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When passing the taglist to gst_audio_decoder_merge_tags, the reference is increased by audiodecoder and the caller should free the taglist being passed. https://bugzilla.gnome.org/show_bug.cgi?id=753903
475 lines
14 KiB
C
475 lines
14 KiB
C
/* GStreamer Wavpack plugin
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* Copyright (c) 2005 Arwed v. Merkatz <v.merkatz@gmx.net>
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* Copyright (c) 2006 Edward Hervey <bilboed@gmail.com>
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* Copyright (c) 2006 Sebastian Dröge <slomo@circular-chaos.org>
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*
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* gstwavpackdec.c: raw Wavpack bitstream decoder
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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-wavpackdec
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*
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* WavpackDec decodes framed (for example by the WavpackParse element)
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* Wavpack streams and decodes them to raw audio.
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* <ulink url="http://www.wavpack.com/">Wavpack</ulink> is an open-source
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* audio codec that features both lossless and lossy encoding.
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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=test.wv ! wavpackparse ! wavpackdec ! audioconvert ! audioresample ! autoaudiosink
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* ]| This pipeline decodes the Wavpack file test.wv into raw audio buffers and
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* tries to play it back using an automatically found audio sink.
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* </refsect2>
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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 <gst/gst.h>
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#include <gst/audio/audio.h>
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#include <math.h>
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#include <string.h>
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#include <wavpack/wavpack.h>
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#include "gstwavpackdec.h"
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#include "gstwavpackcommon.h"
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#include "gstwavpackstreamreader.h"
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GST_DEBUG_CATEGORY_STATIC (gst_wavpack_dec_debug);
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#define GST_CAT_DEFAULT gst_wavpack_dec_debug
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static GstStaticPadTemplate sink_factory = 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, "
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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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);
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static GstStaticPadTemplate src_factory = 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) S8, "
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"layout = (string) interleaved, "
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"channels = (int) [ 1, 8 ], "
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"rate = (int) [ 6000, 192000 ]; "
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"audio/x-raw, "
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"format = (string) " GST_AUDIO_NE (S16) ", "
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"layout = (string) interleaved, "
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"channels = (int) [ 1, 8 ], "
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"rate = (int) [ 6000, 192000 ]; "
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"audio/x-raw, "
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"format = (string) " GST_AUDIO_NE (S32) ", "
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"layout = (string) interleaved, "
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"channels = (int) [ 1, 8 ], " "rate = (int) [ 6000, 192000 ]")
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);
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static gboolean gst_wavpack_dec_start (GstAudioDecoder * dec);
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static gboolean gst_wavpack_dec_stop (GstAudioDecoder * dec);
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static gboolean gst_wavpack_dec_set_format (GstAudioDecoder * dec,
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GstCaps * caps);
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static GstFlowReturn gst_wavpack_dec_handle_frame (GstAudioDecoder * dec,
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GstBuffer * buffer);
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static void gst_wavpack_dec_finalize (GObject * object);
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static void gst_wavpack_dec_post_tags (GstWavpackDec * dec);
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#define gst_wavpack_dec_parent_class parent_class
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G_DEFINE_TYPE (GstWavpackDec, gst_wavpack_dec, GST_TYPE_AUDIO_DECODER);
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static void
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gst_wavpack_dec_class_init (GstWavpackDecClass * klass)
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{
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GObjectClass *gobject_class = (GObjectClass *) klass;
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GstElementClass *element_class = (GstElementClass *) (klass);
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GstAudioDecoderClass *base_class = (GstAudioDecoderClass *) (klass);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&src_factory));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&sink_factory));
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gst_element_class_set_static_metadata (element_class, "Wavpack audio decoder",
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"Codec/Decoder/Audio",
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"Decodes Wavpack audio data",
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"Arwed v. Merkatz <v.merkatz@gmx.net>, "
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"Sebastian Dröge <slomo@circular-chaos.org>");
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gobject_class->finalize = gst_wavpack_dec_finalize;
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base_class->start = GST_DEBUG_FUNCPTR (gst_wavpack_dec_start);
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base_class->stop = GST_DEBUG_FUNCPTR (gst_wavpack_dec_stop);
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base_class->set_format = GST_DEBUG_FUNCPTR (gst_wavpack_dec_set_format);
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base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_wavpack_dec_handle_frame);
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}
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static void
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gst_wavpack_dec_reset (GstWavpackDec * dec)
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{
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dec->wv_id.buffer = NULL;
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dec->wv_id.position = dec->wv_id.length = 0;
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dec->channels = 0;
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dec->channel_mask = 0;
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dec->sample_rate = 0;
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dec->depth = 0;
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}
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static void
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gst_wavpack_dec_init (GstWavpackDec * dec)
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{
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dec->context = NULL;
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dec->stream_reader = gst_wavpack_stream_reader_new ();
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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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gst_wavpack_dec_reset (dec);
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}
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static void
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gst_wavpack_dec_finalize (GObject * object)
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{
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GstWavpackDec *dec = GST_WAVPACK_DEC (object);
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g_free (dec->stream_reader);
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dec->stream_reader = NULL;
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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_dec_start (GstAudioDecoder * dec)
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{
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GST_DEBUG_OBJECT (dec, "start");
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/* never mind a few errors */
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gst_audio_decoder_set_max_errors (dec, 16);
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/* don't bother us with flushing */
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gst_audio_decoder_set_drainable (dec, FALSE);
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/* aim for some perfect timestamping */
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gst_audio_decoder_set_tolerance (dec, 10 * GST_MSECOND);
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return TRUE;
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}
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static gboolean
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gst_wavpack_dec_stop (GstAudioDecoder * dec)
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{
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GstWavpackDec *wpdec = GST_WAVPACK_DEC (dec);
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GST_DEBUG_OBJECT (dec, "stop");
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if (wpdec->context) {
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WavpackCloseFile (wpdec->context);
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wpdec->context = NULL;
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}
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gst_wavpack_dec_reset (wpdec);
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return TRUE;
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}
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static void
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gst_wavpack_dec_negotiate (GstWavpackDec * dec)
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{
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GstAudioInfo info;
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GstAudioFormat fmt;
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GstAudioChannelPosition pos[64] = { GST_AUDIO_CHANNEL_POSITION_INVALID, };
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/* arrange for 1, 2 or 4-byte width == depth output */
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dec->width = dec->depth;
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switch (dec->depth) {
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case 8:
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fmt = GST_AUDIO_FORMAT_S8;
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break;
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case 16:
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fmt = _GST_AUDIO_FORMAT_NE (S16);
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break;
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case 24:
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case 32:
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fmt = _GST_AUDIO_FORMAT_NE (S32);
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dec->width = 32;
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break;
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default:
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fmt = GST_AUDIO_FORMAT_UNKNOWN;
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g_assert_not_reached ();
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break;
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}
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g_assert (dec->channel_mask != 0);
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if (!gst_wavpack_get_channel_positions (dec->channels,
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dec->channel_mask, pos))
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GST_WARNING_OBJECT (dec, "Failed to set channel layout");
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gst_audio_info_init (&info);
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gst_audio_info_set_format (&info, fmt, dec->sample_rate, dec->channels, pos);
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gst_audio_channel_positions_to_valid_order (info.position, info.channels);
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gst_audio_get_channel_reorder_map (info.channels,
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info.position, pos, dec->channel_reorder_map);
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/* should always succeed */
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gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (dec), &info);
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}
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static gboolean
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gst_wavpack_dec_set_format (GstAudioDecoder * bdec, GstCaps * caps)
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{
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/* pretty much nothing to do here,
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* we'll parse it all from the stream and setup then */
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return TRUE;
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}
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static void
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gst_wavpack_dec_post_tags (GstWavpackDec * dec)
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{
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GstTagList *list;
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GstFormat format_time = GST_FORMAT_TIME, format_bytes = GST_FORMAT_BYTES;
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gint64 duration, size;
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/* try to estimate the average bitrate */
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if (gst_pad_peer_query_duration (GST_AUDIO_DECODER_SINK_PAD (dec),
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format_bytes, &size) &&
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gst_pad_peer_query_duration (GST_AUDIO_DECODER_SINK_PAD (dec),
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format_time, &duration) && size > 0 && duration > 0) {
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guint64 bitrate;
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list = gst_tag_list_new_empty ();
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bitrate = gst_util_uint64_scale (size, 8 * GST_SECOND, duration);
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gst_tag_list_add (list, GST_TAG_MERGE_REPLACE, GST_TAG_BITRATE,
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(guint) bitrate, NULL);
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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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}
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}
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static GstFlowReturn
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gst_wavpack_dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buf)
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{
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GstWavpackDec *dec;
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GstBuffer *outbuf = NULL;
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GstFlowReturn ret = GST_FLOW_OK;
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WavpackHeader wph;
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int32_t decoded, unpacked_size;
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gboolean format_changed;
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gint width, depth, i, j, max;
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gint32 *dec_data = NULL;
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guint8 *out_data;
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GstMapInfo map, omap;
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dec = GST_WAVPACK_DEC (bdec);
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g_return_val_if_fail (buf != NULL, GST_FLOW_ERROR);
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gst_buffer_map (buf, &map, GST_MAP_READ);
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/* check input, we only accept framed input with complete chunks */
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if (map.size < sizeof (WavpackHeader))
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goto input_not_framed;
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if (!gst_wavpack_read_header (&wph, map.data))
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goto invalid_header;
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if (map.size < wph.ckSize + 4 * 1 + 4)
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goto input_not_framed;
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if (!(wph.flags & INITIAL_BLOCK))
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goto input_not_framed;
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dec->wv_id.buffer = map.data;
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dec->wv_id.length = map.size;
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dec->wv_id.position = 0;
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/* create a new wavpack context if there is none yet but if there
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* was already one (i.e. caps were set on the srcpad) check whether
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* the new one has the same caps */
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if (!dec->context) {
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gchar error_msg[80];
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dec->context = WavpackOpenFileInputEx (dec->stream_reader,
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&dec->wv_id, NULL, error_msg, OPEN_STREAMING, 0);
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/* expect this to work */
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if (!dec->context) {
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GST_WARNING_OBJECT (dec, "Couldn't decode buffer: %s", error_msg);
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goto context_failed;
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}
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}
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g_assert (dec->context != NULL);
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format_changed =
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(dec->sample_rate != WavpackGetSampleRate (dec->context)) ||
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(dec->channels != WavpackGetNumChannels (dec->context)) ||
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(dec->depth != WavpackGetBytesPerSample (dec->context) * 8) ||
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(dec->channel_mask != WavpackGetChannelMask (dec->context));
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if (!gst_pad_has_current_caps (GST_AUDIO_DECODER_SRC_PAD (dec)) ||
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format_changed) {
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gint channel_mask;
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dec->sample_rate = WavpackGetSampleRate (dec->context);
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dec->channels = WavpackGetNumChannels (dec->context);
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dec->depth = WavpackGetBytesPerSample (dec->context) * 8;
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channel_mask = WavpackGetChannelMask (dec->context);
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if (channel_mask == 0)
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channel_mask = gst_wavpack_get_default_channel_mask (dec->channels);
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dec->channel_mask = channel_mask;
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gst_wavpack_dec_negotiate (dec);
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/* send GST_TAG_AUDIO_CODEC and GST_TAG_BITRATE tags before something
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* is decoded or after the format has changed */
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gst_wavpack_dec_post_tags (dec);
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}
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/* alloc output buffer */
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dec_data = g_malloc (4 * wph.block_samples * dec->channels);
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/* decode */
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decoded = WavpackUnpackSamples (dec->context, dec_data, wph.block_samples);
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if (decoded != wph.block_samples)
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goto decode_error;
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unpacked_size = (dec->width / 8) * wph.block_samples * dec->channels;
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outbuf = gst_buffer_new_and_alloc (unpacked_size);
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/* legacy; pass along offset, whatever that might entail */
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GST_BUFFER_OFFSET (outbuf) = GST_BUFFER_OFFSET (buf);
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gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
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out_data = omap.data;
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width = dec->width;
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depth = dec->depth;
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max = dec->channels * wph.block_samples;
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if (width == 8) {
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gint8 *outbuffer = (gint8 *) out_data;
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gint *reorder_map = dec->channel_reorder_map;
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for (i = 0; i < max; i += dec->channels) {
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for (j = 0; j < dec->channels; j++)
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*outbuffer++ = (gint8) (dec_data[i + reorder_map[j]]);
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}
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} else if (width == 16) {
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gint16 *outbuffer = (gint16 *) out_data;
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gint *reorder_map = dec->channel_reorder_map;
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for (i = 0; i < max; i += dec->channels) {
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for (j = 0; j < dec->channels; j++)
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*outbuffer++ = (gint16) (dec_data[i + reorder_map[j]]);
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}
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} else if (dec->width == 32) {
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gint32 *outbuffer = (gint32 *) out_data;
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gint *reorder_map = dec->channel_reorder_map;
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if (width != depth) {
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for (i = 0; i < max; i += dec->channels) {
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for (j = 0; j < dec->channels; j++)
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*outbuffer++ =
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(gint32) (dec_data[i + reorder_map[j]] << (width - depth));
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}
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} else {
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for (i = 0; i < max; i += dec->channels) {
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for (j = 0; j < dec->channels; j++)
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*outbuffer++ = (gint32) (dec_data[i + reorder_map[j]]);
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}
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}
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} else {
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g_assert_not_reached ();
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}
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gst_buffer_unmap (outbuf, &omap);
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gst_buffer_unmap (buf, &map);
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buf = NULL;
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g_free (dec_data);
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ret = gst_audio_decoder_finish_frame (bdec, outbuf, 1);
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out:
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if (buf)
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gst_buffer_unmap (buf, &map);
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if (G_UNLIKELY (ret != GST_FLOW_OK)) {
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GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (ret));
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}
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return ret;
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/* ERRORS */
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input_not_framed:
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{
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GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("Expected framed input"));
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ret = GST_FLOW_ERROR;
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goto out;
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}
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invalid_header:
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{
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GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("Invalid wavpack header"));
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ret = GST_FLOW_ERROR;
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goto out;
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}
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context_failed:
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{
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GST_AUDIO_DECODER_ERROR (bdec, 1, LIBRARY, INIT, (NULL),
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("error creating Wavpack context"), ret);
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goto out;
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}
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decode_error:
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{
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const gchar *reason = "unknown";
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if (dec->context) {
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reason = WavpackGetErrorMessage (dec->context);
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} else {
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reason = "couldn't create decoder context";
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}
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GST_AUDIO_DECODER_ERROR (bdec, 1, STREAM, DECODE, (NULL),
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("decoding error: %s", reason), ret);
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g_free (dec_data);
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if (ret == GST_FLOW_OK)
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gst_audio_decoder_finish_frame (bdec, NULL, 1);
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goto out;
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}
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}
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gboolean
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gst_wavpack_dec_plugin_init (GstPlugin * plugin)
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{
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if (!gst_element_register (plugin, "wavpackdec",
|
|
GST_RANK_PRIMARY, GST_TYPE_WAVPACK_DEC))
|
|
return FALSE;
|
|
GST_DEBUG_CATEGORY_INIT (gst_wavpack_dec_debug, "wavpackdec", 0,
|
|
"Wavpack decoder");
|
|
return TRUE;
|
|
}
|