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authorDiego Biurrun <diego@biurrun.de>2011-10-30 18:02:42 +0100
committerDiego Biurrun <diego@biurrun.de>2011-11-02 10:42:55 +0100
commit2f5df0b12caea699ba85efa1fdb54fd0b57b4cfd (patch)
tree861355be8a653081a7ea52b43597d5329d50ef5c /libavformat/dv.c
parent20566eb0f0e3b0f538bcf68fca42e67afaa0dec6 (diff)
Replace ffmpeg references with more accurate libav* references.
Diffstat (limited to 'libavformat/dv.c')
-rw-r--r--libavformat/dv.c2
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.