
mp3parse->encoder_padding = encoder_padding;
/* As mentioned in the overview at the beginning of this source
* file, decoders exhibit a delay of 529 samples. */
mp3parse->decoder_delay = 529;
/* Where the actual, non-padding samples start & end, in sample offsets. */
mp3parse->start_of_actual_samples = mp3parse->encoder_delay +
mp3parse->decoder_delay;
mp3parse->end_of_actual_samples = num_xing_samples +
mp3parse->decoder_delay - mp3parse->encoder_padding;
/* Length of padding at the start and at the end of the stream,
* in nanoseconds. */
mp3parse->start_padding_time = gst_util_uint64_scale_int (GST_SECOND,
mp3parse->start_of_actual_samples, mp3parse->rate);
mp3parse->end_padding_time = mp3parse->xing_total_time -
gst_util_uint64_scale_int (mp3parse->end_of_actual_samples,
GST_SECOND, mp3parse->rate);
/* Total length of all combined padding samples, in nanoseconds. */
mp3parse->total_padding_time = gst_util_uint64_scale_int (GST_SECOND,
total_padding_samples, mp3parse->rate);
/* Length of media, in samples, without the number of padding samples. */
actual_num_xing_samples = (num_xing_samples >= total_padding_samples) ?
(num_xing_samples - total_padding_samples) : 0;
/* Length of media, converted to nanoseconds. This is used for setting
* baseparse's duration. */
mp3parse->xing_actual_total_time = gst_util_uint64_scale (GST_SECOND,
actual_num_xing_samples, mp3parse->rate);
GST_DEBUG_OBJECT (mp3parse, "Encoder delay: %u samples",
mp3parse->encoder_delay);
GST_DEBUG_OBJECT (mp3parse, "Encoder padding: %u samples",
mp3parse->encoder_padding);
GST_DEBUG_OBJECT (mp3parse, "Decoder delay: %u samples",
mp3parse->decoder_delay);
GST_DEBUG_OBJECT (mp3parse, "Start of actual samples: %"
G_GUINT64_FORMAT, mp3parse->start_of_actual_samples);
GST_DEBUG_OBJECT (mp3parse, "End of actual samples: %"
G_GUINT64_FORMAT, mp3parse->end_of_actual_samples);
GST_DEBUG_OBJECT (mp3parse, "Total padding samples: %" G_GUINT64_FORMAT,
total_padding_samples);
GST_DEBUG_OBJECT (mp3parse, "Start padding time: %" GST_TIME_FORMAT,
GST_TIME_ARGS (mp3parse->start_padding_time));
GST_DEBUG_OBJECT (mp3parse, "End padding time: %" GST_TIME_FORMAT,
GST_TIME_ARGS (mp3parse->end_padding_time));
GST_DEBUG_OBJECT (mp3parse, "Total padding time: %" GST_TIME_FORMAT,
GST_TIME_ARGS (mp3parse->total_padding_time));
GST_DEBUG_OBJECT (mp3parse, "Actual total media samples: %"
G_GUINT64_FORMAT, actual_num_xing_samples);
GST_DEBUG_OBJECT (mp3parse, "Actual total media length: %"
GST_TIME_FORMAT, GST_TIME_ARGS (mp3parse->xing_actual_total_time));
}
} 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_actual_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_src_query (GstBaseParse * parse, GstQuery * query)
{
gboolean res = FALSE;
GstMpegAudioParse *mp3parse = GST_MPEG_AUDIO_PARSE (parse);
res = GST_BASE_PARSE_CLASS (parent_class)->src_query (parse, query);
if (!res)
return FALSE;
/* If upstream operates in BYTE format then consider any parsed Xing/LAME
* header to remove encoder/decoder delay and padding samples from the
* position query. */
if (mp3parse->upstream_format == GST_FORMAT_BYTES
|| GST_PAD_MODE (GST_BASE_PARSE_SINK_PAD (parse)) == GST_PAD_MODE_PULL) {
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_POSITION:{
GstFormat format;
gint64 position, new_position;
GstClockTime duration_to_skip;
gst_query_parse_position (query, &format, &position);
/* Adjust the position to exclude padding samples. */
if ((position < 0) || (format != GST_FORMAT_TIME))
break;
duration_to_skip = mp3parse->frame_duration +
mp3parse->start_padding_time;
if (position < duration_to_skip)
new_position = 0;
else
new_position = position - duration_to_skip;
if (new_position > (mp3parse->xing_actual_total_time))
new_position = mp3parse->xing_actual_total_time;
GST_LOG_OBJECT (mp3parse, "applying gapless padding info to position "
"query response: %" GST_TIME_FORMAT " -> %" GST_TIME_FORMAT,
GST_TIME_ARGS (position), GST_TIME_ARGS (new_position));
gst_query_set_position (query, GST_FORMAT_TIME, new_position);
break;
}
default:
break;
}
}
return res;
}
static gboolean
gst_mpeg_audio_parse_sink_event (GstBaseParse * parse, GstEvent * event)
{
gboolean res = FALSE;
GstMpegAudioParse *mp3parse = GST_MPEG_AUDIO_PARSE (parse);
res =
GST_BASE_PARSE_CLASS (parent_class)->sink_event (parse,
gst_event_ref (event));
if (!res) {
gst_event_unref (event);
return FALSE;
}
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEGMENT:{
const GstSegment *segment;
gst_event_parse_segment (event, &segment);
mp3parse->upstream_format = segment->format;
}
default:
break;
}
gst_event_unref (event);
return res;
}
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);
}
/* adjust buffer PTS/DTS/durations according to gapless playback info */
if ((mp3parse->upstream_format == GST_FORMAT_BYTES
|| GST_PAD_MODE (GST_BASE_PARSE_SINK_PAD (parse)) ==
GST_PAD_MODE_PULL)
&& GST_CLOCK_TIME_IS_VALID (mp3parse->total_padding_time)) {
guint64 frame_nr;
GstClockTime pts, dts;
gboolean add_clipping_meta = FALSE;
guint32 start_clip = 0, end_clip = 0;
GstClockTime timestamp_decrement;
guint64 sample_pos;
guint64 sample_pos_end;
/* Get the number of the current frame so we can determine where we
* currently are in the MPEG stream.
*
* Gapless playback is best done based on samples, not timestamps,
* to avoid potential rounding errors that can otherwise cause a few
* samples to be incorrectly clipped or not clipped.
*
* TODO: At the moment, there is no dedicated baseparse API for finding
* out what frame we are currently in. The frame number is calculated
* out of the PTS of the current frame. Each frame has the same duration,
* and at this point, the buffer's PTS has not been adjusted to exclude
* the padding samples, so the PTS will be an integer multiple of
* frame_duration. However, this is not an ideal solution. Investigate
* how to properly implement this. */
frame_nr = GST_BUFFER_PTS (frame->buffer) / mp3parse->frame_duration;
GST_LOG_OBJECT (mp3parse, "Handling MP3 frame #%" G_GUINT64_FORMAT,
frame_nr);
/* By default, we subtract the start_padding_time from the timestamps.
* start_padding_time specifies the duration of the padding samples
* at the beginning of the MPEG stream. To factor out these padding
* samples, we have to shift the timestamps back, which is done with
* this decrement. */
timestamp_decrement = mp3parse->start_padding_time;
pts = GST_BUFFER_PTS (frame->buffer);
dts = GST_BUFFER_DTS (frame->buffer);
/* sample_pos specifies the current position of the beginning of the
* current frame, while sample_pos_end specifies the current position
* of 1 samples past the end of the current frame. Both values are
* in samples. */
sample_pos = frame_nr * mp3parse->spf;
sample_pos_end = sample_pos + mp3parse->spf;
/* Check if the frame is not (fully) within the actual playback range. */
if (G_UNLIKELY (sample_pos <= mp3parse->start_of_actual_samples ||
(sample_pos_end >= mp3parse->end_of_actual_samples))) {
if (G_UNLIKELY (frame_nr >= mp3parse->xing_frames)) {
/* Test #1: Check if the current position lies past the length
* that is specified by the Xing frame header. This normally does
* not happen, but does occur with "Frankenstein" streams (see
* the explanation at the beginning of this source file for more).
* Do this first, since the other test may yield false positives
* in this case. */
GST_LOG_OBJECT (mp3parse, "There are frames beyond what the Xing "
"metadata indicates; this is a Frankenstein stream!");
/* The frames past the "officially" last one (= the last one according
* to the Xing header frame) are located past the padding samples
* that follow the actual playback range. The length of these
* padding samples in nanoseconds is stored in end_padding_time.
* We need to shift the PTS to compensate for these padding samples,
* otherwise there would be a timestamp discontinuity between the
* last "official" frame and the first "Frankenstein" frame. */
timestamp_decrement += mp3parse->end_padding_time;
} else if (sample_pos_end <= mp3parse->start_of_actual_samples) {
/* Test #2: Check if the frame lies completely before the actual
* playback range. This happens if the number of padding samples
* at the start of the stream exceeds the size of a frame, meaning
* that the entire frame will be filled with padding samples.
* This has not been observed so far. However, it is in theory
* possible, so handle it here. */
/* We want to clip all samples in the frame. Since this is a frame
* at the start of the stream, set start_clip to the frame size.
* Also set the buffer duration to 0 to make sure baseparse does not
* increment timestamps after this current frame is finished. */
start_clip = mp3parse->spf;
GST_BUFFER_DURATION (frame->buffer) = 0;
add_clipping_meta = TRUE;
} else if (sample_pos <= mp3parse->start_of_actual_samples) {
/* Test #3: Check if a portion of the frame lies before the actual
* playback range. Set the duration to the number of samples that
* remain after clipping. */
start_clip = mp3parse->start_of_actual_samples - sample_pos;
GST_BUFFER_DURATION (frame->buffer) =
gst_util_uint64_scale_int (sample_pos_end -
mp3parse->start_of_actual_samples, GST_SECOND, mp3parse->rate);
add_clipping_meta = TRUE;
} else if (sample_pos >= mp3parse->end_of_actual_samples) {
/* Test #4: Check if the frame lies completely after the actual
* playback range. Similar to test #2, this happens if the number
* of padding samples at the end of the stream exceeds the size of
* a frame, meaning that the entire frame will be filled with padding
* samples. Unlike test #2, this has been observed in mp3s several
* times: The penultimate frame is partially clipped, the final
* frame is fully clipped. */
GstClockTime padding_ns;
/* We want to clip all samples in the frame. Since this is a frame
* at the end of the stream, set end_clip to the frame size.
* Also set the buffer duration to 0 to make sure baseparse does not
* increment timestamps after this current frame is finished. */
end_clip = mp3parse->spf;
GST_BUFFER_DURATION (frame->buffer) = 0;
/* Even though this frame will be fully clipped, we still have to
* make sure its timestamps are not discontinuous with the preceding
* ones. To that end, it is necessary to subtract the time range
* between the current position and the last valid playback range
* position from the PTS and DTS. */
padding_ns = gst_util_uint64_scale_int (sample_pos -
mp3parse->end_of_actual_samples, GST_SECOND, mp3parse->rate);
timestamp_decrement += padding_ns;
add_clipping_meta = TRUE;
} else if (sample_pos_end >= mp3parse->end_of_actual_samples) {
/* Test #5: Check if a portion of the frame lies after the actual
* playback range. Set the duration to the number of samples that
* remain after clipping. */
end_clip = sample_pos_end - mp3parse->end_of_actual_samples;
GST_BUFFER_DURATION (frame->buffer) =
gst_util_uint64_scale_int (mp3parse->end_of_actual_samples -
sample_pos, GST_SECOND, mp3parse->rate);
add_clipping_meta = TRUE;
}
}
if (G_UNLIKELY (add_clipping_meta)) {
GST_DEBUG_OBJECT (mp3parse, "Adding clipping meta: start %"
G_GUINT32_FORMAT " end %" G_GUINT32_FORMAT, start_clip, end_clip);
gst_buffer_add_audio_clipping_meta (frame->buffer, GST_FORMAT_DEFAULT,
start_clip, end_clip);
}
/* Adjust the timestamps by subtracting from them. The decrement
* is computed above. */
GST_BUFFER_PTS (frame->buffer) = (pts >= timestamp_decrement) ? (pts -
timestamp_decrement) : 0;
GST_BUFFER_DTS (frame->buffer) = (dts >= timestamp_decrement) ? (dts -
timestamp_decrement) : 0;
/* NOTE: We do not adjust the size here, just the timestamps and duration.
* We also do not drop fully clipped frames. This is because downstream
* MPEG audio decoders still need the data of the frame, even if it gets
* fully clipped later. They do need these frames for their decoding process.
* If these frames were dropped, the decoders would not fully decode all
* of the data from the MPEG stream. */
/* TODO: Should offset/offset_end also be adjusted? */
}
/* Check if this frame can safely be dropped (for example, because it is an
* empty Xing header frame). */
if (G_UNLIKELY (mp3parse->outgoing_frame_is_xing_header)) {
GST_DEBUG_OBJECT (mp3parse, "Marking frame as decode-only / droppable");
mp3parse->outgoing_frame_is_xing_header = FALSE;
GST_BUFFER_DURATION (frame->buffer) = 0;
GST_BUFFER_FLAG_SET (frame->buffer, GST_BUFFER_FLAG_DECODE_ONLY);
GST_BUFFER_FLAG_SET (frame->buffer, GST_BUFFER_FLAG_DROPPABLE);
}
/* 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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