/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
+ * Copyright (C) <2006> Tim-Philipp Müller <tim centricular net>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* Boston, MA 02111-1307, USA.
*/
-/* 2001/04/03 - Updated parseau to use caps nego
- * Zaheer Merali <zaheer@grid9.net
+/**
+ * SECTION:element-auparse
+ *
+ * Parses .au files mostly originating from sun os based computers.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
+
#include <stdlib.h>
#include <string.h>
-#include <gstauparse.h>
+#include "gstauparse.h"
#include <gst/audio/audio.h>
-/* elementfactory information */
-static GstElementDetails gst_auparse_details = GST_ELEMENT_DETAILS (
- ".au parser",
- "Codec/Parser/Audio",
- "Parse an .au file into raw audio",
- "Erik Walthinsen <omega@cse.ogi.edu>"
-);
-
-static GstStaticPadTemplate gst_auparse_sink_template =
-GST_STATIC_PAD_TEMPLATE (
- "sink",
- GST_PAD_SINK,
- GST_PAD_ALWAYS,
- GST_STATIC_CAPS ( "audio/x-au" )
-);
-
-static GstStaticPadTemplate gst_auparse_src_template =
-GST_STATIC_PAD_TEMPLATE (
- "src",
- GST_PAD_SRC,
- GST_PAD_ALWAYS,
- GST_STATIC_CAPS (
- GST_AUDIO_INT_PAD_TEMPLATE_CAPS "; "
- "audio/x-alaw, "
- "rate = (int) [ 8000, 48000 ], "
- "channels = (int) [ 1, 2 ]"
- )
-);
-
-/* AuParse signals and args */
-enum {
- /* FILL ME */
- LAST_SIGNAL
-};
-
-enum {
- ARG_0,
- /* FILL ME */
-};
-
-static void gst_auparse_base_init (gpointer g_class);
-static void gst_auparse_class_init (GstAuParseClass *klass);
-static void gst_auparse_init (GstAuParse *auparse);
-
-static void gst_auparse_chain (GstPad *pad,GstData *_data);
-
-static GstElementClass *parent_class = NULL;
-/*static guint gst_auparse_signals[LAST_SIGNAL] = { 0 }; */
-
-GType
-gst_auparse_get_type (void)
-{
- static GType auparse_type = 0;
-
- if (!auparse_type) {
- static const GTypeInfo auparse_info = {
- sizeof(GstAuParseClass),
- gst_auparse_base_init,
- NULL,
- (GClassInitFunc) gst_auparse_class_init,
- NULL,
- NULL,
- sizeof(GstAuParse),
- 0,
- (GInstanceInitFunc) gst_auparse_init,
- };
- auparse_type = g_type_register_static (GST_TYPE_ELEMENT, "GstAuParse", &auparse_info, 0);
- }
- return auparse_type;
-}
+GST_DEBUG_CATEGORY_STATIC (auparse_debug);
+#define GST_CAT_DEFAULT (auparse_debug)
+
+static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/x-au")
+ );
+
+#define GST_AU_PARSE_RAW_PAD_TEMPLATE_CAPS \
+ "audio/x-raw, " \
+ "format= (string) { S8, S16LE, S16BE, S24LE, S24BE, " \
+ "S32LE, S32BE, F32LE, F32BE, " \
+ "F64LE, F64BE }, " \
+ "rate = (int) [ 8000, 192000 ], " \
+ "channels = (int) [ 1, 2 ]"
+
+#define GST_AU_PARSE_ALAW_PAD_TEMPLATE_CAPS \
+ "audio/x-alaw, " \
+ "rate = (int) [ 8000, 192000 ], " \
+ "channels = (int) [ 1, 2 ]"
+
+#define GST_AU_PARSE_MULAW_PAD_TEMPLATE_CAPS \
+ "audio/x-mulaw, " \
+ "rate = (int) [ 8000, 192000 ], " \
+ "channels = (int) [ 1, 2 ]"
+
+/* Nothing to decode those ADPCM streams for now */
+#define GST_AU_PARSE_ADPCM_PAD_TEMPLATE_CAPS \
+ "audio/x-adpcm, " \
+ "layout = (string) { g721, g722, g723_3, g723_5 }"
+
+static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS (GST_AU_PARSE_RAW_PAD_TEMPLATE_CAPS "; "
+ GST_AU_PARSE_ALAW_PAD_TEMPLATE_CAPS ";"
+ GST_AU_PARSE_MULAW_PAD_TEMPLATE_CAPS ";"
+ GST_AU_PARSE_ADPCM_PAD_TEMPLATE_CAPS));
+
+
+static void gst_au_parse_dispose (GObject * object);
+static GstFlowReturn gst_au_parse_chain (GstPad * pad, GstBuffer * buf);
+static GstStateChangeReturn gst_au_parse_change_state (GstElement * element,
+ GstStateChange transition);
+static void gst_au_parse_reset (GstAuParse * auparse);
+static gboolean gst_au_parse_src_query (GstPad * pad, GstQuery * query);
+static gboolean gst_au_parse_src_event (GstPad * pad, GstEvent * event);
+static gboolean gst_au_parse_sink_event (GstPad * pad, GstEvent * event);
+static gboolean gst_au_parse_src_convert (GstAuParse * auparse,
+ GstFormat src_format, gint64 srcval, GstFormat dest_format,
+ gint64 * destval);
+
+#define gst_au_parse_parent_class parent_class
+G_DEFINE_TYPE (GstAuParse, gst_au_parse, GST_TYPE_ELEMENT);
static void
-gst_auparse_base_init (gpointer g_class)
+gst_au_parse_class_init (GstAuParseClass * klass)
{
- GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
+ GObjectClass *gobject_class;
+ GstElementClass *gstelement_class;
+
+ GST_DEBUG_CATEGORY_INIT (auparse_debug, "auparse", 0, ".au parser");
+
+ gobject_class = (GObjectClass *) klass;
+ gstelement_class = (GstElementClass *) klass;
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&gst_auparse_sink_template));
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&gst_auparse_src_template));
- gst_element_class_set_details (element_class, &gst_auparse_details);
+ gobject_class->dispose = gst_au_parse_dispose;
+ gstelement_class->change_state =
+ GST_DEBUG_FUNCPTR (gst_au_parse_change_state);
+ gst_element_class_add_pad_template (gstelement_class,
+ gst_static_pad_template_get (&sink_template));
+ gst_element_class_add_pad_template (gstelement_class,
+ gst_static_pad_template_get (&src_template));
+ gst_element_class_set_details_simple (gstelement_class,
+ "AU audio demuxer",
+ "Codec/Demuxer/Audio",
+ "Parse an .au file into raw audio",
+ "Erik Walthinsen <omega@cse.ogi.edu>");
}
static void
-gst_auparse_class_init (GstAuParseClass *klass)
+gst_au_parse_init (GstAuParse * auparse)
{
- GstElementClass *gstelement_class;
+ auparse->sinkpad = gst_pad_new_from_static_template (&sink_template, "sink");
+ gst_pad_set_chain_function (auparse->sinkpad,
+ GST_DEBUG_FUNCPTR (gst_au_parse_chain));
+ gst_pad_set_event_function (auparse->sinkpad,
+ GST_DEBUG_FUNCPTR (gst_au_parse_sink_event));
+ gst_element_add_pad (GST_ELEMENT (auparse), auparse->sinkpad);
- gstelement_class = (GstElementClass*) klass;
+ auparse->srcpad = gst_pad_new_from_static_template (&src_template, "src");
+ gst_pad_set_query_function (auparse->srcpad,
+ GST_DEBUG_FUNCPTR (gst_au_parse_src_query));
+ gst_pad_set_event_function (auparse->srcpad,
+ GST_DEBUG_FUNCPTR (gst_au_parse_src_event));
+ gst_pad_use_fixed_caps (auparse->srcpad);
+ gst_element_add_pad (GST_ELEMENT (auparse), auparse->srcpad);
- parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
+ auparse->adapter = gst_adapter_new ();
+ gst_au_parse_reset (auparse);
}
-static void
-gst_auparse_init (GstAuParse *auparse)
+static void
+gst_au_parse_dispose (GObject * object)
{
- auparse->sinkpad = gst_pad_new_from_template (
- gst_static_pad_template_get (&gst_auparse_sink_template), "sink");
- gst_element_add_pad (GST_ELEMENT (auparse), auparse->sinkpad);
- gst_pad_set_chain_function (auparse->sinkpad, gst_auparse_chain);
+ GstAuParse *au = GST_AU_PARSE (object);
- auparse->srcpad = gst_pad_new_from_template (
- gst_static_pad_template_get (&gst_auparse_src_template), "src");
- gst_element_add_pad (GST_ELEMENT (auparse), auparse->srcpad);
+ if (au->adapter != NULL) {
+ g_object_unref (au->adapter);
+ au->adapter = NULL;
+ }
+ G_OBJECT_CLASS (parent_class)->dispose (object);
+}
+static void
+gst_au_parse_reset (GstAuParse * auparse)
+{
auparse->offset = 0;
- auparse->size = 0;
+ auparse->buffer_offset = 0;
auparse->encoding = 0;
- auparse->frequency = 0;
+ auparse->samplerate = 0;
auparse->channels = 0;
+
+ gst_adapter_clear (auparse->adapter);
+
+ /* gst_segment_init (&auparse->segment, GST_FORMAT_TIME); */
}
-static void
-gst_auparse_chain (GstPad *pad, GstData *_data)
+static void
+gst_au_parse_negotiate_srcpad (GstAuParse * auparse, GstCaps * new_caps)
+{
+ if (auparse->src_caps && gst_caps_is_equal (new_caps, auparse->src_caps)) {
+ GST_LOG_OBJECT (auparse, "same caps, nothing to do");
+ return;
+ }
+
+ gst_caps_replace (&auparse->src_caps, new_caps);
+ GST_DEBUG_OBJECT (auparse, "Changing src pad caps to %" GST_PTR_FORMAT,
+ auparse->src_caps);
+ gst_pad_set_caps (auparse->srcpad, auparse->src_caps);
+
+ return;
+}
+
+static GstFlowReturn
+gst_au_parse_parse_header (GstAuParse * auparse)
{
- GstBuffer *buf = GST_BUFFER (_data);
- GstAuParse *auparse;
- gchar *data;
- glong size;
GstCaps *tempcaps;
- gint law, depth;
- gboolean sign;
+ guint32 size;
+ guint8 *head;
+ gchar layout[7] = { 0, };
+ GstAudioFormat format = GST_AUDIO_FORMAT_UNKNOWN;
+ gint law = 0;
+ guint endianness;
+
+ head = (guint8 *) gst_adapter_map (auparse->adapter, 24);
+ g_assert (head != NULL);
+
+ GST_DEBUG_OBJECT (auparse, "[%c%c%c%c]", head[0], head[1], head[2], head[3]);
+
+ switch (GST_READ_UINT32_BE (head)) {
+ /* normal format is big endian (au is a Sparc format) */
+ case 0x2e736e64:{ /* ".snd" */
+ endianness = G_BIG_ENDIAN;
+ break;
+ }
+ /* and of course, someone had to invent a little endian
+ * version. Used by DEC systems. */
+ case 0x646e732e: /* dns. */
+ case 0x0064732e:{ /* other source say it is "dns." */
+ endianness = G_LITTLE_ENDIAN;
+ break;
+ }
+ default:{
+ goto unknown_header;
+ }
+ }
- g_return_if_fail (pad != NULL);
- g_return_if_fail (GST_IS_PAD (pad));
- g_return_if_fail (buf != NULL);
+ auparse->offset = GST_READ_UINT32_BE (head + 4);
+ /* Do not trust size, could be set to -1 : unknown
+ * otherwise: filesize = size + auparse->offset
+ */
+ size = GST_READ_UINT32_BE (head + 8);
+ auparse->encoding = GST_READ_UINT32_BE (head + 12);
+ auparse->samplerate = GST_READ_UINT32_BE (head + 16);
+ auparse->channels = GST_READ_UINT32_BE (head + 20);
+
+ if (auparse->samplerate < 8000 || auparse->samplerate > 192000)
+ goto unsupported_sample_rate;
+
+ if (auparse->channels < 1 || auparse->channels > 2)
+ goto unsupported_number_of_channels;
+
+ GST_DEBUG_OBJECT (auparse, "offset %" G_GINT64_FORMAT ", size %u, "
+ "encoding %u, frequency %u, channels %u", auparse->offset, size,
+ auparse->encoding, auparse->samplerate, auparse->channels);
+
+ /* Docs:
+ * http://www.opengroup.org/public/pubs/external/auformat.html
+ * http://astronomy.swin.edu.au/~pbourke/dataformats/au/
+ * Solaris headers : /usr/include/audio/au.h
+ * libsndfile : src/au.c
+ *
+ * Samples :
+ * http://www.tsp.ece.mcgill.ca/MMSP/Documents/AudioFormats/AU/Samples.html
+ */
+
+ switch (auparse->encoding) {
+ case 1: /* 8-bit ISDN mu-law G.711 */
+ law = 1;
+ break;
+ case 27: /* 8-bit ISDN A-law G.711 */
+ law = 2;
+ break;
+
+ case 2: /* 8-bit linear PCM, FIXME signed? */
+ format = GST_AUDIO_FORMAT_S8;
+ auparse->sample_size = auparse->channels;
+ break;
+ case 3: /* 16-bit linear PCM */
+ if (endianness == G_LITTLE_ENDIAN)
+ format = GST_AUDIO_FORMAT_S16LE;
+ else
+ format = GST_AUDIO_FORMAT_S16BE;
+ auparse->sample_size = auparse->channels * 2;
+ break;
+ case 4: /* 24-bit linear PCM */
+ if (endianness == G_LITTLE_ENDIAN)
+ format = GST_AUDIO_FORMAT_S24LE;
+ else
+ format = GST_AUDIO_FORMAT_S24BE;
+ auparse->sample_size = auparse->channels * 3;
+ break;
+ case 5: /* 32-bit linear PCM */
+ if (endianness == G_LITTLE_ENDIAN)
+ format = GST_AUDIO_FORMAT_S32LE;
+ else
+ format = GST_AUDIO_FORMAT_S32BE;
+ auparse->sample_size = auparse->channels * 4;
+ break;
+
+ case 6: /* 32-bit IEEE floating point */
+ if (endianness == G_LITTLE_ENDIAN)
+ format = GST_AUDIO_FORMAT_F32LE;
+ else
+ format = GST_AUDIO_FORMAT_F32BE;
+ auparse->sample_size = auparse->channels * 4;
+ break;
+ case 7: /* 64-bit IEEE floating point */
+ if (endianness == G_LITTLE_ENDIAN)
+ format = GST_AUDIO_FORMAT_F64LE;
+ else
+ format = GST_AUDIO_FORMAT_F64BE;
+ auparse->sample_size = auparse->channels * 8;
+ break;
+
+ case 23: /* 4-bit CCITT G.721 ADPCM 32kbps -> modplug/libsndfile (compressed 8-bit mu-law) */
+ strcpy (layout, "g721");
+ break;
+ case 24: /* 8-bit CCITT G.722 ADPCM -> rtp */
+ strcpy (layout, "g722");
+ break;
+ case 25: /* 3-bit CCITT G.723.3 ADPCM 24kbps -> rtp/xine/modplug/libsndfile */
+ strcpy (layout, "g723_3");
+ break;
+ case 26: /* 5-bit CCITT G.723.5 ADPCM 40kbps -> rtp/xine/modplug/libsndfile */
+ strcpy (layout, "g723_5");
+ break;
+
+ case 8: /* Fragmented sample data */
+ case 9: /* AU_ENCODING_NESTED */
+
+ case 10: /* DSP program */
+ case 11: /* DSP 8-bit fixed point */
+ case 12: /* DSP 16-bit fixed point */
+ case 13: /* DSP 24-bit fixed point */
+ case 14: /* DSP 32-bit fixed point */
+
+ case 16: /* AU_ENCODING_DISPLAY : non-audio display data */
+ case 17: /* AU_ENCODING_MULAW_SQUELCH */
+
+ case 18: /* 16-bit linear with emphasis */
+ case 19: /* 16-bit linear compressed (NeXT) */
+ case 20: /* 16-bit linear with emphasis and compression */
+
+ case 21: /* Music kit DSP commands */
+ case 22: /* Music kit DSP commands samples */
+
+ default:
+ goto unknown_format;
+ }
+
+ if (law) {
+ tempcaps =
+ gst_caps_new_simple ((law == 1) ? "audio/x-mulaw" : "audio/x-alaw",
+ "rate", G_TYPE_INT, auparse->samplerate,
+ "channels", G_TYPE_INT, auparse->channels, NULL);
+ auparse->sample_size = auparse->channels;
+ } else if (format != GST_AUDIO_FORMAT_UNKNOWN) {
+ tempcaps = gst_caps_new_simple ("audio/x-raw",
+ "format", G_TYPE_STRING, gst_audio_format_to_string (format),
+ "rate", G_TYPE_INT, auparse->samplerate,
+ "channels", G_TYPE_INT, auparse->channels, NULL);
+ } else if (layout[0]) {
+ tempcaps = gst_caps_new_simple ("audio/x-adpcm",
+ "layout", G_TYPE_STRING, layout, NULL);
+ auparse->sample_size = 0;
+ } else
+ goto unknown_format;
+
+ GST_DEBUG_OBJECT (auparse, "sample_size=%d", auparse->sample_size);
+
+ gst_au_parse_negotiate_srcpad (auparse, tempcaps);
+
+ GST_DEBUG_OBJECT (auparse, "offset=%" G_GINT64_FORMAT, auparse->offset);
+ gst_adapter_unmap (auparse->adapter, auparse->offset);
+
+ gst_caps_unref (tempcaps);
+ return GST_FLOW_OK;
+
+ /* ERRORS */
+unknown_header:
+ {
+ gst_adapter_unmap (auparse->adapter, 0);
+ GST_ELEMENT_ERROR (auparse, STREAM, WRONG_TYPE, (NULL), (NULL));
+ return GST_FLOW_ERROR;
+ }
+unsupported_sample_rate:
+ {
+ gst_adapter_unmap (auparse->adapter, 0);
+ GST_ELEMENT_ERROR (auparse, STREAM, FORMAT, (NULL),
+ ("Unsupported samplerate: %u", auparse->samplerate));
+ return GST_FLOW_ERROR;
+ }
+unsupported_number_of_channels:
+ {
+ gst_adapter_unmap (auparse->adapter, 0);
+ GST_ELEMENT_ERROR (auparse, STREAM, FORMAT, (NULL),
+ ("Unsupported number of channels: %u", auparse->channels));
+ return GST_FLOW_ERROR;
+ }
+unknown_format:
+ {
+ gst_adapter_unmap (auparse->adapter, 0);
+ GST_ELEMENT_ERROR (auparse, STREAM, FORMAT, (NULL),
+ ("Unsupported encoding: %u", auparse->encoding));
+ return GST_FLOW_ERROR;
+ }
+}
+
+#define AU_HEADER_SIZE 24
+
+static GstFlowReturn
+gst_au_parse_chain (GstPad * pad, GstBuffer * buf)
+{
+ GstFlowReturn ret = GST_FLOW_OK;
+ GstAuParse *auparse;
+ gint avail, sendnow = 0;
+ gint64 timestamp;
+ gint64 duration;
+ gint64 offset;
+ GstSegment segment;
- auparse = GST_AUPARSE (gst_pad_get_parent (pad));
-
- GST_DEBUG ("gst_auparse_chain: got buffer in '%s'",
- gst_element_get_name (GST_ELEMENT (auparse)));
+ auparse = GST_AU_PARSE (gst_pad_get_parent (pad));
- data = GST_BUFFER_DATA (buf);
- size = GST_BUFFER_SIZE (buf);
+ GST_LOG_OBJECT (auparse, "got buffer of size %u", gst_buffer_get_size (buf));
+
+ gst_adapter_push (auparse->adapter, buf);
+ buf = NULL;
/* if we haven't seen any data yet... */
- if (auparse->size == 0) {
- GstBuffer *newbuf;
- guint32 *head = (guint32 *)data;
-
- /* normal format is big endian (au is a Sparc format) */
- if (GUINT32_FROM_BE (*head) == 0x2e736e64) {
- head++;
- auparse->le = 0;
- auparse->offset = GUINT32_FROM_BE (*head);
- head++;
- auparse->size = GUINT32_FROM_BE (*head);
- head++;
- auparse->encoding = GUINT32_FROM_BE (*head);
- head++;
- auparse->frequency = GUINT32_FROM_BE (*head);
- head++;
- auparse->channels = GUINT32_FROM_BE (*head);
- head++;
-
- /* and of course, someone had to invent a little endian
- * version. Used by DEC systems. */
- } else if (GUINT32_FROM_LE (*head) == 0x0064732E) {
- auparse->le = 1;
- head++;
- auparse->le = 0;
- auparse->offset = GUINT32_FROM_LE (*head);
- head++;
- auparse->size = GUINT32_FROM_LE (*head);
- head++;
- auparse->encoding = GUINT32_FROM_LE (*head);
- head++;
- auparse->frequency = GUINT32_FROM_LE (*head);
- head++;
- auparse->channels = GUINT32_FROM_LE (*head);
- head++;
-
- } else {
- g_warning ("help, dunno what I'm looking at!\n");
- gst_buffer_unref(buf);
- return;
+ if (!gst_pad_has_current_caps (auparse->srcpad)) {
+ if (gst_adapter_available (auparse->adapter) < AU_HEADER_SIZE) {
+ GST_DEBUG_OBJECT (auparse, "need more data to parse header");
+ ret = GST_FLOW_OK;
+ goto out;
+ }
+
+ ret = gst_au_parse_parse_header (auparse);
+ if (ret != GST_FLOW_OK)
+ goto out;
+
+ gst_segment_init (&segment, GST_FORMAT_TIME);
+ gst_pad_push_event (auparse->srcpad, gst_event_new_segment (&segment));
+ }
+
+ avail = gst_adapter_available (auparse->adapter);
+
+ if (auparse->sample_size > 0) {
+ /* Ensure we push a buffer that's a multiple of the frame size downstream */
+ sendnow = avail - (avail % auparse->sample_size);
+ } else {
+ /* It's something non-trivial (such as ADPCM), we don't understand it, so
+ * just push downstream and assume it will know what to do with it */
+ sendnow = avail;
+ }
+
+ if (sendnow > 0) {
+ GstBuffer *outbuf;
+ gint64 pos;
+
+ outbuf = gst_adapter_take_buffer (auparse->adapter, sendnow);
+ outbuf = gst_buffer_make_writable (outbuf);
+
+ pos = auparse->buffer_offset - auparse->offset;
+ pos = MAX (pos, 0);
+
+ if (auparse->sample_size > 0 && auparse->samplerate > 0) {
+ gst_au_parse_src_convert (auparse, GST_FORMAT_BYTES, pos,
+ GST_FORMAT_DEFAULT, &offset);
+ gst_au_parse_src_convert (auparse, GST_FORMAT_BYTES, pos,
+ GST_FORMAT_TIME, ×tamp);
+ gst_au_parse_src_convert (auparse, GST_FORMAT_BYTES,
+ sendnow, GST_FORMAT_TIME, &duration);
+
+ GST_BUFFER_OFFSET (outbuf) = offset;
+ GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
+ GST_BUFFER_DURATION (outbuf) = duration;
}
- g_print ("offset %ld, size %ld, encoding %ld, frequency %ld, channels %ld\n",
- auparse->offset,auparse->size,auparse->encoding,
- auparse->frequency,auparse->channels);
- GST_DEBUG ("offset %ld, size %ld, encoding %ld, frequency %ld, channels %ld",
- auparse->offset,auparse->size,auparse->encoding,
- auparse->frequency,auparse->channels);
-
- switch (auparse->encoding) {
- case 1:
- law = 1;
- depth = 8;
- sign = FALSE;
- break;
- case 2:
- law = 0;
- depth = 8;
- sign = FALSE;
- break;
- case 3:
- law = 0;
- depth = 16;
- sign = TRUE;
- break;
- default:
- g_warning ("help!, dont know how to deal with this format yet\n");
- return;
+ auparse->buffer_offset += sendnow;
+
+ ret = gst_pad_push (auparse->srcpad, outbuf);
+ }
+
+out:
+
+ gst_object_unref (auparse);
+ return ret;
+}
+
+static gboolean
+gst_au_parse_src_convert (GstAuParse * auparse, GstFormat src_format,
+ gint64 srcval, GstFormat dest_format, gint64 * destval)
+{
+ gboolean ret = TRUE;
+ guint samplesize, rate;
+
+ if (dest_format == src_format) {
+ *destval = srcval;
+ return TRUE;
+ }
+
+ GST_OBJECT_LOCK (auparse);
+ samplesize = auparse->sample_size;
+ rate = auparse->samplerate;
+ GST_OBJECT_UNLOCK (auparse);
+
+ if (samplesize == 0 || rate == 0) {
+ GST_LOG_OBJECT (auparse, "cannot convert, sample_size or rate unknown");
+ return FALSE;
+ }
+
+ switch (src_format) {
+ case GST_FORMAT_BYTES:
+ srcval /= samplesize;
+ /* fallthrough */
+ case GST_FORMAT_DEFAULT:{
+ switch (dest_format) {
+ case GST_FORMAT_DEFAULT:
+ *destval = srcval;
+ break;
+ case GST_FORMAT_BYTES:
+ *destval = srcval * samplesize;
+ break;
+ case GST_FORMAT_TIME:
+ *destval = gst_util_uint64_scale_int (srcval, GST_SECOND, rate);
+ break;
+ default:
+ ret = FALSE;
+ break;
+ }
+ break;
+ }
+ case GST_FORMAT_TIME:{
+ switch (dest_format) {
+ case GST_FORMAT_BYTES:
+ *destval = samplesize *
+ gst_util_uint64_scale_int (srcval, rate, GST_SECOND);
+ break;
+ case GST_FORMAT_DEFAULT:
+ *destval = gst_util_uint64_scale_int (srcval, rate, GST_SECOND);
+ break;
+ default:
+ ret = FALSE;
+ break;
+ }
+ break;
}
+ default:{
+ ret = FALSE;
+ break;
+ }
+ }
+
+ if (!ret) {
+ GST_DEBUG_OBJECT (auparse, "could not convert from %s to %s format",
+ gst_format_get_name (src_format), gst_format_get_name (dest_format));
+ }
- if (law) {
- tempcaps = gst_caps_new_simple ("audio/x-alaw",
- "rate", G_TYPE_INT, auparse->frequency,
- "channels", G_TYPE_INT, auparse->channels, NULL);
- } else {
- tempcaps = gst_caps_new_simple ("audio/x-raw-int",
- "endianness", G_TYPE_INT, G_BIG_ENDIAN,
- "rate", G_TYPE_INT, auparse->frequency,
- "channels", G_TYPE_INT, auparse->channels,
- "depth", G_TYPE_INT, depth,
- "width", G_TYPE_INT, depth,
- "signed", G_TYPE_BOOLEAN, sign, NULL);
+ return ret;
+}
+
+static gboolean
+gst_au_parse_src_query (GstPad * pad, GstQuery * query)
+{
+ GstAuParse *auparse;
+ gboolean ret = FALSE;
+
+ auparse = GST_AU_PARSE (gst_pad_get_parent (pad));
+
+ switch (GST_QUERY_TYPE (query)) {
+ case GST_QUERY_DURATION:{
+ GstFormat format;
+ gint64 len, val;
+
+ gst_query_parse_duration (query, &format, NULL);
+ if (!gst_pad_query_peer_duration (auparse->sinkpad, GST_FORMAT_BYTES,
+ &len)) {
+ GST_DEBUG_OBJECT (auparse, "failed to query upstream length");
+ break;
+ }
+ GST_OBJECT_LOCK (auparse);
+ len -= auparse->offset;
+ GST_OBJECT_UNLOCK (auparse);
+
+ ret =
+ gst_au_parse_src_convert (auparse, GST_FORMAT_BYTES, len, format,
+ &val);
+
+ if (ret) {
+ gst_query_set_duration (query, format, val);
+ }
+ break;
+ }
+ case GST_QUERY_POSITION:{
+ GstFormat format;
+ gint64 pos, val;
+
+ gst_query_parse_position (query, &format, NULL);
+ if (!gst_pad_query_peer_position (auparse->sinkpad, GST_FORMAT_BYTES,
+ &pos)) {
+ GST_DEBUG_OBJECT (auparse, "failed to query upstream position");
+ break;
+ }
+ GST_OBJECT_LOCK (auparse);
+ pos -= auparse->offset;
+ GST_OBJECT_UNLOCK (auparse);
+
+ ret = gst_au_parse_src_convert (auparse, GST_FORMAT_BYTES, pos,
+ format, &val);
+
+ if (ret) {
+ gst_query_set_position (query, format, val);
+ }
+ break;
+ }
+ case GST_QUERY_SEEKING:{
+ GstFormat format;
+
+ gst_query_parse_seeking (query, &format, NULL, NULL, NULL);
+ /* FIXME: query duration in 'format'
+ gst_query_set_seeking (query, format, TRUE, 0, duration);
+ */
+ gst_query_set_seeking (query, format, TRUE, 0, GST_CLOCK_TIME_NONE);
+ ret = TRUE;
+ break;
}
+ default:
+ ret = gst_pad_query_default (pad, query);
+ break;
+ }
+
+ gst_object_unref (auparse);
+ return ret;
+}
- if (gst_pad_try_set_caps (auparse->srcpad, tempcaps) <= 0) {
- gst_buffer_unref (buf);
- gst_element_error (GST_ELEMENT (auparse), "could not set audio caps");
- return;
+static gboolean
+gst_au_parse_handle_seek (GstAuParse * auparse, GstEvent * event)
+{
+ GstSeekType start_type, stop_type;
+ GstSeekFlags flags;
+ GstFormat format;
+ gdouble rate;
+ gint64 start, stop;
+ gboolean res;
+
+ gst_event_parse_seek (event, &rate, &format, &flags, &start_type, &start,
+ &stop_type, &stop);
+
+ if (format != GST_FORMAT_TIME) {
+ GST_DEBUG_OBJECT (auparse, "only support seeks in TIME format");
+ return FALSE;
+ }
+
+ res = gst_au_parse_src_convert (auparse, GST_FORMAT_TIME, start,
+ GST_FORMAT_BYTES, &start);
+
+ if (stop > 0) {
+ res = gst_au_parse_src_convert (auparse, GST_FORMAT_TIME, stop,
+ GST_FORMAT_BYTES, &stop);
+ }
+
+ GST_INFO_OBJECT (auparse,
+ "seeking: %" G_GINT64_FORMAT " ... %" G_GINT64_FORMAT, start, stop);
+
+ event = gst_event_new_seek (rate, GST_FORMAT_BYTES, flags, start_type, start,
+ stop_type, stop);
+ res = gst_pad_push_event (auparse->sinkpad, event);
+ return res;
+}
+
+static gboolean
+gst_au_parse_sink_event (GstPad * pad, GstEvent * event)
+{
+ GstAuParse *auparse;
+ gboolean ret = TRUE;
+
+ auparse = GST_AU_PARSE (gst_pad_get_parent (pad));
+
+ switch (GST_EVENT_TYPE (event)) {
+ case GST_EVENT_CAPS:
+ {
+ /* discard, we'll come up with proper src caps */
+ gst_event_unref (event);
+ break;
+ }
+ case GST_EVENT_SEGMENT:
+ {
+ gint64 start, stop, offset = 0;
+ GstSegment segment;
+ GstEvent *new_event = NULL;
+
+ /* some debug output */
+ gst_event_copy_segment (event, &segment);
+ GST_DEBUG_OBJECT (auparse, "received newsegment %" GST_SEGMENT_FORMAT,
+ &segment);
+
+ start = segment.start;
+ stop = segment.stop;
+ if (auparse->sample_size > 0) {
+ if (start > 0) {
+ offset = start;
+ start -= auparse->offset;
+ start = MAX (start, 0);
+ }
+ if (stop > 0) {
+ stop -= auparse->offset;
+ stop = MAX (stop, 0);
+ }
+ gst_au_parse_src_convert (auparse, GST_FORMAT_BYTES, start,
+ GST_FORMAT_TIME, &start);
+ gst_au_parse_src_convert (auparse, GST_FORMAT_BYTES, stop,
+ GST_FORMAT_TIME, &stop);
+ }
+
+ GST_INFO_OBJECT (auparse,
+ "new segment: %" GST_TIME_FORMAT " ... %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
+
+ gst_segment_init (&segment, GST_FORMAT_TIME);
+ segment.start = segment.time = start;
+ segment.stop = stop;
+ new_event = gst_event_new_segment (&segment);
+
+ ret = gst_pad_push_event (auparse->srcpad, new_event);
+
+ auparse->buffer_offset = offset;
+
+ gst_event_unref (event);
+ break;
}
+ case GST_EVENT_EOS:
+ if (!auparse->srcpad) {
+ GST_ELEMENT_ERROR (auparse, STREAM, WRONG_TYPE,
+ ("No valid input found before end of stream"), (NULL));
+ }
+ /* fall-through */
+ default:
+ ret = gst_pad_event_default (pad, event);
+ break;
+ }
- newbuf = gst_buffer_new ();
- GST_BUFFER_DATA (newbuf) = (gpointer) malloc (size-(auparse->offset));
- memcpy (GST_BUFFER_DATA (newbuf), data+24, size-(auparse->offset));
- GST_BUFFER_SIZE (newbuf) = size-(auparse->offset);
+ gst_object_unref (auparse);
+ return ret;
+}
+
+static gboolean
+gst_au_parse_src_event (GstPad * pad, GstEvent * event)
+{
+ GstAuParse *auparse;
+ gboolean ret;
- gst_buffer_unref (buf);
+ auparse = GST_AU_PARSE (gst_pad_get_parent (pad));
- gst_pad_push (auparse->srcpad, GST_DATA (newbuf));
- return;
+ switch (GST_EVENT_TYPE (event)) {
+ case GST_EVENT_SEEK:
+ ret = gst_au_parse_handle_seek (auparse, event);
+ break;
+ default:
+ ret = gst_pad_event_default (pad, event);
+ break;
}
- gst_pad_push (auparse->srcpad, GST_DATA (buf));
+ gst_object_unref (auparse);
+ return ret;
}
+static GstStateChangeReturn
+gst_au_parse_change_state (GstElement * element, GstStateChange transition)
+{
+ GstAuParse *auparse = GST_AU_PARSE (element);
+ GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
+
+ ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+ if (ret == GST_STATE_CHANGE_FAILURE)
+ return ret;
+
+ switch (transition) {
+ case GST_STATE_CHANGE_PAUSED_TO_READY:
+ gst_au_parse_reset (auparse);
+ default:
+ break;
+ }
+
+ return ret;
+}
static gboolean
-plugin_init (GstPlugin *plugin)
+plugin_init (GstPlugin * plugin)
{
if (!gst_element_register (plugin, "auparse", GST_RANK_SECONDARY,
- GST_TYPE_AUPARSE)) {
+ GST_TYPE_AU_PARSE)) {
return FALSE;
}
return TRUE;
}
-GST_PLUGIN_DEFINE (
- GST_VERSION_MAJOR,
- GST_VERSION_MINOR,
- "auparse",
- "parses au streams",
- plugin_init,
- VERSION,
- "GPL",
- GST_PACKAGE,
- GST_ORIGIN
-)
-
+GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
+ GST_VERSION_MINOR,
+ "auparse",
+ "parses au streams", plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME,
+ GST_PACKAGE_ORIGIN)
projects / platform / upstream / gst-plugins-good.git / blobdiff