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Diffstat (limited to 'ext/wavpack/gstwavpackdec.c')
-rwxr-xr-xext/wavpack/gstwavpackdec.c483
1 files changed, 483 insertions, 0 deletions
diff --git a/ext/wavpack/gstwavpackdec.c b/ext/wavpack/gstwavpackdec.c
new file mode 100755
index 0000000..10659ec
--- /dev/null
+++ b/ext/wavpack/gstwavpackdec.c
@@ -0,0 +1,483 @@
+/* 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., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+/**
+ * SECTION:element-wavpackdec
+ *
+ * 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.
+ *
+ * <refsect2>
+ * <title>Example launch line</title>
+ * |[
+ * gst-launch-1.0 filesrc location=test.wv ! wavpackparse ! wavpackdec ! audioconvert ! audioresample ! autoaudiosink
+ * ]| This pipeline decodes the Wavpack file test.wv into raw audio buffers and
+ * tries to play it back using an automatically found audio sink.
+ * </refsect2>
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#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"
+
+
+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, "
+ "depth = (int) [ 1, 32 ], "
+ "channels = (int) [ 1, 8 ], "
+ "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, "
+ "format = (string) S8, "
+ "layout = (string) interleaved, "
+ "channels = (int) [ 1, 8 ], "
+ "rate = (int) [ 6000, 192000 ]; "
+ "audio/x-raw, "
+ "format = (string) " GST_AUDIO_NE (S16) ", "
+ "layout = (string) interleaved, "
+ "channels = (int) [ 1, 8 ], "
+ "rate = (int) [ 6000, 192000 ]; "
+ "audio/x-raw, "
+ "format = (string) " GST_AUDIO_NE (S32) ", "
+ "layout = (string) interleaved, "
+ "channels = (int) [ 1, 8 ], " "rate = (int) [ 6000, 192000 ]")
+ );
+
+static gboolean gst_wavpack_dec_start (GstAudioDecoder * dec);
+static gboolean gst_wavpack_dec_stop (GstAudioDecoder * dec);
+static gboolean gst_wavpack_dec_set_format (GstAudioDecoder * dec,
+ GstCaps * caps);
+static GstFlowReturn gst_wavpack_dec_handle_frame (GstAudioDecoder * dec,
+ GstBuffer * buffer);
+
+static void gst_wavpack_dec_finalize (GObject * object);
+static void gst_wavpack_dec_post_tags (GstWavpackDec * dec);
+
+#define gst_wavpack_dec_parent_class parent_class
+G_DEFINE_TYPE (GstWavpackDec, gst_wavpack_dec, GST_TYPE_AUDIO_DECODER);
+
+static void
+gst_wavpack_dec_class_init (GstWavpackDecClass * klass)
+{
+ GObjectClass *gobject_class = (GObjectClass *) klass;
+ GstElementClass *element_class = (GstElementClass *) (klass);
+ GstAudioDecoderClass *base_class = (GstAudioDecoderClass *) (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_static_metadata (element_class, "Wavpack audio decoder",
+ "Codec/Decoder/Audio",
+ "Decodes Wavpack audio data",
+ "Arwed v. Merkatz <v.merkatz@gmx.net>, "
+ "Sebastian Dröge <slomo@circular-chaos.org>");
+
+ gobject_class->finalize = gst_wavpack_dec_finalize;
+
+ base_class->start = GST_DEBUG_FUNCPTR (gst_wavpack_dec_start);
+ base_class->stop = GST_DEBUG_FUNCPTR (gst_wavpack_dec_stop);
+ base_class->set_format = GST_DEBUG_FUNCPTR (gst_wavpack_dec_set_format);
+ base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_wavpack_dec_handle_frame);
+}
+
+static void
+gst_wavpack_dec_reset (GstWavpackDec * dec)
+{
+ dec->wv_id.buffer = NULL;
+ dec->wv_id.position = dec->wv_id.length = 0;
+
+ dec->channels = 0;
+ dec->channel_mask = 0;
+ dec->sample_rate = 0;
+ dec->depth = 0;
+}
+
+static void
+gst_wavpack_dec_init (GstWavpackDec * dec)
+{
+ dec->context = NULL;
+ dec->stream_reader = gst_wavpack_stream_reader_new ();
+
+ gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (dec), TRUE);
+
+ 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_start (GstAudioDecoder * dec)
+{
+ GST_DEBUG_OBJECT (dec, "start");
+
+ /* never mind a few errors */
+ gst_audio_decoder_set_max_errors (dec, 16);
+ /* don't bother us with flushing */
+ gst_audio_decoder_set_drainable (dec, FALSE);
+ /* aim for some perfect timestamping */
+ gst_audio_decoder_set_tolerance (dec, 10 * GST_MSECOND);
+
+ return TRUE;
+}
+
+static gboolean
+gst_wavpack_dec_stop (GstAudioDecoder * dec)
+{
+ GstWavpackDec *wpdec = GST_WAVPACK_DEC (dec);
+
+ GST_DEBUG_OBJECT (dec, "stop");
+
+ if (wpdec->context) {
+ WavpackCloseFile (wpdec->context);
+ wpdec->context = NULL;
+ }
+
+ gst_wavpack_dec_reset (wpdec);
+
+ return TRUE;
+}
+
+static void
+gst_wavpack_dec_negotiate (GstWavpackDec * dec)
+{
+ GstAudioInfo info;
+ GstAudioFormat fmt;
+ GstAudioChannelPosition pos[64] = { GST_AUDIO_CHANNEL_POSITION_INVALID, };
+
+ /* arrange for 1, 2 or 4-byte width == depth output */
+ dec->width = dec->depth;
+ switch (dec->depth) {
+ case 8:
+ fmt = GST_AUDIO_FORMAT_S8;
+ break;
+ case 16:
+ fmt = _GST_AUDIO_FORMAT_NE (S16);
+ break;
+ case 24:
+ case 32:
+ fmt = _GST_AUDIO_FORMAT_NE (S32);
+ dec->width = 32;
+ break;
+ default:
+ fmt = GST_AUDIO_FORMAT_UNKNOWN;
+ g_assert_not_reached ();
+ break;
+ }
+
+ g_assert (dec->channel_mask != 0);
+
+ if (!gst_wavpack_get_channel_positions (dec->channels,
+ dec->channel_mask, pos))
+ GST_WARNING_OBJECT (dec, "Failed to set channel layout");
+
+ gst_audio_info_init (&info);
+ gst_audio_info_set_format (&info, fmt, dec->sample_rate, dec->channels, pos);
+
+ gst_audio_channel_positions_to_valid_order (info.position, info.channels);
+ gst_audio_get_channel_reorder_map (info.channels,
+ info.position, pos, dec->channel_reorder_map);
+
+ /* should always succeed */
+ gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (dec), &info);
+}
+
+static gboolean
+gst_wavpack_dec_set_format (GstAudioDecoder * bdec, GstCaps * caps)
+{
+ /* pretty much nothing to do here,
+ * we'll parse it all from the stream and setup then */
+
+ 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;
+
+ /* try to estimate the average bitrate */
+ if (gst_pad_peer_query_duration (GST_AUDIO_DECODER_SINK_PAD (dec),
+ format_bytes, &size) &&
+ gst_pad_peer_query_duration (GST_AUDIO_DECODER_SINK_PAD (dec),
+ format_time, &duration) && size > 0 && duration > 0) {
+ guint64 bitrate;
+
+ list = gst_tag_list_new_empty ();
+
+ 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_audio_decoder_merge_tags (GST_AUDIO_DECODER (dec), list,
+ GST_TAG_MERGE_REPLACE);
+ }
+}
+
+static GstFlowReturn
+gst_wavpack_dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buf)
+{
+ GstWavpackDec *dec;
+ GstBuffer *outbuf = NULL;
+ GstFlowReturn ret = GST_FLOW_OK;
+ WavpackHeader wph;
+ int32_t decoded, unpacked_size;
+ gboolean format_changed;
+ gint width, depth, i, j, max;
+ gint32 *dec_data = NULL;
+ guint8 *out_data;
+ GstMapInfo map, omap;
+
+ dec = GST_WAVPACK_DEC (bdec);
+
+ g_return_val_if_fail (buf != NULL, GST_FLOW_ERROR);
+
+ gst_buffer_map (buf, &map, GST_MAP_READ);
+
+ /* check input, we only accept framed input with complete chunks */
+ if (map.size < sizeof (WavpackHeader))
+ goto input_not_framed;
+
+ if (!gst_wavpack_read_header (&wph, map.data))
+ goto invalid_header;
+
+ if (map.size < wph.ckSize + 4 * 1 + 4)
+ goto input_not_framed;
+
+ if (!(wph.flags & INITIAL_BLOCK))
+ goto input_not_framed;
+
+ dec->wv_id.buffer = map.data;
+ dec->wv_id.length = map.size;
+ 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);
+
+ /* expect this to work */
+ if (!dec->context) {
+ GST_WARNING_OBJECT (dec, "Couldn't decode buffer: %s", error_msg);
+ goto context_failed;
+ }
+ }
+
+ g_assert (dec->context != NULL);
+
+ format_changed =
+ (dec->sample_rate != WavpackGetSampleRate (dec->context)) ||
+ (dec->channels != WavpackGetNumChannels (dec->context)) ||
+ (dec->depth != WavpackGetBytesPerSample (dec->context) * 8) ||
+#ifdef WAVPACK_OLD_API
+ (dec->channel_mask != dec->context->config.channel_mask);
+#else
+ (dec->channel_mask != WavpackGetChannelMask (dec->context));
+#endif
+
+ if (!gst_pad_has_current_caps (GST_AUDIO_DECODER_SRC_PAD (dec)) ||
+ format_changed) {
+ gint channel_mask;
+
+ dec->sample_rate = WavpackGetSampleRate (dec->context);
+ dec->channels = WavpackGetNumChannels (dec->context);
+ dec->depth = WavpackGetBytesPerSample (dec->context) * 8;
+
+#ifdef WAVPACK_OLD_API
+ channel_mask = dec->context->config.channel_mask;
+#else
+ channel_mask = WavpackGetChannelMask (dec->context);
+#endif
+ if (channel_mask == 0)
+ channel_mask = gst_wavpack_get_default_channel_mask (dec->channels);
+
+ dec->channel_mask = channel_mask;
+
+ gst_wavpack_dec_negotiate (dec);
+
+ /* 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 */
+ dec_data = g_malloc (4 * wph.block_samples * dec->channels);
+
+ /* decode */
+ decoded = WavpackUnpackSamples (dec->context, dec_data, wph.block_samples);
+ if (decoded != wph.block_samples)
+ goto decode_error;
+
+ unpacked_size = (dec->width / 8) * wph.block_samples * dec->channels;
+ outbuf = gst_buffer_new_and_alloc (unpacked_size);
+
+ /* legacy; pass along offset, whatever that might entail */
+ GST_BUFFER_OFFSET (outbuf) = GST_BUFFER_OFFSET (buf);
+
+ gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
+ out_data = omap.data;
+
+ width = dec->width;
+ depth = dec->depth;
+ max = dec->channels * wph.block_samples;
+ if (width == 8) {
+ gint8 *outbuffer = (gint8 *) out_data;
+ gint *reorder_map = dec->channel_reorder_map;
+
+ for (i = 0; i < max; i += dec->channels) {
+ for (j = 0; j < dec->channels; j++)
+ *outbuffer++ = (gint8) (dec_data[i + reorder_map[j]]);
+ }
+ } else if (width == 16) {
+ gint16 *outbuffer = (gint16 *) out_data;
+ gint *reorder_map = dec->channel_reorder_map;
+
+ for (i = 0; i < max; i += dec->channels) {
+ for (j = 0; j < dec->channels; j++)
+ *outbuffer++ = (gint16) (dec_data[i + reorder_map[j]]);
+ }
+ } else if (dec->width == 32) {
+ gint32 *outbuffer = (gint32 *) out_data;
+ gint *reorder_map = dec->channel_reorder_map;
+
+ if (width != depth) {
+ for (i = 0; i < max; i += dec->channels) {
+ for (j = 0; j < dec->channels; j++)
+ *outbuffer++ =
+ (gint32) (dec_data[i + reorder_map[j]] << (width - depth));
+ }
+ } else {
+ for (i = 0; i < max; i += dec->channels) {
+ for (j = 0; j < dec->channels; j++)
+ *outbuffer++ = (gint32) (dec_data[i + reorder_map[j]]);
+ }
+ }
+ } else {
+ g_assert_not_reached ();
+ }
+
+ gst_buffer_unmap (outbuf, &omap);
+ gst_buffer_unmap (buf, &map);
+ buf = NULL;
+
+ g_free (dec_data);
+
+ ret = gst_audio_decoder_finish_frame (bdec, outbuf, 1);
+
+out:
+ if (buf)
+ gst_buffer_unmap (buf, &map);
+
+ if (G_UNLIKELY (ret != GST_FLOW_OK)) {
+ GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (ret));
+ }
+
+ return ret;
+
+/* ERRORS */
+input_not_framed:
+ {
+ GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("Expected framed input"));
+ ret = GST_FLOW_ERROR;
+ goto out;
+ }
+invalid_header:
+ {
+ GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("Invalid wavpack header"));
+ ret = GST_FLOW_ERROR;
+ goto out;
+ }
+context_failed:
+ {
+ GST_AUDIO_DECODER_ERROR (bdec, 1, LIBRARY, INIT, (NULL),
+ ("error creating Wavpack context"), ret);
+ goto out;
+ }
+decode_error:
+ {
+ const gchar *reason = "unknown";
+
+ if (dec->context) {
+#ifdef WAVPACK_OLD_API
+ reason = dec->context->error_message;
+#else
+ reason = WavpackGetErrorMessage (dec->context);
+#endif
+ } else {
+ reason = "couldn't create decoder context";
+ }
+ GST_AUDIO_DECODER_ERROR (bdec, 1, STREAM, DECODE, (NULL),
+ ("decoding error: %s", reason), ret);
+ g_free (dec_data);
+ if (ret == GST_FLOW_OK)
+ gst_audio_decoder_finish_frame (bdec, NULL, 1);
+ goto out;
+ }
+}
+
+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;
+}
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-21 20:59:43 +0000