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author | Max Kellermann <max@duempel.org> | 2008-04-12 04:20:30 +0000 |
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committer | Eric Wong <normalperson@yhbt.net> | 2008-04-12 04:20:30 +0000 |
commit | 00fbd7bb4b64d6078b79c1e35d5b1d3bb325a405 (patch) | |
tree | 29c1994070ed34076dd4db29f071927f2716d7ae /src/inputPlugins | |
parent | 3db2b987183d0c325825ce70f0f1c343d22839d4 (diff) | |
download | mpd-00fbd7bb4b64d6078b79c1e35d5b1d3bb325a405.tar.gz mpd-00fbd7bb4b64d6078b79c1e35d5b1d3bb325a405.tar.xz mpd-00fbd7bb4b64d6078b79c1e35d5b1d3bb325a405.zip |
special optimized case for 16bit stereo
Not having to loop for every sample byte (depending on a variable
unknown at compile time) saves a lot of CPU cycles. We could consider
reimplementing this function with liboil...
git-svn-id: https://svn.musicpd.org/mpd/trunk@7330 09075e82-0dd4-0310-85a5-a0d7c8717e4f
Diffstat (limited to '')
-rw-r--r-- | src/inputPlugins/flac_plugin.c | 23 |
1 files changed, 20 insertions, 3 deletions
diff --git a/src/inputPlugins/flac_plugin.c b/src/inputPlugins/flac_plugin.c index 8f4c96884..b52704959 100644 --- a/src/inputPlugins/flac_plugin.c +++ b/src/inputPlugins/flac_plugin.c @@ -206,6 +206,18 @@ static void flacMetadata(const flac_decoder * dec, flac_metadata_common_cb(block, (FlacData *) vdata); } +static void flac_convert_stereo16(unsigned char *dest, + const FLAC__int32 * const buf[], + unsigned int position, unsigned int end) +{ + for (; position < end; ++position) { + *(uint16_t*)dest = buf[0][position]; + dest += 2; + *(uint16_t*)dest = buf[1][position]; + dest += 2; + } +} + static void flac_convert(unsigned char *dest, unsigned int num_channels, unsigned int bytes_per_sample, @@ -266,9 +278,14 @@ static FLAC__StreamDecoderWriteStatus flacWrite(const flac_decoder *dec, if (num_samples > max_samples) num_samples = max_samples; - flac_convert(data->chunk + data->chunk_length, - num_channels, bytes_per_sample, buf, - c_samp, c_samp + num_samples); + if (bytes_per_sample == 2) + flac_convert_stereo16(data->chunk + data->chunk_length, + buf, c_samp, + c_samp + num_samples); + else + flac_convert(data->chunk + data->chunk_length, + num_channels, bytes_per_sample, buf, + c_samp, c_samp + num_samples); data->chunk_length = num_samples; if (flacSendChunk(data) < 0) { |