[0.11] gst-plugins-base: gst: Add new layout field to the raw audio caps
Sebastian Dröge
slomo at kemper.freedesktop.org
Thu Jan 5 01:36:01 PST 2012
Module: gst-plugins-base
Branch: 0.11
Commit: 5bdf6b33838e10be1c23ce5d7d9b38f814f6b8a9
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-base/commit/?id=5bdf6b33838e10be1c23ce5d7d9b38f814f6b8a9
Author: Sebastian Dröge <sebastian.droege at collabora.co.uk>
Date: Sat Dec 31 14:21:27 2011 +0100
gst: Add new layout field to the raw audio caps
---
gst/audioconvert/gstaudioconvert.c | 3 ++-
gst/audioresample/gstaudioresample.c | 6 ++++--
gst/audiotestsrc/gstaudiotestsrc.c | 1 +
3 files changed, 7 insertions(+), 3 deletions(-)
diff --git a/gst/audioconvert/gstaudioconvert.c b/gst/audioconvert/gstaudioconvert.c
index dcb139d..f23e3ab 100644
--- a/gst/audioconvert/gstaudioconvert.c
+++ b/gst/audioconvert/gstaudioconvert.c
@@ -116,7 +116,8 @@ G_DEFINE_TYPE_WITH_CODE (GstAudioConvert, gst_audio_convert,
/*** GSTREAMER PROTOTYPES *****************************************************/
#define STATIC_CAPS \
-GST_STATIC_CAPS (GST_AUDIO_CAPS_MAKE (GST_AUDIO_FORMATS_ALL))
+GST_STATIC_CAPS (GST_AUDIO_CAPS_MAKE (GST_AUDIO_FORMATS_ALL) \
+ ", layout = (string) interleaved")
static GstStaticPadTemplate gst_audio_convert_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
diff --git a/gst/audioresample/gstaudioresample.c b/gst/audioresample/gstaudioresample.c
index a74acc4..0e04371 100644
--- a/gst/audioresample/gstaudioresample.c
+++ b/gst/audioresample/gstaudioresample.c
@@ -67,10 +67,12 @@ enum
#if G_BYTE_ORDER == G_LITTLE_ENDIAN
#define SUPPORTED_CAPS \
- GST_AUDIO_CAPS_MAKE ("{ F32LE, F64LE, S32LE, S24LE, S16LE, S8 }")
+ GST_AUDIO_CAPS_MAKE ("{ F32LE, F64LE, S32LE, S24LE, S16LE, S8 }") \
+ ", layout = (string) { interleaved, non-interleaved }"
#else
#define SUPPORTED_CAPS \
- GST_AUDIO_CAPS_MAKE ("{ F32BE, F64BE, S32BE, S24BE, S16BE, S8 }")
+ GST_AUDIO_CAPS_MAKE ("{ F32BE, F64BE, S32BE, S24BE, S16BE, S8 }") \
+ ", layout = (string) { interleaved, non-interleaved }"
#endif
/* If TRUE integer arithmetic resampling is faster and will be used if appropriate */
diff --git a/gst/audiotestsrc/gstaudiotestsrc.c b/gst/audiotestsrc/gstaudiotestsrc.c
index 6a59884..2a5a474 100644
--- a/gst/audiotestsrc/gstaudiotestsrc.c
+++ b/gst/audiotestsrc/gstaudiotestsrc.c
@@ -89,6 +89,7 @@ GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) " FORMAT_STR ", "
+ "layout = (string) interleaved, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]")
);
More information about the gstreamer-commits
mailing list