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author | Max Kellermann <max@duempel.org> | 2008-10-10 14:40:54 +0200 |
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committer | Max Kellermann <max@duempel.org> | 2008-10-10 14:40:54 +0200 |
commit | de2cb3f37568e7680549057f8d7b6d748c388480 (patch) | |
tree | 46f9f43a1f83b49945c8a4fc77f933fad9230e01 /src/inputPlugins/mp4_plugin.c | |
parent | 6101dc6c768b09dbcdc1840a84b619a5a6a20129 (diff) | |
download | mpd-de2cb3f37568e7680549057f8d7b6d748c388480.tar.gz mpd-de2cb3f37568e7680549057f8d7b6d748c388480.tar.xz mpd-de2cb3f37568e7680549057f8d7b6d748c388480.zip |
audio_format: renamed sampleRate to sample_rate
The last bit of CamelCase in audio_format.h. Additionally, rename a
bunch of local variables.
Diffstat (limited to '')
-rw-r--r-- | src/inputPlugins/mp4_plugin.c | 8 |
1 files changed, 4 insertions, 4 deletions
diff --git a/src/inputPlugins/mp4_plugin.c b/src/inputPlugins/mp4_plugin.c index 6a2d167b2..d284313d4 100644 --- a/src/inputPlugins/mp4_plugin.c +++ b/src/inputPlugins/mp4_plugin.c @@ -92,7 +92,7 @@ static int mp4_decode(struct decoder * mpd_decoder, InputStream * inStream) struct audio_format audio_format; unsigned char *mp4Buffer; unsigned int mp4BufferSize; - uint32_t sampleRate; + uint32_t sample_rate; unsigned char channels; long sampleId; long numSamples; @@ -149,7 +149,7 @@ static int mp4_decode(struct decoder * mpd_decoder, InputStream * inStream) mp4ff_get_decoder_config(mp4fh, track, &mp4Buffer, &mp4BufferSize); if (faacDecInit2 - (decoder, mp4Buffer, mp4BufferSize, &sampleRate, &channels) < 0) { + (decoder, mp4Buffer, mp4BufferSize, &sample_rate, &channels) < 0) { ERROR("Error not a AAC stream.\n"); faacDecClose(decoder); mp4ff_close(mp4fh); @@ -157,7 +157,7 @@ static int mp4_decode(struct decoder * mpd_decoder, InputStream * inStream) return -1; } - audio_format.sampleRate = sampleRate; + audio_format.sample_rate = sample_rate; audio_format.channels = channels; file_time = mp4ff_get_track_duration_use_offsets(mp4fh, track); scale = mp4ff_time_scale(mp4fh, track); @@ -255,7 +255,7 @@ static int mp4_decode(struct decoder * mpd_decoder, InputStream * inStream) #ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE scale = frameInfo.samplerate; #endif - audio_format.sampleRate = scale; + audio_format.sample_rate = scale; audio_format.channels = frameInfo.channels; decoder_initialized(mpd_decoder, &audio_format, total_time); |