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| author | Piotr Fusik <fox@scene.pl> | 2011-09-13 21:10:43 +0200 |
|---|---|---|
| committer | Stefan Sauer <ensonic@users.sf.net> | 2011-12-02 12:03:27 +0100 |
| commit | 14644457b06f48b26f32f88ef91e1286a48ebe24 (patch) | |
| tree | 5216c737500ee2cc4e0032e0c841f6d64660c0cd /gst/audioresample | |
| parent | 0cce8ab97d614ef53970292bd403e7f4460d79f9 (diff) | |
| download | gstreamer-plugins-base-14644457b06f48b26f32f88ef91e1286a48ebe24.tar.gz | |
various: typo fixes
Fix typos in code and docs. Fixes. #658984
Diffstat (limited to 'gst/audioresample')
| -rw-r--r-- | gst/audioresample/gstaudioresample.c | 6 | ||||
| -rw-r--r-- | gst/audioresample/resample.c | 2 |
2 files changed, 4 insertions, 4 deletions
diff --git a/gst/audioresample/gstaudioresample.c b/gst/audioresample/gstaudioresample.c index 80988cb7f..418a77c95 100644 --- a/gst/audioresample/gstaudioresample.c +++ b/gst/audioresample/gstaudioresample.c @@ -103,7 +103,7 @@ GST_STATIC_CAPS ( \ "signed = (boolean) true" \ ) -/* If TRUE integer arithmetic resampling is faster and will be used if appropiate */ +/* If TRUE integer arithmetic resampling is faster and will be used if appropriate */ #if defined AUDIORESAMPLE_FORMAT_INT static gboolean gst_audio_resample_use_int = TRUE; #elif defined AUDIORESAMPLE_FORMAT_FLOAT @@ -187,7 +187,7 @@ gst_audio_resample_class_init (GstAudioResampleClass * klass) * * Length of the resample filter * - * Deprectated: Use #GstAudioResample:quality property instead + * Deprecated: Use #GstAudioResample:quality property instead */ g_object_class_install_property (gobject_class, PROP_FILTER_LENGTH, g_param_spec_int ("filter-length", "Filter length", @@ -1554,7 +1554,7 @@ _benchmark_integer_resampling (void) resample_int_resampler_destroy (stb); if (av > bv) - GST_INFO ("Using integer resampler if appropiate: %lf < %lf", bv, av); + GST_INFO ("Using integer resampler if appropriate: %lf < %lf", bv, av); else GST_INFO ("Using float resampler for everything: %lf <= %lf", av, bv); diff --git a/gst/audioresample/resample.c b/gst/audioresample/resample.c index 7cc04d66c..490eebcc4 100644 --- a/gst/audioresample/resample.c +++ b/gst/audioresample/resample.c @@ -461,7 +461,7 @@ resampler_basic_direct_single (SpeexResamplerState * st, sum += MULT16_16 (sinc[j], iptr[j]); /* This code is slower on most DSPs which have only 2 accumulators. - Plus this this forces truncation to 32 bits and you lose the HW guard bits. + Plus this forces truncation to 32 bits and you lose the HW guard bits. I think we can trust the compiler and let it vectorize and/or unroll itself. spx_word32_t accum[4] = {0,0,0,0}; for(j=0;j<N;j+=4) { |