1 files changed, 512 insertions, 0 deletions

diff --git a/ext/wavpack/gstwavpackdec.c b/ext/wavpack/gstwavpackdec.c
new file mode 100644
index 0000000..4ba242d
--- /dev/null
+++ b/ext/wavpack/gstwavpackdec.c
@@ -0,0 +1,512 @@
+/* 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
+ *
+ * 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 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 <gst/audio/multichannel.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, 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-int, "
+        "width = (int) 32, "
+        "depth = (int) [ 1, 32 ], "
+        "channels = (int) [ 1, 8 ], "
+        "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 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)
+{
+  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+
+  gst_element_class_add_static_pad_template (element_class, &src_factory);
+  gst_element_class_add_static_pad_template (element_class, &sink_factory);
+  gst_element_class_set_details_simple (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>");
+}
+
+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_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->channel_mask = 0;
+  dec->sample_rate = 0;
+  dec->depth = 0;
+
+  gst_segment_init (&dec->segment, GST_FORMAT_TIME);
+  dec->next_block_index = 0;
+}
+
+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;
+    GstAudioChannelPosition *pos;
+
+    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);
+
+    /* If we already have the channel layout set from upstream
+     * take this */
+    if (gst_structure_has_field (structure, "channel-positions")) {
+      pos = gst_audio_get_channel_positions (structure);
+      if (pos != NULL && dec->channels > 2) {
+        GstStructure *new_str = gst_caps_get_structure (caps, 0);
+
+        gst_audio_set_channel_positions (new_str, pos);
+        dec->channel_mask =
+            gst_wavpack_get_channel_mask_from_positions (pos, dec->channels);
+      }
+
+      if (pos != NULL)
+        g_free (pos);
+    }
+
+    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 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 = NULL;
+  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;
+
+  if (!(wph.flags & INITIAL_BLOCK))
+    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)) ||
+#ifdef WAVPACK_OLD_API
+      (dec->channel_mask != dec->context->config.channel_mask);
+#else
+      (dec->channel_mask != WavpackGetChannelMask (dec->context));
+#endif
+
+  if (!GST_PAD_CAPS (dec->srcpad) || format_changed) {
+    GstCaps *caps;
+    gint channel_mask;
+
+    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);
+
+#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;
+
+    /* Only set the channel layout for more than two channels
+     * otherwise things break unfortunately */
+    if (channel_mask != 0 && dec->channels > 2)
+      if (!gst_wavpack_set_channel_layout (caps, channel_mask))
+        GST_WARNING_OBJECT (dec, "Failed to set channel layout");
+
+    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_copy_metadata (outbuf, buf, GST_BUFFER_COPY_TIMESTAMPS);
+
+  /* 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) || dec->next_block_index != wph.block_index)
+    GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
+
+  dec->next_block_index = wph.block_index + wph.block_samples;
+
+  /* decode */
+  decoded = WavpackUnpackSamples (dec->context,
+      (int32_t *) GST_BUFFER_DATA (outbuf), wph.block_samples);
+  if (decoded != wph.block_samples)
+    goto decode_error;
+
+  if ((outbuf = gst_audio_buffer_clip (outbuf, &dec->segment,
+              dec->sample_rate, 4 * dec->channels))) {
+    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);
+  }
+
+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:
+  {
+    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_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL),
+        ("Failed to decode wavpack stream: %s", reason));
+    if (outbuf)
+      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_TIME);
+      }
+      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;
+}