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gst/mpegaudioparse/gstmpegaudioparse.*: Put timestamps on buffers.

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Original commit message from CVS:
* gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_init),
(gst_mp3parse_sink_event), (gst_mp3parse_chain):
* gst/mpegaudioparse/gstmpegaudioparse.h:
Put timestamps on buffers.
This commit is contained in:
Wim Taymans 2005-09-26 14:31:53 +00:00
parent a1c1d43036
commit edd3ce1d23
4 changed files with 68 additions and 9 deletions
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7
ChangeLog
View file

@ -1,3 +1,10 @@
2005-09-26 Wim Taymans <wim@fluendo.com>
* gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_init),
(gst_mp3parse_sink_event), (gst_mp3parse_chain):
* gst/mpegaudioparse/gstmpegaudioparse.h:
Put timestamps on buffers.
2005-09-21 Flavio Oliveira <flavio.oliveira@indt.org.br>
* ext/amrnb/amrnbenc.c: (gst_amrnbenc_base_init):

2
common

@ -1 +1 @@
Subproject commit cd4da6a319d9f92d28f7b8a3b412577e6de50b64
Subproject commit 7caeee4b949b4388927fec7fcf25f767429bde30

66
gst/mpegaudioparse/gstmpegaudioparse.c
View file

@ -67,6 +67,7 @@ static void gst_mp3parse_class_init (GstMPEGAudioParseClass * klass);
static void gst_mp3parse_base_init (GstMPEGAudioParseClass * klass);
static void gst_mp3parse_init (GstMPEGAudioParse * mp3parse);
static gboolean gst_mp3parse_sink_event (GstPad * pad, GstEvent * event);
static GstFlowReturn gst_mp3parse_chain (GstPad * pad, GstBuffer * buffer);
static int head_check (unsigned long head);
@ -261,6 +262,7 @@ gst_mp3parse_init (GstMPEGAudioParse * mp3parse)
mp3parse->sinkpad =
gst_pad_new_from_template (gst_static_pad_template_get
(&mp3_sink_template), "sink");
gst_pad_set_event_function (mp3parse->sinkpad, gst_mp3parse_sink_event);
gst_pad_set_chain_function (mp3parse->sinkpad, gst_mp3parse_chain);
gst_element_add_pad (GST_ELEMENT (mp3parse), mp3parse->sinkpad);
@ -278,6 +280,38 @@ gst_mp3parse_init (GstMPEGAudioParse * mp3parse)
mp3parse->rate = mp3parse->channels = mp3parse->layer = -1;
}
static gboolean
gst_mp3parse_sink_event (GstPad * pad, GstEvent * event)
{
gboolean res;
GstMPEGAudioParse *mp3parse;
mp3parse = GST_MP3PARSE (gst_pad_get_parent (pad));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_NEWSEGMENT:
{
GstFormat format;
gst_event_parse_newsegment (event, NULL, &format, NULL, NULL, NULL);
if (format != GST_FORMAT_TIME)
mp3parse->last_ts = 0;
else
/* we will be receiving timestamps */
mp3parse->last_ts = -1;
break;
}
default:
break;
}
res = gst_pad_push_event (mp3parse->srcpad, event);
gst_object_unref (mp3parse);
return res;
}
/* FIXME, use adapter */
static GstFlowReturn
gst_mp3parse_chain (GstPad * pad, GstBuffer * buf)
@ -288,13 +322,16 @@ gst_mp3parse_chain (GstPad * pad, GstBuffer * buf)
guint32 header;
int bpf;
GstBuffer *outbuf;
guint64 last_ts;
GstClockTime timestamp;
mp3parse = GST_MP3PARSE (gst_pad_get_parent (pad));
GST_DEBUG ("mp3parse: received buffer of %d bytes", GST_BUFFER_SIZE (buf));
last_ts = GST_BUFFER_TIMESTAMP (buf);
timestamp = GST_BUFFER_TIMESTAMP (buf);
if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
mp3parse->last_ts = timestamp;
}
/* if we have something left from the previous frame */
if (mp3parse->partialbuf) {
@ -383,8 +420,9 @@ gst_mp3parse_chain (GstPad * pad, GstBuffer * buf)
if (channels != mp3parse->channels ||
rate != mp3parse->rate ||
layer != mp3parse->layer || bitrate != mp3parse->bit_rate) {
GstCaps *caps = mp3_caps_create (layer, channels, bitrate, rate);
GstCaps *caps;
caps = mp3_caps_create (layer, channels, bitrate, rate);
gst_pad_set_caps (mp3parse->srcpad, caps);
gst_caps_unref (caps);
@ -398,15 +436,27 @@ gst_mp3parse_chain (GstPad * pad, GstBuffer * buf)
offset += bpf;
if (mp3parse->skip == 0) {
gint spf; /* samples fer frame */
GST_DEBUG ("mp3parse: pushing buffer of %d bytes",
GST_BUFFER_SIZE (outbuf));
GST_BUFFER_TIMESTAMP (outbuf) = last_ts;
if (mp3parse->layer == 1) {
GST_BUFFER_DURATION (outbuf) = 384 * GST_SECOND / mp3parse->rate;
} else {
GST_BUFFER_DURATION (outbuf) = 1152 * GST_SECOND / mp3parse->rate;
GST_BUFFER_TIMESTAMP (outbuf) = mp3parse->last_ts;
/* see http://www.codeproject.com/audio/MPEGAudioInfo.asp */
if (mp3parse->layer == 1)
spf = 384;
else if (mp3parse->layer == 2)
spf = 1152;
else {
if (mp3parse->rate < 32100)
spf = 576;
else
spf = 1152;
}
GST_BUFFER_DURATION (outbuf) = spf * GST_SECOND / mp3parse->rate;
mp3parse->last_ts += GST_BUFFER_DURATION (outbuf);
gst_buffer_set_caps (outbuf, GST_PAD_CAPS (pad));

2
gst/mpegaudioparse/gstmpegaudioparse.h
View file

@ -45,6 +45,8 @@ struct _GstMPEGAudioParse {
GstPad *sinkpad,*srcpad;
guint64 last_ts;
GstBuffer *partialbuf; /* previous buffer (if carryover) */
guint skip; /* number of frames to skip */
guint bit_rate;