From 2f5df0b12caea699ba85efa1fdb54fd0b57b4cfd Mon Sep 17 00:00:00 2001 From: Diego Biurrun Date: Sun, 30 Oct 2011 18:02:42 +0100 Subject: Replace ffmpeg references with more accurate libav* references. --- libavformat/dv.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'libavformat/dv.c') 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. -- cgit v1.2.3