diff options
| author | Stéphane Loeuillet <gstreamer@leroutier.net> | 2004-05-13 21:27:14 +0000 |
|---|---|---|
| committer | Stéphane Loeuillet <gstreamer@leroutier.net> | 2004-05-13 21:27:14 +0000 |
| commit | 61a021fbba546c6fe5bc1faecfcf3be1c9f8cde9 (patch) | |
| tree | d257e239c3b356f38e57061670486bf5856dccc6 /gst/rtp | |
| parent | 59f3c16b0c1a42b056e58fd967d0c72b21ed52ac (diff) | |
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
Diffstat (limited to 'gst/rtp')
| -rw-r--r-- | gst/rtp/gstrtpL16enc.c | 2 | ||||
| -rw-r--r-- | gst/rtp/gstrtpL16pay.c | 2 | ||||
| -rw-r--r-- | gst/rtp/gstrtpgsmenc.c | 2 | ||||
| -rw-r--r-- | gst/rtp/gstrtpgsmpay.c | 2 |
4 files changed, 4 insertions, 4 deletions
diff --git a/gst/rtp/gstrtpL16enc.c b/gst/rtp/gstrtpL16enc.c index 373d2d68..1e3088e2 100644 --- a/gst/rtp/gstrtpL16enc.c +++ b/gst/rtp/gstrtpL16enc.c @@ -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>" }; diff --git a/gst/rtp/gstrtpL16pay.c b/gst/rtp/gstrtpL16pay.c index 373d2d68..1e3088e2 100644 --- a/gst/rtp/gstrtpL16pay.c +++ b/gst/rtp/gstrtpL16pay.c @@ -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>" }; diff --git a/gst/rtp/gstrtpgsmenc.c b/gst/rtp/gstrtpgsmenc.c index f95cc7e3..ae441e14 100644 --- a/gst/rtp/gstrtpgsmenc.c +++ b/gst/rtp/gstrtpgsmenc.c @@ -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>" }; diff --git a/gst/rtp/gstrtpgsmpay.c b/gst/rtp/gstrtpgsmpay.c index f95cc7e3..ae441e14 100644 --- a/gst/rtp/gstrtpgsmpay.c +++ b/gst/rtp/gstrtpgsmpay.c @@ -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>" }; |