Commit 61a021fb authored by Stéphane Loeuillet's avatar Stéphane Loeuillet

ext/mad/gstid3tag.c: move from "Codec/(Dem/M)uxer" to "Codec/(Dem/M)uxer/Audio"

Original commit message from CVS:
* ext/mad/gstid3tag.c : move from "Codec/(Dem/M)uxer" to "Codec/(Dem/M)uxer/Audio"
* gst/wavenc/gstwavenc.c : move from "Codec/Encoder/Audio" to "Codec/Muxer/Audio"

* gst/auparse/gstauparse.c :
- add code (commented for now) to support audio/x-adpcm on src pad
(we have no decoder for those layout yet)

* gst/cdxaparse/gstcdxaparse.c :
* gst/cdxaparse/gstcdxaparse.h :
- partial rewrite using RiffRead (ripped iain's wavparse code)

* gst/rtp/gstrtpL16enc.c : typo
* gst/rtp/gstrtpgsmenc.c : typo
parent 59f3c16b
2004-05-13 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
* ext/mad/gstid3tag.c : move from "Codec/(Dem/M)uxer" to "Codec/(Dem/M)uxer/Audio"
* gst/wavenc/gstwavenc.c : move from "Codec/Encoder/Audio" to "Codec/Muxer/Audio"
* gst/auparse/gstauparse.c :
- add code (commented for now) to support audio/x-adpcm on src pad
(we have no decoder for those layout yet)
* gst/cdxaparse/gstcdxaparse.c :
* gst/cdxaparse/gstcdxaparse.h :
- partial rewrite using RiffRead (ripped iain's wavparse code)
* gst/rtp/gstrtpL16enc.c : typo
* gst/rtp/gstrtpgsmenc.c : typo
2004-05-13 Benjamin Otte <otte@gnome.org>
* configure.ac:
......
......@@ -51,11 +51,10 @@ static GstStaticPadTemplate gst_auparse_src_template =
GST_PAD_SOMETIMES, /* FIXME: spider */
GST_STATIC_CAPS (GST_AUDIO_INT_PAD_TEMPLATE_CAPS "; " /* 24-bit PCM is barely supported by gstreamer actually */
GST_AUDIO_FLOAT_PAD_TEMPLATE_CAPS "; " /* 64-bit float is barely supported by gstreamer actually */
"audio/x-alaw, "
"rate = (int) [ 8000, 192000 ], "
"channels = (int) [ 1, 2 ]; "
"audio/x-mulaw, "
"rate = (int) [ 8000, 192000 ], " "channels = (int) [ 1, 2 ]")
"audio/x-alaw, " "rate = (int) [ 8000, 192000 ], " "channels = (int) [ 1, 2 ]; " "audio/x-mulaw, " "rate = (int) [ 8000, 192000 ], " "channels = (int) [ 1, 2 ]" /*"; "
"audio/x-adpcm, "
"layout = (string) { g721, g722, g723_3, g723_5 }" */ )
/* Nothing to decode those ADPCM streams for now */
);
/* AuParse signals and args */
......@@ -314,6 +313,11 @@ Samples :
"width", G_TYPE_INT, depth,
"endianness", G_TYPE_INT,
auparse->le ? G_LITTLE_ENDIAN : G_BIG_ENDIAN, NULL);
/*
} else if (layout) {
tempcaps = gst_caps_new_simple ("audio/x-adpcm",
"layout", G_TYPE_STRING, layout, NULL);
*/
} else {
tempcaps = gst_caps_new_simple ("audio/x-raw-int",
"endianness", G_TYPE_INT,
......
......@@ -28,7 +28,7 @@
static GstElementDetails gst_rtpL16enc_details = {
"RTP RAW Audio Encoder",
"Codec/Encoder/Network",
"Encodes Raw Audio into an RTP packet",
"Encodes Raw Audio into a RTP packet",
"Zeeshan Ali <zak147@yahoo.com>"
};
......
......@@ -28,7 +28,7 @@
static GstElementDetails gst_rtpL16enc_details = {
"RTP RAW Audio Encoder",
"Codec/Encoder/Network",
"Encodes Raw Audio into an RTP packet",
"Encodes Raw Audio into a RTP packet",
"Zeeshan Ali <zak147@yahoo.com>"
};
......
......@@ -29,7 +29,7 @@
static GstElementDetails gst_rtpgsmenc_details = {
"RTP GSM Audio Encoder",
"Codec/Encoder/Network",
"Encodes GSM audio into an RTP packet",
"Encodes GSM audio into a RTP packet",
"Zeeshan Ali <zak147@yahoo.com>"
};
......
......@@ -29,7 +29,7 @@
static GstElementDetails gst_rtpgsmenc_details = {
"RTP GSM Audio Encoder",
"Codec/Encoder/Network",
"Encodes GSM audio into an RTP packet",
"Encodes GSM audio into a RTP packet",
"Zeeshan Ali <zak147@yahoo.com>"
};
......
......@@ -75,7 +75,7 @@ struct wave_header
static GstElementDetails gst_wavenc_details =
GST_ELEMENT_DETAILS ("WAV encoder",
"Codec/Encoder/Audio",
"Codec/Muxer/Audio",
"Encode raw audio into WAV",
"Iain Holmes <iain@prettypeople.org>");
......
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