/* GStreamer Wavpack plugin
- * (c) 2005 Arwed v. Merkatz <v.merkatz@gmx.net>
+ * 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
*
*/
#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"
+
+
+#define WAVPACK_DEC_MAX_ERRORS 16
GST_DEBUG_CATEGORY_STATIC (gst_wavpack_dec_debug);
#define GST_CAT_DEFAULT gst_wavpack_dec_debug
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-wavpack, "
"width = (int) { 8, 16, 24, 32 }, "
- "channels = (int) { 1, 2 }, "
+ "channels = (int) [ 1, 2 ], "
"rate = (int) [ 6000, 192000 ], " "framed = (boolean) true")
);
+#if 0
static GstStaticPadTemplate wvc_sink_factory =
GST_STATIC_PAD_TEMPLATE ("wvcsink",
GST_PAD_SINK,
GST_PAD_REQUEST,
GST_STATIC_CAPS ("audio/x-wavpack-correction, " "framed = (boolean) true")
);
+#endif
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_STATIC_CAPS ("audio/x-raw-int, "
"width = (int) { 8, 16, 24, 32 }, "
"depth = (int) { 8, 16, 24, 32 }, "
- "channels = (int) { 1, 2 }, "
+ "channels = (int) [ 1, 2 ], "
"rate = (int) [ 6000, 192000 ], "
"endianness = (int) LITTLE_ENDIAN, " "signed = (boolean) true")
-/*
- "audio/x-raw-float, "
- "width = (int) 32, "
- "channels = (int) { 1, 2 }, "
- "rate = (int) [ 6000, 192000 ], " "endianness = (int) LITTLE_ENDIAN"
-*/
);
static GstFlowReturn gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buffer);
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 gboolean gst_wavpack_dec_sink_event (GstPad * pad, GstEvent * event);
-GST_BOILERPLATE (GstWavpackDec, gst_wavpack_dec, GstElement, GST_TYPE_ELEMENT);
-
-static gboolean
-gst_wavpack_dec_setcaps (GstPad * pad, GstCaps * caps)
-{
- GstWavpackDec *wavpackdec = GST_WAVPACK_DEC (gst_pad_get_parent (pad));
- GstStructure *structure;
- GstCaps *srccaps;
- gint bits, rate, channels;
-
- structure = gst_caps_get_structure (caps, 0);
- if (!gst_structure_get_int (structure, "rate", &rate) ||
- !gst_structure_get_int (structure, "channels", &channels) ||
- !gst_structure_get_int (structure, "width", &bits)) {
- return FALSE;
- }
-
- wavpackdec->samplerate = rate;
- wavpackdec->channels = channels;
- wavpackdec->width = bits;
-
-/* 32-bit output seems to be in fact 32 bit int (e.g. Prod_Girls.wv) */
-/* if (bits != 32) { */
- srccaps = gst_caps_new_simple ("audio/x-raw-int",
- "rate", G_TYPE_INT, wavpackdec->samplerate,
- "channels", G_TYPE_INT, wavpackdec->channels,
- "depth", G_TYPE_INT, bits,
- "width", G_TYPE_INT, bits,
- "endianness", G_TYPE_INT, G_LITTLE_ENDIAN,
- "signed", G_TYPE_BOOLEAN, TRUE, NULL);
-/*
- } else {
- srccaps = gst_caps_new_simple ("audio/x-raw-float",
- "rate", G_TYPE_INT, wavpackdec->samplerate,
- "channels", G_TYPE_INT, wavpackdec->channels,
- "width", G_TYPE_INT, 32,
- "endianness", G_TYPE_INT, G_LITTLE_ENDIAN, NULL);
- }
-*/
-/* gst_pad_set_caps (wavpackdec->sinkpad, caps); */
-
- gst_pad_set_caps (wavpackdec->srcpad, srccaps);
- gst_pad_use_fixed_caps (wavpackdec->srcpad);
+#if 0
+static GstPad *gst_wavpack_dec_request_new_pad (GstElement * element,
+ GstPadTemplate * templ, const gchar * name);
+#endif
- return TRUE;
-}
+GST_BOILERPLATE (GstWavpackDec, gst_wavpack_dec, GstElement, GST_TYPE_ELEMENT);
#if 0
static GstPadLinkReturn
GST_ELEMENT_DETAILS ("WavePack audio decoder",
"Codec/Decoder/Audio",
"Decode Wavpack audio data",
- "Arwed v. Merkatz <v.merkatz@gmx.net>");
+ "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));
+#if 0
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&wvc_sink_factory));
+#endif
gst_element_class_set_details (element_class, &plugin_details);
}
static void
-gst_wavpack_dec_dispose (GObject * object)
-{
- GstWavpackDec *wavpackdec = GST_WAVPACK_DEC (object);
-
- g_free (wavpackdec->decodebuf);
- wavpackdec->decodebuf = NULL;
-
- /* FIXME: what about wavpackdec->stream and wavpackdec->context? (tpm) */
-
- G_OBJECT_CLASS (parent_class)->dispose (object);
-}
-
-static void
gst_wavpack_dec_class_init (GstWavpackDecClass * klass)
{
- GObjectClass *gobject_class;
- GstElementClass *gstelement_class;
-
- gobject_class = (GObjectClass *) klass;
- gstelement_class = (GstElementClass *) klass;
-
- gobject_class->dispose = gst_wavpack_dec_dispose;
-}
-
-static gboolean
-gst_wavpack_dec_src_query (GstPad * pad, GstQuery * query)
-{
- return gst_pad_query_default (pad, query);
-}
-
-static gboolean
-gst_wavpack_dec_sink_event (GstPad * pad, GstEvent * event)
-{
- GstWavpackDec *dec;
-
- 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:{
- /* TODO: save current segment so we can do clipping, for now
- * we'll just leave the clipping to the audio sink */
- break;
- }
- default:
- break;
- }
-
- gst_object_unref (dec);
+ GObjectClass *gobject_class = (GObjectClass *) klass;
+ GstElementClass *gstelement_class = (GstElementClass *) klass;
- return gst_pad_event_default (pad, event);
+ gstelement_class->change_state =
+ GST_DEBUG_FUNCPTR (gst_wavpack_dec_change_state);
+#if 0
+ gstelement_class->request_new_pad =
+ GST_DEBUG_FUNCPTR (gst_wavpack_dec_request_new_pad);
+#endif
+ gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_wavpack_dec_finalize);
}
static void
wavpackdec->sinkpad =
gst_pad_new_from_template (gst_element_class_get_pad_template (klass,
"sink"), "sink");
- gst_element_add_pad (GST_ELEMENT (wavpackdec), wavpackdec->sinkpad);
-
gst_pad_set_chain_function (wavpackdec->sinkpad,
GST_DEBUG_FUNCPTR (gst_wavpack_dec_chain));
- gst_pad_set_setcaps_function (wavpackdec->sinkpad,
- GST_DEBUG_FUNCPTR (gst_wavpack_dec_setcaps));
gst_pad_set_event_function (wavpackdec->sinkpad,
GST_DEBUG_FUNCPTR (gst_wavpack_dec_sink_event));
-
-#if 0
- wavpackdec->wvcsinkpad =
- gst_pad_new_from_template (gst_element_class_get_pad_template (klass,
- "wvcsink"), "wvcsink");
- gst_pad_set_link_function (wavpackdec->wvcsinkpad, gst_wavpack_dec_wvclink);
- gst_element_add_pad (GST_ELEMENT (wavpackdec), wavpackdec->wvcsinkpad);
-#endif
-
+ gst_element_add_pad (GST_ELEMENT (wavpackdec), wavpackdec->sinkpad);
wavpackdec->srcpad =
gst_pad_new_from_template (gst_element_class_get_pad_template (klass,
"src"), "src");
gst_pad_use_fixed_caps (wavpackdec->srcpad);
-
- gst_pad_set_query_function (wavpackdec->srcpad,
- GST_DEBUG_FUNCPTR (gst_wavpack_dec_src_query));
-
gst_element_add_pad (GST_ELEMENT (wavpackdec), wavpackdec->srcpad);
- wavpackdec->decodebuf = NULL;
- wavpackdec->decodebuf_size = 0;
- wavpackdec->stream = (WavpackStream *) g_malloc0 (sizeof (WavpackStream));
- wavpackdec->context = (WavpackContext *) g_malloc0 (sizeof (WavpackContext));
-}
+ wavpackdec->context = NULL;
+ wavpackdec->stream_reader = gst_wavpack_stream_reader_new ();
-static void
-gst_wavpack_dec_setup_context (GstWavpackDec * wavpackdec, guchar * data,
- guchar * cdata)
-{
- WavpackContext *context = wavpackdec->context;
- WavpackStream *stream = wavpackdec->stream;
- guint buffer_size;
+ wavpackdec->wv_id.buffer = NULL;
+ wavpackdec->wv_id.position = wavpackdec->wv_id.length = 0;
- memset (context, 0, sizeof (context));
+/*
+ wavpackdec->wvc_id.buffer = NULL;
+ wavpackdec->wvc_id.position = wavpackdec->wvc_id.length = 0;
+ wavpackdec->wvcsinkpad = NULL;
+*/
- context->open_flags = 0;
- context->current_stream = 0;
- context->num_streams = 1;
+ wavpackdec->error_count = 0;
- memset (stream, 0, sizeof (stream));
- context->streams[0] = stream;
- gst_wavpack_read_header (&stream->wphdr, data);
- stream->blockbuff = data;
+ wavpackdec->channels = 0;
+ wavpackdec->sample_rate = 0;
+ wavpackdec->width = 0;
- if (cdata) {
- context->wvc_flag = TRUE;
- gst_wavpack_read_header (&stream->wphdr, cdata);
- stream->block2buff = cdata;
- } else {
- context->wvc_flag = FALSE;
- }
+ gst_segment_init (&wavpackdec->segment, GST_FORMAT_UNDEFINED);
+}
- buffer_size =
- stream->wphdr.block_samples * wavpackdec->channels * sizeof (int32_t);
- if (wavpackdec->decodebuf_size < buffer_size) {
- wavpackdec->decodebuf =
- (int32_t *) g_realloc (wavpackdec->decodebuf, buffer_size);
- wavpackdec->decodebuf_size = buffer_size;
- }
+static void
+gst_wavpack_dec_finalize (GObject * object)
+{
+ GstWavpackDec *wavpackdec = GST_WAVPACK_DEC (object);
+
+ g_free (wavpackdec->stream_reader);
+ wavpackdec->stream_reader = NULL;
- unpack_init (context);
+ G_OBJECT_CLASS (parent_class)->finalize (object);
}
-static GstBuffer *
-gst_wavpack_dec_format_samples (GstWavpackDec * wavpackdec, int32_t * samples,
- guint num_samples)
+static void
+gst_wavpack_dec_format_samples (GstWavpackDec * wavpackdec, guint8 * dst,
+ int32_t * samples, guint num_samples)
{
- GstBuffer *buf;
gint i;
- guint8 *dst;
int32_t temp;
- buf =
- gst_buffer_new_and_alloc (num_samples * wavpackdec->width / 8 *
- wavpackdec->channels);
-
- dst = (guint8 *) GST_BUFFER_DATA (buf);
-
switch (wavpackdec->width) {
case 8:
for (i = 0; i < num_samples * wavpackdec->channels; ++i)
- *dst++ = *samples++ + 128;
+ *dst++ = (guint8) (*samples++);
break;
case 16:
for (i = 0; i < num_samples * wavpackdec->channels; ++i) {
default:
break;
}
+}
+
+static gboolean
+gst_wavpack_dec_clip_outgoing_buffer (GstWavpackDec * wavpackdec,
+ GstBuffer * buf)
+{
+ gint64 start, stop, cstart, cstop, diff;
- return buf;
+ if (wavpackdec->segment.format != GST_FORMAT_TIME)
+ return TRUE;
+
+ start = GST_BUFFER_TIMESTAMP (buf);
+ stop = start + GST_BUFFER_DURATION (buf);
+
+ if (gst_segment_clip (&wavpackdec->segment, GST_FORMAT_TIME,
+ start, stop, &cstart, &cstop)) {
+
+ diff = cstart - start;
+ if (diff > 0) {
+ GST_BUFFER_TIMESTAMP (buf) = cstart;
+ GST_BUFFER_DURATION (buf) -= diff;
+
+ diff = ((wavpackdec->width + 7) >> 3) * wavpackdec->channels
+ * GST_CLOCK_TIME_TO_FRAMES (diff, wavpackdec->sample_rate);
+ GST_BUFFER_DATA (buf) += diff;
+ GST_BUFFER_SIZE (buf) -= diff;
+ }
+
+ diff = cstop - stop;
+ if (diff > 0) {
+ GST_BUFFER_DURATION (buf) -= diff;
+
+ diff = ((wavpackdec->width + 7) >> 3) * wavpackdec->channels
+ * GST_CLOCK_TIME_TO_FRAMES (diff, wavpackdec->sample_rate);
+ GST_BUFFER_SIZE (buf) -= diff;
+ }
+ } else {
+ GST_DEBUG_OBJECT (wavpackdec, "buffer is outside configured segment");
+ return FALSE;
+ }
+
+ return TRUE;
}
static GstFlowReturn
gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buf)
{
- GstWavpackDec *wavpackdec = GST_WAVPACK_DEC (gst_pad_get_parent (pad));
- GstBuffer *outbuf, *cbuf = NULL;
+ GstWavpackDec *wavpackdec;
+ GstBuffer *outbuf;
+ GstBuffer *cbuf = NULL;
GstFlowReturn ret = GST_FLOW_OK;
+ WavpackHeader wph;
+ int32_t *unpack_buf;
+ int32_t unpacked_sample_count;
+
+ wavpackdec = GST_WAVPACK_DEC (GST_PAD_PARENT (pad));
+
+ /* we only accept framed input with complete chunks */
+ g_assert (GST_BUFFER_SIZE (buf) >= sizeof (WavpackHeader));
+ gst_wavpack_read_header (&wph, GST_BUFFER_DATA (buf));
+ g_assert (GST_BUFFER_SIZE (buf) ==
+ wph.ckSize + 4 * sizeof (char) + sizeof (uint32_t));
+
+ wavpackdec->wv_id.buffer = GST_BUFFER_DATA (buf);
+ wavpackdec->wv_id.length = GST_BUFFER_SIZE (buf);
+ wavpackdec->wv_id.position = 0;
#if 0
+ /* check whether the correction pad is linked and we can get
+ * the correction chunk that corresponds to our current data */
if (gst_pad_is_linked (wavpackdec->wvcsinkpad)) {
if (GST_FLOW_OK != gst_pad_pull_range (wavpackdec->wvcsinkpad,
- wavpackdec->wvcflushed_bytes, -1, &cbuf)) {
+ GST_BUFFER_OFFSET (buf), -1, &cbuf)) {
cbuf = NULL;
} else {
- wavpackdec->wvcflushed_bytes += GST_BUFFER_SIZE (cbuf);
+ /* this won't work (tpm) */
+ if (!(GST_BUFFER_TIMESTAMP (cbuf) == GST_BUFFER_TIMESTAMP (buf)) ||
+ !(GST_BUFFER_DURATION (cbuf) == GST_BUFFER_DURATION (buf)) ||
+ !(GST_BUFFER_OFFSET (cbuf) == GST_BUFFER_OFFSET (buf)) ||
+ !(GST_BUFFER_OFFSET_END (cbuf) == GST_BUFFER_OFFSET (buf))) {
+ gst_buffer_unref (cbuf);
+ cbuf = NULL;
+ } else {
+ wavpackdec->wvc_id.buffer = GST_BUFFER_DATA (cbuf);
+ wavpackdec->wvc_id.length = GST_BUFFER_SIZE (cbuf);
+ wavpackdec->wvc_id.position = 0;
+ }
}
-
}
#endif
- gst_wavpack_dec_setup_context (wavpackdec, GST_BUFFER_DATA (buf),
- cbuf ? GST_BUFFER_DATA (cbuf) : NULL);
- unpack_samples (wavpackdec->context, wavpackdec->decodebuf,
- wavpackdec->context->streams[0]->wphdr.block_samples);
- outbuf =
- gst_wavpack_dec_format_samples (wavpackdec, wavpackdec->decodebuf,
- wavpackdec->context->streams[0]->wphdr.block_samples);
+ /* 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 (!wavpackdec->context) {
+ gchar error_msg[80];
- gst_buffer_stamp (outbuf, buf);
+/*
+ wavpackdec->context =
+ WavpackOpenFileInputEx (wavpackdec->stream_reader, &wavpackdec->wv_id,
+ (cbuf) ? &wavpackdec->wvc_id : NULL, error_msg, OPEN_STREAMING, 0);
+*/
+
+ wavpackdec->context = WavpackOpenFileInputEx (wavpackdec->stream_reader,
+ &wavpackdec->wv_id, NULL, error_msg, OPEN_STREAMING, 0);
+
+ if (!wavpackdec->context) {
+ wavpackdec->error_count++;
+ GST_ELEMENT_WARNING (wavpackdec, LIBRARY, INIT, (NULL),
+ ("Couldn't open buffer for decoding: %s", error_msg));
+ if (wavpackdec->error_count <= WAVPACK_DEC_MAX_ERRORS) {
+ ret = GST_FLOW_OK;
+ } else {
+ ret = GST_FLOW_ERROR;
+ }
+ gst_buffer_unref (buf);
+ if (cbuf) {
+ gst_buffer_unref (cbuf);
+ }
+ return ret;
+ }
+
+ if (GST_PAD_CAPS (wavpackdec->srcpad)) {
+ if ((wavpackdec->sample_rate !=
+ WavpackGetSampleRate (wavpackdec->context))
+ || (wavpackdec->channels !=
+ WavpackGetNumChannels (wavpackdec->context))
+ || (wavpackdec->width !=
+ WavpackGetBitsPerSample (wavpackdec->context))) {
+ gst_buffer_unref (buf);
+ if (cbuf) {
+ gst_buffer_unref (cbuf);
+ }
+
+ /* FIXME: use the right error */
+ GST_ELEMENT_ERROR (wavpackdec, LIBRARY, INIT, (NULL),
+ ("Got Wavpack chunk with changed format settings!"));
+ return GST_FLOW_ERROR;
+ }
+ }
+ }
+ wavpackdec->error_count = 0;
+
+ if (!GST_PAD_CAPS (wavpackdec->srcpad)) {
+ GstCaps *caps;
+
+ g_assert (wavpackdec->context);
+ wavpackdec->sample_rate = WavpackGetSampleRate (wavpackdec->context);
+ wavpackdec->channels = WavpackGetNumChannels (wavpackdec->context);
+ wavpackdec->width = WavpackGetBitsPerSample (wavpackdec->context);
+ caps = gst_caps_new_simple ("audio/x-raw-int",
+ "rate", G_TYPE_INT, wavpackdec->sample_rate,
+ "channels", G_TYPE_INT, wavpackdec->channels,
+ "depth", G_TYPE_INT, wavpackdec->width,
+ "width", G_TYPE_INT, wavpackdec->width,
+ "endianness", G_TYPE_INT, G_LITTLE_ENDIAN,
+ "signed", G_TYPE_BOOLEAN, TRUE, NULL);
+
+ if (!gst_pad_set_caps (wavpackdec->srcpad, caps)) {
+ gst_caps_unref (caps);
+ WavpackCloseFile (wavpackdec->context);
+ wavpackdec->context = NULL;
+ gst_buffer_unref (buf);
+ if (cbuf) {
+ gst_buffer_unref (cbuf);
+ }
+ /* FIXME: use the right error */
+ GST_ELEMENT_ERROR (wavpackdec, LIBRARY, INIT, (NULL),
+ ("Couldn't set caps on source pad: %" GST_PTR_FORMAT, caps));
+ return GST_FLOW_ERROR;
+ }
+ gst_caps_unref (caps);
+ gst_pad_use_fixed_caps (wavpackdec->srcpad);
+ }
+
+ g_assert (wavpackdec->context);
+ unpack_buf =
+ (int32_t *) g_malloc (sizeof (int32_t) * wph.block_samples *
+ wavpackdec->channels);
+ unpacked_sample_count =
+ WavpackUnpackSamples (wavpackdec->context, unpack_buf, wph.block_samples);
+ g_assert (unpacked_sample_count == wph.block_samples);
+
+ ret =
+ gst_pad_alloc_buffer_and_set_caps (wavpackdec->srcpad,
+ GST_BUFFER_OFFSET (buf),
+ wph.block_samples * ((wavpackdec->width +
+ 7) >> 3) * wavpackdec->channels,
+ GST_PAD_CAPS (wavpackdec->srcpad), &outbuf);
+
+ if (GST_FLOW_IS_FATAL (ret)) {
+ WavpackCloseFile (wavpackdec->context);
+ wavpackdec->context = NULL;
+ g_free (unpack_buf);
+ gst_buffer_unref (buf);
+ if (cbuf) {
+ gst_buffer_unref (cbuf);
+ }
+ return ret;
+ } else if (ret != GST_FLOW_OK) {
+ g_free (unpack_buf);
+ gst_buffer_unref (buf);
+ if (cbuf) {
+ gst_buffer_unref (cbuf);
+ }
+ return ret;
+ }
+
+ gst_wavpack_dec_format_samples (wavpackdec, GST_BUFFER_DATA (outbuf),
+ unpack_buf, wph.block_samples);
+ g_free (unpack_buf);
+ gst_buffer_stamp (outbuf, buf);
gst_buffer_unref (buf);
if (cbuf) {
gst_buffer_unref (cbuf);
}
- gst_buffer_set_caps (outbuf, GST_PAD_CAPS (wavpackdec->srcpad));
+ if (gst_wavpack_dec_clip_outgoing_buffer (wavpackdec, outbuf)) {
+ GST_LOG_OBJECT (wavpackdec, "pushing buffer with time %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)));
+ ret = gst_pad_push (wavpackdec->srcpad, outbuf);
+ if (ret != GST_FLOW_OK) {
+ GST_DEBUG_OBJECT (wavpackdec, "pad_push: %s", gst_flow_get_name (ret));
+ }
+ } else {
+ gst_buffer_unref (outbuf);
+ }
- GST_LOG_OBJECT (wavpackdec, "pushing buffer with time %" GST_TIME_FORMAT,
- GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)));
+ return ret;
+}
+
+static gboolean
+gst_wavpack_dec_sink_event (GstPad * pad, GstEvent * event)
+{
+ GstWavpackDec *wavpackdec = GST_WAVPACK_DEC (gst_pad_get_parent (pad));
- ret = gst_pad_push (wavpackdec->srcpad, outbuf);
- if (ret != GST_FLOW_OK) {
- GST_DEBUG_OBJECT (wavpackdec, "pad_push: %s", gst_flow_get_name (ret));
+ GST_LOG_OBJECT (wavpackdec, "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 (&wavpackdec->segment, is_update, rate, fmt,
+ start, end, base);
+ } else {
+ gst_segment_init (&wavpackdec->segment, GST_FORMAT_UNDEFINED);
+ }
+ break;
+ }
+ default:
+ break;
+ }
+
+ gst_object_unref (wavpackdec);
+ 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 *wavpackdec = GST_WAVPACK_DEC (element);
+
+ switch (transition) {
+ case GST_STATE_CHANGE_NULL_TO_READY:
+ break;
+ case GST_STATE_CHANGE_READY_TO_PAUSED:
+ gst_segment_init (&wavpackdec->segment, GST_FORMAT_UNDEFINED);
+ 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 (wavpackdec->context) {
+ WavpackCloseFile (wavpackdec->context);
+ wavpackdec->context = NULL;
+ }
+ wavpackdec->wv_id.buffer = NULL;
+ wavpackdec->wv_id.position = 0;
+ wavpackdec->wv_id.length = 0;
+ /*
+ wavpackdec->wvc_id.buffer = NULL;
+ wavpackdec->wvc_id.position = 0;
+ wavpackdec->wvc_id.length = 0;
+ wavpackdec->error_count = 0;
+ wavpackdec->wvcsinkpad = NULL;
+ */
+ wavpackdec->channels = 0;
+ wavpackdec->sample_rate = 0;
+ wavpackdec->width = 0;
+ break;
+ case GST_STATE_CHANGE_READY_TO_NULL:
+ break;
+ default:
+ break;
}
return ret;
}
+#if 0
+static GstPad *
+gst_wavpack_dec_request_new_pad (GstElement * element,
+ GstPadTemplate * templ, const gchar * name)
+{
+ GstWavpackDec *wavpackdec = GST_WAVPACK_DEC (element);
+ GstPad *pad;
+
+ if (wavpackdec->wvcsinkpad == NULL) {
+ wavpackdec->wvcsinkpad = gst_pad_new_from_template (template, name);
+ gst_pad_set_link_function (wavpackdec->wvcsinkpad, gst_wavpack_dec_wvclink);
+ gst_pad_use_fixed_caps (wavpackdec->wvcsinkpad);
+ gst_element_add_pad (GST_ELEMENT (wavpackdec), wavpackdec->wvcsinkpad);
+ gst_element_no_more_pads (GST_ELEMENT (wavpackdec));
+ } else {
+ pad = NULL;
+ }
+
+ return pad;
+}
+#endif
+
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;
}
/* GStreamer wavpack plugin
- * (c) 2005 Arwed v. Merkatz <v.merkatz@gmx.net>
- * (c) 2006 Tim-Philipp Müller <tim centricular net>
+ * Copyright (c) 2005 Arwed v. Merkatz <v.merkatz@gmx.net>
+ * Copyright (c) 2006 Tim-Philipp Müller <tim centricular net>
+ * Copyright (c) 2006 Sebastian Dröge <slomo@circular-chaos.org>
*
* gstwavpackparse.c: wavpack file parser
*
#include <wavpack/wavpack.h>
#include "gstwavpackparse.h"
+#include "gstwavpackstreamreader.h"
#include "gstwavpackcommon.h"
GST_DEBUG_CATEGORY_STATIC (gst_wavpack_parse_debug);
GST_PAD_SOMETIMES,
GST_STATIC_CAPS ("audio/x-wavpack, "
"width = (int) { 8, 16, 24, 32 }, "
- "channels = (int) { 1, 2 }, "
+ "channels = (int) [ 1, 2 ], "
"rate = (int) [ 6000, 192000 ], " "framed = (boolean) true")
);
GST_ELEMENT_DETAILS ("WavePack parser",
"Codec/Demuxer/Audio",
"Parses Wavpack files",
- "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,
}
static void
-gst_wavpack_parse_dispose (GObject * object)
+gst_wavpack_parse_finalize (GObject * object)
{
gst_wavpack_parse_reset (GST_WAVPACK_PARSE (object));
- G_OBJECT_CLASS (parent_class)->dispose (object);
+
+ G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
- gobject_class->dispose = gst_wavpack_parse_dispose;
+ gobject_class->finalize = gst_wavpack_parse_finalize;
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_wavpack_parse_change_state);
}
entry = &g_array_index (wvparse->entries, GstWavpackParseIndexEntry, i);
GST_LOG_OBJECT (wvparse, "Index entry %03u: sample %" G_GINT64_FORMAT " @"
- " byte %" G_GINT64_FORMAT, entry->sample_offset, entry->byte_offset);
+ " byte %" G_GINT64_FORMAT, i, entry->sample_offset, entry->byte_offset);
if (entry->sample_offset <= sample_offset &&
sample_offset < entry->sample_offset_end) {
GST_LOG_OBJECT (wvparse, "found match");
return entry;
}
+
+ /* as the list is sorted and we first look at the latest entry
+ * we can abort searching for an entry if the sample we want is
+ * after the latest one */
+ if (sample_offset >= entry->sample_offset_end)
+ break;
}
GST_LOG_OBJECT (wvparse, "no match in index");
return NULL;
/* now scan forward until we find the chunk we're looking for or hit EOS */
do {
- WavpackHeader header = { {0,}
- , 0,
- };
+ WavpackHeader header;
GstBuffer *buf;
buf = gst_wavpack_parse_pull_buffer (parse, off, sizeof (WavpackHeader),
stop = gst_util_uint64_scale_int (stop, rate, GST_SECOND);
}
+ /* if seek is to something after the end of the stream seek only
+ * to the end. this can be caused by rounding errors */
+ if (start >= wvparse->total_samples)
+ start = wvparse->total_samples;
+
flush = ((seek_flags & GST_SEEK_FLAG_FLUSH) != 0);
if (start < 0) {
gst_wavpack_parse_create_src_pad (GstWavpackParse * wvparse, GstBuffer * buf,
WavpackHeader * header)
{
- WavpackMetadata meta;
+ GstWavpackMetadata meta;
GstCaps *caps = NULL;
guchar *bufptr;
bufptr = GST_BUFFER_DATA (buf) + sizeof (WavpackHeader);
- while (read_metadata_buff (&meta, GST_BUFFER_DATA (buf), &bufptr)) {
+ while (gst_wavpack_read_metadata (&meta, GST_BUFFER_DATA (buf), &bufptr)) {
switch (meta.id) {
case ID_WVC_BITSTREAM:{
caps = gst_caps_new_simple ("audio/x-wavpack-correction",
(GST_ELEMENT_GET_CLASS (wvparse), "wvcsrc"), "wvcsrc");
break;
}
- case ID_RIFF_HEADER:{
- WaveHeader wheader;
-
- /* skip RiffChunkHeader and ChunkHeader */
- g_memmove (&wheader, meta.data + 20, sizeof (WaveHeader));
- little_endian_to_native (&wheader, WaveHeaderFormat);
- wvparse->samplerate = wheader.SampleRate;
- wvparse->channels = wheader.NumChannels;
+ case ID_WV_BITSTREAM:
+ case ID_WVX_BITSTREAM:{
+ WavpackStreamReader *stream_reader = gst_wavpack_stream_reader_new ();
+ WavpackContext *wpc;
+ gchar error_msg[80];
+ read_id rid;
+
+ rid.buffer = GST_BUFFER_DATA (buf);
+ rid.length = GST_BUFFER_SIZE (buf);
+ rid.position = 0;
+
+ wpc =
+ WavpackOpenFileInputEx (stream_reader, &rid, NULL, error_msg, 0, 0);
+
+ if (!wpc)
+ return FALSE;
+
+ wvparse->samplerate = WavpackGetSampleRate (wpc);
+ wvparse->channels = WavpackGetNumChannels (wpc);
wvparse->total_samples = header->total_samples;
+ if (wvparse->total_samples == (int32_t) - 1)
+ wvparse->total_samples = 0;
+ else
+ wvparse->total_samples--;
+
caps = gst_caps_new_simple ("audio/x-wavpack",
- "width", G_TYPE_INT, wheader.BitsPerSample,
+ "width", G_TYPE_INT, WavpackGetBitsPerSample (wpc),
"channels", G_TYPE_INT, wvparse->channels,
"rate", G_TYPE_INT, wvparse->samplerate,
"framed", G_TYPE_BOOLEAN, TRUE, NULL);
wvparse->srcpad =
gst_pad_new_from_template (gst_element_class_get_pad_template
(GST_ELEMENT_GET_CLASS (wvparse), "src"), "src");
+ WavpackCloseFile (wpc);
+ g_free (stream_reader);
break;
}
default:{
GST_DEBUG_FUNCPTR (gst_wavpack_parse_src_event));
gst_pad_set_caps (wvparse->srcpad, caps);
+ gst_caps_unref (caps);
gst_pad_use_fixed_caps (wvparse->srcpad);
gst_object_ref (wvparse->srcpad);
projects / platform / upstream / gst-plugins-good.git / commitdiff