mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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8dd45c0027
Original commit message from CVS: Patch by: Sebastian Dröge <slomo at circular-chaos.org> * configure.ac: Check for wavpack version and define WAVPACK_OLD_API if necessary. * ext/wavpack/Makefile.am: * ext/wavpack/gstwavpackcommon.c: (gst_wavpack_read_header), (gst_wavpack_read_metadata): * ext/wavpack/gstwavpackcommon.h: * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_base_init), (gst_wavpack_dec_class_init), (gst_wavpack_dec_init), (gst_wavpack_dec_finalize), (gst_wavpack_dec_format_samples), (gst_wavpack_dec_clip_outgoing_buffer), (gst_wavpack_dec_chain), (gst_wavpack_dec_sink_event), (gst_wavpack_dec_change_state), (gst_wavpack_dec_request_new_pad), (gst_wavpack_dec_plugin_init): * ext/wavpack/gstwavpackdec.h: * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_class_init), (gst_wavpack_enc_init), (gst_wavpack_enc_finalize), (gst_wavpack_enc_set_wp_config): * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_base_init), (gst_wavpack_parse_finalize), (gst_wavpack_parse_class_init), (gst_wavpack_parse_index_get_entry_from_sample), (gst_wavpack_parse_scan_to_find_sample), (gst_wavpack_parse_handle_seek_event), (gst_wavpack_parse_create_src_pad): * ext/wavpack/gstwavpackstreamreader.c: * ext/wavpack/gstwavpackstreamreader.h: Port to new/official wavpack API, don't use API that was exported in wavpack header files and in the lib but meant to be private, at least not for recent wavpack versions; misc. 'cleanups' (#347443).
570 lines
17 KiB
C
570 lines
17 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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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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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#define WAVPACK_DEC_MAX_ERRORS 16
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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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"width = (int) { 8, 16, 24, 32 }, "
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"channels = (int) [ 1, 2 ], "
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"rate = (int) [ 6000, 192000 ], " "framed = (boolean) true")
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);
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#if 0
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static GstStaticPadTemplate wvc_sink_factory =
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GST_STATIC_PAD_TEMPLATE ("wvcsink",
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GST_PAD_SINK,
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GST_PAD_REQUEST,
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GST_STATIC_CAPS ("audio/x-wavpack-correction, " "framed = (boolean) true")
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);
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#endif
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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-int, "
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"width = (int) { 8, 16, 24, 32 }, "
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"depth = (int) { 8, 16, 24, 32 }, "
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"channels = (int) [ 1, 2 ], "
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"rate = (int) [ 6000, 192000 ], "
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"endianness = (int) LITTLE_ENDIAN, " "signed = (boolean) true")
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);
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static GstFlowReturn gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buffer);
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static gboolean gst_wavpack_dec_sink_event (GstPad * pad, GstEvent * event);
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static void gst_wavpack_dec_finalize (GObject * object);
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static GstStateChangeReturn gst_wavpack_dec_change_state (GstElement * element,
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GstStateChange transition);
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static gboolean gst_wavpack_dec_sink_event (GstPad * pad, GstEvent * event);
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#if 0
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static GstPad *gst_wavpack_dec_request_new_pad (GstElement * element,
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GstPadTemplate * templ, const gchar * name);
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#endif
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GST_BOILERPLATE (GstWavpackDec, gst_wavpack_dec, GstElement, GST_TYPE_ELEMENT);
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#if 0
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static GstPadLinkReturn
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gst_wavpack_dec_wvclink (GstPad * pad, GstPad * peer)
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{
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if (!gst_caps_is_fixed (GST_PAD_CAPS (peer)))
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return GST_PAD_LINK_REFUSED;
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return GST_PAD_LINK_OK;
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}
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#endif
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static void
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gst_wavpack_dec_base_init (gpointer klass)
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{
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static const GstElementDetails plugin_details =
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GST_ELEMENT_DETAILS ("WavePack audio decoder",
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"Codec/Decoder/Audio",
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"Decode 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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GstElementClass *element_class = GST_ELEMENT_CLASS (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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#if 0
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&wvc_sink_factory));
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#endif
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gst_element_class_set_details (element_class, &plugin_details);
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}
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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 *gstelement_class = (GstElementClass *) klass;
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gstelement_class->change_state =
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GST_DEBUG_FUNCPTR (gst_wavpack_dec_change_state);
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#if 0
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gstelement_class->request_new_pad =
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GST_DEBUG_FUNCPTR (gst_wavpack_dec_request_new_pad);
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#endif
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gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_wavpack_dec_finalize);
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}
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static void
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gst_wavpack_dec_init (GstWavpackDec * wavpackdec, GstWavpackDecClass * gklass)
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{
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GstElementClass *klass = GST_ELEMENT_GET_CLASS (wavpackdec);
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wavpackdec->sinkpad =
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gst_pad_new_from_template (gst_element_class_get_pad_template (klass,
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"sink"), "sink");
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gst_pad_set_chain_function (wavpackdec->sinkpad,
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GST_DEBUG_FUNCPTR (gst_wavpack_dec_chain));
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gst_pad_set_event_function (wavpackdec->sinkpad,
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GST_DEBUG_FUNCPTR (gst_wavpack_dec_sink_event));
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gst_element_add_pad (GST_ELEMENT (wavpackdec), wavpackdec->sinkpad);
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wavpackdec->srcpad =
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gst_pad_new_from_template (gst_element_class_get_pad_template (klass,
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"src"), "src");
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gst_pad_use_fixed_caps (wavpackdec->srcpad);
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gst_element_add_pad (GST_ELEMENT (wavpackdec), wavpackdec->srcpad);
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wavpackdec->context = NULL;
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wavpackdec->stream_reader = gst_wavpack_stream_reader_new ();
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wavpackdec->wv_id.buffer = NULL;
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wavpackdec->wv_id.position = wavpackdec->wv_id.length = 0;
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/*
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wavpackdec->wvc_id.buffer = NULL;
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wavpackdec->wvc_id.position = wavpackdec->wvc_id.length = 0;
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wavpackdec->wvcsinkpad = NULL;
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*/
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wavpackdec->error_count = 0;
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wavpackdec->channels = 0;
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wavpackdec->sample_rate = 0;
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wavpackdec->width = 0;
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gst_segment_init (&wavpackdec->segment, GST_FORMAT_UNDEFINED);
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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 *wavpackdec = GST_WAVPACK_DEC (object);
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g_free (wavpackdec->stream_reader);
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wavpackdec->stream_reader = NULL;
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static void
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gst_wavpack_dec_format_samples (GstWavpackDec * wavpackdec, guint8 * dst,
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int32_t * samples, guint num_samples)
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{
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gint i;
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int32_t temp;
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switch (wavpackdec->width) {
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case 8:
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for (i = 0; i < num_samples * wavpackdec->channels; ++i)
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*dst++ = (guint8) (*samples++);
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break;
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case 16:
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for (i = 0; i < num_samples * wavpackdec->channels; ++i) {
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*dst++ = (guint8) (temp = *samples++);
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*dst++ = (guint8) (temp >> 8);
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}
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break;
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case 24:
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for (i = 0; i < num_samples * wavpackdec->channels; ++i) {
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*dst++ = (guint8) (temp = *samples++);
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*dst++ = (guint8) (temp >> 8);
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*dst++ = (guint8) (temp >> 16);
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}
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break;
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case 32:
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for (i = 0; i < num_samples * wavpackdec->channels; ++i) {
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*dst++ = (guint8) (temp = *samples++);
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*dst++ = (guint8) (temp >> 8);
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*dst++ = (guint8) (temp >> 16);
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*dst++ = (guint8) (temp >> 24);
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}
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break;
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default:
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break;
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}
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}
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static gboolean
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gst_wavpack_dec_clip_outgoing_buffer (GstWavpackDec * wavpackdec,
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GstBuffer * buf)
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{
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gint64 start, stop, cstart, cstop, diff;
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if (wavpackdec->segment.format != GST_FORMAT_TIME)
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return TRUE;
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start = GST_BUFFER_TIMESTAMP (buf);
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stop = start + GST_BUFFER_DURATION (buf);
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if (gst_segment_clip (&wavpackdec->segment, GST_FORMAT_TIME,
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start, stop, &cstart, &cstop)) {
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diff = cstart - start;
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if (diff > 0) {
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GST_BUFFER_TIMESTAMP (buf) = cstart;
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GST_BUFFER_DURATION (buf) -= diff;
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diff = ((wavpackdec->width + 7) >> 3) * wavpackdec->channels
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* GST_CLOCK_TIME_TO_FRAMES (diff, wavpackdec->sample_rate);
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GST_BUFFER_DATA (buf) += diff;
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GST_BUFFER_SIZE (buf) -= diff;
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}
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diff = cstop - stop;
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if (diff > 0) {
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GST_BUFFER_DURATION (buf) -= diff;
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diff = ((wavpackdec->width + 7) >> 3) * wavpackdec->channels
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* GST_CLOCK_TIME_TO_FRAMES (diff, wavpackdec->sample_rate);
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GST_BUFFER_SIZE (buf) -= diff;
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}
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} else {
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GST_DEBUG_OBJECT (wavpackdec, "buffer is outside configured segment");
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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 GstFlowReturn
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gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buf)
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{
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GstWavpackDec *wavpackdec;
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GstBuffer *outbuf;
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GstBuffer *cbuf = NULL;
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GstFlowReturn ret = GST_FLOW_OK;
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WavpackHeader wph;
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int32_t *unpack_buf;
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int32_t unpacked_sample_count;
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wavpackdec = GST_WAVPACK_DEC (GST_PAD_PARENT (pad));
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/* we only accept framed input with complete chunks */
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g_assert (GST_BUFFER_SIZE (buf) >= sizeof (WavpackHeader));
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gst_wavpack_read_header (&wph, GST_BUFFER_DATA (buf));
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g_assert (GST_BUFFER_SIZE (buf) ==
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wph.ckSize + 4 * sizeof (char) + sizeof (uint32_t));
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wavpackdec->wv_id.buffer = GST_BUFFER_DATA (buf);
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wavpackdec->wv_id.length = GST_BUFFER_SIZE (buf);
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wavpackdec->wv_id.position = 0;
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#if 0
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/* check whether the correction pad is linked and we can get
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* the correction chunk that corresponds to our current data */
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if (gst_pad_is_linked (wavpackdec->wvcsinkpad)) {
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if (GST_FLOW_OK != gst_pad_pull_range (wavpackdec->wvcsinkpad,
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GST_BUFFER_OFFSET (buf), -1, &cbuf)) {
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cbuf = NULL;
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} else {
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/* this won't work (tpm) */
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if (!(GST_BUFFER_TIMESTAMP (cbuf) == GST_BUFFER_TIMESTAMP (buf)) ||
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!(GST_BUFFER_DURATION (cbuf) == GST_BUFFER_DURATION (buf)) ||
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!(GST_BUFFER_OFFSET (cbuf) == GST_BUFFER_OFFSET (buf)) ||
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!(GST_BUFFER_OFFSET_END (cbuf) == GST_BUFFER_OFFSET (buf))) {
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gst_buffer_unref (cbuf);
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cbuf = NULL;
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} else {
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wavpackdec->wvc_id.buffer = GST_BUFFER_DATA (cbuf);
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wavpackdec->wvc_id.length = GST_BUFFER_SIZE (cbuf);
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wavpackdec->wvc_id.position = 0;
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}
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}
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}
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#endif
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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 (!wavpackdec->context) {
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gchar error_msg[80];
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/*
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wavpackdec->context =
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WavpackOpenFileInputEx (wavpackdec->stream_reader, &wavpackdec->wv_id,
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(cbuf) ? &wavpackdec->wvc_id : NULL, error_msg, OPEN_STREAMING, 0);
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*/
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wavpackdec->context = WavpackOpenFileInputEx (wavpackdec->stream_reader,
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&wavpackdec->wv_id, NULL, error_msg, OPEN_STREAMING, 0);
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if (!wavpackdec->context) {
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wavpackdec->error_count++;
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GST_ELEMENT_WARNING (wavpackdec, LIBRARY, INIT, (NULL),
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("Couldn't open buffer for decoding: %s", error_msg));
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if (wavpackdec->error_count <= WAVPACK_DEC_MAX_ERRORS) {
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ret = GST_FLOW_OK;
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} else {
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ret = GST_FLOW_ERROR;
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}
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gst_buffer_unref (buf);
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if (cbuf) {
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gst_buffer_unref (cbuf);
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}
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return ret;
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}
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if (GST_PAD_CAPS (wavpackdec->srcpad)) {
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if ((wavpackdec->sample_rate !=
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WavpackGetSampleRate (wavpackdec->context))
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|| (wavpackdec->channels !=
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WavpackGetNumChannels (wavpackdec->context))
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|| (wavpackdec->width !=
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WavpackGetBitsPerSample (wavpackdec->context))) {
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gst_buffer_unref (buf);
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if (cbuf) {
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gst_buffer_unref (cbuf);
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}
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/* FIXME: use the right error */
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GST_ELEMENT_ERROR (wavpackdec, LIBRARY, INIT, (NULL),
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("Got Wavpack chunk with changed format settings!"));
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return GST_FLOW_ERROR;
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}
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}
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}
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wavpackdec->error_count = 0;
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if (!GST_PAD_CAPS (wavpackdec->srcpad)) {
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GstCaps *caps;
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g_assert (wavpackdec->context);
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wavpackdec->sample_rate = WavpackGetSampleRate (wavpackdec->context);
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wavpackdec->channels = WavpackGetNumChannels (wavpackdec->context);
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wavpackdec->width = WavpackGetBitsPerSample (wavpackdec->context);
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caps = gst_caps_new_simple ("audio/x-raw-int",
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"rate", G_TYPE_INT, wavpackdec->sample_rate,
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"channels", G_TYPE_INT, wavpackdec->channels,
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"depth", G_TYPE_INT, wavpackdec->width,
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"width", G_TYPE_INT, wavpackdec->width,
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"endianness", G_TYPE_INT, G_LITTLE_ENDIAN,
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"signed", G_TYPE_BOOLEAN, TRUE, NULL);
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if (!gst_pad_set_caps (wavpackdec->srcpad, caps)) {
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gst_caps_unref (caps);
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WavpackCloseFile (wavpackdec->context);
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wavpackdec->context = NULL;
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gst_buffer_unref (buf);
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if (cbuf) {
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gst_buffer_unref (cbuf);
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}
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/* FIXME: use the right error */
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GST_ELEMENT_ERROR (wavpackdec, LIBRARY, INIT, (NULL),
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("Couldn't set caps on source pad: %" GST_PTR_FORMAT, caps));
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return GST_FLOW_ERROR;
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}
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gst_caps_unref (caps);
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gst_pad_use_fixed_caps (wavpackdec->srcpad);
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}
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g_assert (wavpackdec->context);
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unpack_buf =
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(int32_t *) g_malloc (sizeof (int32_t) * wph.block_samples *
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wavpackdec->channels);
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unpacked_sample_count =
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WavpackUnpackSamples (wavpackdec->context, unpack_buf, wph.block_samples);
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g_assert (unpacked_sample_count == wph.block_samples);
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ret =
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gst_pad_alloc_buffer_and_set_caps (wavpackdec->srcpad,
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GST_BUFFER_OFFSET (buf),
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wph.block_samples * ((wavpackdec->width +
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7) >> 3) * wavpackdec->channels,
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GST_PAD_CAPS (wavpackdec->srcpad), &outbuf);
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if (GST_FLOW_IS_FATAL (ret)) {
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WavpackCloseFile (wavpackdec->context);
|
|
wavpackdec->context = NULL;
|
|
g_free (unpack_buf);
|
|
gst_buffer_unref (buf);
|
|
if (cbuf) {
|
|
gst_buffer_unref (cbuf);
|
|
}
|
|
return ret;
|
|
} else if (ret != GST_FLOW_OK) {
|
|
g_free (unpack_buf);
|
|
gst_buffer_unref (buf);
|
|
if (cbuf) {
|
|
gst_buffer_unref (cbuf);
|
|
}
|
|
return ret;
|
|
}
|
|
|
|
gst_wavpack_dec_format_samples (wavpackdec, GST_BUFFER_DATA (outbuf),
|
|
unpack_buf, wph.block_samples);
|
|
g_free (unpack_buf);
|
|
gst_buffer_stamp (outbuf, buf);
|
|
gst_buffer_unref (buf);
|
|
if (cbuf) {
|
|
gst_buffer_unref (cbuf);
|
|
}
|
|
|
|
if (gst_wavpack_dec_clip_outgoing_buffer (wavpackdec, outbuf)) {
|
|
GST_LOG_OBJECT (wavpackdec, "pushing buffer with time %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)));
|
|
ret = gst_pad_push (wavpackdec->srcpad, outbuf);
|
|
if (ret != GST_FLOW_OK) {
|
|
GST_DEBUG_OBJECT (wavpackdec, "pad_push: %s", gst_flow_get_name (ret));
|
|
}
|
|
} else {
|
|
gst_buffer_unref (outbuf);
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static gboolean
|
|
gst_wavpack_dec_sink_event (GstPad * pad, GstEvent * event)
|
|
{
|
|
GstWavpackDec *wavpackdec = GST_WAVPACK_DEC (gst_pad_get_parent (pad));
|
|
|
|
GST_LOG_OBJECT (wavpackdec, "Received %s event", GST_EVENT_TYPE_NAME (event));
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_NEWSEGMENT:{
|
|
GstFormat fmt;
|
|
gboolean is_update;
|
|
gint64 start, end, base;
|
|
gdouble rate;
|
|
|
|
gst_event_parse_new_segment (event, &is_update, &rate, &fmt, &start,
|
|
&end, &base);
|
|
if (fmt == GST_FORMAT_TIME) {
|
|
GST_DEBUG ("Got NEWSEGMENT event in GST_FORMAT_TIME, passing on (%"
|
|
GST_TIME_FORMAT " - %" GST_TIME_FORMAT ")", GST_TIME_ARGS (start),
|
|
GST_TIME_ARGS (end));
|
|
gst_segment_set_newsegment (&wavpackdec->segment, is_update, rate, fmt,
|
|
start, end, base);
|
|
} else {
|
|
gst_segment_init (&wavpackdec->segment, GST_FORMAT_UNDEFINED);
|
|
}
|
|
break;
|
|
}
|
|
default:
|
|
break;
|
|
}
|
|
|
|
gst_object_unref (wavpackdec);
|
|
return gst_pad_event_default (pad, event);
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_wavpack_dec_change_state (GstElement * element, GstStateChange transition)
|
|
{
|
|
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
|
|
GstWavpackDec *wavpackdec = GST_WAVPACK_DEC (element);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_NULL_TO_READY:
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
gst_segment_init (&wavpackdec->segment, GST_FORMAT_UNDEFINED);
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
if (wavpackdec->context) {
|
|
WavpackCloseFile (wavpackdec->context);
|
|
wavpackdec->context = NULL;
|
|
}
|
|
wavpackdec->wv_id.buffer = NULL;
|
|
wavpackdec->wv_id.position = 0;
|
|
wavpackdec->wv_id.length = 0;
|
|
/*
|
|
wavpackdec->wvc_id.buffer = NULL;
|
|
wavpackdec->wvc_id.position = 0;
|
|
wavpackdec->wvc_id.length = 0;
|
|
wavpackdec->error_count = 0;
|
|
wavpackdec->wvcsinkpad = NULL;
|
|
*/
|
|
wavpackdec->channels = 0;
|
|
wavpackdec->sample_rate = 0;
|
|
wavpackdec->width = 0;
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_NULL:
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
#if 0
|
|
static GstPad *
|
|
gst_wavpack_dec_request_new_pad (GstElement * element,
|
|
GstPadTemplate * templ, const gchar * name)
|
|
{
|
|
GstWavpackDec *wavpackdec = GST_WAVPACK_DEC (element);
|
|
GstPad *pad;
|
|
|
|
if (wavpackdec->wvcsinkpad == NULL) {
|
|
wavpackdec->wvcsinkpad = gst_pad_new_from_template (template, name);
|
|
gst_pad_set_link_function (wavpackdec->wvcsinkpad, gst_wavpack_dec_wvclink);
|
|
gst_pad_use_fixed_caps (wavpackdec->wvcsinkpad);
|
|
gst_element_add_pad (GST_ELEMENT (wavpackdec), wavpackdec->wvcsinkpad);
|
|
gst_element_no_more_pads (GST_ELEMENT (wavpackdec));
|
|
} else {
|
|
pad = NULL;
|
|
}
|
|
|
|
return pad;
|
|
}
|
|
#endif
|
|
|
|
gboolean
|
|
gst_wavpack_dec_plugin_init (GstPlugin * plugin)
|
|
{
|
|
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;
|
|
}
|