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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/wavpack_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 'src/inputPlugins/wavpack_plugin.c')
-rw-r--r-- | src/inputPlugins/wavpack_plugin.c | 9 |
1 files changed, 4 insertions, 5 deletions
diff --git a/src/inputPlugins/wavpack_plugin.c b/src/inputPlugins/wavpack_plugin.c index af7c3a2f3..3e99980bd 100644 --- a/src/inputPlugins/wavpack_plugin.c +++ b/src/inputPlugins/wavpack_plugin.c @@ -140,7 +140,7 @@ static void wavpack_decode(struct decoder * decoder, int position, outsamplesize; int Bps; - audio_format.sampleRate = WavpackGetSampleRate(wpc); + audio_format.sample_rate = WavpackGetSampleRate(wpc); audio_format.channels = WavpackGetReducedChannels(wpc); audio_format.bits = WavpackGetBitsPerSample(wpc); @@ -168,7 +168,7 @@ static void wavpack_decode(struct decoder * decoder, samplesreq = sizeof(chunk) / (4 * audio_format.channels); decoder_initialized(decoder, &audio_format, - (float)allsamples / audio_format.sampleRate); + (float)allsamples / audio_format.sample_rate); position = 0; @@ -180,7 +180,7 @@ static void wavpack_decode(struct decoder * decoder, decoder_clear(decoder); where = decoder_seek_where(decoder) * - audio_format.sampleRate; + audio_format.sample_rate; if (WavpackSeekSample(wpc, where)) { position = where; decoder_command_finished(decoder); @@ -200,8 +200,7 @@ static void wavpack_decode(struct decoder * decoder, int bitrate = (int)(WavpackGetInstantBitrate(wpc) / 1000 + 0.5); position += samplesgot; - file_time = (float)position / - audio_format.sampleRate; + file_time = (float)position / audio_format.sample_rate; format_samples(Bps, chunk, samplesgot * audio_format.channels); |