mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2025-01-23 15:48:23 +00:00
wavpackdec: port to 0.11
This commit is contained in:
parent
e0c7004667
commit
2f7d8f55a4
3 changed files with 109 additions and 114 deletions
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@ -316,7 +316,6 @@ GST_PLUGINS_NONPORTED="deinterlace interleave flx \
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videobox \
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cairo cairo_gobject dv1394 \
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oss oss4 \
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wavpack \
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osx_video osx_audio "
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AC_SUBST(GST_PLUGINS_NONPORTED)
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@ -44,7 +44,6 @@
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#include <gst/gst.h>
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#include <gst/audio/audio.h>
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#include <gst/audio/multichannel.h>
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#include <math.h>
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#include <string.h>
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@ -62,7 +61,7 @@ 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) [ 1, 32 ], "
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"depth = (int) [ 1, 32 ], "
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"channels = (int) [ 1, 8 ], "
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"rate = (int) [ 6000, 192000 ], " "framed = (boolean) true")
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);
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@ -70,21 +69,20 @@ static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
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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, depth = (int) 8, "
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GST_STATIC_CAPS ("audio/x-raw, "
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"format = (string) S8, "
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"layout = (string) interleaved, "
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"channels = (int) [ 1, 8 ], "
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"rate = (int) [ 6000, 192000 ], "
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"endianness = (int) BYTE_ORDER, " "signed = (boolean) true; "
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"audio/x-raw-int, "
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"width = (int) 16, depth = (int) 16, "
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"rate = (int) [ 6000, 192000 ]; "
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"audio/x-raw, "
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"format = (string) " GST_AUDIO_NE (S16) ", "
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"layout = (string) interleaved, "
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"channels = (int) [ 1, 8 ], "
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"rate = (int) [ 6000, 192000 ], "
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"endianness = (int) BYTE_ORDER, " "signed = (boolean) true; "
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"audio/x-raw-int, "
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"width = (int) 32, depth = (int) 32, "
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"channels = (int) [ 1, 8 ], "
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"rate = (int) [ 6000, 192000 ], "
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"endianness = (int) BYTE_ORDER, " "signed = (boolean) true; ")
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"rate = (int) [ 6000, 192000 ]; "
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"audio/x-raw, "
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"format = (string) " GST_AUDIO_NE (S32) ", "
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"layout = (string) interleaved, "
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"channels = (int) [ 1, 8 ], " "rate = (int) [ 6000, 192000 ]")
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);
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static gboolean gst_wavpack_dec_start (GstAudioDecoder * dec);
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@ -97,13 +95,15 @@ static GstFlowReturn gst_wavpack_dec_handle_frame (GstAudioDecoder * dec,
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static void gst_wavpack_dec_finalize (GObject * object);
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static void gst_wavpack_dec_post_tags (GstWavpackDec * dec);
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GST_BOILERPLATE (GstWavpackDec, gst_wavpack_dec, GstAudioDecoder,
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GST_TYPE_AUDIO_DECODER);
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#define gst_wavpack_dec_parent_class parent_class
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G_DEFINE_TYPE (GstWavpackDec, gst_wavpack_dec, GST_TYPE_AUDIO_DECODER);
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static void
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gst_wavpack_dec_base_init (gpointer klass)
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gst_wavpack_dec_class_init (GstWavpackDecClass * klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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GObjectClass *gobject_class = (GObjectClass *) klass;
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GstElementClass *element_class = (GstElementClass *) (klass);
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GstAudioDecoderClass *base_class = (GstAudioDecoderClass *) (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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@ -114,13 +114,6 @@ gst_wavpack_dec_base_init (gpointer klass)
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"Decodes 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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}
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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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GstAudioDecoderClass *base_class = (GstAudioDecoderClass *) (klass);
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gobject_class->finalize = gst_wavpack_dec_finalize;
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@ -143,7 +136,7 @@ gst_wavpack_dec_reset (GstWavpackDec * dec)
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}
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static void
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gst_wavpack_dec_init (GstWavpackDec * dec, GstWavpackDecClass * gklass)
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gst_wavpack_dec_init (GstWavpackDec * dec)
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{
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dec->context = NULL;
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dec->stream_reader = gst_wavpack_stream_reader_new ();
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@ -197,69 +190,51 @@ gst_wavpack_dec_stop (GstAudioDecoder * dec)
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static void
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gst_wavpack_dec_negotiate (GstWavpackDec * dec)
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{
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GstCaps *caps;
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GstAudioInfo info;
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GstAudioFormat fmt;
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GstAudioChannelPosition pos[64] = { GST_AUDIO_CHANNEL_POSITION_INVALID, };
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/* arrange for 1, 2 or 4-byte width == depth output */
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if (dec->depth == 24)
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dec->width = 32;
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else
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dec->width = dec->depth;
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dec->width = dec->depth;
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switch (dec->depth) {
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case 8:
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fmt = GST_AUDIO_FORMAT_S8;
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break;
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case 16:
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fmt = _GST_AUDIO_FORMAT_NE (S16);
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break;
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case 24:
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case 32:
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fmt = _GST_AUDIO_FORMAT_NE (S32);
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dec->width = 32;
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break;
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default:
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g_assert_not_reached ();
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break;
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}
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caps = gst_caps_new_simple ("audio/x-raw-int",
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"rate", G_TYPE_INT, dec->sample_rate,
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"channels", G_TYPE_INT, dec->channels,
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"depth", G_TYPE_INT, dec->width,
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"width", G_TYPE_INT, dec->width,
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"endianness", G_TYPE_INT, G_BYTE_ORDER,
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"signed", G_TYPE_BOOLEAN, TRUE, NULL);
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g_assert (dec->channel_mask != 0);
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/* Only set the channel layout for more than two channels
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* otherwise things break unfortunately */
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if (dec->channel_mask != 0 && dec->channels > 2)
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if (!gst_wavpack_set_channel_layout (caps, dec->channel_mask))
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GST_WARNING_OBJECT (dec, "Failed to set channel layout");
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if (!gst_wavpack_get_channel_positions (dec->channels,
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dec->channel_mask, pos))
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GST_WARNING_OBJECT (dec, "Failed to set channel layout");
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GST_DEBUG_OBJECT (dec, "setting caps %" GST_PTR_FORMAT, caps);
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gst_audio_info_init (&info);
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gst_audio_info_set_format (&info, fmt, dec->sample_rate, dec->channels, pos);
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gst_audio_channel_positions_to_valid_order (info.position, info.channels);
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gst_audio_get_channel_reorder_map (info.channels,
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info.position, pos, dec->channel_reorder_map);
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/* should always succeed */
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gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), caps);
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gst_caps_unref (caps);
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gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (dec), &info);
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}
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static gboolean
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gst_wavpack_dec_set_format (GstAudioDecoder * bdec, GstCaps * caps)
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{
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GstWavpackDec *dec = GST_WAVPACK_DEC (bdec);
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GstStructure *structure = gst_caps_get_structure (caps, 0);
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/* Check if we can set the caps here already */
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if (gst_structure_get_int (structure, "channels", &dec->channels) &&
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gst_structure_get_int (structure, "rate", &dec->sample_rate) &&
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gst_structure_get_int (structure, "width", &dec->depth)) {
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GstAudioChannelPosition *pos;
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/* If we already have the channel layout set from upstream
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* take this */
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if (gst_structure_has_field (structure, "channel-positions")) {
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pos = gst_audio_get_channel_positions (structure);
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if (pos != NULL && dec->channels > 2) {
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GstStructure *new_str = gst_caps_get_structure (caps, 0);
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gst_audio_set_channel_positions (new_str, pos);
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dec->channel_mask =
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gst_wavpack_get_channel_mask_from_positions (pos, dec->channels);
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}
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if (pos != NULL)
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g_free (pos);
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}
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gst_wavpack_dec_negotiate (dec);
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/* send GST_TAG_AUDIO_CODEC and GST_TAG_BITRATE tags before something
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* is decoded or after the format has changed */
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gst_wavpack_dec_post_tags (dec);
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}
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/* pretty much nothing to do here,
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* we'll parse it all from the stream and setup then */
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return TRUE;
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}
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@ -272,13 +247,13 @@ gst_wavpack_dec_post_tags (GstWavpackDec * dec)
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gint64 duration, size;
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/* try to estimate the average bitrate */
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if (gst_pad_query_peer_duration (GST_AUDIO_DECODER_SINK_PAD (dec),
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&format_bytes, &size) &&
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gst_pad_query_peer_duration (GST_AUDIO_DECODER_SINK_PAD (dec),
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&format_time, &duration) && size > 0 && duration > 0) {
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if (gst_pad_peer_query_duration (GST_AUDIO_DECODER_SINK_PAD (dec),
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format_bytes, &size) &&
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gst_pad_peer_query_duration (GST_AUDIO_DECODER_SINK_PAD (dec),
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format_time, &duration) && size > 0 && duration > 0) {
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guint64 bitrate;
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list = gst_tag_list_new ();
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list = gst_tag_list_new_empty ();
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bitrate = gst_util_uint64_scale (size, 8 * GST_SECOND, duration);
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gst_tag_list_add (list, GST_TAG_MERGE_REPLACE, GST_TAG_BITRATE,
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@ -298,29 +273,32 @@ gst_wavpack_dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buf)
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WavpackHeader wph;
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int32_t decoded, unpacked_size;
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gboolean format_changed;
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gint width, depth, i, max;
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gint width, depth, i, j, max;
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gint32 *dec_data = NULL;
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guint8 *out_data;
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GstMapInfo map, omap;
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dec = GST_WAVPACK_DEC (bdec);
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g_return_val_if_fail (buf != NULL, GST_FLOW_ERROR);
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gst_buffer_map (buf, &map, GST_MAP_READ);
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/* check input, we only accept framed input with complete chunks */
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if (GST_BUFFER_SIZE (buf) < sizeof (WavpackHeader))
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if (map.size < sizeof (WavpackHeader))
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goto input_not_framed;
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if (!gst_wavpack_read_header (&wph, GST_BUFFER_DATA (buf)))
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if (!gst_wavpack_read_header (&wph, map.data))
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goto invalid_header;
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if (GST_BUFFER_SIZE (buf) < wph.ckSize + 4 * 1 + 4)
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if (map.size < wph.ckSize + 4 * 1 + 4)
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goto input_not_framed;
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if (!(wph.flags & INITIAL_BLOCK))
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goto input_not_framed;
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dec->wv_id.buffer = GST_BUFFER_DATA (buf);
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dec->wv_id.length = GST_BUFFER_SIZE (buf);
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dec->wv_id.buffer = map.data;
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dec->wv_id.length = map.size;
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dec->wv_id.position = 0;
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/* create a new wavpack context if there is none yet but if there
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@ -351,7 +329,8 @@ gst_wavpack_dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buf)
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(dec->channel_mask != WavpackGetChannelMask (dec->context));
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#endif
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if (!GST_PAD_CAPS (GST_AUDIO_DECODER_SRC_PAD (dec)) || format_changed) {
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if (!gst_pad_has_current_caps (GST_AUDIO_DECODER_SRC_PAD (dec)) ||
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format_changed) {
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gint channel_mask;
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dec->sample_rate = WavpackGetSampleRate (dec->context);
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@ -376,58 +355,72 @@ gst_wavpack_dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buf)
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}
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/* alloc output buffer */
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unpacked_size = (dec->width / 8) * wph.block_samples * dec->channels;
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ret = gst_pad_alloc_buffer (GST_AUDIO_DECODER_SRC_PAD (dec),
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GST_BUFFER_OFFSET (buf), unpacked_size,
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GST_PAD_CAPS (GST_AUDIO_DECODER_SRC_PAD (dec)), &outbuf);
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if (ret != GST_FLOW_OK)
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goto out;
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dec_data = g_malloc (4 * wph.block_samples * dec->channels);
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out_data = GST_BUFFER_DATA (outbuf);
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/* decode */
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decoded = WavpackUnpackSamples (dec->context, dec_data, wph.block_samples);
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if (decoded != wph.block_samples)
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goto decode_error;
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unpacked_size = (dec->width / 8) * wph.block_samples * dec->channels;
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outbuf = gst_buffer_new_and_alloc (unpacked_size);
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/* legacy; pass along offset, whatever that might entail */
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GST_BUFFER_OFFSET (outbuf) = GST_BUFFER_OFFSET (buf);
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gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
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out_data = omap.data;
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width = dec->width;
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depth = dec->depth;
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max = dec->channels * wph.block_samples;
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if (width == 8) {
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gint8 *outbuffer = (gint8 *) out_data;
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gint *reorder_map = dec->channel_reorder_map;
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for (i = 0; i < max; i++) {
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*outbuffer++ = (gint8) (dec_data[i]);
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for (i = 0; i < max; i += dec->channels) {
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for (j = 0; j < dec->channels; j++)
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*outbuffer++ = (gint8) (dec_data[i + reorder_map[j]]);
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}
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} else if (width == 16) {
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gint16 *outbuffer = (gint16 *) out_data;
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gint *reorder_map = dec->channel_reorder_map;
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for (i = 0; i < max; i++) {
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*outbuffer++ = (gint16) (dec_data[i]);
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for (i = 0; i < max; i += dec->channels) {
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for (j = 0; j < dec->channels; j++)
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*outbuffer++ = (gint16) (dec_data[i + reorder_map[j]]);
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}
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} else if (dec->width == 32) {
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gint32 *outbuffer = (gint32 *) out_data;
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gint *reorder_map = dec->channel_reorder_map;
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if (width != depth) {
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for (i = 0; i < max; i++) {
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*outbuffer++ = (gint32) (dec_data[i] << (width - depth));
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for (i = 0; i < max; i += dec->channels) {
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for (j = 0; j < dec->channels; j++)
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*outbuffer++ =
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(gint32) (dec_data[i + reorder_map[j]] << (width - depth));
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}
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} else {
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for (i = 0; i < max; i++) {
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*outbuffer++ = (gint32) dec_data[i];
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for (i = 0; i < max; i += dec->channels) {
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for (j = 0; j < dec->channels; j++)
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*outbuffer++ = (gint32) (dec_data[i + reorder_map[j]]);
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}
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}
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} else {
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g_assert_not_reached ();
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}
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gst_buffer_unmap (outbuf, &omap);
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gst_buffer_unmap (buf, &map);
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buf = NULL;
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g_free (dec_data);
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ret = gst_audio_decoder_finish_frame (bdec, outbuf, 1);
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out:
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if (buf)
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gst_buffer_unmap (buf, &map);
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if (G_UNLIKELY (ret != GST_FLOW_OK)) {
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GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (ret));
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@ -439,12 +432,14 @@ out:
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input_not_framed:
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{
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GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("Expected framed input"));
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return GST_FLOW_ERROR;
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ret = GST_FLOW_ERROR;
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goto out;
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}
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invalid_header:
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{
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GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("Invalid wavpack header"));
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return GST_FLOW_ERROR;
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ret = GST_FLOW_ERROR;
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goto out;
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}
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context_failed:
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{
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@ -468,11 +463,9 @@ decode_error:
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GST_AUDIO_DECODER_ERROR (bdec, 1, STREAM, DECODE, (NULL),
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("decoding error: %s", reason), ret);
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g_free (dec_data);
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if (outbuf)
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gst_buffer_unref (outbuf);
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if (ret == GST_FLOW_OK)
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gst_audio_decoder_finish_frame (bdec, NULL, 1);
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return ret;
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goto out;
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}
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}
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@ -482,7 +475,7 @@ gst_wavpack_dec_plugin_init (GstPlugin * plugin)
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if (!gst_element_register (plugin, "wavpackdec",
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GST_RANK_PRIMARY, GST_TYPE_WAVPACK_DEC))
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return FALSE;
|
||||
GST_DEBUG_CATEGORY_INIT (gst_wavpack_dec_debug, "wavpack_dec", 0,
|
||||
GST_DEBUG_CATEGORY_INIT (gst_wavpack_dec_debug, "wavpackdec", 0,
|
||||
"Wavpack decoder");
|
||||
return TRUE;
|
||||
}
|
||||
|
|
|
@ -60,6 +60,9 @@ struct _GstWavpackDec
|
|||
gint width;
|
||||
gint channels;
|
||||
gint channel_mask;
|
||||
|
||||
gint channel_reorder_map[64];
|
||||
|
||||
};
|
||||
|
||||
struct _GstWavpackDecClass
|
||||
|
|
Loading…
Reference in a new issue