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| author | Max Kellermann <max@duempel.org> | 2008-10-10 14:40:54 +0200 |
|---|---|---|
| committer | Max Kellermann <max@duempel.org> | 2008-10-10 14:40:54 +0200 |
| commit | de2cb3f37568e7680549057f8d7b6d748c388480 (patch) | |
| tree | 46f9f43a1f83b49945c8a4fc77f933fad9230e01 /src/inputPlugins/audiofile_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/audiofile_plugin.c | 8 |
1 files changed, 4 insertions, 4 deletions
diff --git a/src/inputPlugins/audiofile_plugin.c b/src/inputPlugins/audiofile_plugin.c index 421cdf354..4c08074c4 100644 --- a/src/inputPlugins/audiofile_plugin.c +++ b/src/inputPlugins/audiofile_plugin.c @@ -71,14 +71,14 @@ static int audiofile_decode(struct decoder * decoder, char *path) AF_SAMPFMT_TWOSCOMP, 16); afGetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits); audio_format.bits = (uint8_t)bits; - audio_format.sampleRate = + audio_format.sample_rate = (unsigned int)afGetRate(af_fp, AF_DEFAULT_TRACK); audio_format.channels = (uint8_t)afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK); frame_count = afGetFrameCount(af_fp, AF_DEFAULT_TRACK); - total_time = ((float)frame_count / (float)audio_format.sampleRate); + total_time = ((float)frame_count / (float)audio_format.sample_rate); bitRate = (uint16_t)(st.st_size * 8.0 / total_time / 1000.0 + 0.5); @@ -97,7 +97,7 @@ static int audiofile_decode(struct decoder * decoder, char *path) if (decoder_get_command(decoder) == DECODE_COMMAND_SEEK) { decoder_clear(decoder); current = decoder_seek_where(decoder) * - audio_format.sampleRate; + audio_format.sample_rate; afSeekFrame(af_fp, AF_DEFAULT_TRACK, current); decoder_command_finished(decoder); } @@ -110,7 +110,7 @@ static int audiofile_decode(struct decoder * decoder, char *path) current += ret; decoder_data(decoder, NULL, 1, chunk, ret * fs, - (float)current / (float)audio_format.sampleRate, + (float)current / (float)audio_format.sample_rate, bitRate, NULL); } while (decoder_get_command(decoder) != DECODE_COMMAND_STOP); |