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| author | Diego Biurrun <diego@biurrun.de> | 2011-10-30 18:02:42 +0100 |
|---|---|---|
| committer | Diego Biurrun <diego@biurrun.de> | 2011-11-02 10:42:55 +0100 |
| commit | 2f5df0b12caea699ba85efa1fdb54fd0b57b4cfd (patch) | |
| tree | 861355be8a653081a7ea52b43597d5329d50ef5c /libavformat/dv.c | |
| parent | 20566eb0f0e3b0f538bcf68fca42e67afaa0dec6 (diff) | |
| download | ffmpeg-2f5df0b12caea699ba85efa1fdb54fd0b57b4cfd.tar.gz | |
Replace ffmpeg references with more accurate libav* references.
Diffstat (limited to 'libavformat/dv.c')
| -rw-r--r-- | libavformat/dv.c | 2 |
1 files changed, 1 insertions, 1 deletions
diff --git a/libavformat/dv.c b/libavformat/dv.c index 2813bc3b67..21823752f5 100644 --- a/libavformat/dv.c +++ b/libavformat/dv.c @@ -96,7 +96,7 @@ static const uint8_t* dv_extract_pack(uint8_t* frame, enum dv_pack_type t) /* * There's a couple of assumptions being made here: * 1. By default we silence erroneous (0x8000/16bit 0x800/12bit) audio samples. - * We can pass them upwards when ffmpeg will be ready to deal with them. + * We can pass them upwards when libavcodec will be ready to deal with them. * 2. We don't do software emphasis. * 3. Audio is always returned as 16bit linear samples: 12bit nonlinear samples * are converted into 16bit linear ones. |