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Previous patch would try to unref a buffer that was pushed downstream. Instead only unref when/if needed and keep usage of the cleanup: goto block
617 lines
19 KiB
C
617 lines
19 KiB
C
/* GStreamer DCA parser
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* Copyright (C) 2010 Tim-Philipp Müller <tim centricular net>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:element-dcaparse
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* @short_description: DCA (DTS Coherent Acoustics) parser
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* @see_also: #GstAmrParse, #GstAACParse, #GstAc3Parse
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*
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* This is a DCA (DTS Coherent Acoustics) parser.
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*
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* <refsect2>
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* <title>Example launch line</title>
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* |[
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* gst-launch-1.0 filesrc location=abc.dts ! dcaparse ! dtsdec ! audioresample ! audioconvert ! autoaudiosink
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* ]|
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* </refsect2>
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*/
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/* TODO:
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* - should accept framed and unframed input (needs decodebin fixes first)
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* - seeking in raw .dts files doesn't seem to work, but duration estimate ok
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*
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* - if frames have 'odd' durations, the frame durations (plus timestamps)
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* aren't adjusted up occasionally to make up for rounding error gaps.
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* (e.g. if 512 samples per frame @ 48kHz = 10.666666667 ms/frame)
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <string.h>
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#include "gstdcaparse.h"
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#include <gst/base/base.h>
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#include <gst/pbutils/pbutils.h>
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GST_DEBUG_CATEGORY_STATIC (dca_parse_debug);
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#define GST_CAT_DEFAULT dca_parse_debug
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static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-dts,"
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" framed = (boolean) true,"
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" channels = (int) [ 1, 8 ],"
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" rate = (int) [ 8000, 192000 ],"
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" depth = (int) { 14, 16 },"
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" endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, "
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" block-size = (int) [ 1, MAX], " " frame-size = (int) [ 1, MAX]"));
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static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-dts; " "audio/x-private1-dts"));
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static void gst_dca_parse_finalize (GObject * object);
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static gboolean gst_dca_parse_start (GstBaseParse * parse);
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static gboolean gst_dca_parse_stop (GstBaseParse * parse);
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static GstFlowReturn gst_dca_parse_handle_frame (GstBaseParse * parse,
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GstBaseParseFrame * frame, gint * skipsize);
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static GstFlowReturn gst_dca_parse_pre_push_frame (GstBaseParse * parse,
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GstBaseParseFrame * frame);
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static GstCaps *gst_dca_parse_get_sink_caps (GstBaseParse * parse,
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GstCaps * filter);
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static gboolean gst_dca_parse_set_sink_caps (GstBaseParse * parse,
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GstCaps * caps);
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#define gst_dca_parse_parent_class parent_class
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G_DEFINE_TYPE (GstDcaParse, gst_dca_parse, GST_TYPE_BASE_PARSE);
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static void
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gst_dca_parse_class_init (GstDcaParseClass * klass)
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{
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GstBaseParseClass *parse_class = GST_BASE_PARSE_CLASS (klass);
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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GObjectClass *object_class = G_OBJECT_CLASS (klass);
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GST_DEBUG_CATEGORY_INIT (dca_parse_debug, "dcaparse", 0,
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"DCA audio stream parser");
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object_class->finalize = gst_dca_parse_finalize;
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parse_class->start = GST_DEBUG_FUNCPTR (gst_dca_parse_start);
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parse_class->stop = GST_DEBUG_FUNCPTR (gst_dca_parse_stop);
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parse_class->handle_frame = GST_DEBUG_FUNCPTR (gst_dca_parse_handle_frame);
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parse_class->pre_push_frame =
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GST_DEBUG_FUNCPTR (gst_dca_parse_pre_push_frame);
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parse_class->get_sink_caps = GST_DEBUG_FUNCPTR (gst_dca_parse_get_sink_caps);
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parse_class->set_sink_caps = GST_DEBUG_FUNCPTR (gst_dca_parse_set_sink_caps);
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gst_element_class_add_static_pad_template (element_class, &sink_template);
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gst_element_class_add_static_pad_template (element_class, &src_template);
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gst_element_class_set_static_metadata (element_class,
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"DTS Coherent Acoustics audio stream parser", "Codec/Parser/Audio",
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"DCA parser", "Tim-Philipp Müller <tim centricular net>");
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}
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static void
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gst_dca_parse_reset (GstDcaParse * dcaparse)
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{
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dcaparse->channels = -1;
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dcaparse->rate = -1;
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dcaparse->depth = -1;
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dcaparse->endianness = -1;
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dcaparse->block_size = -1;
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dcaparse->frame_size = -1;
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dcaparse->last_sync = 0;
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dcaparse->sent_codec_tag = FALSE;
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}
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static void
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gst_dca_parse_init (GstDcaParse * dcaparse)
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{
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gst_base_parse_set_min_frame_size (GST_BASE_PARSE (dcaparse),
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DCA_MIN_FRAMESIZE);
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gst_dca_parse_reset (dcaparse);
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dcaparse->baseparse_chainfunc =
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GST_BASE_PARSE_SINK_PAD (GST_BASE_PARSE (dcaparse))->chainfunc;
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GST_PAD_SET_ACCEPT_INTERSECT (GST_BASE_PARSE_SINK_PAD (dcaparse));
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GST_PAD_SET_ACCEPT_TEMPLATE (GST_BASE_PARSE_SINK_PAD (dcaparse));
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}
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static void
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gst_dca_parse_finalize (GObject * object)
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{
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static gboolean
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gst_dca_parse_start (GstBaseParse * parse)
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{
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GstDcaParse *dcaparse = GST_DCA_PARSE (parse);
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GST_DEBUG_OBJECT (parse, "starting");
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gst_dca_parse_reset (dcaparse);
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return TRUE;
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}
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static gboolean
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gst_dca_parse_stop (GstBaseParse * parse)
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{
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GST_DEBUG_OBJECT (parse, "stopping");
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return TRUE;
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}
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static gboolean
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gst_dca_parse_parse_header (GstDcaParse * dcaparse,
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const GstByteReader * reader, guint * frame_size,
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guint * sample_rate, guint * channels, guint * depth,
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gint * endianness, guint * num_blocks, guint * samples_per_block,
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gboolean * terminator)
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{
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static const int sample_rates[16] = { 0, 8000, 16000, 32000, 0, 0, 11025,
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22050, 44100, 0, 0, 12000, 24000, 48000, 96000, 192000
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};
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static const guint8 channels_table[16] = { 1, 2, 2, 2, 2, 3, 3, 4, 4, 5,
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6, 6, 6, 7, 8, 8
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};
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GstByteReader r = *reader;
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guint16 hdr[8];
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guint32 marker;
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guint chans, lfe, i;
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if (gst_byte_reader_get_remaining (&r) < (4 + sizeof (hdr)))
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return FALSE;
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marker = gst_byte_reader_peek_uint32_be_unchecked (&r);
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/* raw big endian or 14-bit big endian */
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if (marker == 0x7FFE8001 || marker == 0x1FFFE800) {
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for (i = 0; i < G_N_ELEMENTS (hdr); ++i)
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hdr[i] = gst_byte_reader_get_uint16_be_unchecked (&r);
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} else
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/* raw little endian or 14-bit little endian */
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if (marker == 0xFE7F0180 || marker == 0xFF1F00E8) {
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for (i = 0; i < G_N_ELEMENTS (hdr); ++i)
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hdr[i] = gst_byte_reader_get_uint16_le_unchecked (&r);
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} else {
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return FALSE;
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}
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GST_LOG_OBJECT (dcaparse, "dts sync marker 0x%08x at offset %u", marker,
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gst_byte_reader_get_pos (reader));
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/* 14-bit mode */
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if (marker == 0x1FFFE800 || marker == 0xFF1F00E8) {
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if ((hdr[2] & 0xFFF0) != 0x07F0)
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return FALSE;
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/* discard top 2 bits (2 void), shift in 2 */
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hdr[0] = (hdr[0] << 2) | ((hdr[1] >> 12) & 0x0003);
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/* discard top 4 bits (2 void, 2 shifted into hdr[0]), shift in 4 etc. */
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hdr[1] = (hdr[1] << 4) | ((hdr[2] >> 10) & 0x000F);
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hdr[2] = (hdr[2] << 6) | ((hdr[3] >> 8) & 0x003F);
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hdr[3] = (hdr[3] << 8) | ((hdr[4] >> 6) & 0x00FF);
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hdr[4] = (hdr[4] << 10) | ((hdr[5] >> 4) & 0x03FF);
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hdr[5] = (hdr[5] << 12) | ((hdr[6] >> 2) & 0x0FFF);
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hdr[6] = (hdr[6] << 14) | ((hdr[7] >> 0) & 0x3FFF);
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g_assert (hdr[0] == 0x7FFE && hdr[1] == 0x8001);
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}
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GST_LOG_OBJECT (dcaparse, "frame header: %04x%04x%04x%04x",
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hdr[2], hdr[3], hdr[4], hdr[5]);
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*terminator = (hdr[2] & 0x80) ? FALSE : TRUE;
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*samples_per_block = ((hdr[2] >> 10) & 0x1f) + 1;
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*num_blocks = ((hdr[2] >> 2) & 0x7F) + 1;
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*frame_size = (((hdr[2] & 0x03) << 12) | (hdr[3] >> 4)) + 1;
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chans = ((hdr[3] & 0x0F) << 2) | (hdr[4] >> 14);
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*sample_rate = sample_rates[(hdr[4] >> 10) & 0x0F];
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lfe = (hdr[5] >> 9) & 0x03;
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GST_TRACE_OBJECT (dcaparse, "frame size %u, num_blocks %u, rate %u, "
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"samples per block %u", *frame_size, *num_blocks, *sample_rate,
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*samples_per_block);
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if (*num_blocks < 6 || *frame_size < 96 || *sample_rate == 0)
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return FALSE;
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if (marker == 0x1FFFE800 || marker == 0xFF1F00E8)
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*frame_size = (*frame_size * 16) / 14; /* FIXME: round up? */
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if (chans < G_N_ELEMENTS (channels_table))
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*channels = channels_table[chans] + ((lfe) ? 1 : 0);
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else
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*channels = 0;
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if (depth)
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*depth = (marker == 0x1FFFE800 || marker == 0xFF1F00E8) ? 14 : 16;
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if (endianness)
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*endianness = (marker == 0xFE7F0180 || marker == 0xFF1F00E8) ?
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G_LITTLE_ENDIAN : G_BIG_ENDIAN;
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GST_TRACE_OBJECT (dcaparse, "frame size %u, channels %u, rate %u, "
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"num_blocks %u, samples_per_block %u", *frame_size, *channels,
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*sample_rate, *num_blocks, *samples_per_block);
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return TRUE;
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}
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static gint
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gst_dca_parse_find_sync (GstDcaParse * dcaparse, GstByteReader * reader,
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gsize bufsize, guint32 * sync)
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{
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guint32 best_sync = 0;
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guint best_offset = G_MAXUINT;
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gint off;
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/* FIXME: verify syncs via _parse_header() here already */
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/* Raw little endian */
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off = gst_byte_reader_masked_scan_uint32 (reader, 0xffffffff, 0xfe7f0180,
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0, bufsize);
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if (off >= 0 && off < best_offset) {
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best_offset = off;
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best_sync = 0xfe7f0180;
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}
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/* Raw big endian */
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off = gst_byte_reader_masked_scan_uint32 (reader, 0xffffffff, 0x7ffe8001,
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0, bufsize);
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if (off >= 0 && off < best_offset) {
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best_offset = off;
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best_sync = 0x7ffe8001;
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}
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/* FIXME: check next 2 bytes as well for 14-bit formats (but then don't
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* forget to adjust the *skipsize= in _check_valid_frame() */
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/* 14-bit little endian */
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off = gst_byte_reader_masked_scan_uint32 (reader, 0xffffffff, 0xff1f00e8,
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0, bufsize);
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if (off >= 0 && off < best_offset) {
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best_offset = off;
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best_sync = 0xff1f00e8;
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}
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/* 14-bit big endian */
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off = gst_byte_reader_masked_scan_uint32 (reader, 0xffffffff, 0x1fffe800,
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0, bufsize);
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if (off >= 0 && off < best_offset) {
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best_offset = off;
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best_sync = 0x1fffe800;
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}
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if (best_offset == G_MAXUINT)
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return -1;
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*sync = best_sync;
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return best_offset;
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}
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static GstFlowReturn
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gst_dca_parse_handle_frame (GstBaseParse * parse,
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GstBaseParseFrame * frame, gint * skipsize)
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{
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GstDcaParse *dcaparse = GST_DCA_PARSE (parse);
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GstBuffer *buf = frame->buffer;
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GstByteReader r;
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gboolean parser_in_sync;
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gboolean terminator;
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guint32 sync = 0;
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guint size = 0, rate, chans, num_blocks, samples_per_block, depth;
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gint block_size;
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gint endianness;
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gint off = -1;
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GstMapInfo map;
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GstFlowReturn ret = GST_FLOW_EOS;
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gsize extra_size = 0;
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gst_buffer_map (buf, &map, GST_MAP_READ);
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if (G_UNLIKELY (map.size < 16)) {
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*skipsize = 1;
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goto cleanup;
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}
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parser_in_sync = !GST_BASE_PARSE_LOST_SYNC (parse);
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gst_byte_reader_init (&r, map.data, map.size);
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if (G_LIKELY (parser_in_sync && dcaparse->last_sync != 0)) {
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off = gst_byte_reader_masked_scan_uint32 (&r, 0xffffffff,
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dcaparse->last_sync, 0, map.size);
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}
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if (G_UNLIKELY (off < 0)) {
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off = gst_dca_parse_find_sync (dcaparse, &r, map.size, &sync);
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}
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/* didn't find anything that looks like a sync word, skip */
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if (off < 0) {
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*skipsize = map.size - 3;
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GST_DEBUG_OBJECT (dcaparse, "no sync, skipping %d bytes", *skipsize);
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goto cleanup;
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}
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GST_LOG_OBJECT (parse, "possible sync %08x at buffer offset %d", sync, off);
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/* possible frame header, but not at offset 0? skip bytes before sync */
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if (off > 0) {
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*skipsize = off;
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goto cleanup;
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}
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/* make sure the values in the frame header look sane */
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if (!gst_dca_parse_parse_header (dcaparse, &r, &size, &rate, &chans, &depth,
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&endianness, &num_blocks, &samples_per_block, &terminator)) {
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*skipsize = 4;
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goto cleanup;
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}
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GST_LOG_OBJECT (parse, "got frame, sync %08x, size %u, rate %d, channels %d",
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sync, size, rate, chans);
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dcaparse->last_sync = sync;
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/* FIXME: Don't look for a second syncword, there are streams out there
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* that consistently contain garbage between every frame so we never ever
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* find a second consecutive syncword.
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* See https://bugzilla.gnome.org/show_bug.cgi?id=738237
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*/
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#if 0
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parser_draining = GST_BASE_PARSE_DRAINING (parse);
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if (!parser_in_sync && !parser_draining) {
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/* check for second frame to be sure */
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GST_DEBUG_OBJECT (dcaparse, "resyncing; checking next frame syncword");
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if (map.size >= (size + 16)) {
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guint s2, r2, c2, n2, s3;
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gboolean t;
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GST_MEMDUMP ("buf", map.data, size + 16);
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gst_byte_reader_init (&r, map.data, map.size);
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gst_byte_reader_skip_unchecked (&r, size);
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if (!gst_dca_parse_parse_header (dcaparse, &r, &s2, &r2, &c2, NULL, NULL,
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&n2, &s3, &t)) {
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GST_DEBUG_OBJECT (dcaparse, "didn't find second syncword");
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*skipsize = 4;
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goto cleanup;
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}
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/* ok, got sync now, let's assume constant frame size */
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gst_base_parse_set_min_frame_size (parse, size);
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} else {
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/* wait for some more data */
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GST_LOG_OBJECT (dcaparse,
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"next sync out of reach (%" G_GSIZE_FORMAT " < %u)", map.size,
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size + 16);
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goto cleanup;
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}
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}
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#endif
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/* found frame */
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ret = GST_FLOW_OK;
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/* metadata handling */
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block_size = num_blocks * samples_per_block;
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if (G_UNLIKELY (dcaparse->rate != rate || dcaparse->channels != chans
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|| dcaparse->depth != depth || dcaparse->endianness != endianness
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|| (!terminator && dcaparse->block_size != block_size)
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|| (size != dcaparse->frame_size))) {
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GstCaps *caps;
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caps = gst_caps_new_simple ("audio/x-dts",
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"framed", G_TYPE_BOOLEAN, TRUE,
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"rate", G_TYPE_INT, rate, "channels", G_TYPE_INT, chans,
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"endianness", G_TYPE_INT, endianness, "depth", G_TYPE_INT, depth,
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|
"block-size", G_TYPE_INT, block_size, "frame-size", G_TYPE_INT, size,
|
|
NULL);
|
|
gst_pad_set_caps (GST_BASE_PARSE_SRC_PAD (parse), caps);
|
|
gst_caps_unref (caps);
|
|
|
|
dcaparse->rate = rate;
|
|
dcaparse->channels = chans;
|
|
dcaparse->depth = depth;
|
|
dcaparse->endianness = endianness;
|
|
dcaparse->block_size = block_size;
|
|
dcaparse->frame_size = size;
|
|
|
|
gst_base_parse_set_frame_rate (parse, rate, block_size, 0, 0);
|
|
}
|
|
|
|
cleanup:
|
|
/* it is possible that DTS HD substream after DTS core */
|
|
if (parse->flags & GST_BASE_PARSE_FLAG_DRAINING || map.size >= size + 9) {
|
|
extra_size = 0;
|
|
if (map.size >= size + 9) {
|
|
const guint8 *next = map.data + size;
|
|
/* Check for DTS_SYNCWORD_SUBSTREAM */
|
|
if (next[0] == 0x64 && next[1] == 0x58 && next[2] == 0x20
|
|
&& next[3] == 0x25) {
|
|
/* 7.4.1 Extension Substream Header */
|
|
GstBitReader reader;
|
|
gst_bit_reader_init (&reader, next + 4, 5);
|
|
gst_bit_reader_skip (&reader, 8 + 2); /* skip UserDefinedBits and nExtSSIndex) */
|
|
if (gst_bit_reader_get_bits_uint8_unchecked (&reader, 1) == 0) {
|
|
gst_bit_reader_skip (&reader, 8);
|
|
extra_size =
|
|
gst_bit_reader_get_bits_uint32_unchecked (&reader, 16) + 1;
|
|
} else {
|
|
gst_bit_reader_skip (&reader, 12);
|
|
extra_size =
|
|
gst_bit_reader_get_bits_uint32_unchecked (&reader, 20) + 1;
|
|
}
|
|
}
|
|
}
|
|
gst_buffer_unmap (buf, &map);
|
|
if (ret == GST_FLOW_OK && size + extra_size <= map.size) {
|
|
ret = gst_base_parse_finish_frame (parse, frame, size + extra_size);
|
|
} else {
|
|
ret = GST_FLOW_OK;
|
|
}
|
|
} else {
|
|
gst_buffer_unmap (buf, &map);
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
/*
|
|
* MPEG-PS private1 streams add a 2 bytes "Audio Substream Headers" for each
|
|
* buffer (not each frame) with the offset of the next frame's start.
|
|
* These 2 bytes can be dropped safely as they do not include any timing
|
|
* information, only the offset to the start of the next frame.
|
|
* See gstac3parse.c for a more detailed description.
|
|
* */
|
|
|
|
static GstFlowReturn
|
|
gst_dca_parse_chain_priv (GstPad * pad, GstObject * parent, GstBuffer * buffer)
|
|
{
|
|
GstDcaParse *dcaparse = GST_DCA_PARSE (parent);
|
|
GstFlowReturn ret;
|
|
GstBuffer *newbuf;
|
|
gsize size;
|
|
|
|
size = gst_buffer_get_size (buffer);
|
|
if (size >= 2) {
|
|
newbuf = gst_buffer_copy_region (buffer, GST_BUFFER_COPY_ALL, 2, size - 2);
|
|
gst_buffer_unref (buffer);
|
|
ret = dcaparse->baseparse_chainfunc (pad, parent, newbuf);
|
|
} else {
|
|
gst_buffer_unref (buffer);
|
|
ret = GST_FLOW_OK;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
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_dca_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 gboolean
|
|
gst_dca_parse_set_sink_caps (GstBaseParse * parse, GstCaps * caps)
|
|
{
|
|
GstStructure *s;
|
|
GstDcaParse *dcaparse = GST_DCA_PARSE (parse);
|
|
|
|
s = gst_caps_get_structure (caps, 0);
|
|
if (gst_structure_has_name (s, "audio/x-private1-dts")) {
|
|
gst_pad_set_chain_function (parse->sinkpad, gst_dca_parse_chain_priv);
|
|
} else {
|
|
gst_pad_set_chain_function (parse->sinkpad, dcaparse->baseparse_chainfunc);
|
|
}
|
|
return TRUE;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_dca_parse_pre_push_frame (GstBaseParse * parse, GstBaseParseFrame * frame)
|
|
{
|
|
GstDcaParse *dcaparse = GST_DCA_PARSE (parse);
|
|
|
|
if (!dcaparse->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 */
|
|
dcaparse->sent_codec_tag = TRUE;
|
|
}
|
|
|
|
return GST_FLOW_OK;
|
|
}
|