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Diffstat (limited to 'gst/mpegaudioparse/gstmpegaudioparse.c')
-rw-r--r--gst/mpegaudioparse/gstmpegaudioparse.c421
1 files changed, 218 insertions, 203 deletions
diff --git a/gst/mpegaudioparse/gstmpegaudioparse.c b/gst/mpegaudioparse/gstmpegaudioparse.c
index aeed48f1..1e77bf42 100644
--- a/gst/mpegaudioparse/gstmpegaudioparse.c
+++ b/gst/mpegaudioparse/gstmpegaudioparse.c
@@ -32,35 +32,30 @@ static GstElementDetails mp3parse_details = {
"Erik Walthinsen <omega@cse.ogi.edu>"
};
-static GstStaticPadTemplate mp3_src_template =
-GST_STATIC_PAD_TEMPLATE (
- "src",
+static GstStaticPadTemplate mp3_src_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/mpeg, "
- "mpegversion = (int) 1, "
- "layer = (int) [ 1, 3 ], "
- "rate = (int) [ 8000, 48000], "
- "channels = (int) [ 1, 2 ]")
-);
-
-static GstStaticPadTemplate mp3_sink_template =
-GST_STATIC_PAD_TEMPLATE (
- "sink",
+ "mpegversion = (int) 1, "
+ "layer = (int) [ 1, 3 ], "
+ "rate = (int) [ 8000, 48000], " "channels = (int) [ 1, 2 ]")
+ );
+
+static GstStaticPadTemplate mp3_sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
- GST_STATIC_CAPS ("audio/mpeg, "
- "mpegversion = (int) 1"
- )
-);
+ GST_STATIC_CAPS ("audio/mpeg, " "mpegversion = (int) 1")
+ );
/* GstMPEGAudioParse signals and args */
-enum {
+enum
+{
/* FILL ME */
LAST_SIGNAL
};
-enum {
+enum
+{
ARG_0,
ARG_SKIP,
ARG_BIT_RATE,
@@ -68,63 +63,64 @@ enum {
};
-static void gst_mp3parse_class_init (GstMPEGAudioParseClass *klass);
-static void gst_mp3parse_base_init (GstMPEGAudioParseClass *klass);
-static void gst_mp3parse_init (GstMPEGAudioParse *mp3parse);
+static void gst_mp3parse_class_init (GstMPEGAudioParseClass * klass);
+static void gst_mp3parse_base_init (GstMPEGAudioParseClass * klass);
+static void gst_mp3parse_init (GstMPEGAudioParse * mp3parse);
-static void gst_mp3parse_chain (GstPad *pad,GstData *_data);
-static long bpf_from_header (GstMPEGAudioParse *parse, unsigned long header);
-static int head_check (unsigned long head);
+static void gst_mp3parse_chain (GstPad * pad, GstData * _data);
+static long bpf_from_header (GstMPEGAudioParse * parse, unsigned long header);
+static int head_check (unsigned long head);
-static void gst_mp3parse_set_property (GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec);
-static void gst_mp3parse_get_property (GObject *object, guint prop_id, GValue *value, GParamSpec *pspec);
-static GstElementStateReturn
- gst_mp3parse_change_state (GstElement *element);
+static void gst_mp3parse_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec);
+static void gst_mp3parse_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec);
+static GstElementStateReturn gst_mp3parse_change_state (GstElement * element);
static GstElementClass *parent_class = NULL;
+
/*static guint gst_mp3parse_signals[LAST_SIGNAL] = { 0 }; */
GType
-gst_mp3parse_get_type(void) {
+gst_mp3parse_get_type (void)
+{
static GType mp3parse_type = 0;
if (!mp3parse_type) {
static const GTypeInfo mp3parse_info = {
- sizeof(GstMPEGAudioParseClass),
- (GBaseInitFunc)gst_mp3parse_base_init,
+ sizeof (GstMPEGAudioParseClass),
+ (GBaseInitFunc) gst_mp3parse_base_init,
NULL,
- (GClassInitFunc)gst_mp3parse_class_init,
+ (GClassInitFunc) gst_mp3parse_class_init,
NULL,
NULL,
- sizeof(GstMPEGAudioParse),
+ sizeof (GstMPEGAudioParse),
0,
- (GInstanceInitFunc)gst_mp3parse_init,
+ (GInstanceInitFunc) gst_mp3parse_init,
};
mp3parse_type = g_type_register_static (GST_TYPE_ELEMENT,
- "GstMPEGAudioParse",
- &mp3parse_info, 0);
+ "GstMPEGAudioParse", &mp3parse_info, 0);
}
return mp3parse_type;
}
static guint mp3types_bitrates[2][3][16] =
-{ { {0, 32, 64, 96, 128, 160, 192, 224, 256, 288, 320, 352, 384, 416, 448, },
- {0, 32, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384, },
- {0, 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, } },
- { {0, 32, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224, 256, },
- {0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160, },
- {0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160, } },
+ { {{0, 32, 64, 96, 128, 160, 192, 224, 256, 288, 320, 352, 384, 416, 448,},
+ {0, 32, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384,},
+ {0, 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320,}},
+{{0, 32, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224, 256,},
+ {0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160,},
+ {0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160,}},
};
-static guint mp3types_freqs[3][3] =
-{ {44100, 48000, 32000},
- {22050, 24000, 16000},
- {11025, 12000, 8000}};
+static guint mp3types_freqs[3][3] = { {44100, 48000, 32000},
+{22050, 24000, 16000},
+{11025, 12000, 8000}
+};
static inline guint
-mp3_type_frame_length_from_header (guint32 header, guint *put_layer,
- guint *put_channels, guint *put_bitrate,
- guint *put_samplerate)
+mp3_type_frame_length_from_header (guint32 header, guint * put_layer,
+ guint * put_channels, guint * put_bitrate, guint * put_samplerate)
{
guint length;
gulong mode, samplerate, bitrate, layer, channels, padding;
@@ -163,7 +159,7 @@ mp3_type_frame_length_from_header (guint32 header, guint *put_layer,
GST_DEBUG ("Calculated mp3 frame length of %u bytes", length);
GST_DEBUG ("samplerate = %lu, bitrate = %lu, layer = %lu, channels = %lu",
- samplerate, bitrate, layer, channels);
+ samplerate, bitrate, layer, channels);
if (put_layer)
*put_layer = layer;
@@ -204,8 +200,7 @@ mp3_type_frame_length_from_header (guint32 header, guint *put_layer,
#define GST_MP3_TYPEFIND_MIN_DATA (1440 * (GST_MP3_TYPEFIND_MIN_HEADERS + 1) - 1 + 3)
static GstCaps *
-mp3_caps_create (guint layer, guint channels,
- guint bitrate, guint samplerate)
+mp3_caps_create (guint layer, guint channels, guint bitrate, guint samplerate)
{
GstCaps *new;
@@ -216,15 +211,14 @@ mp3_caps_create (guint layer, guint channels,
new = gst_caps_new_simple ("audio/mpeg",
"mpegversion", G_TYPE_INT, 1,
- "layer", G_TYPE_INT, layer,
- "rate", G_TYPE_INT, samplerate,
- "channels", G_TYPE_INT, channels, NULL);
+ "layer", G_TYPE_INT, layer,
+ "rate", G_TYPE_INT, samplerate, "channels", G_TYPE_INT, channels, NULL);
return new;
}
static void
-gst_mp3parse_base_init (GstMPEGAudioParseClass *klass)
+gst_mp3parse_base_init (GstMPEGAudioParseClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
@@ -236,22 +230,18 @@ gst_mp3parse_base_init (GstMPEGAudioParseClass *klass)
}
static void
-gst_mp3parse_class_init (GstMPEGAudioParseClass *klass)
+gst_mp3parse_class_init (GstMPEGAudioParseClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
- gobject_class = (GObjectClass*)klass;
- gstelement_class = (GstElementClass*)klass;
+ gobject_class = (GObjectClass *) klass;
+ gstelement_class = (GstElementClass *) klass;
- g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_SKIP,
- g_param_spec_int("skip","skip","skip",
- G_MININT,G_MAXINT,0,G_PARAM_READWRITE)); /* CHECKME */
- g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_BIT_RATE,
- g_param_spec_int("bitrate","Bitrate","Bit Rate",
- G_MININT,G_MAXINT,0,G_PARAM_READABLE)); /* CHECKME */
+ g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_SKIP, g_param_spec_int ("skip", "skip", "skip", G_MININT, G_MAXINT, 0, G_PARAM_READWRITE)); /* CHECKME */
+ g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_BIT_RATE, g_param_spec_int ("bitrate", "Bitrate", "Bit Rate", G_MININT, G_MAXINT, 0, G_PARAM_READABLE)); /* CHECKME */
- parent_class = g_type_class_ref(GST_TYPE_ELEMENT);
+ parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
gobject_class->set_property = gst_mp3parse_set_property;
gobject_class->get_property = gst_mp3parse_get_property;
@@ -260,18 +250,20 @@ gst_mp3parse_class_init (GstMPEGAudioParseClass *klass)
}
static void
-gst_mp3parse_init (GstMPEGAudioParse *mp3parse)
+gst_mp3parse_init (GstMPEGAudioParse * mp3parse)
{
- mp3parse->sinkpad = gst_pad_new_from_template(
- gst_static_pad_template_get (&mp3_sink_template), "sink");
- gst_element_add_pad(GST_ELEMENT(mp3parse),mp3parse->sinkpad);
-
- gst_pad_set_chain_function(mp3parse->sinkpad,gst_mp3parse_chain);
- gst_element_set_loop_function (GST_ELEMENT(mp3parse),NULL);
-
- mp3parse->srcpad = gst_pad_new_from_template(
- gst_static_pad_template_get (&mp3_src_template), "src");
- gst_element_add_pad(GST_ELEMENT(mp3parse),mp3parse->srcpad);
+ mp3parse->sinkpad =
+ gst_pad_new_from_template (gst_static_pad_template_get
+ (&mp3_sink_template), "sink");
+ gst_element_add_pad (GST_ELEMENT (mp3parse), mp3parse->sinkpad);
+
+ gst_pad_set_chain_function (mp3parse->sinkpad, gst_mp3parse_chain);
+ gst_element_set_loop_function (GST_ELEMENT (mp3parse), NULL);
+
+ mp3parse->srcpad =
+ gst_pad_new_from_template (gst_static_pad_template_get
+ (&mp3_src_template), "src");
+ gst_element_add_pad (GST_ELEMENT (mp3parse), mp3parse->srcpad);
gst_pad_use_explicit_caps (mp3parse->srcpad);
/*gst_pad_set_type_id(mp3parse->srcpad, mp3frametype); */
@@ -283,71 +275,71 @@ gst_mp3parse_init (GstMPEGAudioParse *mp3parse)
}
static void
-gst_mp3parse_chain (GstPad *pad, GstData *_data)
+gst_mp3parse_chain (GstPad * pad, GstData * _data)
{
GstBuffer *buf = GST_BUFFER (_data);
GstMPEGAudioParse *mp3parse;
guchar *data;
- glong size,offset = 0;
+ glong size, offset = 0;
guint32 header;
int bpf;
GstBuffer *outbuf;
guint64 last_ts;
- g_return_if_fail(pad != NULL);
- g_return_if_fail(GST_IS_PAD(pad));
- g_return_if_fail(buf != NULL);
+ g_return_if_fail (pad != NULL);
+ g_return_if_fail (GST_IS_PAD (pad));
+ g_return_if_fail (buf != NULL);
/* g_return_if_fail(GST_IS_BUFFER(buf)); */
mp3parse = GST_MP3PARSE (gst_pad_get_parent (pad));
- GST_DEBUG ("mp3parse: received buffer of %d bytes",GST_BUFFER_SIZE(buf));
+ GST_DEBUG ("mp3parse: received buffer of %d bytes", GST_BUFFER_SIZE (buf));
- last_ts = GST_BUFFER_TIMESTAMP(buf);
+ last_ts = GST_BUFFER_TIMESTAMP (buf);
/* FIXME, do flush */
/*
- if (mp3parse->partialbuf) {
- gst_buffer_unref(mp3parse->partialbuf);
- mp3parse->partialbuf = NULL;
- }
- mp3parse->in_flush = TRUE;
- */
+ if (mp3parse->partialbuf) {
+ gst_buffer_unref(mp3parse->partialbuf);
+ mp3parse->partialbuf = NULL;
+ }
+ mp3parse->in_flush = TRUE;
+ */
/* if we have something left from the previous frame */
if (mp3parse->partialbuf) {
GstBuffer *newbuf;
- newbuf = gst_buffer_merge(mp3parse->partialbuf, buf);
+ newbuf = gst_buffer_merge (mp3parse->partialbuf, buf);
/* and the one we received.. */
- gst_buffer_unref(buf);
- gst_buffer_unref(mp3parse->partialbuf);
+ gst_buffer_unref (buf);
+ gst_buffer_unref (mp3parse->partialbuf);
mp3parse->partialbuf = newbuf;
- }
- else {
+ } else {
mp3parse->partialbuf = buf;
}
- size = GST_BUFFER_SIZE(mp3parse->partialbuf);
- data = GST_BUFFER_DATA(mp3parse->partialbuf);
+ size = GST_BUFFER_SIZE (mp3parse->partialbuf);
+ data = GST_BUFFER_DATA (mp3parse->partialbuf);
/* while we still have bytes left -4 for the header */
- while (offset < size-4) {
+ while (offset < size - 4) {
int skipped = 0;
- GST_DEBUG ("mp3parse: offset %ld, size %ld ",offset, size);
+ GST_DEBUG ("mp3parse: offset %ld, size %ld ", offset, size);
/* search for a possible start byte */
- for (;((data[offset] != 0xff) && (offset < size));offset++) skipped++;
+ for (; ((data[offset] != 0xff) && (offset < size)); offset++)
+ skipped++;
if (skipped && !mp3parse->in_flush) {
- GST_DEBUG ("mp3parse: **** now at %ld skipped %d bytes",offset,skipped);
+ GST_DEBUG ("mp3parse: **** now at %ld skipped %d bytes", offset, skipped);
}
/* construct the header word */
- header = GUINT32_FROM_BE(*((guint32 *)(data+offset)));
+ header = GUINT32_FROM_BE (*((guint32 *) (data + offset)));
/* if it's a valid header, go ahead and send off the frame */
- if (head_check(header)) {
+ if (head_check (header)) {
/* calculate the bpf of the frame */
- bpf = bpf_from_header(mp3parse, header);
+ bpf = bpf_from_header (mp3parse, header);
/********************************************************************************
* robust seek support
@@ -361,107 +353,116 @@ gst_mp3parse_chain (GstPad *pad, GstData *_data)
* from previous frames. In this case, seeking may be more complicated because
* the frames are not independently coded.
********************************************************************************/
- if ( mp3parse->in_flush ) {
- guint32 header2;
+ if (mp3parse->in_flush) {
+ guint32 header2;
- if ((size-offset)<(bpf+4)) { if (mp3parse->in_flush) break; } /* wait until we have the the entire current frame as well as the next frame header */
-
- header2 = GUINT32_FROM_BE(*((guint32 *)(data+offset+bpf)));
- GST_DEBUG ("mp3parse: header=%08X, header2=%08X, bpf=%d", (unsigned int)header, (unsigned int)header2, bpf );
+ if ((size - offset) < (bpf + 4)) {
+ if (mp3parse->in_flush)
+ break;
+ }
+ /* wait until we have the the entire current frame as well as the next frame header */
+ header2 = GUINT32_FROM_BE (*((guint32 *) (data + offset + bpf)));
+ GST_DEBUG ("mp3parse: header=%08X, header2=%08X, bpf=%d",
+ (unsigned int) header, (unsigned int) header2, bpf);
/* mask the bits which are allowed to differ between frames */
#define HDRMASK ~((0xF << 12) /* bitrate */ | \
(0x1 << 9) /* padding */ | \
- (0x3 << 4)) /*mode extension*/
-
- if ( (header2&HDRMASK) != (header&HDRMASK) ) { /* require 2 matching headers in a row */
- GST_DEBUG ("mp3parse: next header doesn't match (header=%08X, header2=%08X, bpf=%d)", (unsigned int)header, (unsigned int)header2, bpf );
- offset++; /* This frame is invalid. Start looking for a valid frame at the next position in the stream */
- continue;
- }
+ (0x3 << 4)) /*mode extension */
+
+ if ((header2 & HDRMASK) != (header & HDRMASK)) { /* require 2 matching headers in a row */
+ GST_DEBUG
+ ("mp3parse: next header doesn't match (header=%08X, header2=%08X, bpf=%d)",
+ (unsigned int) header, (unsigned int) header2, bpf);
+ offset++; /* This frame is invalid. Start looking for a valid frame at the next position in the stream */
+ continue;
+ }
}
/* if we don't have the whole frame... */
if ((size - offset) < bpf) {
- GST_DEBUG ("mp3parse: partial buffer needed %ld < %d ",(size-offset), bpf);
+ GST_DEBUG ("mp3parse: partial buffer needed %ld < %d ", (size - offset),
+ bpf);
break;
} else {
- guint bitrate, layer, rate, channels;
-
- if (!mp3_type_frame_length_from_header (header, &layer,
- &channels,
- &bitrate, &rate)) {
- g_error("Header failed internal error");
- }
- if (channels != mp3parse->channels ||
- rate != mp3parse->rate ||
- layer != mp3parse->layer ||
- bitrate != mp3parse->bit_rate) {
- GstCaps *caps = mp3_caps_create (layer, channels, bitrate, rate);
-
- gst_pad_set_explicit_caps(mp3parse->srcpad, caps);
-
- mp3parse->channels = channels;
- mp3parse->layer = layer;
- mp3parse->rate = rate;
- mp3parse->bit_rate = bitrate;
- }
-
- outbuf = gst_buffer_create_sub(mp3parse->partialbuf,offset,bpf);
-
- offset += bpf;
+ guint bitrate, layer, rate, channels;
+
+ if (!mp3_type_frame_length_from_header (header, &layer,
+ &channels, &bitrate, &rate)) {
+ g_error ("Header failed internal error");
+ }
+ if (channels != mp3parse->channels ||
+ rate != mp3parse->rate ||
+ layer != mp3parse->layer || bitrate != mp3parse->bit_rate) {
+ GstCaps *caps = mp3_caps_create (layer, channels, bitrate, rate);
+
+ gst_pad_set_explicit_caps (mp3parse->srcpad, caps);
+
+ mp3parse->channels = channels;
+ mp3parse->layer = layer;
+ mp3parse->rate = rate;
+ mp3parse->bit_rate = bitrate;
+ }
+
+ outbuf = gst_buffer_create_sub (mp3parse->partialbuf, offset, bpf);
+
+ offset += bpf;
if (mp3parse->skip == 0) {
- GST_DEBUG ("mp3parse: pushing buffer of %d bytes",GST_BUFFER_SIZE(outbuf));
+ GST_DEBUG ("mp3parse: pushing buffer of %d bytes",
+ GST_BUFFER_SIZE (outbuf));
if (mp3parse->in_flush) {
/* FIXME do some sort of flush event */
mp3parse->in_flush = FALSE;
}
- GST_BUFFER_TIMESTAMP(outbuf) = last_ts;
- GST_BUFFER_DURATION(outbuf) = 8 * (GST_SECOND/1000) * GST_BUFFER_SIZE(outbuf) / mp3parse->bit_rate;
-
- if (GST_PAD_CAPS (mp3parse->srcpad) != NULL) {
- gst_pad_push(mp3parse->srcpad,GST_DATA (outbuf));
- } else {
- GST_DEBUG ("No capsnego yet, delaying buffer push");
- gst_buffer_unref (outbuf);
- }
- }
- else {
- GST_DEBUG ("mp3parse: skipping buffer of %d bytes",GST_BUFFER_SIZE(outbuf));
- gst_buffer_unref(outbuf);
+ GST_BUFFER_TIMESTAMP (outbuf) = last_ts;
+ GST_BUFFER_DURATION (outbuf) =
+ 8 * (GST_SECOND / 1000) * GST_BUFFER_SIZE (outbuf) /
+ mp3parse->bit_rate;
+
+ if (GST_PAD_CAPS (mp3parse->srcpad) != NULL) {
+ gst_pad_push (mp3parse->srcpad, GST_DATA (outbuf));
+ } else {
+ GST_DEBUG ("No capsnego yet, delaying buffer push");
+ gst_buffer_unref (outbuf);
+ }
+ } else {
+ GST_DEBUG ("mp3parse: skipping buffer of %d bytes",
+ GST_BUFFER_SIZE (outbuf));
+ gst_buffer_unref (outbuf);
mp3parse->skip--;
}
}
} else {
offset++;
- if (!mp3parse->in_flush) GST_DEBUG ("mp3parse: *** wrong header, skipping byte (FIXME?)");
+ if (!mp3parse->in_flush)
+ GST_DEBUG ("mp3parse: *** wrong header, skipping byte (FIXME?)");
}
}
/* if we have processed this block and there are still */
/* bytes left not in a partial block, copy them over. */
- if (size-offset > 0) {
+ if (size - offset > 0) {
glong remainder = (size - offset);
- GST_DEBUG ("mp3parse: partial buffer needed %ld for trailing bytes",remainder);
- outbuf = gst_buffer_create_sub(mp3parse->partialbuf,offset,remainder);
- gst_buffer_unref(mp3parse->partialbuf);
+ GST_DEBUG ("mp3parse: partial buffer needed %ld for trailing bytes",
+ remainder);
+
+ outbuf = gst_buffer_create_sub (mp3parse->partialbuf, offset, remainder);
+ gst_buffer_unref (mp3parse->partialbuf);
mp3parse->partialbuf = outbuf;
- }
- else {
- gst_buffer_unref(mp3parse->partialbuf);
+ } else {
+ gst_buffer_unref (mp3parse->partialbuf);
mp3parse->partialbuf = NULL;
}
}
static long
-bpf_from_header (GstMPEGAudioParse *parse, unsigned long header)
+bpf_from_header (GstMPEGAudioParse * parse, unsigned long header)
{
guint bitrate, layer, rate, channels, length;
if (!(length = mp3_type_frame_length_from_header (header, &layer,
- &channels,
- &bitrate, &rate))) {
+ &channels, &bitrate, &rate))) {
return 0;
}
@@ -471,40 +472,57 @@ bpf_from_header (GstMPEGAudioParse *parse, unsigned long header)
static gboolean
head_check (unsigned long head)
{
- GST_DEBUG ("checking mp3 header 0x%08lx",head);
+ GST_DEBUG ("checking mp3 header 0x%08lx", head);
/* if it's not a valid sync */
if ((head & 0xffe00000) != 0xffe00000) {
- GST_DEBUG ("invalid sync");return FALSE; }
+ GST_DEBUG ("invalid sync");
+ return FALSE;
+ }
/* if it's an invalid MPEG version */
if (((head >> 19) & 3) == 0x1) {
- GST_DEBUG ("invalid MPEG version");return FALSE; }
+ GST_DEBUG ("invalid MPEG version");
+ return FALSE;
+ }
/* if it's an invalid layer */
if (!((head >> 17) & 3)) {
- GST_DEBUG ("invalid layer");return FALSE; }
+ GST_DEBUG ("invalid layer");
+ return FALSE;
+ }
/* if it's an invalid bitrate */
if (((head >> 12) & 0xf) == 0x0) {
- GST_DEBUG ("invalid bitrate");return FALSE; }
+ GST_DEBUG ("invalid bitrate");
+ return FALSE;
+ }
if (((head >> 12) & 0xf) == 0xf) {
- GST_DEBUG ("invalid bitrate");return FALSE; }
+ GST_DEBUG ("invalid bitrate");
+ return FALSE;
+ }
/* if it's an invalid samplerate */
if (((head >> 10) & 0x3) == 0x3) {
- GST_DEBUG ("invalid samplerate");return FALSE; }
- if ((head & 0xffff0000) == 0xfffe0000) {
- GST_DEBUG ("invalid sync");return FALSE; }
+ GST_DEBUG ("invalid samplerate");
+ return FALSE;
+ }
+ if ((head & 0xffff0000) == 0xfffe0000) {
+ GST_DEBUG ("invalid sync");
+ return FALSE;
+ }
if (head & 0x00000002) {
- GST_DEBUG ("invalid emphasis");return FALSE; }
+ GST_DEBUG ("invalid emphasis");
+ return FALSE;
+ }
return TRUE;
}
static void
-gst_mp3parse_set_property (GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec)
+gst_mp3parse_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec)
{
GstMPEGAudioParse *src;
/* it's not null if we got it, but it might not be ours */
- g_return_if_fail(GST_IS_MP3PARSE(object));
- src = GST_MP3PARSE(object);
+ g_return_if_fail (GST_IS_MP3PARSE (object));
+ src = GST_MP3PARSE (object);
switch (prop_id) {
case ARG_SKIP:
@@ -516,13 +534,14 @@ gst_mp3parse_set_property (GObject *object, guint prop_id, const GValue *value,
}
static void
-gst_mp3parse_get_property (GObject *object, guint prop_id, GValue *value, GParamSpec *pspec)
+gst_mp3parse_get_property (GObject * object, guint prop_id, GValue * value,
+ GParamSpec * pspec)
{
GstMPEGAudioParse *src;
/* it's not null if we got it, but it might not be ours */
- g_return_if_fail(GST_IS_MP3PARSE(object));
- src = GST_MP3PARSE(object);
+ g_return_if_fail (GST_IS_MP3PARSE (object));
+ src = GST_MP3PARSE (object);
switch (prop_id) {
case ARG_SKIP:
@@ -537,43 +556,39 @@ gst_mp3parse_get_property (GObject *object, guint prop_id, GValue *value, GParam
}
}
-static GstElementStateReturn
-gst_mp3parse_change_state (GstElement *element)
+static GstElementStateReturn
+gst_mp3parse_change_state (GstElement * element)
{
GstMPEGAudioParse *src;
- g_return_val_if_fail(GST_IS_MP3PARSE(element), GST_STATE_FAILURE);
- src = GST_MP3PARSE(element);
+ g_return_val_if_fail (GST_IS_MP3PARSE (element), GST_STATE_FAILURE);
+ src = GST_MP3PARSE (element);
switch (GST_STATE_TRANSITION (element)) {
case GST_STATE_PAUSED_TO_READY:
- src->channels = -1; src->rate = -1; src->layer = -1;
+ src->channels = -1;
+ src->rate = -1;
+ src->layer = -1;
break;
default:
break;
}
- if (GST_ELEMENT_CLASS(parent_class)->change_state)
- return GST_ELEMENT_CLASS(parent_class)->change_state(element);
+ if (GST_ELEMENT_CLASS (parent_class)->change_state)
+ return GST_ELEMENT_CLASS (parent_class)->change_state (element);
return GST_STATE_SUCCESS;
}
static gboolean
-plugin_init (GstPlugin *plugin)
+plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "mp3parse",
- GST_RANK_NONE, GST_TYPE_MP3PARSE);
+ GST_RANK_NONE, GST_TYPE_MP3PARSE);
}
-GST_PLUGIN_DEFINE (
- GST_VERSION_MAJOR,
- GST_VERSION_MINOR,
- "mpegaudioparse",
- "MPEG-1 layer 1/2/3 audio parser",
- plugin_init,
- VERSION,
- "LGPL",
- GST_PACKAGE,
- GST_ORIGIN
-)
+GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
+ GST_VERSION_MINOR,
+ "mpegaudioparse",
+ "MPEG-1 layer 1/2/3 audio parser",
+ plugin_init, VERSION, "LGPL", GST_PACKAGE, GST_ORIGIN)
generated by cgit v1.2.1 (git 2.18.0) at 2024-06-30 15:58:45 +0000