mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-09-09 05:38:58 +00:00
The pre_push_frame default clipping behaviour was introduced in 2010 with commit30be03004e
and modified with commit4163969a24
in 2011, when most parsers didn't implement a pre_push_frame yet. Not having it meant that clipping was done by default. Those that did implement a pre_push_frame (flacparse and mpegaudioparse) at the time, had the flag adjusted as part of the 2011 refactor work. All other parsers got a pre_push_frame vfunc implementation only in 2013, but seem to have forgot to keep the clipping behaviour, as was done automatically when a pre_push_frame implementation doesn't exist for the parser. aacparse lost it with commit91d4abcea
in July 2013; the others in Dec 2013 as part of AUDIO_CODEC tag posting in commits6f89b430e
,d2ab5199b
,29f2cae12
,753d3c23a
and292780574
.
710 lines
21 KiB
C
710 lines
21 KiB
C
/* GStreamer Wavpack parser
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* Copyright (C) 2012 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
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* Copyright (C) 2012 Nokia Corporation. All rights reserved.
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* Contact: Stefan Kost <stefan.kost@nokia.com>
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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-wavpackparse
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* @title: wavpackparse
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* @short_description: Wavpack parser
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* @see_also: #GstAmrParse, #GstAACParse
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*
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* This is an Wavpack parser.
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*
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* ## Example launch line
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* |[
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* gst-launch-1.0 filesrc location=abc.wavpack ! wavpackparse ! wavpackdec ! audioresample ! audioconvert ! autoaudiosink
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* ]|
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*
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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 "gstwavpackparse.h"
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#include <gst/base/base.h>
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#include <gst/pbutils/pbutils.h>
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#include <gst/audio/audio.h>
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GST_DEBUG_CATEGORY_STATIC (wavpack_parse_debug);
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#define GST_CAT_DEFAULT wavpack_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-wavpack, "
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"depth = (int) [ 1, 32 ], "
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"channels = (int) [ 1, 8 ], "
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"rate = (int) [ 6000, 192000 ], " "framed = (boolean) TRUE; "
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"audio/x-wavpack-correction, " "framed = (boolean) TRUE")
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);
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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-wavpack; audio/x-wavpack-correction"));
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static void gst_wavpack_parse_finalize (GObject * object);
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static gboolean gst_wavpack_parse_start (GstBaseParse * parse);
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static gboolean gst_wavpack_parse_stop (GstBaseParse * parse);
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static GstFlowReturn gst_wavpack_parse_handle_frame (GstBaseParse * parse,
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GstBaseParseFrame * frame, gint * skipsize);
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static GstCaps *gst_wavpack_parse_get_sink_caps (GstBaseParse * parse,
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GstCaps * filter);
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static GstFlowReturn gst_wavpack_parse_pre_push_frame (GstBaseParse * parse,
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GstBaseParseFrame * frame);
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#define gst_wavpack_parse_parent_class parent_class
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G_DEFINE_TYPE (GstWavpackParse, gst_wavpack_parse, GST_TYPE_BASE_PARSE);
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static void
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gst_wavpack_parse_class_init (GstWavpackParseClass * 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 (wavpack_parse_debug, "wavpackparse", 0,
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"Wavpack audio stream parser");
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object_class->finalize = gst_wavpack_parse_finalize;
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parse_class->start = GST_DEBUG_FUNCPTR (gst_wavpack_parse_start);
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parse_class->stop = GST_DEBUG_FUNCPTR (gst_wavpack_parse_stop);
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parse_class->handle_frame =
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GST_DEBUG_FUNCPTR (gst_wavpack_parse_handle_frame);
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parse_class->get_sink_caps =
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GST_DEBUG_FUNCPTR (gst_wavpack_parse_get_sink_caps);
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parse_class->pre_push_frame =
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GST_DEBUG_FUNCPTR (gst_wavpack_parse_pre_push_frame);
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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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"Wavpack audio stream parser", "Codec/Parser/Audio",
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"Wavpack parser", "Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>");
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}
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static void
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gst_wavpack_parse_reset (GstWavpackParse * wvparse)
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{
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wvparse->channels = -1;
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wvparse->channel_mask = 0;
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wvparse->sample_rate = -1;
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wvparse->width = -1;
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wvparse->total_samples = 0;
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wvparse->sent_codec_tag = FALSE;
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}
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static void
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gst_wavpack_parse_init (GstWavpackParse * wvparse)
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{
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gst_wavpack_parse_reset (wvparse);
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GST_PAD_SET_ACCEPT_INTERSECT (GST_BASE_PARSE_SINK_PAD (wvparse));
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GST_PAD_SET_ACCEPT_TEMPLATE (GST_BASE_PARSE_SINK_PAD (wvparse));
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}
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static void
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gst_wavpack_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_wavpack_parse_start (GstBaseParse * parse)
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{
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GstWavpackParse *wvparse = GST_WAVPACK_PARSE (parse);
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GST_DEBUG_OBJECT (parse, "starting");
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gst_wavpack_parse_reset (wvparse);
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/* need header at least */
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gst_base_parse_set_min_frame_size (GST_BASE_PARSE (wvparse),
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sizeof (WavpackHeader));
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/* inform baseclass we can come up with ts, based on counters in packets */
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gst_base_parse_set_has_timing_info (GST_BASE_PARSE_CAST (wvparse), TRUE);
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gst_base_parse_set_syncable (GST_BASE_PARSE_CAST (wvparse), TRUE);
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return TRUE;
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}
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static gboolean
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gst_wavpack_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 gint
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gst_wavpack_get_default_channel_mask (gint nchannels)
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{
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gint channel_mask = 0;
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/* Set the default channel mask for the given number of channels.
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* It's the same as for WAVE_FORMAT_EXTENDED:
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* http://www.microsoft.com/whdc/device/audio/multichaud.mspx
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*/
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switch (nchannels) {
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case 11:
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channel_mask |= 0x00400;
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channel_mask |= 0x00200;
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case 9:
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channel_mask |= 0x00100;
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case 8:
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channel_mask |= 0x00080;
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channel_mask |= 0x00040;
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case 6:
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channel_mask |= 0x00020;
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channel_mask |= 0x00010;
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case 4:
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channel_mask |= 0x00008;
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case 3:
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channel_mask |= 0x00004;
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case 2:
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channel_mask |= 0x00002;
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channel_mask |= 0x00001;
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break;
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case 1:
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/* For mono use front center */
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channel_mask |= 0x00004;
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break;
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}
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return channel_mask;
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}
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static const struct
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{
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const guint32 ms_mask;
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const GstAudioChannelPosition gst_pos;
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} layout_mapping[] = {
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{
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0x00001, GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT}, {
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0x00002, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT}, {
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0x00004, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER}, {
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0x00008, GST_AUDIO_CHANNEL_POSITION_LFE1}, {
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0x00010, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT}, {
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0x00020, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT}, {
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0x00040, GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER}, {
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0x00080, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER}, {
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0x00100, GST_AUDIO_CHANNEL_POSITION_REAR_CENTER}, {
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0x00200, GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT}, {
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0x00400, GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT}, {
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0x00800, GST_AUDIO_CHANNEL_POSITION_TOP_CENTER}, {
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0x01000, GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_LEFT}, {
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0x02000, GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_CENTER}, {
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0x04000, GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_RIGHT}, {
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0x08000, GST_AUDIO_CHANNEL_POSITION_TOP_REAR_LEFT}, {
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0x10000, GST_AUDIO_CHANNEL_POSITION_TOP_REAR_CENTER}, {
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0x20000, GST_AUDIO_CHANNEL_POSITION_TOP_REAR_RIGHT}
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};
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#define MAX_CHANNEL_POSITIONS G_N_ELEMENTS (layout_mapping)
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static gboolean
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gst_wavpack_get_channel_positions (gint num_channels, gint layout,
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GstAudioChannelPosition * pos)
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{
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gint i, p;
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if (num_channels == 1 && layout == 0x00004) {
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pos[0] = GST_AUDIO_CHANNEL_POSITION_MONO;
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return TRUE;
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}
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p = 0;
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for (i = 0; i < MAX_CHANNEL_POSITIONS; ++i) {
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if ((layout & layout_mapping[i].ms_mask) != 0) {
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if (p >= num_channels) {
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GST_WARNING ("More bits set in the channel layout map than there "
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"are channels! Broken file");
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return FALSE;
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}
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if (layout_mapping[i].gst_pos == GST_AUDIO_CHANNEL_POSITION_INVALID) {
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GST_WARNING ("Unsupported channel position (mask 0x%08x) in channel "
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"layout map - ignoring those channels", layout_mapping[i].ms_mask);
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/* what to do? just ignore it and let downstream deal with a channel
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* layout that has INVALID positions in it for now ... */
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}
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pos[p] = layout_mapping[i].gst_pos;
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++p;
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}
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}
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if (p != num_channels) {
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GST_WARNING ("Only %d bits set in the channel layout map, but there are "
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"supposed to be %d channels! Broken file", p, num_channels);
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return FALSE;
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}
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return TRUE;
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}
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static const guint32 sample_rates[] = {
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6000, 8000, 9600, 11025, 12000, 16000, 22050,
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24000, 32000, 44100, 48000, 64000, 88200, 96000, 192000
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};
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#define CHECK(call) { \
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if (!call) \
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goto read_failed; \
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}
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/* caller ensures properly sync'ed with enough data */
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static gboolean
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gst_wavpack_parse_frame_metadata (GstWavpackParse * parse, GstBuffer * buf,
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gint skip, WavpackHeader * wph, WavpackInfo * wpi)
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{
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GstByteReader br;
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gint i;
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GstMapInfo map;
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g_return_val_if_fail (wph != NULL || wpi != NULL, FALSE);
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g_return_val_if_fail (gst_buffer_get_size (buf) >=
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skip + sizeof (WavpackHeader), FALSE);
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gst_buffer_map (buf, &map, GST_MAP_READ);
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gst_byte_reader_init (&br, map.data + skip, wph->ckSize + 8);
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/* skip past header */
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gst_byte_reader_skip_unchecked (&br, sizeof (WavpackHeader));
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/* get some basics from header */
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i = (wph->flags >> 23) & 0xF;
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if (!wpi->rate)
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wpi->rate = (i < G_N_ELEMENTS (sample_rates)) ? sample_rates[i] : 44100;
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wpi->width = ((wph->flags & 0x3) + 1) * 8;
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if (!wpi->channels)
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wpi->channels = (wph->flags & 0x4) ? 1 : 2;
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if (!wpi->channel_mask)
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wpi->channel_mask = 5 - wpi->channels;
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/* need to dig metadata blocks for some more */
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while (gst_byte_reader_get_remaining (&br)) {
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gint size = 0;
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guint16 size2 = 0;
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guint8 c, id;
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const guint8 *data;
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GstByteReader mbr;
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CHECK (gst_byte_reader_get_uint8 (&br, &id));
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CHECK (gst_byte_reader_get_uint8 (&br, &c));
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if (id & ID_LARGE)
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CHECK (gst_byte_reader_get_uint16_le (&br, &size2));
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size = size2;
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size <<= 8;
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size += c;
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size <<= 1;
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if (id & ID_ODD_SIZE)
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size--;
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CHECK (gst_byte_reader_get_data (&br, size + (size & 1), &data));
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gst_byte_reader_init (&mbr, data, size);
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/* 0x1f is the metadata id mask and 0x20 flag is for later extensions
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* that do not need to be handled by the decoder */
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switch (id & 0x3f) {
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case ID_WVC_BITSTREAM:
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GST_LOG_OBJECT (parse, "correction bitstream");
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wpi->correction = TRUE;
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break;
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case ID_WV_BITSTREAM:
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case ID_WVX_BITSTREAM:
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break;
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case ID_SAMPLE_RATE:
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if (size == 3) {
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CHECK (gst_byte_reader_get_uint24_le (&mbr, &wpi->rate));
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GST_LOG_OBJECT (parse, "updated with custom rate %d", wpi->rate);
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} else {
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GST_DEBUG_OBJECT (parse, "unexpected size for SAMPLE_RATE metadata");
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}
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break;
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case ID_CHANNEL_INFO:
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{
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guint16 channels;
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guint32 mask = 0;
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if (size == 6) {
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CHECK (gst_byte_reader_get_uint16_le (&mbr, &channels));
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channels = channels & 0xFFF;
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CHECK (gst_byte_reader_get_uint24_le (&mbr, &mask));
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} else if (size) {
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CHECK (gst_byte_reader_get_uint8 (&mbr, &c));
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channels = c;
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while (gst_byte_reader_get_uint8 (&mbr, &c))
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mask |= (((guint32) c) << 8);
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} else {
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GST_DEBUG_OBJECT (parse, "unexpected size for CHANNEL_INFO metadata");
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break;
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}
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wpi->channels = channels;
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wpi->channel_mask = mask;
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break;
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}
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default:
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GST_LOG_OBJECT (parse, "unparsed ID 0x%x", id);
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break;
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}
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}
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gst_buffer_unmap (buf, &map);
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return TRUE;
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/* ERRORS */
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read_failed:
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{
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gst_buffer_unmap (buf, &map);
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GST_DEBUG_OBJECT (parse, "short read while parsing metadata");
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/* let's look the other way anyway */
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return TRUE;
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}
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}
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/* caller ensures properly sync'ed with enough data */
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static gboolean
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gst_wavpack_parse_frame_header (GstWavpackParse * parse, GstBuffer * buf,
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gint skip, WavpackHeader * _wph)
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{
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GstByteReader br;
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WavpackHeader wph = { {0,}, 0, };
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GstMapInfo map;
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gboolean hdl = TRUE;
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g_return_val_if_fail (gst_buffer_get_size (buf) >=
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skip + sizeof (WavpackHeader), FALSE);
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gst_buffer_map (buf, &map, GST_MAP_READ);
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gst_byte_reader_init (&br, map.data, map.size);
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/* marker */
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gst_byte_reader_skip_unchecked (&br, skip + 4);
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/* read */
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hdl &= gst_byte_reader_get_uint32_le (&br, &wph.ckSize);
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hdl &= gst_byte_reader_get_uint16_le (&br, &wph.version);
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hdl &= gst_byte_reader_get_uint8 (&br, &wph.track_no);
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hdl &= gst_byte_reader_get_uint8 (&br, &wph.index_no);
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hdl &= gst_byte_reader_get_uint32_le (&br, &wph.total_samples);
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hdl &= gst_byte_reader_get_uint32_le (&br, &wph.block_index);
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hdl &= gst_byte_reader_get_uint32_le (&br, &wph.block_samples);
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hdl &= gst_byte_reader_get_uint32_le (&br, &wph.flags);
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hdl &= gst_byte_reader_get_uint32_le (&br, &wph.crc);
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if (!hdl)
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GST_WARNING_OBJECT (parse, "Error reading header");
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/* dump */
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GST_LOG_OBJECT (parse, "size %d", wph.ckSize);
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GST_LOG_OBJECT (parse, "version 0x%x", wph.version);
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GST_LOG_OBJECT (parse, "total samples %d", wph.total_samples);
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GST_LOG_OBJECT (parse, "block index %d", wph.block_index);
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GST_LOG_OBJECT (parse, "block samples %d", wph.block_samples);
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GST_LOG_OBJECT (parse, "flags 0x%x", wph.flags);
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GST_LOG_OBJECT (parse, "crc 0x%x", wph.flags);
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if (!parse->total_samples && wph.block_index == 0 && wph.total_samples != -1) {
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GST_DEBUG_OBJECT (parse, "determined duration of %u samples",
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wph.total_samples);
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parse->total_samples = wph.total_samples;
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}
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if (_wph)
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*_wph = wph;
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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;
|
|
}
|