/* -*- Mode: C; tab-width: 2; indent-tabs-mode: t; c-basic-offset: 2 -*- */
/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
- * Copyright (C) <2006> Nokia Corporation.
+ * Copyright (C) <2006> Nokia Corporation, Stefan Kost <stefan.kost@nokia.com>.
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* <title>Example launch line</title>
* <para>
* <programlisting>
- * gst-launch filesrc sine.wav ! wavparse ! audioconvert ! alsasink
+ * gst-launch filesrc location=sine.wav ! queue ! wavparse ! audioconvert ! alsasink
* </programlisting>
* Read a wav file and output to the soundcard using the ALSA element. The
* wav file is assumed to contain raw uncompressed samples.
* </para>
+ * <para>
+ * <programlisting>
+ * gst-launch gnomevfssrc location=http://www.example.org/sine.wav ! wavparse ! audioconvert ! alsasink
+ * </programlisting>
+ * Stream data from
+ * </para>
+ *
* </refsect2>
*
* Last reviewed on 2006-03-03 (0.10.3)
static void gst_wavparse_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
+static const GstElementDetails gst_wavparse_details =
+GST_ELEMENT_DETAILS ("WAV audio demuxer",
+ "Codec/Demuxer/Audio",
+ "Parse a .wav file into raw audio",
+ "Erik Walthinsen <omega@cse.ogi.edu>");
+
static GstStaticPadTemplate sink_template_factory =
GST_STATIC_PAD_TEMPLATE ("wavparse_sink",
GST_PAD_SINK,
gst_wavparse_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
- GstPadTemplate *templ;
- static const GstElementDetails gst_wavparse_details =
- GST_ELEMENT_DETAILS ("WAV audio demuxer",
- "Codec/Demuxer/Audio",
- "Parse a .wav file into raw audio",
- "Erik Walthinsen <omega@cse.ogi.edu>");
-
- gst_element_class_set_details (element_class, &gst_wavparse_details);
/* register src pads */
- templ = gst_static_pad_template_get (&sink_template_factory);
- gst_element_class_add_pad_template (element_class, templ);
- gst_object_unref (templ);
- templ = gst_static_pad_template_get (&src_template_factory);
- gst_element_class_add_pad_template (element_class, templ);
- gst_object_unref (templ);
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&sink_template_factory));
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&src_template_factory));
+ gst_element_class_set_details (element_class, &gst_wavparse_details);
}
static void
gstelement_class->change_state = gst_wavparse_change_state;
gstelement_class->send_event = gst_wavparse_send_event;
-
- GST_DEBUG_CATEGORY_INIT (wavparse_debug, "wavparse", 0, "WAV parser");
}
wavparse->datasize = 0;
wavparse->datastart = 0;
wavparse->got_fmt = FALSE;
+ wavparse->first = TRUE;
if (wavparse->seek_event)
gst_event_unref (wavparse->seek_event);
gst_pad_set_chain_function (wavparse->sinkpad,
GST_DEBUG_FUNCPTR (gst_wavparse_chain));
gst_element_add_pad (GST_ELEMENT (wavparse), wavparse->sinkpad);
+
+ /* src, will be created later */
+ wavparse->srcpad = NULL;
}
static void
GST_DEBUG_FUNCPTR (gst_wavparse_pad_query));
gst_pad_set_event_function (wavparse->srcpad,
GST_DEBUG_FUNCPTR (gst_wavparse_srcpad_event));
+
+ GST_DEBUG_OBJECT (wavparse, "srcpad created");
}
static void
}
}
+
+
#if 0
static void
gst_wavparse_parse_adtl (GstWavParse * wavparse, int len)
g_object_notify (G_OBJECT (wavparse), "metadata");
}
-#endif
-#if 0
static void
gst_wavparse_parse_cues (GstWavParse * wavparse, int len)
{
g_object_notify (G_OBJECT (wavparse), "metadata");
}
-#endif
-
-static gboolean
-gst_wavparse_parse_file_header (GstElement * element, GstBuffer * buf)
-{
- guint32 doctype;
-
- if (!gst_riff_parse_file_header (element, buf, &doctype))
- return FALSE;
-
- if (doctype != GST_RIFF_RIFF_WAVE)
- goto not_wav;
-
- return TRUE;
-
- /* ERRORS */
-not_wav:
- {
- GST_ELEMENT_ERROR (element, STREAM, WRONG_TYPE, (NULL),
- ("File is not an WAVE file: %" GST_FOURCC_FORMAT,
- GST_FOURCC_ARGS (doctype)));
- return FALSE;
- }
-}
-
-static GstFlowReturn
-gst_wavparse_stream_init (GstWavParse * wav)
-{
- GstFlowReturn res;
- GstBuffer *buf = NULL;
-
- if ((res = gst_pad_pull_range (wav->sinkpad,
- wav->offset, 12, &buf)) != GST_FLOW_OK)
- return res;
- else if (!gst_wavparse_parse_file_header (GST_ELEMENT (wav), buf))
- return GST_FLOW_ERROR;
-
- wav->offset += 12;
-
- return GST_FLOW_OK;
-}
-#if 0
/* Read 'fmt ' header */
static gboolean
gst_wavparse_fmt (GstWavParse * wav)
}
#endif
+
+
+static gboolean
+gst_wavparse_parse_file_header (GstElement * element, GstBuffer * buf)
+{
+ guint32 doctype;
+
+ if (!gst_riff_parse_file_header (element, buf, &doctype))
+ return FALSE;
+
+ if (doctype != GST_RIFF_RIFF_WAVE)
+ goto not_wav;
+
+ return TRUE;
+
+ /* ERRORS */
+not_wav:
+ {
+ GST_ELEMENT_ERROR (element, STREAM, WRONG_TYPE, (NULL),
+ ("File is not an WAVE file: %" GST_FOURCC_FORMAT,
+ GST_FOURCC_ARGS (doctype)));
+ return FALSE;
+ }
+}
+
+static GstFlowReturn
+gst_wavparse_stream_init (GstWavParse * wav)
+{
+ GstFlowReturn res;
+ GstBuffer *buf = NULL;
+
+ if ((res = gst_pad_pull_range (wav->sinkpad,
+ wav->offset, 12, &buf)) != GST_FLOW_OK)
+ return res;
+ else if (!gst_wavparse_parse_file_header (GST_ELEMENT (wav), buf))
+ return GST_FLOW_ERROR;
+
+ wav->offset += 12;
+
+ return GST_FLOW_OK;
+}
+
/* This function is used to perform seeks on the element in
* pull mode.
*
flush = flags & GST_SEEK_FLAG_FLUSH;
- if (flush) {
+ if (flush && wav->srcpad) {
GST_DEBUG_OBJECT (wav, "sending flush start");
gst_pad_push_event (wav->srcpad, gst_event_new_flush_start ());
} else {
GST_TIME_ARGS (seeksegment.start), GST_TIME_ARGS (stop));
/* prepare for streaming again */
- if (flush) {
- GST_DEBUG_OBJECT (wav, "sending flush stop");
- gst_pad_push_event (wav->srcpad, gst_event_new_flush_stop ());
- } else if (wav->segment_running) {
- /* we are running the current segment and doing a non-flushing seek,
- * close the segment first based on the last_stop. */
- GST_DEBUG_OBJECT (wav, "closing running segment %" G_GINT64_FORMAT
- " to %" G_GINT64_FORMAT, wav->segment.start, wav->segment.last_stop);
-
- gst_pad_push_event (wav->srcpad,
- gst_event_new_new_segment (TRUE,
- wav->segment.rate, wav->segment.format,
- wav->segment.start, wav->segment.last_stop, wav->segment.time));
+ if (wav->srcpad) {
+ if (flush) {
+ GST_DEBUG_OBJECT (wav, "sending flush stop");
+ gst_pad_push_event (wav->srcpad, gst_event_new_flush_stop ());
+ } else if (wav->segment_running) {
+ /* we are running the current segment and doing a non-flushing seek,
+ * close the segment first based on the last_stop. */
+ GST_DEBUG_OBJECT (wav, "closing running segment %" G_GINT64_FORMAT
+ " to %" G_GINT64_FORMAT, wav->segment.start, wav->segment.last_stop);
+
+ gst_pad_push_event (wav->srcpad,
+ gst_event_new_new_segment (TRUE,
+ wav->segment.rate, wav->segment.format,
+ wav->segment.start, wav->segment.last_stop, wav->segment.time));
+ }
}
memcpy (&wav->segment, &seeksegment, sizeof (GstSegment));
/* loop headers until we get data */
while (!gotdata) {
-
if (wav->streaming) {
if (!gst_wavparse_peek_chunk_info (wav, &tag, &size))
return GST_FLOW_OK;
#define MAX_BUFFER_SIZE 4096
static GstFlowReturn
-gst_wavparse_stream_data (GstWavParse * wav, gboolean first)
+gst_wavparse_stream_data (GstWavParse * wav)
{
GstBuffer *buf = NULL;
GstFlowReturn res = GST_FLOW_OK;
guint64 pos, nextpos;
iterate_adapter:
- GST_LOG_OBJECT (wav, "offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT,
- wav->offset, wav->end_offset);
+ GST_LOG_OBJECT (wav,
+ "offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT " , dataleft: %"
+ G_GINT64_FORMAT, wav->offset, wav->end_offset, wav->dataleft);
/* Get the next n bytes and output them */
if (wav->dataleft == 0 || wav->dataleft < wav->blockalign)
"from the sinkpad", desired);
if (wav->streaming) {
- if (gst_adapter_available (wav->adapter) < desired)
+ guint avail = gst_adapter_available (wav->adapter);
+
+ if (avail < desired) {
+ GST_LOG_OBJECT (wav, "Got only %d bytes of data from the sinkpad", avail);
return GST_FLOW_OK;
+ }
buf = gst_buffer_new ();
GST_BUFFER_DATA (buf) = gst_adapter_take (wav->adapter, desired);
goto pull_error;
}
- obtained = GST_BUFFER_SIZE (buf);
-
/* first chunk of data? create the source pad. We do this only here so
* we can detect broken .wav files with dts disguised as raw PCM (sigh) */
- if (first) {
+ if (G_UNLIKELY (wav->first)) {
+ wav->first = FALSE;
gst_wavparse_add_src_pad (wav, buf);
}
+ obtained = GST_BUFFER_SIZE (buf);
+
/* our positions */
pos = wav->offset - wav->datastart;
nextpos = pos + obtained;
if ((res = gst_pad_push (wav->srcpad, buf)) != GST_FLOW_OK)
goto push_error;
} else {
- GST_DEBUG ("Srcpad not linked!");
gst_buffer_unref (buf);
+ goto not_linked;
}
if (obtained < wav->dataleft) {
wav->dataleft -= obtained;
- wav->offset += obtained;
+ //wav->offset += obtained;
} else {
wav->dataleft = 0;
}
+ wav->offset += obtained;
/* Iterate until need more data, so adapter size won't grow */
if (wav->streaming) {
GST_LOG_OBJECT (wav,
GST_DEBUG_OBJECT (wav, "Error pushing on srcpad");
return res;
}
+not_linked:
+ {
+ GST_DEBUG_OBJECT (wav, "Srcpad not linked!");
+ return GST_FLOW_ERROR;
+ }
}
static void
GstFlowReturn ret;
GstWavParse *wav = GST_WAVPARSE (GST_PAD_PARENT (pad));
+ GST_LOG_OBJECT (wav, "process data");
+
switch (wav->state) {
case GST_WAVPARSE_START:
+ GST_DEBUG_OBJECT (wav, "GST_WAVPARSE_START");
if ((ret = gst_wavparse_stream_init (wav)) != GST_FLOW_OK)
goto pause;
/* fall-through */
case GST_WAVPARSE_HEADER:
+ GST_DEBUG_OBJECT (wav, "GST_WAVPARSE_HEADER");
if ((ret = gst_wavparse_stream_headers (wav)) != GST_FLOW_OK)
goto pause;
wav->state = GST_WAVPARSE_DATA;
- if ((ret = gst_wavparse_stream_data (wav, TRUE)) != GST_FLOW_OK)
- goto pause;
- break;
+ /* fall-through */
+
case GST_WAVPARSE_DATA:
- if ((ret = gst_wavparse_stream_data (wav, FALSE)) != GST_FLOW_OK)
+ if ((ret = gst_wavparse_stream_data (wav)) != GST_FLOW_OK)
goto pause;
break;
default:
GstFlowReturn ret;
GstWavParse *wav = GST_WAVPARSE (GST_PAD_PARENT (pad));
+ GST_LOG_OBJECT (wav, "adapter_push %" G_GINT64_FORMAT " bytes",
+ GST_BUFFER_SIZE (buf));
+
gst_adapter_push (wav->adapter, buf);
switch (wav->state) {
case GST_WAVPARSE_START:
+ GST_DEBUG_OBJECT (wav, "GST_WAVPARSE_START");
if ((ret = gst_wavparse_parse_stream_init (wav)) != GST_FLOW_OK)
goto pause;
+
+ wav->state = GST_WAVPARSE_HEADER;
/* fall-through */
case GST_WAVPARSE_HEADER:
+ GST_DEBUG_OBJECT (wav, "GST_WAVPARSE_HEADER");
if ((ret = gst_wavparse_stream_headers (wav)) != GST_FLOW_OK)
goto pause;
wav->state = GST_WAVPARSE_DATA;
- if ((ret = gst_wavparse_stream_data (wav, TRUE)) != GST_FLOW_OK)
- goto pause;
- break;
+ /* fall-through */
+
case GST_WAVPARSE_DATA:
- if ((ret = gst_wavparse_stream_data (wav, FALSE)) != GST_FLOW_OK)
+ if ((ret = gst_wavparse_stream_data (wav)) != GST_FLOW_OK)
goto pause;
break;
default:
{
gst_riff_init ();
+ GST_DEBUG_CATEGORY_INIT (wavparse_debug, "wavparse", 0, "WAV parser");
+
return gst_element_register (plugin, "wavparse", GST_RANK_PRIMARY,
GST_TYPE_WAVPARSE);
}
projects / platform / upstream / gst-plugins-good.git / commitdiff