From 4d5f8819ad6176fc29cca51ac618c4994b40ee92 Mon Sep 17 00:00:00 2001 From: Edward Hervey Date: Thu, 8 May 2008 11:15:52 +0000 Subject: [PATCH] ext/ffmpeg/gstffmpegaudioresample.c: small gst-indent run. 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. --- ChangeLog | 12 +++++ ext/ffmpeg/gstffmpegaudioresample.c | 91 +++++++++++++++++++------------------ 2 files changed, 59 insertions(+), 44 deletions(-) diff --git a/ChangeLog b/ChangeLog index cb1c9e8..52ae857 100644 --- a/ChangeLog +++ b/ChangeLog @@ -1,5 +1,17 @@ 2008-05-08 Edward Hervey + * 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 + * gst-libs/ext/Makefile.am: Use 'make clean' and not 'make dist-clean' for local cleanups. Fixes #519235 diff --git a/ext/ffmpeg/gstffmpegaudioresample.c b/ext/ffmpeg/gstffmpegaudioresample.c index 63017f4..fa750a5 100644 --- a/ext/ffmpeg/gstffmpegaudioresample.c +++ b/ext/ffmpeg/gstffmpegaudioresample.c @@ -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; } -- 2.7.4