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GST-Tensordecoder-ov_ep/subprojects/gst-plugins-good/gst/audioparsers/gstwavpackparse.c

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/* GStreamer Wavpack parser
* Copyright (C) 2012 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* Copyright (C) 2012 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-wavpackparse
* @title: wavpackparse
* @short_description: Wavpack parser
* @see_also: #GstAmrParse, #GstAACParse
*
* This is an Wavpack parser.
*
* ## Example launch line
* |[
* gst-launch-1.0 filesrc location=abc.wavpack ! wavpackparse ! wavpackdec ! audioresample ! audioconvert ! autoaudiosink
* ]|
*
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include "gstaudioparserselements.h"
#include "gstwavpackparse.h"
#include <gst/base/base.h>
#include <gst/pbutils/pbutils.h>
#include <gst/audio/audio.h>
GST_DEBUG_CATEGORY_STATIC (wavpack_parse_debug);
#define GST_CAT_DEFAULT wavpack_parse_debug
static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-wavpack, "
"depth = (int) [ 1, 32 ], "
"channels = (int) [ 1, 8 ], "
"rate = (int) [ 6000, 192000 ], " "framed = (boolean) TRUE; "
"audio/x-wavpack-correction, " "framed = (boolean) TRUE")
);
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-wavpack; audio/x-wavpack-correction"));
static void gst_wavpack_parse_finalize (GObject * object);
static gboolean gst_wavpack_parse_start (GstBaseParse * parse);
static gboolean gst_wavpack_parse_stop (GstBaseParse * parse);
static GstFlowReturn gst_wavpack_parse_handle_frame (GstBaseParse * parse,
GstBaseParseFrame * frame, gint * skipsize);
static GstCaps *gst_wavpack_parse_get_sink_caps (GstBaseParse * parse,
GstCaps * filter);
static GstFlowReturn gst_wavpack_parse_pre_push_frame (GstBaseParse * parse,
GstBaseParseFrame * frame);
#define gst_wavpack_parse_parent_class parent_class
G_DEFINE_TYPE (GstWavpackParse, gst_wavpack_parse, GST_TYPE_BASE_PARSE);
GST_ELEMENT_REGISTER_DEFINE (wavpackparse, "wavpackparse",
GST_RANK_PRIMARY + 1, GST_TYPE_WAVPACK_PARSE);
static void
gst_wavpack_parse_class_init (GstWavpackParseClass * 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 (wavpack_parse_debug, "wavpackparse", 0,
"Wavpack audio stream parser");
object_class->finalize = gst_wavpack_parse_finalize;
parse_class->start = GST_DEBUG_FUNCPTR (gst_wavpack_parse_start);
parse_class->stop = GST_DEBUG_FUNCPTR (gst_wavpack_parse_stop);
parse_class->handle_frame =
GST_DEBUG_FUNCPTR (gst_wavpack_parse_handle_frame);
parse_class->get_sink_caps =
GST_DEBUG_FUNCPTR (gst_wavpack_parse_get_sink_caps);
parse_class->pre_push_frame =
GST_DEBUG_FUNCPTR (gst_wavpack_parse_pre_push_frame);
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,
"Wavpack audio stream parser", "Codec/Parser/Audio",
"Wavpack parser", "Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>");
}
static void
gst_wavpack_parse_reset (GstWavpackParse * wvparse)
{
wvparse->channels = -1;
wvparse->channel_mask = 0;
wvparse->sample_rate = -1;
wvparse->width = -1;
wvparse->total_samples = 0;
wvparse->sent_codec_tag = FALSE;
}
static void
gst_wavpack_parse_init (GstWavpackParse * wvparse)
{
gst_wavpack_parse_reset (wvparse);
GST_PAD_SET_ACCEPT_INTERSECT (GST_BASE_PARSE_SINK_PAD (wvparse));
GST_PAD_SET_ACCEPT_TEMPLATE (GST_BASE_PARSE_SINK_PAD (wvparse));
}
static void
gst_wavpack_parse_finalize (GObject * object)
{
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_wavpack_parse_start (GstBaseParse * parse)
{
GstWavpackParse *wvparse = GST_WAVPACK_PARSE (parse);
GST_DEBUG_OBJECT (parse, "starting");
gst_wavpack_parse_reset (wvparse);
/* need header at least */
gst_base_parse_set_min_frame_size (GST_BASE_PARSE (wvparse),
sizeof (WavpackHeader));
/* inform baseclass we can come up with ts, based on counters in packets */
gst_base_parse_set_has_timing_info (GST_BASE_PARSE_CAST (wvparse), TRUE);
gst_base_parse_set_syncable (GST_BASE_PARSE_CAST (wvparse), TRUE);
return TRUE;
}
static gboolean
gst_wavpack_parse_stop (GstBaseParse * parse)
{
GST_DEBUG_OBJECT (parse, "stopping");
return TRUE;
}
static gint
gst_wavpack_get_default_channel_mask (gint nchannels)
{
gint channel_mask = 0;
/* Set the default channel mask for the given number of channels.
* It's the same as for WAVE_FORMAT_EXTENDED:
* http://www.microsoft.com/whdc/device/audio/multichaud.mspx
*/
switch (nchannels) {
case 11:
channel_mask |= 0x00400;
channel_mask |= 0x00200;
/* FALLTHROUGH */
case 9:
channel_mask |= 0x00100;
/* FALLTHROUGH */
case 8:
channel_mask |= 0x00080;
channel_mask |= 0x00040;
/* FALLTHROUGH */
case 6:
channel_mask |= 0x00020;
channel_mask |= 0x00010;
/* FALLTHROUGH */
case 4:
channel_mask |= 0x00008;
/* FALLTHROUGH */
case 3:
channel_mask |= 0x00004;
/* FALLTHROUGH */
case 2:
channel_mask |= 0x00002;
channel_mask |= 0x00001;
break;
case 1:
/* For mono use front center */
channel_mask |= 0x00004;
break;
}
return channel_mask;
}
static const struct
{
const guint32 ms_mask;
const GstAudioChannelPosition gst_pos;
} layout_mapping[] = {
{
0x00001, GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT}, {
0x00002, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT}, {
0x00004, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER}, {
0x00008, GST_AUDIO_CHANNEL_POSITION_LFE1}, {
0x00010, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT}, {
0x00020, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT}, {
0x00040, GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER}, {
0x00080, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER}, {
0x00100, GST_AUDIO_CHANNEL_POSITION_REAR_CENTER}, {
0x00200, GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT}, {
0x00400, GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT}, {
0x00800, GST_AUDIO_CHANNEL_POSITION_TOP_CENTER}, {
0x01000, GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_LEFT}, {
0x02000, GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_CENTER}, {
0x04000, GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_RIGHT}, {
0x08000, GST_AUDIO_CHANNEL_POSITION_TOP_REAR_LEFT}, {
0x10000, GST_AUDIO_CHANNEL_POSITION_TOP_REAR_CENTER}, {
0x20000, GST_AUDIO_CHANNEL_POSITION_TOP_REAR_RIGHT}
};
#define MAX_CHANNEL_POSITIONS G_N_ELEMENTS (layout_mapping)
static gboolean
gst_wavpack_get_channel_positions (gint num_channels, gint layout,
GstAudioChannelPosition * pos)
{
gint i, p;
if (num_channels == 1 && layout == 0x00004) {
pos[0] = GST_AUDIO_CHANNEL_POSITION_MONO;
return TRUE;
}
p = 0;
for (i = 0; i < MAX_CHANNEL_POSITIONS; ++i) {
if ((layout & layout_mapping[i].ms_mask) != 0) {
if (p >= num_channels) {
GST_WARNING ("More bits set in the channel layout map than there "
"are channels! Broken file");
return FALSE;
}
if (layout_mapping[i].gst_pos == GST_AUDIO_CHANNEL_POSITION_INVALID) {
GST_WARNING ("Unsupported channel position (mask 0x%08x) in channel "
"layout map - ignoring those channels", layout_mapping[i].ms_mask);
/* what to do? just ignore it and let downstream deal with a channel
* layout that has INVALID positions in it for now ... */
}
pos[p] = layout_mapping[i].gst_pos;
++p;
}
}
if (p != num_channels) {
GST_WARNING ("Only %d bits set in the channel layout map, but there are "
"supposed to be %d channels! Broken file", p, num_channels);
return FALSE;
}
return TRUE;
}
static const guint32 sample_rates[] = {
6000, 8000, 9600, 11025, 12000, 16000, 22050,
24000, 32000, 44100, 48000, 64000, 88200, 96000, 192000
};
#define CHECK(call) { \
if (!call) \
goto read_failed; \
}
/* caller ensures properly sync'ed with enough data */
static gboolean
gst_wavpack_parse_frame_metadata (GstWavpackParse * parse, GstBuffer * buf,
gint skip, WavpackHeader * wph, WavpackInfo * wpi)
{
GstByteReader br;
gint i;
GstMapInfo map;
g_return_val_if_fail (wph != NULL || wpi != NULL, FALSE);
g_return_val_if_fail (gst_buffer_get_size (buf) >=
skip + sizeof (WavpackHeader), FALSE);
gst_buffer_map (buf, &map, GST_MAP_READ);
gst_byte_reader_init (&br, map.data + skip, wph->ckSize + 8);
/* skip past header */
gst_byte_reader_skip_unchecked (&br, sizeof (WavpackHeader));
/* get some basics from header */
i = (wph->flags >> 23) & 0xF;
if (!wpi->rate)
wpi->rate = (i < G_N_ELEMENTS (sample_rates)) ? sample_rates[i] : 44100;
wpi->width = ((wph->flags & 0x3) + 1) * 8;
if (!wpi->channels)
wpi->channels = (wph->flags & 0x4) ? 1 : 2;
if (!wpi->channel_mask)
wpi->channel_mask = 5 - wpi->channels;
/* need to dig metadata blocks for some more */
while (gst_byte_reader_get_remaining (&br)) {
guint32 size = 0;
guint16 size2 = 0;
guint8 c, id;
const guint8 *data;
GstByteReader mbr;
CHECK (gst_byte_reader_get_uint8 (&br, &id));
CHECK (gst_byte_reader_get_uint8 (&br, &c));
if (id & ID_LARGE)
CHECK (gst_byte_reader_get_uint16_le (&br, &size2));
size = size2;
size <<= 8;
size += c;
size <<= 1;
if (id & ID_ODD_SIZE) {
if (size == 0)
goto read_failed;
size--;
}
CHECK (gst_byte_reader_get_data (&br, size + (size & 1), &data));
gst_byte_reader_init (&mbr, data, size);
/* 0x1f is the metadata id mask and 0x20 flag is for later extensions
* that do not need to be handled by the decoder */
switch (id & 0x3f) {
case ID_WVC_BITSTREAM:
GST_LOG_OBJECT (parse, "correction bitstream");
wpi->correction = TRUE;
break;
case ID_WV_BITSTREAM:
case ID_WVX_BITSTREAM:
case ID_WVX_NEW_BITSTREAM:
break;
case ID_SAMPLE_RATE:
if (size == 3) {
CHECK (gst_byte_reader_get_uint24_le (&mbr, &wpi->rate));
GST_LOG_OBJECT (parse, "updated with custom rate %d", wpi->rate);
} else {
GST_DEBUG_OBJECT (parse, "unexpected size for SAMPLE_RATE metadata");
}
break;
case ID_CHANNEL_INFO:
{
guint16 channels;
guint32 mask = 0;
if (size == 6) {
CHECK (gst_byte_reader_get_uint16_le (&mbr, &channels));
channels = channels & 0xFFF;
CHECK (gst_byte_reader_get_uint24_le (&mbr, &mask));
} else if (size) {
CHECK (gst_byte_reader_get_uint8 (&mbr, &c));
channels = c;
while (gst_byte_reader_get_uint8 (&mbr, &c))
mask |= (((guint32) c) << 8);
} else {
GST_DEBUG_OBJECT (parse, "unexpected size for CHANNEL_INFO metadata");
break;
}
wpi->channels = channels;
wpi->channel_mask = mask;
break;
}
default:
GST_LOG_OBJECT (parse, "unparsed ID 0x%x", id);
break;
}
}
gst_buffer_unmap (buf, &map);
return TRUE;
/* ERRORS */
read_failed:
{
gst_buffer_unmap (buf, &map);
GST_DEBUG_OBJECT (parse, "short read while parsing metadata");
/* let's look the other way anyway */
return TRUE;
}
}
/* caller ensures properly sync'ed with enough data */
static gboolean
gst_wavpack_parse_frame_header (GstWavpackParse * parse, GstBuffer * buf,
gint skip, WavpackHeader * _wph)
{
GstByteReader br;
WavpackHeader wph = { {0,}, 0, };
GstMapInfo map;
gboolean hdl = TRUE;
g_return_val_if_fail (gst_buffer_get_size (buf) >=
skip + sizeof (WavpackHeader), FALSE);
gst_buffer_map (buf, &map, GST_MAP_READ);
gst_byte_reader_init (&br, map.data, map.size);
/* marker */
gst_byte_reader_skip_unchecked (&br, skip + 4);
/* read */
hdl &= gst_byte_reader_get_uint32_le (&br, &wph.ckSize);
hdl &= gst_byte_reader_get_uint16_le (&br, &wph.version);
hdl &= gst_byte_reader_get_uint8 (&br, &wph.track_no);
hdl &= gst_byte_reader_get_uint8 (&br, &wph.index_no);
hdl &= gst_byte_reader_get_uint32_le (&br, &wph.total_samples);
hdl &= gst_byte_reader_get_uint32_le (&br, &wph.block_index);
hdl &= gst_byte_reader_get_uint32_le (&br, &wph.block_samples);
hdl &= gst_byte_reader_get_uint32_le (&br, &wph.flags);
hdl &= gst_byte_reader_get_uint32_le (&br, &wph.crc);
if (!hdl)
GST_WARNING_OBJECT (parse, "Error reading header");
/* dump */
GST_LOG_OBJECT (parse, "size %d", wph.ckSize);
GST_LOG_OBJECT (parse, "version 0x%x", wph.version);
GST_LOG_OBJECT (parse, "total samples %d", wph.total_samples);
GST_LOG_OBJECT (parse, "block index %d", wph.block_index);
GST_LOG_OBJECT (parse, "block samples %d", wph.block_samples);
GST_LOG_OBJECT (parse, "flags 0x%x", wph.flags);
GST_LOG_OBJECT (parse, "crc 0x%x", wph.flags);
if (!parse->total_samples && wph.block_index == 0 && wph.total_samples != -1) {
GST_DEBUG_OBJECT (parse, "determined duration of %u samples",
wph.total_samples);
parse->total_samples = wph.total_samples;
}
if (_wph)
*_wph = wph;
gst_buffer_unmap (buf, &map);
return TRUE;
}
static GstFlowReturn
gst_wavpack_parse_handle_frame (GstBaseParse * parse,
GstBaseParseFrame * frame, gint * skipsize)
{
GstWavpackParse *wvparse = GST_WAVPACK_PARSE (parse);
GstBuffer *buf = frame->buffer;
GstByteReader reader;
gint off;
guint rate, chans, width, mask;
gboolean lost_sync, draining, final;
guint frmsize = 0;
WavpackHeader wph;
WavpackInfo wpi = { 0, };
GstMapInfo map;
if (G_UNLIKELY (gst_buffer_get_size (buf) < sizeof (WavpackHeader)))
return FALSE;
gst_buffer_map (buf, &map, GST_MAP_READ);
gst_byte_reader_init (&reader, map.data, map.size);
/* scan for 'wvpk' marker */
off = gst_byte_reader_masked_scan_uint32 (&reader, 0xffffffff, 0x7776706b,
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 skip;
}
/* possible frame header, but not at offset 0? skip bytes before sync */
if (off > 0) {
*skipsize = off;
goto skip;
}
/* make sure the values in the frame header look sane */
gst_wavpack_parse_frame_header (wvparse, buf, 0, &wph);
frmsize = wph.ckSize + 8;
/* need the entire frame for parsing */
if (gst_byte_reader_get_remaining (&reader) < frmsize)
goto more;
/* got a frame, now we can dig for some more metadata */
GST_LOG_OBJECT (parse, "got frame");
gst_wavpack_parse_frame_metadata (wvparse, buf, 0, &wph, &wpi);
lost_sync = GST_BASE_PARSE_LOST_SYNC (parse);
draining = GST_BASE_PARSE_DRAINING (parse);
while (!(final = (wph.flags & FLAG_FINAL_BLOCK)) || (lost_sync && !draining)) {
guint32 word = 0;
GST_LOG_OBJECT (wvparse, "checking next frame syncword; "
"lost_sync: %d, draining: %d, final: %d", lost_sync, draining, final);
if (!gst_byte_reader_skip (&reader, wph.ckSize + 8) ||
!gst_byte_reader_peek_uint32_be (&reader, &word)) {
GST_DEBUG_OBJECT (wvparse, "... but not sufficient data");
frmsize += 4;
goto more;
} else {
if (word != 0x7776706b) {
GST_DEBUG_OBJECT (wvparse, "0x%x not OK", word);
*skipsize = off + 2;
goto skip;
}
/* need to parse each frame/block for metadata if several ones */
if (!final) {
gint av;
GST_LOG_OBJECT (wvparse, "checking frame at offset %d (0x%x)",
frmsize, frmsize);
av = gst_byte_reader_get_remaining (&reader);
if (av < sizeof (WavpackHeader)) {
frmsize += sizeof (WavpackHeader);
goto more;
}
gst_wavpack_parse_frame_header (wvparse, buf, frmsize, &wph);
off = frmsize;
frmsize += wph.ckSize + 8;
if (av < wph.ckSize + 8)
goto more;
gst_wavpack_parse_frame_metadata (wvparse, buf, off, &wph, &wpi);
/* could also check for matching block_index and block_samples ?? */
}
}
/* resynced if we make it here */
lost_sync = FALSE;
}
rate = wpi.rate;
width = wpi.width;
chans = wpi.channels;
mask = wpi.channel_mask;
GST_LOG_OBJECT (parse, "rate: %u, width: %u, chans: %u", rate, width, chans);
GST_BUFFER_PTS (buf) =
gst_util_uint64_scale_int (wph.block_index, GST_SECOND, rate);
GST_BUFFER_DTS (buf) = GST_BUFFER_PTS (buf);
GST_BUFFER_DURATION (buf) =
gst_util_uint64_scale_int (wph.block_index + wph.block_samples,
GST_SECOND, rate) - GST_BUFFER_PTS (buf);
if (G_UNLIKELY (wvparse->sample_rate != rate || wvparse->channels != chans
|| wvparse->width != width || wvparse->channel_mask != mask)) {
GstCaps *caps;
if (wpi.correction) {
caps = gst_caps_new_simple ("audio/x-wavpack-correction",
"framed", G_TYPE_BOOLEAN, TRUE, NULL);
} else {
caps = gst_caps_new_simple ("audio/x-wavpack",
"channels", G_TYPE_INT, chans,
"rate", G_TYPE_INT, rate,
"depth", G_TYPE_INT, width, "framed", G_TYPE_BOOLEAN, TRUE, NULL);
if (!mask)
mask = gst_wavpack_get_default_channel_mask (wvparse->channels);
if (mask != 0) {
GstAudioChannelPosition pos[64] =
{ GST_AUDIO_CHANNEL_POSITION_INVALID, };
guint64 gmask;
if (!gst_wavpack_get_channel_positions (chans, mask, pos)) {
GST_WARNING_OBJECT (wvparse, "Failed to determine channel layout");
} else {
gst_audio_channel_positions_to_mask (pos, chans, FALSE, &gmask);
if (gmask)
gst_caps_set_simple (caps,
"channel-mask", GST_TYPE_BITMASK, gmask, NULL);
}
}
}
gst_pad_set_caps (GST_BASE_PARSE_SRC_PAD (parse), caps);
gst_caps_unref (caps);
wvparse->sample_rate = rate;
wvparse->channels = chans;
wvparse->width = width;
wvparse->channel_mask = mask;
if (wvparse->total_samples) {
GST_DEBUG_OBJECT (wvparse, "setting duration");
gst_base_parse_set_duration (GST_BASE_PARSE (wvparse),
GST_FORMAT_TIME, gst_util_uint64_scale_int (wvparse->total_samples,
GST_SECOND, wvparse->sample_rate), 0);
}
}
/* return to normal size */
gst_base_parse_set_min_frame_size (parse, sizeof (WavpackHeader));
gst_buffer_unmap (buf, &map);
return gst_base_parse_finish_frame (parse, frame, frmsize);
skip:
gst_buffer_unmap (buf, &map);
GST_LOG_OBJECT (wvparse, "skipping %d", *skipsize);
return GST_FLOW_OK;
more:
gst_buffer_unmap (buf, &map);
GST_LOG_OBJECT (wvparse, "need at least %u", frmsize);
gst_base_parse_set_min_frame_size (parse, frmsize);
*skipsize = 0;
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, "framed");
}
}
static GstCaps *
gst_wavpack_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 framed 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;
}
static GstFlowReturn
gst_wavpack_parse_pre_push_frame (GstBaseParse * parse,
GstBaseParseFrame * frame)
{
GstWavpackParse *wavpackparse = GST_WAVPACK_PARSE (parse);
if (!wavpackparse->sent_codec_tag) {
GstTagList *taglist;
GstCaps *caps;
/* codec tag */
caps = gst_pad_get_current_caps (GST_BASE_PARSE_SRC_PAD (parse));
if (G_UNLIKELY (caps == NULL)) {
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;
}
}
taglist = gst_tag_list_new_empty ();
gst_pb_utils_add_codec_description_to_tag_list (taglist,
GST_TAG_AUDIO_CODEC, caps);
gst_caps_unref (caps);
gst_base_parse_merge_tags (parse, taglist, GST_TAG_MERGE_REPLACE);
gst_tag_list_unref (taglist);
/* also signals the end of first-frame processing */
wavpackparse->sent_codec_tag = TRUE;
}
frame->flags |= GST_BASE_PARSE_FRAME_FLAG_CLIP;
return GST_FLOW_OK;
}
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