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Diffstat (limited to 'ext/wavpack/gstwavpackdec.c')
-rw-r--r--ext/wavpack/gstwavpackdec.c497
1 files changed, 0 insertions, 497 deletions
diff --git a/ext/wavpack/gstwavpackdec.c b/ext/wavpack/gstwavpackdec.c
deleted file mode 100644
index 5adaf1be..00000000
--- a/ext/wavpack/gstwavpackdec.c
+++ /dev/null
@@ -1,497 +0,0 @@
-/* GStreamer Wavpack plugin
- * Copyright (c) 2005 Arwed v. Merkatz <v.merkatz@gmx.net>
- * Copyright (c) 2006 Edward Hervey <bilboed@gmail.com>
- * Copyright (c) 2006 Sebastian Dröge <slomo@circular-chaos.org>
- *
- * gstwavpackdec.c: raw Wavpack bitstream decoder
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- */
-
-/**
- * SECTION:element-wavpackdec
- *
- * <refsect2>
- * WavpackDec decodes framed (for example by the WavpackParse element)
- * Wavpack streams and decodes them to raw audio.
- * <ulink url="http://www.wavpack.com/">Wavpack</ulink> is an open-source
- * audio codec that features both lossless and lossy encoding.
- * <title>Example launch line</title>
- * <para>
- * <programlisting>
- * gst-launch filesrc location=test.wv ! wavpackparse ! wavpackdec ! audioconvert ! audioresample ! autoaudiosink
- * </programlisting>
- * This pipeline decodes the Wavpack file test.wv into raw audio buffers and
- * tries to play it back using an automatically found audio sink.
- * </para>
- * </refsect2>
- */
-
-#include <gst/gst.h>
-#include <gst/audio/audio.h>
-
-#include <math.h>
-#include <string.h>
-
-#include <wavpack/wavpack.h>
-#include "gstwavpackdec.h"
-#include "gstwavpackcommon.h"
-#include "gstwavpackstreamreader.h"
-
-
-#define WAVPACK_DEC_MAX_ERRORS 16
-
-GST_DEBUG_CATEGORY_STATIC (gst_wavpack_dec_debug);
-#define GST_CAT_DEFAULT gst_wavpack_dec_debug
-
-static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
- GST_PAD_SINK,
- GST_PAD_ALWAYS,
- GST_STATIC_CAPS ("audio/x-wavpack, "
- "width = (int) [ 1, 32 ], "
- "channels = (int) [ 1, 2 ], "
- "rate = (int) [ 6000, 192000 ], " "framed = (boolean) true")
- );
-
-static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
- GST_PAD_SRC,
- GST_PAD_ALWAYS,
- GST_STATIC_CAPS ("audio/x-raw-int, "
- "width = (int) 32, "
- "depth = (int) [ 1, 32 ], "
- "channels = (int) [ 1, 2 ], "
- "rate = (int) [ 6000, 192000 ], "
- "endianness = (int) BYTE_ORDER, " "signed = (boolean) true")
- );
-
-static GstFlowReturn gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buffer);
-static gboolean gst_wavpack_dec_sink_set_caps (GstPad * pad, GstCaps * caps);
-static gboolean gst_wavpack_dec_sink_event (GstPad * pad, GstEvent * event);
-static void gst_wavpack_dec_finalize (GObject * object);
-static GstStateChangeReturn gst_wavpack_dec_change_state (GstElement * element,
- GstStateChange transition);
-static gboolean gst_wavpack_dec_sink_event (GstPad * pad, GstEvent * event);
-static void gst_wavpack_dec_post_tags (GstWavpackDec * dec);
-
-GST_BOILERPLATE (GstWavpackDec, gst_wavpack_dec, GstElement, GST_TYPE_ELEMENT);
-
-static void
-gst_wavpack_dec_base_init (gpointer klass)
-{
- static const GstElementDetails plugin_details =
- GST_ELEMENT_DETAILS ("Wavpack audio decoder",
- "Codec/Decoder/Audio",
- "Decodes Wavpack audio data",
- "Arwed v. Merkatz <v.merkatz@gmx.net>, "
- "Sebastian Dröge <slomo@circular-chaos.org>");
- GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
-
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&src_factory));
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&sink_factory));
- gst_element_class_set_details (element_class, &plugin_details);
-}
-
-static void
-gst_wavpack_dec_class_init (GstWavpackDecClass * klass)
-{
- GObjectClass *gobject_class = (GObjectClass *) klass;
- GstElementClass *gstelement_class = (GstElementClass *) klass;
-
- gstelement_class->change_state =
- GST_DEBUG_FUNCPTR (gst_wavpack_dec_change_state);
- gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_wavpack_dec_finalize);
-}
-
-static void
-gst_wavpack_dec_reset (GstWavpackDec * dec)
-{
- dec->wv_id.buffer = NULL;
- dec->wv_id.position = dec->wv_id.length = 0;
-
- dec->error_count = 0;
-
- dec->channels = 0;
- dec->sample_rate = 0;
- dec->depth = 0;
-
- gst_segment_init (&dec->segment, GST_FORMAT_UNDEFINED);
-}
-
-static void
-gst_wavpack_dec_init (GstWavpackDec * dec, GstWavpackDecClass * gklass)
-{
- dec->sinkpad = gst_pad_new_from_static_template (&sink_factory, "sink");
- gst_pad_set_chain_function (dec->sinkpad,
- GST_DEBUG_FUNCPTR (gst_wavpack_dec_chain));
- gst_pad_set_setcaps_function (dec->sinkpad,
- GST_DEBUG_FUNCPTR (gst_wavpack_dec_sink_set_caps));
- gst_pad_set_event_function (dec->sinkpad,
- GST_DEBUG_FUNCPTR (gst_wavpack_dec_sink_event));
- gst_element_add_pad (GST_ELEMENT (dec), dec->sinkpad);
-
- dec->srcpad = gst_pad_new_from_static_template (&src_factory, "src");
- gst_pad_use_fixed_caps (dec->srcpad);
- gst_element_add_pad (GST_ELEMENT (dec), dec->srcpad);
-
- dec->context = NULL;
- dec->stream_reader = gst_wavpack_stream_reader_new ();
-
- gst_wavpack_dec_reset (dec);
-}
-
-static void
-gst_wavpack_dec_finalize (GObject * object)
-{
- GstWavpackDec *dec = GST_WAVPACK_DEC (object);
-
- g_free (dec->stream_reader);
- dec->stream_reader = NULL;
-
- G_OBJECT_CLASS (parent_class)->finalize (object);
-}
-
-static gboolean
-gst_wavpack_dec_sink_set_caps (GstPad * pad, GstCaps * caps)
-{
- GstWavpackDec *dec = GST_WAVPACK_DEC (gst_pad_get_parent (pad));
- GstStructure *structure = gst_caps_get_structure (caps, 0);
-
- /* Check if we can set the caps here already */
- if (gst_structure_get_int (structure, "channels", &dec->channels) &&
- gst_structure_get_int (structure, "rate", &dec->sample_rate) &&
- gst_structure_get_int (structure, "width", &dec->depth)) {
- GstCaps *caps;
-
- caps = gst_caps_new_simple ("audio/x-raw-int",
- "rate", G_TYPE_INT, dec->sample_rate,
- "channels", G_TYPE_INT, dec->channels,
- "depth", G_TYPE_INT, dec->depth,
- "width", G_TYPE_INT, 32,
- "endianness", G_TYPE_INT, G_BYTE_ORDER,
- "signed", G_TYPE_BOOLEAN, TRUE, NULL);
-
- GST_DEBUG_OBJECT (dec, "setting caps %" GST_PTR_FORMAT, caps);
-
- /* should always succeed */
- gst_pad_set_caps (dec->srcpad, caps);
- gst_caps_unref (caps);
-
- /* send GST_TAG_AUDIO_CODEC and GST_TAG_BITRATE tags before something
- * is decoded or after the format has changed */
- gst_wavpack_dec_post_tags (dec);
- }
-
- gst_object_unref (dec);
-
- return TRUE;
-}
-
-static gboolean
-gst_wavpack_dec_clip_outgoing_buffer (GstWavpackDec * dec, GstBuffer * buf)
-{
- gint64 start, stop, cstart, cstop, diff;
-
- if (dec->segment.format != GST_FORMAT_TIME)
- return TRUE;
-
- start = GST_BUFFER_TIMESTAMP (buf);
- stop = start + GST_BUFFER_DURATION (buf);
-
- if (gst_segment_clip (&dec->segment, GST_FORMAT_TIME,
- start, stop, &cstart, &cstop)) {
-
- diff = cstart - start;
- if (diff > 0) {
- GST_BUFFER_TIMESTAMP (buf) = cstart;
- GST_BUFFER_DURATION (buf) -= diff;
-
- diff = 4 * dec->channels
- * GST_CLOCK_TIME_TO_FRAMES (diff, dec->sample_rate);
- GST_BUFFER_DATA (buf) += diff;
- GST_BUFFER_SIZE (buf) -= diff;
- }
-
- diff = cstop - stop;
- if (diff > 0) {
- GST_BUFFER_DURATION (buf) -= diff;
-
- diff = 4 * dec->channels
- * GST_CLOCK_TIME_TO_FRAMES (diff, dec->sample_rate);
- GST_BUFFER_SIZE (buf) -= diff;
- }
- } else {
- GST_DEBUG_OBJECT (dec, "buffer is outside configured segment");
- return FALSE;
- }
-
- return TRUE;
-}
-
-static void
-gst_wavpack_dec_post_tags (GstWavpackDec * dec)
-{
- GstTagList *list;
- GstFormat format_time = GST_FORMAT_TIME, format_bytes = GST_FORMAT_BYTES;
- gint64 duration, size;
-
- list = gst_tag_list_new ();
-
- gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
- GST_TAG_AUDIO_CODEC, "Wavpack", NULL);
-
- /* try to estimate the average bitrate */
- if (gst_pad_query_peer_duration (dec->sinkpad, &format_bytes, &size) &&
- gst_pad_query_peer_duration (dec->sinkpad, &format_time, &duration) &&
- size > 0 && duration > 0) {
- guint64 bitrate;
-
- bitrate = gst_util_uint64_scale (size, 8 * GST_SECOND, duration);
- gst_tag_list_add (list, GST_TAG_MERGE_REPLACE, GST_TAG_BITRATE,
- (guint) bitrate, NULL);
- }
-
- gst_element_post_message (GST_ELEMENT (dec),
- gst_message_new_tag (GST_OBJECT (dec), list));
-}
-
-static GstFlowReturn
-gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buf)
-{
- GstWavpackDec *dec;
- GstBuffer *outbuf;
- GstFlowReturn ret = GST_FLOW_OK;
- WavpackHeader wph;
- int32_t decoded, unpacked_size;
- gboolean format_changed;
-
- dec = GST_WAVPACK_DEC (GST_PAD_PARENT (pad));
-
- /* check input, we only accept framed input with complete chunks */
- if (GST_BUFFER_SIZE (buf) < sizeof (WavpackHeader))
- goto input_not_framed;
-
- if (!gst_wavpack_read_header (&wph, GST_BUFFER_DATA (buf)))
- goto invalid_header;
-
- if (GST_BUFFER_SIZE (buf) != wph.ckSize + 4 * 1 + 4)
- goto input_not_framed;
-
- dec->wv_id.buffer = GST_BUFFER_DATA (buf);
- dec->wv_id.length = GST_BUFFER_SIZE (buf);
- dec->wv_id.position = 0;
-
- /* create a new wavpack context if there is none yet but if there
- * was already one (i.e. caps were set on the srcpad) check whether
- * the new one has the same caps */
- if (!dec->context) {
- gchar error_msg[80];
-
- dec->context = WavpackOpenFileInputEx (dec->stream_reader,
- &dec->wv_id, NULL, error_msg, OPEN_STREAMING, 0);
-
- if (!dec->context) {
- GST_WARNING ("Couldn't decode buffer: %s", error_msg);
- dec->error_count++;
- if (dec->error_count <= WAVPACK_DEC_MAX_ERRORS) {
- goto out; /* just return OK for now */
- } else {
- goto decode_error;
- }
- }
- }
-
- g_assert (dec->context != NULL);
-
- dec->error_count = 0;
-
- format_changed =
- (dec->sample_rate != WavpackGetSampleRate (dec->context)) ||
- (dec->channels != WavpackGetNumChannels (dec->context)) ||
- (dec->depth != WavpackGetBitsPerSample (dec->context));
-
- if (!GST_PAD_CAPS (dec->srcpad) || format_changed) {
- GstCaps *caps;
-
- dec->sample_rate = WavpackGetSampleRate (dec->context);
- dec->channels = WavpackGetNumChannels (dec->context);
- dec->depth = WavpackGetBitsPerSample (dec->context);
-
- caps = gst_caps_new_simple ("audio/x-raw-int",
- "rate", G_TYPE_INT, dec->sample_rate,
- "channels", G_TYPE_INT, dec->channels,
- "depth", G_TYPE_INT, dec->depth,
- "width", G_TYPE_INT, 32,
- "endianness", G_TYPE_INT, G_BYTE_ORDER,
- "signed", G_TYPE_BOOLEAN, TRUE, NULL);
-
- GST_DEBUG_OBJECT (dec, "setting caps %" GST_PTR_FORMAT, caps);
-
- /* should always succeed */
- gst_pad_set_caps (dec->srcpad, caps);
- gst_caps_unref (caps);
-
- /* send GST_TAG_AUDIO_CODEC and GST_TAG_BITRATE tags before something
- * is decoded or after the format has changed */
- gst_wavpack_dec_post_tags (dec);
- }
-
- /* alloc output buffer */
- unpacked_size = 4 * wph.block_samples * dec->channels;
- ret = gst_pad_alloc_buffer (dec->srcpad, GST_BUFFER_OFFSET (buf),
- unpacked_size, GST_PAD_CAPS (dec->srcpad), &outbuf);
-
- if (ret != GST_FLOW_OK)
- goto out;
-
- gst_buffer_stamp (outbuf, buf);
- /* If we got a DISCONT buffer forward the flag. Nothing else
- * has to be done as libwavpack doesn't store state between
- * Wavpack blocks */
- if (GST_BUFFER_IS_DISCONT (buf))
- GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
-
- /* decode */
- decoded = WavpackUnpackSamples (dec->context,
- (int32_t *) GST_BUFFER_DATA (outbuf), wph.block_samples);
- if (decoded != wph.block_samples)
- goto decode_error;
-
- if (gst_wavpack_dec_clip_outgoing_buffer (dec, outbuf)) {
- GST_LOG_OBJECT (dec, "pushing buffer with time %" GST_TIME_FORMAT,
- GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)));
- ret = gst_pad_push (dec->srcpad, outbuf);
- } else {
- gst_buffer_unref (outbuf);
- }
-
-out:
-
- if (G_UNLIKELY (ret != GST_FLOW_OK)) {
- GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (ret));
- }
-
- gst_buffer_unref (buf);
-
- return ret;
-
-/* ERRORS */
-input_not_framed:
- {
- GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("Expected framed input"));
- gst_buffer_unref (buf);
- return GST_FLOW_ERROR;
- }
-invalid_header:
- {
- GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("Invalid wavpack header"));
- gst_buffer_unref (buf);
- return GST_FLOW_ERROR;
- }
-decode_error:
- {
- GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL),
- ("Failed to decode wavpack stream"));
- gst_buffer_unref (outbuf);
- gst_buffer_unref (buf);
- return GST_FLOW_ERROR;
- }
-}
-
-static gboolean
-gst_wavpack_dec_sink_event (GstPad * pad, GstEvent * event)
-{
- GstWavpackDec *dec = GST_WAVPACK_DEC (gst_pad_get_parent (pad));
-
- GST_LOG_OBJECT (dec, "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 (&dec->segment, is_update, rate, fmt,
- start, end, base);
- } else {
- gst_segment_init (&dec->segment, GST_FORMAT_UNDEFINED);
- }
- break;
- }
- default:
- break;
- }
-
- gst_object_unref (dec);
- 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 *dec = GST_WAVPACK_DEC (element);
-
- switch (transition) {
- case GST_STATE_CHANGE_NULL_TO_READY:
- break;
- case GST_STATE_CHANGE_READY_TO_PAUSED:
- 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 (dec->context) {
- WavpackCloseFile (dec->context);
- dec->context = NULL;
- }
-
- gst_wavpack_dec_reset (dec);
- break;
- case GST_STATE_CHANGE_READY_TO_NULL:
- break;
- default:
- break;
- }
-
- return ret;
-}
-
-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, "wavpack_dec", 0,
- "Wavpack decoder");
- return TRUE;
-}
generated by cgit v1.2.1 (git 2.18.0) at 2025-02-07 12:33:27 +0000