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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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1b8df15b78
Use speex_header_free() to free memory which was allocated by library. Cross-CRT issue should not happen on 1.17 Cerbero build but might happen custom build or so. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/606>
573 lines
16 KiB
C
573 lines
16 KiB
C
/* GStreamer
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* Copyright (C) 2004 Wim Taymans <wim@fluendo.com>
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* Copyright (C) 2006 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-speexdec
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* @title: speexdec
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* @see_also: speexenc, oggdemux
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*
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* This element decodes a Speex stream to raw integer audio.
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* [Speex](http://www.speex.org/) is a royalty-free
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* audio codec maintained by the [Xiph.org Foundation](http://www.xiph.org/).
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*
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* ## Example pipelines
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* |[
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* gst-launch-1.0 -v filesrc location=speex.ogg ! oggdemux ! speexdec ! audioconvert ! audioresample ! alsasink
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* ]| Decode an Ogg/Speex file. To create an Ogg/Speex file refer to the
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* documentation of speexenc.
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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 "gstspeexdec.h"
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#include <stdlib.h>
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#include <string.h>
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#include <gst/tag/tag.h>
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#include <gst/audio/audio.h>
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GST_DEBUG_CATEGORY_STATIC (speexdec_debug);
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#define GST_CAT_DEFAULT speexdec_debug
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#define DEFAULT_ENH TRUE
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enum
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{
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ARG_0,
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ARG_ENH
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};
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#define FORMAT_STR GST_AUDIO_NE(S16)
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static GstStaticPadTemplate speex_dec_src_factory =
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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-raw, "
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"format = (string) " FORMAT_STR ", "
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"layout = (string) interleaved, "
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"rate = (int) [ 6000, 48000 ], " "channels = (int) [ 1, 2 ]")
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);
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static GstStaticPadTemplate speex_dec_sink_factory =
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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-speex")
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);
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#define gst_speex_dec_parent_class parent_class
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G_DEFINE_TYPE (GstSpeexDec, gst_speex_dec, GST_TYPE_AUDIO_DECODER);
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static gboolean gst_speex_dec_start (GstAudioDecoder * dec);
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static gboolean gst_speex_dec_stop (GstAudioDecoder * dec);
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static gboolean gst_speex_dec_set_format (GstAudioDecoder * bdec,
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GstCaps * caps);
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static GstFlowReturn gst_speex_dec_handle_frame (GstAudioDecoder * dec,
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GstBuffer * buffer);
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static void gst_speex_dec_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static void gst_speex_dec_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void
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gst_speex_dec_class_init (GstSpeexDecClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstAudioDecoderClass *base_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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base_class = (GstAudioDecoderClass *) klass;
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gobject_class->set_property = gst_speex_dec_set_property;
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gobject_class->get_property = gst_speex_dec_get_property;
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base_class->start = GST_DEBUG_FUNCPTR (gst_speex_dec_start);
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base_class->stop = GST_DEBUG_FUNCPTR (gst_speex_dec_stop);
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base_class->set_format = GST_DEBUG_FUNCPTR (gst_speex_dec_set_format);
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base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_speex_dec_handle_frame);
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g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_ENH,
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g_param_spec_boolean ("enh", "Enh", "Enable perceptual enhancement",
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DEFAULT_ENH, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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gst_element_class_add_static_pad_template (gstelement_class,
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&speex_dec_src_factory);
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gst_element_class_add_static_pad_template (gstelement_class,
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&speex_dec_sink_factory);
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gst_element_class_set_static_metadata (gstelement_class,
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"Speex audio decoder", "Codec/Decoder/Audio",
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"decode speex streams to audio", "Wim Taymans <wim@fluendo.com>");
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GST_DEBUG_CATEGORY_INIT (speexdec_debug, "speexdec", 0,
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"speex decoding element");
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}
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static void
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gst_speex_dec_reset (GstSpeexDec * dec)
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{
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dec->packetno = 0;
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dec->frame_size = 0;
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dec->frame_duration = 0;
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dec->mode = NULL;
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speex_header_free (dec->header);
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dec->header = NULL;
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speex_bits_destroy (&dec->bits);
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speex_bits_set_bit_buffer (&dec->bits, NULL, 0);
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gst_buffer_replace (&dec->streamheader, NULL);
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gst_buffer_replace (&dec->vorbiscomment, NULL);
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if (dec->stereo) {
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speex_stereo_state_destroy (dec->stereo);
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dec->stereo = NULL;
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}
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if (dec->state) {
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speex_decoder_destroy (dec->state);
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dec->state = NULL;
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}
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}
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static void
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gst_speex_dec_init (GstSpeexDec * dec)
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{
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gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (dec), TRUE);
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gst_audio_decoder_set_use_default_pad_acceptcaps (GST_AUDIO_DECODER_CAST
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(dec), TRUE);
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GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_DECODER_SINK_PAD (dec));
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dec->enh = DEFAULT_ENH;
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gst_speex_dec_reset (dec);
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}
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static gboolean
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gst_speex_dec_start (GstAudioDecoder * dec)
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{
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GstSpeexDec *sd = GST_SPEEX_DEC (dec);
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GST_DEBUG_OBJECT (dec, "start");
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gst_speex_dec_reset (sd);
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/* we know about concealment */
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gst_audio_decoder_set_plc_aware (dec, TRUE);
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return TRUE;
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}
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static gboolean
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gst_speex_dec_stop (GstAudioDecoder * dec)
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{
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GstSpeexDec *sd = GST_SPEEX_DEC (dec);
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GST_DEBUG_OBJECT (dec, "stop");
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gst_speex_dec_reset (sd);
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return TRUE;
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}
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static GstFlowReturn
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gst_speex_dec_parse_header (GstSpeexDec * dec, GstBuffer * buf)
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{
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GstMapInfo map;
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GstAudioInfo info;
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static const GstAudioChannelPosition chan_pos[2][2] = {
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{GST_AUDIO_CHANNEL_POSITION_MONO},
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{GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT}
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};
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/* get the header */
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gst_buffer_map (buf, &map, GST_MAP_READ);
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dec->header = speex_packet_to_header ((gchar *) map.data, map.size);
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gst_buffer_unmap (buf, &map);
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if (!dec->header)
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goto no_header;
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if (dec->header->mode >= SPEEX_NB_MODES || dec->header->mode < 0)
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goto mode_too_old;
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dec->mode = speex_lib_get_mode (dec->header->mode);
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/* initialize the decoder */
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dec->state = speex_decoder_init (dec->mode);
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if (!dec->state)
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goto init_failed;
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speex_decoder_ctl (dec->state, SPEEX_SET_ENH, &dec->enh);
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speex_decoder_ctl (dec->state, SPEEX_GET_FRAME_SIZE, &dec->frame_size);
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if (dec->header->nb_channels != 1) {
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dec->stereo = speex_stereo_state_init ();
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dec->callback.callback_id = SPEEX_INBAND_STEREO;
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dec->callback.func = speex_std_stereo_request_handler;
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dec->callback.data = dec->stereo;
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speex_decoder_ctl (dec->state, SPEEX_SET_HANDLER, &dec->callback);
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}
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speex_decoder_ctl (dec->state, SPEEX_SET_SAMPLING_RATE, &dec->header->rate);
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dec->frame_duration = gst_util_uint64_scale_int (dec->frame_size,
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GST_SECOND, dec->header->rate);
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speex_bits_init (&dec->bits);
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/* set caps */
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gst_audio_info_init (&info);
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gst_audio_info_set_format (&info,
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GST_AUDIO_FORMAT_S16,
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dec->header->rate,
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dec->header->nb_channels, chan_pos[dec->header->nb_channels - 1]);
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if (!gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (dec), &info))
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goto nego_failed;
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return GST_FLOW_OK;
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/* ERRORS */
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no_header:
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{
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GST_ELEMENT_ERROR (GST_ELEMENT (dec), STREAM, DECODE,
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(NULL), ("couldn't read header"));
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return GST_FLOW_ERROR;
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}
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mode_too_old:
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{
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GST_ELEMENT_ERROR (GST_ELEMENT (dec), STREAM, DECODE,
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(NULL),
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("Mode number %d does not (yet/any longer) exist in this version",
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dec->header->mode));
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return GST_FLOW_ERROR;
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}
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init_failed:
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{
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GST_ELEMENT_ERROR (GST_ELEMENT (dec), STREAM, DECODE,
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(NULL), ("couldn't initialize decoder"));
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return GST_FLOW_ERROR;
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}
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nego_failed:
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{
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GST_ELEMENT_ERROR (GST_ELEMENT (dec), STREAM, DECODE,
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(NULL), ("couldn't negotiate format"));
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return GST_FLOW_NOT_NEGOTIATED;
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}
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}
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static GstFlowReturn
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gst_speex_dec_parse_comments (GstSpeexDec * dec, GstBuffer * buf)
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{
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GstTagList *list;
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gchar *ver, *encoder = NULL;
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list = gst_tag_list_from_vorbiscomment_buffer (buf, NULL, 0, &encoder);
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if (!list) {
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GST_WARNING_OBJECT (dec, "couldn't decode comments");
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list = gst_tag_list_new_empty ();
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}
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if (encoder) {
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gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
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GST_TAG_ENCODER, encoder, NULL);
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}
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gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
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GST_TAG_AUDIO_CODEC, "Speex", NULL);
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ver = g_strndup (dec->header->speex_version, SPEEX_HEADER_VERSION_LENGTH);
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g_strstrip (ver);
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if (ver != NULL && *ver != '\0') {
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gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
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GST_TAG_ENCODER_VERSION, ver, NULL);
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}
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if (dec->header->bitrate > 0) {
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gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
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GST_TAG_BITRATE, (guint) dec->header->bitrate, NULL);
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}
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GST_INFO_OBJECT (dec, "tags: %" GST_PTR_FORMAT, list);
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gst_audio_decoder_merge_tags (GST_AUDIO_DECODER (dec), list,
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GST_TAG_MERGE_REPLACE);
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gst_tag_list_unref (list);
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g_free (encoder);
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g_free (ver);
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return GST_FLOW_OK;
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}
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static gboolean
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gst_speex_dec_set_format (GstAudioDecoder * bdec, GstCaps * caps)
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{
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GstSpeexDec *dec = GST_SPEEX_DEC (bdec);
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gboolean ret = TRUE;
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GstStructure *s;
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const GValue *streamheader;
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s = gst_caps_get_structure (caps, 0);
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if ((streamheader = gst_structure_get_value (s, "streamheader")) &&
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G_VALUE_HOLDS (streamheader, GST_TYPE_ARRAY) &&
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gst_value_array_get_size (streamheader) >= 2) {
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const GValue *header, *vorbiscomment;
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GstBuffer *buf;
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GstFlowReturn res = GST_FLOW_OK;
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header = gst_value_array_get_value (streamheader, 0);
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if (header && G_VALUE_HOLDS (header, GST_TYPE_BUFFER)) {
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buf = gst_value_get_buffer (header);
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res = gst_speex_dec_parse_header (dec, buf);
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if (res != GST_FLOW_OK)
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goto done;
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gst_buffer_replace (&dec->streamheader, buf);
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}
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vorbiscomment = gst_value_array_get_value (streamheader, 1);
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if (vorbiscomment && G_VALUE_HOLDS (vorbiscomment, GST_TYPE_BUFFER)) {
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buf = gst_value_get_buffer (vorbiscomment);
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res = gst_speex_dec_parse_comments (dec, buf);
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if (res != GST_FLOW_OK)
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goto done;
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gst_buffer_replace (&dec->vorbiscomment, buf);
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}
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}
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done:
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return ret;
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}
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static GstFlowReturn
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gst_speex_dec_parse_data (GstSpeexDec * dec, GstBuffer * buf)
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{
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GstFlowReturn res = GST_FLOW_OK;
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gint i, fpp;
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SpeexBits *bits;
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GstMapInfo map;
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if (!dec->frame_duration)
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goto not_negotiated;
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if (G_LIKELY (gst_buffer_get_size (buf))) {
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/* send data to the bitstream */
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gst_buffer_map (buf, &map, GST_MAP_READ);
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speex_bits_read_from (&dec->bits, (gchar *) map.data, map.size);
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gst_buffer_unmap (buf, &map);
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fpp = dec->header->frames_per_packet;
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bits = &dec->bits;
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GST_DEBUG_OBJECT (dec, "received buffer of size %" G_GSIZE_FORMAT
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", fpp %d, %d bits", map.size, fpp, speex_bits_remaining (bits));
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} else {
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/* FIXME ? actually consider how much concealment is needed */
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/* concealment data, pass NULL as the bits parameters */
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GST_DEBUG_OBJECT (dec, "creating concealment data");
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fpp = dec->header->frames_per_packet;
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bits = NULL;
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}
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/* now decode each frame, catering for unknown number of them (e.g. rtp) */
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for (i = 0; i < fpp; i++) {
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GstBuffer *outbuf;
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gboolean corrupted = FALSE;
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gint ret;
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GST_LOG_OBJECT (dec, "decoding frame %d/%d, %d bits remaining", i, fpp,
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bits ? speex_bits_remaining (bits) : -1);
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#if 0
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res =
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gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec),
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GST_BUFFER_OFFSET_NONE, dec->frame_size * dec->header->nb_channels * 2,
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GST_PAD_CAPS (GST_AUDIO_DECODER_SRC_PAD (dec)), &outbuf);
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if (res != GST_FLOW_OK) {
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GST_DEBUG_OBJECT (dec, "buf alloc flow: %s", gst_flow_get_name (res));
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return res;
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}
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#endif
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/* FIXME, we can use a bufferpool because we have fixed size buffers. We
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* could also use an allocator */
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outbuf =
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gst_buffer_new_allocate (NULL,
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dec->frame_size * dec->header->nb_channels * 2, NULL);
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gst_buffer_map (outbuf, &map, GST_MAP_WRITE);
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ret = speex_decode_int (dec->state, bits, (spx_int16_t *) map.data);
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if (ret == -1) {
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/* uh? end of stream */
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GST_WARNING_OBJECT (dec, "Unexpected end of stream found");
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corrupted = TRUE;
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} else if (ret == -2) {
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GST_WARNING_OBJECT (dec, "Decoding error: corrupted stream?");
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corrupted = TRUE;
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}
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if (bits && speex_bits_remaining (bits) < 0) {
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GST_WARNING_OBJECT (dec, "Decoding overflow: corrupted stream?");
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corrupted = TRUE;
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}
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if (dec->header->nb_channels == 2)
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speex_decode_stereo_int ((spx_int16_t *) map.data, dec->frame_size,
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dec->stereo);
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gst_buffer_unmap (outbuf, &map);
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if (!corrupted) {
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res = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (dec), outbuf, 1);
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} else {
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res = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (dec), NULL, 1);
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gst_buffer_unref (outbuf);
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}
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if (res != GST_FLOW_OK) {
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GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (res));
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break;
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}
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}
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return res;
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/* ERRORS */
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not_negotiated:
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{
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GST_ELEMENT_ERROR (dec, CORE, NEGOTIATION, (NULL),
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("decoder not initialized"));
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return GST_FLOW_NOT_NEGOTIATED;
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}
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}
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static gboolean
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memcmp_buffers (GstBuffer * buf1, GstBuffer * buf2)
|
|
{
|
|
GstMapInfo map;
|
|
gsize size1, size2;
|
|
gboolean res;
|
|
|
|
size1 = gst_buffer_get_size (buf1);
|
|
size2 = gst_buffer_get_size (buf2);
|
|
|
|
if (size1 != size2)
|
|
return FALSE;
|
|
|
|
gst_buffer_map (buf1, &map, GST_MAP_READ);
|
|
res = gst_buffer_memcmp (buf2, 0, map.data, map.size) == 0;
|
|
gst_buffer_unmap (buf1, &map);
|
|
|
|
return res;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_speex_dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buf)
|
|
{
|
|
GstFlowReturn res;
|
|
GstSpeexDec *dec;
|
|
|
|
/* no fancy draining */
|
|
if (G_UNLIKELY (!buf))
|
|
return GST_FLOW_OK;
|
|
|
|
dec = GST_SPEEX_DEC (bdec);
|
|
|
|
/* If we have the streamheader and vorbiscomment from the caps already
|
|
* ignore them here */
|
|
if (dec->streamheader && dec->vorbiscomment) {
|
|
if (memcmp_buffers (dec->streamheader, buf)) {
|
|
GST_DEBUG_OBJECT (dec, "found streamheader");
|
|
gst_audio_decoder_finish_frame (bdec, NULL, 1);
|
|
res = GST_FLOW_OK;
|
|
} else if (memcmp_buffers (dec->vorbiscomment, buf)) {
|
|
GST_DEBUG_OBJECT (dec, "found vorbiscomments");
|
|
gst_audio_decoder_finish_frame (bdec, NULL, 1);
|
|
res = GST_FLOW_OK;
|
|
} else {
|
|
res = gst_speex_dec_parse_data (dec, buf);
|
|
}
|
|
} else {
|
|
/* Otherwise fall back to packet counting and assume that the
|
|
* first two packets are the headers. */
|
|
switch (dec->packetno) {
|
|
case 0:
|
|
GST_DEBUG_OBJECT (dec, "counted streamheader");
|
|
res = gst_speex_dec_parse_header (dec, buf);
|
|
gst_audio_decoder_finish_frame (bdec, NULL, 1);
|
|
break;
|
|
case 1:
|
|
GST_DEBUG_OBJECT (dec, "counted vorbiscomments");
|
|
res = gst_speex_dec_parse_comments (dec, buf);
|
|
gst_audio_decoder_finish_frame (bdec, NULL, 1);
|
|
break;
|
|
default:
|
|
{
|
|
res = gst_speex_dec_parse_data (dec, buf);
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
|
|
dec->packetno++;
|
|
|
|
return res;
|
|
}
|
|
|
|
static void
|
|
gst_speex_dec_get_property (GObject * object, guint prop_id,
|
|
GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstSpeexDec *speexdec;
|
|
|
|
speexdec = GST_SPEEX_DEC (object);
|
|
|
|
switch (prop_id) {
|
|
case ARG_ENH:
|
|
g_value_set_boolean (value, speexdec->enh);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_speex_dec_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstSpeexDec *speexdec;
|
|
|
|
speexdec = GST_SPEEX_DEC (object);
|
|
|
|
switch (prop_id) {
|
|
case ARG_ENH:
|
|
speexdec->enh = g_value_get_boolean (value);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|