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wavpackenc: port to 0.11

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This commit is contained in:
Mark Nauwelaerts 2012-03-05 12:17:39 +01:00
parent 8721817f6d
commit e0c7004667
3 changed files with 66 additions and 78 deletions
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40
ext/wavpack/gstwavpackcommon.c
View file

@ -29,7 +29,6 @@
#include <string.h> #include <string.h>
#include <gst/gst.h> #include <gst/gst.h>
#include <gst/audio/multichannel.h>
GST_DEBUG_CATEGORY_EXTERN (wavpack_debug); GST_DEBUG_CATEGORY_EXTERN (wavpack_debug);
#define GST_CAT_DEFAULT wavpack_debug #define GST_CAT_DEFAULT wavpack_debug
@ -147,7 +146,7 @@ static const struct
0x00001, GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT}, { 0x00001, GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT}, {
0x00002, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT}, { 0x00002, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT}, {
0x00004, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER}, { 0x00004, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER}, {
0x00008, GST_AUDIO_CHANNEL_POSITION_LFE}, { 0x00008, GST_AUDIO_CHANNEL_POSITION_LFE1}, {
0x00010, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT}, { 0x00010, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT}, {
0x00020, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT}, { 0x00020, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT}, {
0x00040, GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER}, { 0x00040, GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER}, {
@ -155,37 +154,25 @@ static const struct
0x00100, GST_AUDIO_CHANNEL_POSITION_REAR_CENTER}, { 0x00100, GST_AUDIO_CHANNEL_POSITION_REAR_CENTER}, {
0x00200, GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT}, { 0x00200, GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT}, {
0x00400, GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT}, { 0x00400, GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT}, {
0x00800, GST_AUDIO_CHANNEL_POSITION_INVALID}, /* TOP_CENTER */ 0x00800, GST_AUDIO_CHANNEL_POSITION_TOP_CENTER}, {
{ 0x01000, GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_LEFT}, {
0x01000, GST_AUDIO_CHANNEL_POSITION_INVALID}, /* TOP_FRONT_LEFT */ 0x02000, GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_CENTER}, {
{ 0x04000, GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_RIGHT}, {
0x02000, GST_AUDIO_CHANNEL_POSITION_INVALID}, /* TOP_FRONT_CENTER */ 0x08000, GST_AUDIO_CHANNEL_POSITION_TOP_REAR_LEFT}, {
{ 0x10000, GST_AUDIO_CHANNEL_POSITION_TOP_REAR_CENTER}, {
0x04000, GST_AUDIO_CHANNEL_POSITION_INVALID}, /* TOP_FRONT_RIGHT */ 0x20000, GST_AUDIO_CHANNEL_POSITION_TOP_REAR_RIGHT}
{
0x08000, GST_AUDIO_CHANNEL_POSITION_INVALID}, /* TOP_BACK_LEFT */
{
0x10000, GST_AUDIO_CHANNEL_POSITION_INVALID}, /* TOP_BACK_CENTER */
{
0x20000, GST_AUDIO_CHANNEL_POSITION_INVALID} /* TOP_BACK_RIGHT */
}; };
#define MAX_CHANNEL_POSITIONS G_N_ELEMENTS (layout_mapping) #define MAX_CHANNEL_POSITIONS G_N_ELEMENTS (layout_mapping)
gboolean gboolean
gst_wavpack_set_channel_layout (GstCaps * caps, gint layout) gst_wavpack_get_channel_positions (gint num_channels, gint layout,
GstAudioChannelPosition * pos)
{ {
GstAudioChannelPosition pos[MAX_CHANNEL_POSITIONS]; gint i, p;
GstStructure *s;
gint num_channels, i, p;
s = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_int (s, "channels", &num_channels))
g_return_val_if_reached (FALSE);
if (num_channels == 1 && layout == 0x00004) { if (num_channels == 1 && layout == 0x00004) {
pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_MONO; pos[0] = GST_AUDIO_CHANNEL_POSITION_MONO;
gst_audio_set_channel_positions (s, pos);
return TRUE; return TRUE;
} }
@ -214,7 +201,6 @@ gst_wavpack_set_channel_layout (GstCaps * caps, gint layout)
return FALSE; return FALSE;
} }
gst_audio_set_channel_positions (s, pos);
return TRUE; return TRUE;
} }
@ -242,7 +228,7 @@ gst_wavpack_get_channel_mask_from_positions (GstAudioChannelPosition * pos,
gint channel_mask = 0; gint channel_mask = 0;
gint i, j; gint i, j;
if (nchannels == 1 && pos[0] == GST_AUDIO_CHANNEL_POSITION_FRONT_MONO) { if (nchannels == 1 && pos[0] == GST_AUDIO_CHANNEL_POSITION_MONO) {
channel_mask = 0x00000004; channel_mask = 0x00000004;
return channel_mask; return channel_mask;
} }

4
ext/wavpack/gstwavpackcommon.h
View file

@ -24,7 +24,7 @@
#define __GST_WAVPACK_COMMON_H__ #define __GST_WAVPACK_COMMON_H__
#include <gst/gst.h> #include <gst/gst.h>
#include <gst/audio/multichannel.h> #include <gst/audio/audio.h>
#include <wavpack/wavpack.h> #include <wavpack/wavpack.h>
typedef struct typedef struct
@ -67,7 +67,7 @@ gboolean gst_wavpack_read_header (WavpackHeader * header, guint8 * buf);
gboolean gst_wavpack_read_metadata (GstWavpackMetadata * meta, gboolean gst_wavpack_read_metadata (GstWavpackMetadata * meta,
guint8 * header_data, guint8 ** p_data); guint8 * header_data, guint8 ** p_data);
gint gst_wavpack_get_default_channel_mask (gint nchannels); gint gst_wavpack_get_default_channel_mask (gint nchannels);
gboolean gst_wavpack_set_channel_layout (GstCaps * caps, gint layout); gboolean gst_wavpack_get_channel_positions (gint nchannels, gint layout, GstAudioChannelPosition *pos);
GstAudioChannelPosition *gst_wavpack_get_default_channel_positions (gint nchannels); GstAudioChannelPosition *gst_wavpack_get_default_channel_positions (gint nchannels);
gint gst_wavpack_get_channel_mask_from_positions (GstAudioChannelPosition *pos, gint nchannels); gint gst_wavpack_get_channel_mask_from_positions (GstAudioChannelPosition *pos, gint nchannels);
gboolean gst_wavpack_set_channel_mapping (GstAudioChannelPosition *pos, gint nchannels, gint8 *channel_mapping); gboolean gst_wavpack_set_channel_mapping (GstAudioChannelPosition *pos, gint nchannels, gint8 *channel_mapping);

100
ext/wavpack/gstwavpackenc.c
View file

@ -31,7 +31,7 @@
* |[ * |[
* gst-launch audiotestsrc num-buffers=500 ! audioconvert ! wavpackenc ! filesink location=sinewave.wv * gst-launch audiotestsrc num-buffers=500 ! audioconvert ! wavpackenc ! filesink location=sinewave.wv
* ]| This pipeline encodes audio from audiotestsrc into a Wavpack file. The audioconvert element is needed * ]| This pipeline encodes audio from audiotestsrc into a Wavpack file. The audioconvert element is needed
* as the Wavpack encoder only accepts input with 32 bit width (and every depth between 1 and 32 bits). * as the Wavpack encoder only accepts input with 32 bit width.
* |[ * |[
* gst-launch cdda://1 ! audioconvert ! wavpackenc ! filesink location=track1.wv * gst-launch cdda://1 ! audioconvert ! wavpackenc ! filesink location=track1.wv
* ]| This pipeline encodes audio from an audio CD into a Wavpack file using * ]| This pipeline encodes audio from an audio CD into a Wavpack file using
@ -90,19 +90,17 @@ GST_DEBUG_CATEGORY_STATIC (gst_wavpack_enc_debug);
static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink", static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK, GST_PAD_SINK,
GST_PAD_ALWAYS, GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, " GST_STATIC_CAPS ("audio/x-raw, "
"width = (int) 32, " "format = (string) " GST_AUDIO_NE (S32) ", "
"depth = (int) { 24, 32 }, " "layout = (string) interleaved, "
"endianness = (int) BYTE_ORDER, " "channels = (int) [ 1, 8 ], " "rate = (int) [ 6000, 192000 ]")
"channels = (int) [ 1, 8 ], "
"rate = (int) [ 6000, 192000 ]," "signed = (boolean) TRUE")
); );
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src", static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC, GST_PAD_SRC,
GST_PAD_ALWAYS, GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-wavpack, " GST_STATIC_CAPS ("audio/x-wavpack, "
"width = (int) [ 1, 32 ], " "depth = (int) [ 1, 32 ], "
"channels = (int) [ 1, 8 ], " "channels = (int) [ 1, 8 ], "
"rate = (int) [ 6000, 192000 ], " "framed = (boolean) TRUE") "rate = (int) [ 6000, 192000 ], " "framed = (boolean) TRUE")
); );
@ -202,13 +200,15 @@ gst_wavpack_enc_joint_stereo_mode_get_type (void)
return qtype; return qtype;
} }
GST_BOILERPLATE (GstWavpackEnc, gst_wavpack_enc, GstAudioEncoder, #define gst_wavpack_enc_parent_class parent_class
GST_TYPE_AUDIO_ENCODER); G_DEFINE_TYPE (GstWavpackEnc, gst_wavpack_enc, GST_TYPE_AUDIO_ENCODER);
static void static void
gst_wavpack_enc_base_init (gpointer klass) gst_wavpack_enc_class_init (GstWavpackEncClass * klass)
{ {
GstElementClass *element_class = GST_ELEMENT_CLASS (klass); GObjectClass *gobject_class = (GObjectClass *) klass;
GstElementClass *element_class = (GstElementClass *) (klass);
GstAudioEncoderClass *base_class = (GstAudioEncoderClass *) (klass);
/* add pad templates */ /* add pad templates */
gst_element_class_add_pad_template (element_class, gst_element_class_add_pad_template (element_class,
@ -223,15 +223,6 @@ gst_wavpack_enc_base_init (gpointer klass)
"Codec/Encoder/Audio", "Codec/Encoder/Audio",
"Encodes audio with the Wavpack lossless/lossy audio codec", "Encodes audio with the Wavpack lossless/lossy audio codec",
"Sebastian Dröge <slomo@circular-chaos.org>"); "Sebastian Dröge <slomo@circular-chaos.org>");
}
static void
gst_wavpack_enc_class_init (GstWavpackEncClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
GstAudioEncoderClass *base_class = (GstAudioEncoderClass *) (klass);
parent_class = g_type_class_peek_parent (klass);
/* set property handlers */ /* set property handlers */
gobject_class->set_property = gst_wavpack_enc_set_property; gobject_class->set_property = gst_wavpack_enc_set_property;
@ -327,7 +318,7 @@ gst_wavpack_enc_reset (GstWavpackEnc * enc)
} }
static void static void
gst_wavpack_enc_init (GstWavpackEnc * enc, GstWavpackEncClass * gclass) gst_wavpack_enc_init (GstWavpackEnc * enc)
{ {
GstAudioEncoder *benc = GST_AUDIO_ENCODER (enc); GstAudioEncoder *benc = GST_AUDIO_ENCODER (enc);
@ -383,7 +374,9 @@ gst_wavpack_enc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
{ {
GstWavpackEnc *enc = GST_WAVPACK_ENC (benc); GstWavpackEnc *enc = GST_WAVPACK_ENC (benc);
GstAudioChannelPosition *pos; GstAudioChannelPosition *pos;
GstAudioChannelPosition opos[64] = { GST_AUDIO_CHANNEL_POSITION_INVALID, };
GstCaps *caps; GstCaps *caps;
guint64 mask = 0;
/* we may be configured again, but that change should have cleanup context */ /* we may be configured again, but that change should have cleanup context */
g_assert (enc->wp_context == NULL); g_assert (enc->wp_context == NULL);
@ -406,14 +399,17 @@ gst_wavpack_enc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
gst_wavpack_set_channel_mapping (pos, enc->channels, gst_wavpack_set_channel_mapping (pos, enc->channels,
enc->channel_mapping); enc->channel_mapping);
/* wavpack caps hold gst mask, not wavpack mask */
gst_audio_channel_positions_to_mask (opos, enc->channels, &mask);
/* set fixed src pad caps now that we know what we will get */ /* set fixed src pad caps now that we know what we will get */
caps = gst_caps_new_simple ("audio/x-wavpack", caps = gst_caps_new_simple ("audio/x-wavpack",
"channels", G_TYPE_INT, enc->channels, "channels", G_TYPE_INT, enc->channels,
"rate", G_TYPE_INT, enc->samplerate, "rate", G_TYPE_INT, enc->samplerate,
"width", G_TYPE_INT, enc->depth, "framed", G_TYPE_BOOLEAN, TRUE, NULL); "depth", G_TYPE_INT, enc->depth, "framed", G_TYPE_BOOLEAN, TRUE, NULL);
if (!gst_wavpack_set_channel_layout (caps, enc->channel_mask)) if (mask)
GST_WARNING_OBJECT (enc, "setting channel layout failed"); gst_caps_set_simple (caps, "channel-mask", GST_TYPE_BITMASK, mask, NULL);
if (!gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (enc), caps)) if (!gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (enc), caps))
goto setting_src_caps_failed; goto setting_src_caps_failed;
@ -562,11 +558,11 @@ gst_wavpack_enc_push_block (void *id, void *data, int32_t count)
pad = (wid->correction) ? enc->wvcsrcpad : GST_AUDIO_ENCODER_SRC_PAD (enc); pad = (wid->correction) ? enc->wvcsrcpad : GST_AUDIO_ENCODER_SRC_PAD (enc);
flow = flow =
(wid->correction) ? &enc->wvcsrcpad_last_return : &enc-> (wid->correction) ? &enc->
srcpad_last_return; wvcsrcpad_last_return : &enc->srcpad_last_return;
buffer = gst_buffer_new_and_alloc (count); buffer = gst_buffer_new_and_alloc (count);
g_memmove (GST_BUFFER_DATA (buffer), block, count); gst_buffer_fill (buffer, 0, data, count);
if (count > sizeof (WavpackHeader) && memcmp (block, "wvpk", 4) == 0) { if (count > sizeof (WavpackHeader) && memcmp (block, "wvpk", 4) == 0) {
/* if it's a Wavpack block set buffer timestamp and duration, etc */ /* if it's a Wavpack block set buffer timestamp and duration, etc */
@ -612,9 +608,12 @@ gst_wavpack_enc_push_block (void *id, void *data, int32_t count)
/* save header for later reference, so we can re-send it later on /* save header for later reference, so we can re-send it later on
* EOS with fixed up values for total sample count etc. */ * EOS with fixed up values for total sample count etc. */
if (enc->first_block == NULL && !wid->correction) { if (enc->first_block == NULL && !wid->correction) {
enc->first_block = GstMapInfo map;
g_memdup (GST_BUFFER_DATA (buffer), GST_BUFFER_SIZE (buffer));
enc->first_block_size = GST_BUFFER_SIZE (buffer); gst_buffer_map (buffer, &map, GST_MAP_READ);
enc->first_block = g_memdup (map.data, map.size);
enc->first_block_size = map.size;
gst_buffer_unmap (buffer, &map);
} }
} }
} }
@ -634,11 +633,11 @@ gst_wavpack_enc_push_block (void *id, void *data, int32_t count)
if (wid->correction || wid->passthrough) { if (wid->correction || wid->passthrough) {
/* push the buffer and forward errors */ /* push the buffer and forward errors */
GST_DEBUG_OBJECT (enc, "pushing buffer with %d bytes", GST_DEBUG_OBJECT (enc, "pushing buffer with %d bytes",
GST_BUFFER_SIZE (buffer)); gst_buffer_get_size (buffer));
*flow = gst_pad_push (pad, buffer); *flow = gst_pad_push (pad, buffer);
} else { } else {
GST_DEBUG_OBJECT (enc, "handing frame of %d bytes", GST_DEBUG_OBJECT (enc, "handing frame of %d bytes",
GST_BUFFER_SIZE (buffer)); gst_buffer_get_size (buffer));
*flow = gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (enc), buffer, *flow = gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (enc), buffer,
samples); samples);
} }
@ -676,6 +675,7 @@ gst_wavpack_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
GstWavpackEnc *enc = GST_WAVPACK_ENC (benc); GstWavpackEnc *enc = GST_WAVPACK_ENC (benc);
uint32_t sample_count; uint32_t sample_count;
GstFlowReturn ret; GstFlowReturn ret;
GstMapInfo map;
/* base class ensures configuration */ /* base class ensures configuration */
g_return_val_if_fail (enc->depth != 0, GST_FLOW_NOT_NEGOTIATED); g_return_val_if_fail (enc->depth != 0, GST_FLOW_NOT_NEGOTIATED);
@ -688,7 +688,7 @@ gst_wavpack_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
if (G_UNLIKELY (!buf)) if (G_UNLIKELY (!buf))
return gst_wavpack_enc_drain (enc); return gst_wavpack_enc_drain (enc);
sample_count = GST_BUFFER_SIZE (buf) / 4; sample_count = gst_buffer_get_size (buf) / 4;
GST_DEBUG_OBJECT (enc, "got %u raw samples", sample_count); GST_DEBUG_OBJECT (enc, "got %u raw samples", sample_count);
/* check if we already have a valid WavpackContext, otherwise make one */ /* check if we already have a valid WavpackContext, otherwise make one */
@ -716,27 +716,28 @@ gst_wavpack_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
if (enc->need_channel_remap) { if (enc->need_channel_remap) {
buf = gst_buffer_make_writable (buf); buf = gst_buffer_make_writable (buf);
gst_wavpack_enc_fix_channel_order (enc, (gint32 *) GST_BUFFER_DATA (buf), gst_buffer_map (buf, &map, GST_MAP_WRITE);
sample_count); gst_wavpack_enc_fix_channel_order (enc, (gint32 *) map.data, sample_count);
gst_buffer_unmap (buf, &map);
} }
gst_buffer_map (buf, &map, GST_MAP_READ);
/* if we want to append the MD5 sum to the stream update it here /* if we want to append the MD5 sum to the stream update it here
* with the current raw samples */ * with the current raw samples */
if (enc->md5) { if (enc->md5) {
g_checksum_update (enc->md5_context, GST_BUFFER_DATA (buf), g_checksum_update (enc->md5_context, map.data, map.size);
GST_BUFFER_SIZE (buf));
} }
/* encode and handle return values from encoding */ /* encode and handle return values from encoding */
if (WavpackPackSamples (enc->wp_context, (int32_t *) GST_BUFFER_DATA (buf), if (WavpackPackSamples (enc->wp_context, (int32_t *) map.data,
sample_count / enc->channels)) { sample_count / enc->channels)) {
GST_DEBUG_OBJECT (enc, "encoding samples successful"); GST_DEBUG_OBJECT (enc, "encoding samples successful");
gst_buffer_unmap (buf, &map);
ret = GST_FLOW_OK; ret = GST_FLOW_OK;
} else { } else {
if ((enc->srcpad_last_return == GST_FLOW_RESEND) || gst_buffer_unmap (buf, &map);
(enc->wvcsrcpad_last_return == GST_FLOW_RESEND)) { if ((enc->srcpad_last_return == GST_FLOW_OK) ||
ret = GST_FLOW_RESEND;
} else if ((enc->srcpad_last_return == GST_FLOW_OK) ||
(enc->wvcsrcpad_last_return == GST_FLOW_OK)) { (enc->wvcsrcpad_last_return == GST_FLOW_OK)) {
ret = GST_FLOW_OK; ret = GST_FLOW_OK;
} else if ((enc->srcpad_last_return == GST_FLOW_NOT_LINKED) && } else if ((enc->srcpad_last_return == GST_FLOW_NOT_LINKED) &&
@ -780,8 +781,7 @@ context_failed:
static void static void
gst_wavpack_enc_rewrite_first_block (GstWavpackEnc * enc) gst_wavpack_enc_rewrite_first_block (GstWavpackEnc * enc)
{ {
GstEvent *event = gst_event_new_new_segment (TRUE, 1.0, GST_FORMAT_BYTES, GstSegment segment;
0, GST_BUFFER_OFFSET_NONE, 0);
gboolean ret; gboolean ret;
g_return_if_fail (enc); g_return_if_fail (enc);
@ -791,7 +791,9 @@ gst_wavpack_enc_rewrite_first_block (GstWavpackEnc * enc)
WavpackUpdateNumSamples (enc->wp_context, enc->first_block); WavpackUpdateNumSamples (enc->wp_context, enc->first_block);
/* try to seek to the beginning of the output */ /* try to seek to the beginning of the output */
ret = gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (enc), event); gst_segment_init (&segment, GST_FORMAT_BYTES);
ret = gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (enc),
gst_event_new_segment (&segment));
if (ret) { if (ret) {
/* try to rewrite the first block */ /* try to rewrite the first block */
GST_DEBUG_OBJECT (enc, "rewriting first block ..."); GST_DEBUG_OBJECT (enc, "rewriting first block ...");
@ -861,7 +863,7 @@ gst_wavpack_enc_sink_event (GstAudioEncoder * benc, GstEvent * event)
GST_EVENT_TYPE_NAME (event)); GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) { switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_NEWSEGMENT: case GST_EVENT_SEGMENT:
if (enc->wp_context) { if (enc->wp_context) {
GST_WARNING_OBJECT (enc, "got NEWSEGMENT after encoding " GST_WARNING_OBJECT (enc, "got NEWSEGMENT after encoding "
"already started"); "already started");
@ -876,7 +878,7 @@ gst_wavpack_enc_sink_event (GstAudioEncoder * benc, GstEvent * event)
} }
/* baseclass handles rest */ /* baseclass handles rest */
return FALSE; return GST_AUDIO_ENCODER_CLASS (parent_class)->event (benc, event);
} }
static void static void
@ -980,7 +982,7 @@ gst_wavpack_enc_plugin_init (GstPlugin * plugin)
GST_RANK_NONE, GST_TYPE_WAVPACK_ENC)) GST_RANK_NONE, GST_TYPE_WAVPACK_ENC))
return FALSE; return FALSE;
GST_DEBUG_CATEGORY_INIT (gst_wavpack_enc_debug, "wavpack_enc", 0, GST_DEBUG_CATEGORY_INIT (gst_wavpack_enc_debug, "wavpackenc", 0,
"Wavpack encoder"); "Wavpack encoder");
return TRUE; return TRUE;