diff options
| -rw-r--r-- | ChangeLog | 16 | ||||
| -rw-r--r-- | gst/auparse/gstauparse.c | 14 | ||||
| -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 | ||||
| -rw-r--r-- | gst/wavenc/gstwavenc.c | 2 |
7 files changed, 30 insertions, 10 deletions
@@ -1,3 +1,19 @@ +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: diff --git a/gst/auparse/gstauparse.c b/gst/auparse/gstauparse.c index 5e16d3e4..f3dacd05 100644 --- a/gst/auparse/gstauparse.c +++ b/gst/auparse/gstauparse.c @@ -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, 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>" }; diff --git a/gst/wavenc/gstwavenc.c b/gst/wavenc/gstwavenc.c index 99e34ce1..7c3e9de8 100644 --- a/gst/wavenc/gstwavenc.c +++ b/gst/wavenc/gstwavenc.c @@ -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>"); |