* 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 "gstwavpackstreamreader.h"
-#define WAVPACK_DEC_MAX_ERRORS 16
-
GST_DEBUG_CATEGORY_STATIC (gst_wavpack_dec_debug);
#define GST_CAT_DEFAULT gst_wavpack_dec_debug
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-wavpack, "
- "width = (int) { 8, 16, 24, 32 }, "
- "channels = (int) [ 1, 2 ], "
+ "width = (int) [ 1, 32 ], "
+ "channels = (int) [ 1, 8 ], "
"rate = (int) [ 6000, 192000 ], " "framed = (boolean) true")
);
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
- "width = (int) { 8, 16, 24, 32 }, "
- "depth = (int) { 8, 16, 24, 32 }, "
- "channels = (int) [ 1, 2 ], "
+ "width = (int) 8, depth = (int) 8, "
+ "channels = (int) [ 1, 8 ], "
+ "rate = (int) [ 6000, 192000 ], "
+ "endianness = (int) BYTE_ORDER, " "signed = (boolean) true; "
+ "audio/x-raw-int, "
+ "width = (int) 16, depth = (int) 16, "
+ "channels = (int) [ 1, 8 ], "
"rate = (int) [ 6000, 192000 ], "
- "endianness = (int) LITTLE_ENDIAN, " "signed = (boolean) true")
+ "endianness = (int) BYTE_ORDER, " "signed = (boolean) true; "
+ "audio/x-raw-int, "
+ "width = (int) 32, depth = (int) 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_event (GstPad * pad, GstEvent * event);
+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 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);
+GST_BOILERPLATE (GstWavpackDec, gst_wavpack_dec, GstAudioDecoder,
+ GST_TYPE_AUDIO_DECODER);
static void
gst_wavpack_dec_base_init (gpointer klass)
{
- static const GstElementDetails plugin_details =
- GST_ELEMENT_DETAILS ("WavePack audio decoder",
- "Codec/Decoder/Audio",
- "Decode 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);
+ 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;
+ GstAudioDecoderClass *base_class = (GstAudioDecoderClass *) (klass);
- gstelement_class->change_state =
- GST_DEBUG_FUNCPTR (gst_wavpack_dec_change_state);
- gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_wavpack_dec_finalize);
+ 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_init (GstWavpackDec * dec, GstWavpackDecClass * gklass)
+gst_wavpack_dec_reset (GstWavpackDec * dec)
{
- 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_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 ();
-
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->width = 0;
+ dec->depth = 0;
+}
- gst_segment_init (&dec->segment, GST_FORMAT_UNDEFINED);
+static void
+gst_wavpack_dec_init (GstWavpackDec * dec, GstWavpackDecClass * gklass)
+{
+ dec->context = NULL;
+ dec->stream_reader = gst_wavpack_stream_reader_new ();
+
+ gst_wavpack_dec_reset (dec);
}
static void
G_OBJECT_CLASS (parent_class)->finalize (object);
}
-static void
-gst_wavpack_dec_format_samples (GstWavpackDec * dec, guint8 * dst,
- int32_t * samples, guint num_samples)
+static gboolean
+gst_wavpack_dec_start (GstAudioDecoder * dec)
{
- gint i;
- int32_t temp;
-
- switch (dec->width) {
- case 8:
- for (i = 0; i < num_samples * dec->channels; ++i)
- *dst++ = (guint8) (*samples++);
- break;
- case 16:
- for (i = 0; i < num_samples * dec->channels; ++i) {
- *dst++ = (guint8) (temp = *samples++);
- *dst++ = (guint8) (temp >> 8);
- }
- break;
- case 24:
- for (i = 0; i < num_samples * dec->channels; ++i) {
- *dst++ = (guint8) (temp = *samples++);
- *dst++ = (guint8) (temp >> 8);
- *dst++ = (guint8) (temp >> 16);
- }
- break;
- case 32:
- for (i = 0; i < num_samples * dec->channels; ++i) {
- *dst++ = (guint8) (temp = *samples++);
- *dst++ = (guint8) (temp >> 8);
- *dst++ = (guint8) (temp >> 16);
- *dst++ = (guint8) (temp >> 24);
- }
- break;
- default:
- break;
- }
+ 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);
+
+ return TRUE;
}
static gboolean
-gst_wavpack_dec_clip_outgoing_buffer (GstWavpackDec * dec, GstBuffer * buf)
+gst_wavpack_dec_stop (GstAudioDecoder * dec)
{
- gint64 start, stop, cstart, cstop, diff;
+ GstWavpackDec *wpdec = GST_WAVPACK_DEC (dec);
- if (dec->segment.format != GST_FORMAT_TIME)
- return TRUE;
+ GST_DEBUG_OBJECT (dec, "stop");
- start = GST_BUFFER_TIMESTAMP (buf);
- stop = start + GST_BUFFER_DURATION (buf);
+ if (wpdec->context) {
+ WavpackCloseFile (wpdec->context);
+ wpdec->context = NULL;
+ }
- if (gst_segment_clip (&dec->segment, GST_FORMAT_TIME,
- start, stop, &cstart, &cstop)) {
+ gst_wavpack_dec_reset (wpdec);
- diff = cstart - start;
- if (diff > 0) {
- GST_BUFFER_TIMESTAMP (buf) = cstart;
- GST_BUFFER_DURATION (buf) -= diff;
+ return TRUE;
+}
- diff = ((dec->width + 7) >> 3) * dec->channels
- * GST_CLOCK_TIME_TO_FRAMES (diff, dec->sample_rate);
- GST_BUFFER_DATA (buf) += diff;
- GST_BUFFER_SIZE (buf) -= diff;
- }
+static void
+gst_wavpack_dec_negotiate (GstWavpackDec * dec)
+{
+ GstCaps *caps;
+
+ /* arrange for 1, 2 or 4-byte width == depth output */
+ if (dec->depth == 24)
+ dec->width = 32;
+ else
+ dec->width = dec->depth;
+
+ 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->width,
+ "width", G_TYPE_INT, dec->width,
+ "endianness", G_TYPE_INT, G_BYTE_ORDER,
+ "signed", G_TYPE_BOOLEAN, TRUE, NULL);
+
+ /* Only set the channel layout for more than two channels
+ * otherwise things break unfortunately */
+ if (dec->channel_mask != 0 && dec->channels > 2)
+ if (!gst_wavpack_set_channel_layout (caps, dec->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 (GST_AUDIO_DECODER_SRC_PAD (dec), caps);
+ gst_caps_unref (caps);
+}
- diff = cstop - stop;
- if (diff > 0) {
- GST_BUFFER_DURATION (buf) -= diff;
+static gboolean
+gst_wavpack_dec_set_format (GstAudioDecoder * bdec, GstCaps * caps)
+{
+ GstWavpackDec *dec = GST_WAVPACK_DEC (bdec);
+ 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)) {
+ GstAudioChannelPosition *pos;
+
+ /* 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);
+ }
- diff = ((dec->width + 7) >> 3) * dec->channels
- * GST_CLOCK_TIME_TO_FRAMES (diff, dec->sample_rate);
- GST_BUFFER_SIZE (buf) -= diff;
+ if (pos != NULL)
+ g_free (pos);
}
- } else {
- GST_DEBUG_OBJECT (dec, "buffer is outside configured segment");
- return FALSE;
+
+ 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);
}
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_query_peer_duration (GST_AUDIO_DECODER_SINK_PAD (dec),
+ &format_bytes, &size) &&
+ gst_pad_query_peer_duration (GST_AUDIO_DECODER_SINK_PAD (dec),
+ &format_time, &duration) && size > 0 && duration > 0) {
+ guint64 bitrate;
+
+ list = gst_tag_list_new ();
+
+ 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)
+gst_wavpack_dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buf)
{
GstWavpackDec *dec;
- GstBuffer *outbuf;
+ GstBuffer *outbuf = NULL;
GstFlowReturn ret = GST_FLOW_OK;
WavpackHeader wph;
- int32_t *unpack_buf = NULL;
int32_t decoded, unpacked_size;
gboolean format_changed;
+ gint width, depth, i, max;
+ gint32 *dec_data = NULL;
+ guint8 *out_data;
+
+ dec = GST_WAVPACK_DEC (bdec);
- dec = GST_WAVPACK_DEC (GST_PAD_PARENT (pad));
+ g_return_val_if_fail (buf != NULL, GST_FLOW_ERROR);
/* check input, we only accept framed input with complete chunks */
if (GST_BUFFER_SIZE (buf) < sizeof (WavpackHeader))
if (!gst_wavpack_read_header (&wph, GST_BUFFER_DATA (buf)))
goto invalid_header;
- if (GST_BUFFER_SIZE (buf) != wph.ckSize + 4 * 1 + 4)
+ 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->context = WavpackOpenFileInputEx (dec->stream_reader,
&dec->wv_id, NULL, error_msg, OPEN_STREAMING, 0);
+ /* expect this to work */
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;
- }
+ GST_WARNING_OBJECT (dec, "Couldn't decode buffer: %s", error_msg);
+ goto context_failed;
}
}
g_assert (dec->context != NULL);
- dec->error_count = 0;
-
format_changed =
(dec->sample_rate != WavpackGetSampleRate (dec->context)) ||
(dec->channels != WavpackGetNumChannels (dec->context)) ||
- (dec->width != WavpackGetBitsPerSample (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_CAPS (dec->srcpad) || format_changed) {
- GstCaps *caps;
+ if (!GST_PAD_CAPS (GST_AUDIO_DECODER_SRC_PAD (dec)) || format_changed) {
+ gint channel_mask;
dec->sample_rate = WavpackGetSampleRate (dec->context);
dec->channels = WavpackGetNumChannels (dec->context);
- dec->width = WavpackGetBitsPerSample (dec->context);
+ dec->depth = WavpackGetBytesPerSample (dec->context) * 8;
- 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->width,
- "width", G_TYPE_INT, dec->width,
- "endianness", G_TYPE_INT, G_LITTLE_ENDIAN,
- "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);
- GST_DEBUG_OBJECT (dec, "setting caps %" GST_PTR_FORMAT, caps);
+ dec->channel_mask = channel_mask;
- /* should always succeed */
- gst_pad_set_caps (dec->srcpad, caps);
- gst_caps_unref (caps);
- }
+ gst_wavpack_dec_negotiate (dec);
- unpacked_size = wph.block_samples * ((dec->width + 7) >> 3) * dec->channels;
+ /* 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 buffer */
- ret = gst_pad_alloc_buffer (dec->srcpad, GST_BUFFER_OFFSET (buf),
- unpacked_size, GST_PAD_CAPS (dec->srcpad), &outbuf);
+ /* alloc output buffer */
+ unpacked_size = (dec->width / 8) * wph.block_samples * dec->channels;
+ ret = gst_pad_alloc_buffer (GST_AUDIO_DECODER_SRC_PAD (dec),
+ GST_BUFFER_OFFSET (buf), unpacked_size,
+ GST_PAD_CAPS (GST_AUDIO_DECODER_SRC_PAD (dec)), &outbuf);
if (ret != GST_FLOW_OK)
goto out;
+ dec_data = g_malloc (4 * wph.block_samples * dec->channels);
+ out_data = GST_BUFFER_DATA (outbuf);
+
/* decode */
- unpack_buf = g_new (int32_t, wph.block_samples * dec->channels);
- decoded = WavpackUnpackSamples (dec->context, unpack_buf, wph.block_samples);
+ decoded = WavpackUnpackSamples (dec->context, dec_data, wph.block_samples);
if (decoded != wph.block_samples)
goto decode_error;
- /* put samples into outbuf buffer */
- gst_wavpack_dec_format_samples (dec, GST_BUFFER_DATA (outbuf),
- unpack_buf, wph.block_samples);
- gst_buffer_stamp (outbuf, buf);
+ width = dec->width;
+ depth = dec->depth;
+ max = dec->channels * wph.block_samples;
+ if (width == 8) {
+ gint8 *outbuffer = (gint8 *) out_data;
- 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);
+ for (i = 0; i < max; i--) {
+ *outbuffer++ = (gint8) (dec_data[i]);
+ }
+ } else if (width == 16) {
+ gint16 *outbuffer = (gint16 *) out_data;
+
+ for (i = 0; i < max; i++) {
+ *outbuffer++ = (gint8) (dec_data[i]);
+ }
+ } else if (dec->width == 32) {
+ gint32 *outbuffer = (gint32 *) out_data;
+
+ if (width != depth) {
+ for (i = 0; i < max; i++) {
+ *outbuffer++ = (gint32) (dec_data[i] << (width - depth));
+ }
+ } else {
+ for (i = 0; i < max; i++) {
+ *outbuffer++ = (gint32) dec_data[i];
+ }
+ }
} else {
- gst_buffer_unref (outbuf);
+ g_assert_not_reached ();
}
+ g_free (dec_data);
+
+ ret = gst_audio_decoder_finish_frame (bdec, outbuf, 1);
+
out:
if (G_UNLIKELY (ret != GST_FLOW_OK)) {
GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (ret));
}
- g_free (unpack_buf);
- 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:
+context_failed:
{
- GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL),
- ("Failed to decode wavpack stream"));
- g_free (unpack_buf);
- gst_buffer_unref (buf);
- return GST_FLOW_ERROR;
+ GST_AUDIO_DECODER_ERROR (bdec, 1, LIBRARY, INIT, (NULL),
+ ("error creating Wavpack context"), ret);
+ goto out;
}
-}
-
-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;
+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";
}
- 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:
- gst_segment_init (&dec->segment, GST_FORMAT_UNDEFINED);
- break;
- case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
- break;
- default:
- break;
+ GST_AUDIO_DECODER_ERROR (bdec, 1, STREAM, DECODE, (NULL),
+ ("decoding error: %s", reason), ret);
+ g_free (dec_data);
+ if (outbuf)
+ gst_buffer_unref (outbuf);
+ if (ret == GST_FLOW_OK)
+ gst_audio_decoder_finish_frame (bdec, NULL, 1);
+ return ret;
}
-
- 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;
- }
- dec->wv_id.buffer = NULL;
- dec->wv_id.position = 0;
- dec->wv_id.length = 0;
- dec->channels = 0;
- dec->sample_rate = 0;
- dec->width = 0;
- break;
- case GST_STATE_CHANGE_READY_TO_NULL:
- break;
- default:
- break;
- }
-
- return ret;
}
gboolean
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");
+ GST_DEBUG_CATEGORY_INIT (gst_wavpack_dec_debug, "wavpack_dec", 0,
+ "Wavpack decoder");
return TRUE;
}
projects / platform / upstream / gstreamer.git / blobdiff