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author | Max Kellermann <max@duempel.org> | 2008-04-12 04:20:13 +0000 |
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committer | Eric Wong <normalperson@yhbt.net> | 2008-04-12 04:20:13 +0000 |
commit | ca1090f93b10c87a5ae3d7234688699569af5b01 (patch) | |
tree | f5ceb0319e3393acf606ebd53c1410cdf3f03236 /src/inputPlugins | |
parent | 0673c9a84dbbca60ed4627154e824dbfdeed269b (diff) | |
download | mpd-ca1090f93b10c87a5ae3d7234688699569af5b01.tar.gz mpd-ca1090f93b10c87a5ae3d7234688699569af5b01.tar.xz mpd-ca1090f93b10c87a5ae3d7234688699569af5b01.zip |
convert blocks until the buffer is full
Move the inner loop which converts samples to flac_convert(). There
it is isolated and easier to optimize. This function does not have to
worry about buffer boundaries; the caller (i.e. flacWrite())
calculates how much is left and is responsible for flushing. That
saves a lot of superfluous range checks within the loop.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7327 09075e82-0dd4-0310-85a5-a0d7c8717e4f
Diffstat (limited to '')
-rw-r--r-- | src/inputPlugins/flac_plugin.c | 66 |
1 files changed, 43 insertions, 23 deletions
diff --git a/src/inputPlugins/flac_plugin.c b/src/inputPlugins/flac_plugin.c index 16f63dda0..8e2dbf6be 100644 --- a/src/inputPlugins/flac_plugin.c +++ b/src/inputPlugins/flac_plugin.c @@ -206,6 +206,27 @@ static void flacMetadata(const flac_decoder * dec, flac_metadata_common_cb(block, (FlacData *) vdata); } +static void flac_convert(unsigned char *dest, + const FLAC__Frame * frame, + unsigned int bytes_per_sample, + const FLAC__int32 * const buf[], + unsigned int position, unsigned int end) +{ + unsigned int c_chan, i; + FLAC__uint16 u16; + unsigned char *uc; + + for (; position < end; ++position) { + for (c_chan = 0; c_chan < frame->header.channels; c_chan++) { + u16 = buf[c_chan][position]; + uc = (unsigned char *)&u16; + for (i = 0; i < bytes_per_sample; i++) { + *dest++ = *uc++; + } + } + } +} + static FLAC__StreamDecoderWriteStatus flacWrite(const flac_decoder *dec, const FLAC__Frame * frame, const FLAC__int32 * const buf[], @@ -213,13 +234,11 @@ static FLAC__StreamDecoderWriteStatus flacWrite(const flac_decoder *dec, { FlacData *data = (FlacData *) vdata; FLAC__uint32 samples = frame->header.blocksize; - FLAC__uint16 u16; - unsigned char *uc; - unsigned int c_samp, c_chan; + unsigned int c_samp; const unsigned int bytes_per_sample = (data->dc->audioFormat.bits / 8); const unsigned int bytes_per_channel = bytes_per_sample * frame->header.channels; - unsigned int i; + unsigned int num_samples, max_samples; float timeChange; FLAC__uint64 newPosition = 0; @@ -239,26 +258,27 @@ static FLAC__StreamDecoderWriteStatus flacWrite(const flac_decoder *dec, } data->position = newPosition; - for (c_samp = 0; c_samp < frame->header.blocksize; c_samp++) { - if (data->chunk_length + bytes_per_channel >= FLAC_CHUNK_SIZE) { - if (flacSendChunk(data) < 0) { - return - FLAC__STREAM_DECODER_WRITE_STATUS_ABORT; - } - data->chunk_length = 0; - if (data->dc->seek) { - return - FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE; - } + for (c_samp = 0; c_samp < frame->header.blocksize; + c_samp += num_samples) { + num_samples = frame->header.blocksize - c_samp; + max_samples = (FLAC_CHUNK_SIZE - data->chunk_length) / + bytes_per_channel; + if (num_samples > max_samples) + num_samples = max_samples; + + flac_convert(data->chunk + data->chunk_length, + frame, bytes_per_sample, buf, + c_samp, c_samp + num_samples); + data->chunk_length += num_samples; + + if (flacSendChunk(data) < 0) { + return + FLAC__STREAM_DECODER_WRITE_STATUS_ABORT; } - - for (c_chan = 0; c_chan < frame->header.channels; - c_chan++) { - u16 = buf[c_chan][c_samp]; - uc = (unsigned char *)&u16; - for (i = 0; i < bytes_per_sample; i++) { - data->chunk[data->chunk_length++] = *(uc++); - } + data->chunk_length = 0; + if (data->dc->seek) { + return + FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE; } } |