mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-07-03 05:05:52 +00:00
Code Issues 1 Projects Releases Wiki Activity

ext/ffmpeg/gstffmpegaudioresample.c: small gst-indent run.

Browse source 
Original commit message from CVS:
* ext/ffmpeg/gstffmpegaudioresample.c:
(gst_ffmpegaudioresample_class_init),
(gst_ffmpegaudioresample_init),
(gst_ffmpegaudioresample_transform_caps),
(gst_ffmpegaudioresample_transform_size),
(gst_ffmpegaudioresample_get_unit_size),
(gst_ffmpegaudioresample_set_caps),
(gst_ffmpegaudioresample_transform):
small gst-indent run.
This commit is contained in:
Edward Hervey 2008-05-08 11:15:52 +00:00
parent 176b11e6ff
commit 4d5f8819ad
2 changed files with 59 additions and 44 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

12
ChangeLog
View file

@ -1,3 +1,15 @@
2008-05-08 Edward Hervey <edward.hervey@collabora.co.uk>
* ext/ffmpeg/gstffmpegaudioresample.c:
(gst_ffmpegaudioresample_class_init),
(gst_ffmpegaudioresample_init),
(gst_ffmpegaudioresample_transform_caps),
(gst_ffmpegaudioresample_transform_size),
(gst_ffmpegaudioresample_get_unit_size),
(gst_ffmpegaudioresample_set_caps),
(gst_ffmpegaudioresample_transform):
small gst-indent run.
2008-05-08 Edward Hervey <edward.hervey@collabora.co.uk>
* gst-libs/ext/Makefile.am:

91
ext/ffmpeg/gstffmpegaudioresample.c
View file

@ -68,31 +68,33 @@ typedef struct _GstFFMpegAudioResampleClass
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, endianness = (int) BYTE_ORDER, signed = (boolean) true, width = (int) 16, depth = (int) 16, channels = (int) { 1 , 2 }, rate = (int) [1, MAX ]")
GST_STATIC_CAPS
("audio/x-raw-int, endianness = (int) BYTE_ORDER, signed = (boolean) true, width = (int) 16, depth = (int) 16, channels = (int) { 1 , 2 }, rate = (int) [1, MAX ]")
);
static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, endianness = (int) BYTE_ORDER, signed = (boolean) true, width = (int) 16, depth = (int) 16, channels = (int) [ 1 , 6 ], rate = (int) [1, MAX ]")
GST_STATIC_CAPS
("audio/x-raw-int, endianness = (int) BYTE_ORDER, signed = (boolean) true, width = (int) 16, depth = (int) 16, channels = (int) [ 1 , 6 ], rate = (int) [1, MAX ]")
);
GST_BOILERPLATE (GstFFMpegAudioResample, gst_ffmpegaudioresample, GstBaseTransform,
GST_TYPE_BASE_TRANSFORM);
GST_BOILERPLATE (GstFFMpegAudioResample, gst_ffmpegaudioresample,
GstBaseTransform, GST_TYPE_BASE_TRANSFORM);
static void gst_ffmpegaudioresample_finalize (GObject * object);
static GstCaps *gst_ffmpegaudioresample_transform_caps (GstBaseTransform * trans,
GstPadDirection direction, GstCaps * caps);
static gboolean gst_ffmpegaudioresample_transform_size (GstBaseTransform * trans,
GstPadDirection direction, GstCaps * caps, guint size, GstCaps *othercaps,
guint * othersize);
static GstCaps *gst_ffmpegaudioresample_transform_caps (GstBaseTransform *
trans, GstPadDirection direction, GstCaps * caps);
static gboolean gst_ffmpegaudioresample_transform_size (GstBaseTransform *
trans, GstPadDirection direction, GstCaps * caps, guint size,
GstCaps * othercaps, guint * othersize);
static gboolean gst_ffmpegaudioresample_get_unit_size (GstBaseTransform * trans,
GstCaps * caps, guint * size);
static gboolean gst_ffmpegaudioresample_set_caps (GstBaseTransform * trans,
GstCaps * incaps, GstCaps * outcaps);
static GstFlowReturn gst_ffmpegaudioresample_transform (GstBaseTransform * trans,
GstBuffer * inbuf, GstBuffer * outbuf);
static GstFlowReturn gst_ffmpegaudioresample_transform (GstBaseTransform *
trans, GstBuffer * inbuf, GstBuffer * outbuf);
static void
gst_ffmpegaudioresample_base_init (gpointer g_class)
@ -125,14 +127,17 @@ gst_ffmpegaudioresample_class_init (GstFFMpegAudioResampleClass * klass)
trans_class->get_unit_size =
GST_DEBUG_FUNCPTR (gst_ffmpegaudioresample_get_unit_size);
trans_class->set_caps = GST_DEBUG_FUNCPTR (gst_ffmpegaudioresample_set_caps);
trans_class->transform = GST_DEBUG_FUNCPTR (gst_ffmpegaudioresample_transform);
trans_class->transform_size = GST_DEBUG_FUNCPTR (gst_ffmpegaudioresample_transform_size);
trans_class->transform =
GST_DEBUG_FUNCPTR (gst_ffmpegaudioresample_transform);
trans_class->transform_size =
GST_DEBUG_FUNCPTR (gst_ffmpegaudioresample_transform_size);
trans_class->passthrough_on_same_caps = TRUE;
}
static void
gst_ffmpegaudioresample_init (GstFFMpegAudioResample * resample, GstFFMpegAudioResampleClass * klass)
gst_ffmpegaudioresample_init (GstFFMpegAudioResample * resample,
GstFFMpegAudioResampleClass * klass)
{
GstBaseTransform *trans = GST_BASE_TRANSFORM (resample);
@ -157,21 +162,21 @@ gst_ffmpegaudioresample_transform_caps (GstBaseTransform * trans,
GstPadDirection direction, GstCaps * caps)
{
GstCaps *retcaps;
GstStructure * struc;
GstStructure *struc;
retcaps = gst_caps_copy (caps);
struc = gst_caps_get_structure (retcaps, 0);
gst_structure_set (struc, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT, NULL);
GST_LOG_OBJECT (trans, "returning caps %" GST_PTR_FORMAT,
retcaps);
GST_LOG_OBJECT (trans, "returning caps %" GST_PTR_FORMAT, retcaps);
return retcaps;
}
static gboolean gst_ffmpegaudioresample_transform_size (GstBaseTransform * trans,
GstPadDirection direction, GstCaps * caps, guint size, GstCaps *othercaps,
guint * othersize)
static gboolean
gst_ffmpegaudioresample_transform_size (GstBaseTransform * trans,
GstPadDirection direction, GstCaps * caps, guint size, GstCaps * othercaps,
guint * othersize)
{
gint inrate, outrate;
gint inchanns, outchanns;
@ -191,12 +196,10 @@ static gboolean gst_ffmpegaudioresample_transform_size (GstBaseTransform * trans
if (!ret)
return FALSE;
conv = gst_util_uint64_scale(size, outrate * outchanns,
inrate * inchanns);
conv = gst_util_uint64_scale (size, outrate * outchanns, inrate * inchanns);
*othersize = (guint) conv;
GST_DEBUG_OBJECT (trans, "Transformed size from %d to %d",
size, *othersize);
GST_DEBUG_OBJECT (trans, "Transformed size from %d to %d", size, *othersize);
return TRUE;
}
@ -206,7 +209,7 @@ gst_ffmpegaudioresample_get_unit_size (GstBaseTransform * trans, GstCaps * caps,
guint * size)
{
gint channels;
GstStructure * structure;
GstStructure *structure;
gboolean ret;
g_assert (size);
@ -228,24 +231,24 @@ gst_ffmpegaudioresample_set_caps (GstBaseTransform * trans, GstCaps * incaps,
GstStructure *instructure = gst_caps_get_structure (incaps, 0);
GstStructure *outstructure = gst_caps_get_structure (outcaps, 0);
GST_LOG_OBJECT (resample, "incaps:%"GST_PTR_FORMAT,
incaps);
GST_LOG_OBJECT (resample, "incaps:%" GST_PTR_FORMAT, incaps);
GST_LOG_OBJECT (resample, "outcaps:%"GST_PTR_FORMAT,
outcaps);
GST_LOG_OBJECT (resample, "outcaps:%" GST_PTR_FORMAT, outcaps);
if (!gst_structure_get_int (instructure, "channels", &resample->in_channels))
return FALSE;
if (!gst_structure_get_int (instructure, "rate", &resample->in_rate))
return FALSE;
if (!gst_structure_get_int (outstructure, "channels", &resample->out_channels))
if (!gst_structure_get_int (outstructure, "channels",
&resample->out_channels))
return FALSE;
if (!gst_structure_get_int (outstructure, "rate", &resample->out_rate))
return FALSE;
resample->res = audio_resample_init (resample->out_channels, resample->in_channels,
resample->out_rate, resample->in_rate);
resample->res =
audio_resample_init (resample->out_channels, resample->in_channels,
resample->out_rate, resample->in_rate);
if (resample->res == NULL)
return FALSE;
@ -263,25 +266,25 @@ gst_ffmpegaudioresample_transform (GstBaseTransform * trans, GstBuffer * inbuf,
gst_buffer_copy_metadata (outbuf, inbuf, GST_BUFFER_COPY_TIMESTAMPS);
nbsamples = GST_BUFFER_SIZE (inbuf) / (2 * resample->in_channels);
GST_LOG_OBJECT (resample, "input buffer duration:%"GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_DURATION (inbuf)));
GST_LOG_OBJECT (resample, "input buffer duration:%" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_DURATION (inbuf)));
GST_DEBUG_OBJECT (resample, "audio_resample(ctx, output:%p [size:%d], input:%p [size:%d], nbsamples:%d",
GST_BUFFER_DATA (outbuf), GST_BUFFER_SIZE (outbuf),
GST_BUFFER_DATA (inbuf), GST_BUFFER_SIZE (inbuf),
nbsamples);
GST_DEBUG_OBJECT (resample,
"audio_resample(ctx, output:%p [size:%d], input:%p [size:%d], nbsamples:%d",
GST_BUFFER_DATA (outbuf), GST_BUFFER_SIZE (outbuf),
GST_BUFFER_DATA (inbuf), GST_BUFFER_SIZE (inbuf), nbsamples);
ret = audio_resample (resample->res, (short *) GST_BUFFER_DATA(outbuf),
(short *) GST_BUFFER_DATA (inbuf), nbsamples);
ret = audio_resample (resample->res, (short *) GST_BUFFER_DATA (outbuf),
(short *) GST_BUFFER_DATA (inbuf), nbsamples);
GST_DEBUG_OBJECT (resample, "audio_resample returned %d", ret);
GST_BUFFER_DURATION(outbuf) = gst_util_uint64_scale (ret, GST_SECOND,
resample->out_rate);
GST_BUFFER_DURATION (outbuf) = gst_util_uint64_scale (ret, GST_SECOND,
resample->out_rate);
GST_BUFFER_SIZE (outbuf) = ret * 2 * resample->out_channels;
GST_LOG_OBJECT (resample, "Output buffer duration:%"GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)));
GST_LOG_OBJECT (resample, "Output buffer duration:%" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)));
return GST_FLOW_OK;
}