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| author | Tim-Philipp Müller <tim.muller@collabora.co.uk> | 2011-12-02 11:10:17 +0000 |
|---|---|---|
| committer | Tim-Philipp Müller <tim.muller@collabora.co.uk> | 2011-12-02 11:10:17 +0000 |
| commit | 177525f89f6a93d12c7d69a39e198e888140fb5b (patch) | |
| tree | bad83715dd58a2a318983114871915824bf8c123 /gst/audioresample | |
| parent | ec0d3566bf15b6daa18f0df01b044b7cecfd9e45 (diff) | |
| parent | 14644457b06f48b26f32f88ef91e1286a48ebe24 (diff) | |
| download | gstreamer-plugins-base-177525f89f6a93d12c7d69a39e198e888140fb5b.tar.gz | |
Merge remote-tracking branch 'origin/master' into 0.11
Conflicts:
gst-libs/gst/netbuffer/gstnetbuffer.c
gst/ffmpegcolorspace/avcodec.h
gst/ffmpegcolorspace/gstffmpegcodecmap.c
gst/ffmpegcolorspace/imgconvert.c
gst/ffmpegcolorspace/imgconvert_template.h
gst/ffmpegcolorspace/mem.c
gst/playback/README
gst/playback/gstplaybasebin.c
gst/playback/gstplaybasebin.h
gst/playback/gstplaybin.c
sys/v4l/v4lmjpegsrc_calls.c
sys/v4l/videodev_mjpeg.h
tests/check/elements/gnomevfssink.c
Diffstat (limited to 'gst/audioresample')
| -rw-r--r-- | gst/audioresample/gstaudioresample.c | 4 | ||||
| -rw-r--r-- | gst/audioresample/resample.c | 2 |
2 files changed, 3 insertions, 3 deletions
diff --git a/gst/audioresample/gstaudioresample.c b/gst/audioresample/gstaudioresample.c index 6def60157..67cc05d69 100644 --- a/gst/audioresample/gstaudioresample.c +++ b/gst/audioresample/gstaudioresample.c @@ -73,7 +73,7 @@ enum GST_AUDIO_CAPS_MAKE ("{ F32BE, F64BE, S32BE, S24BE, S16BE, S8 }") #endif -/* 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 @@ -1395,7 +1395,7 @@ _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 66c2c6dc5..fefa0c536 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) { |