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author | Max Kellermann <max@duempel.org> | 2008-09-23 23:59:55 +0200 |
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committer | Eric Wong <normalperson@yhbt.net> | 2008-09-29 01:54:51 -0700 |
commit | 09edfb67ec85096ec148852e673f370d39b05b9d (patch) | |
tree | 26dd25252ac07945e57607a96e27243e2a21fe79 /src/inputPlugins | |
parent | 3c2c329e8450f59ccafb6ab973948d5da05951e5 (diff) | |
download | mpd-09edfb67ec85096ec148852e673f370d39b05b9d.tar.gz mpd-09edfb67ec85096ec148852e673f370d39b05b9d.tar.xz mpd-09edfb67ec85096ec148852e673f370d39b05b9d.zip |
flac: removed generic sample size support
Support for bit rates except 16 bits (and 8 bits on little endian) has
always been broken. Since we added optimized functions for 8, 16,
24/32 bits, we can remove the generic flac_convert() function.
Instead of removing it, convert it to a wrapper function for
flac_convert_*().
Diffstat (limited to 'src/inputPlugins')
-rw-r--r-- | src/inputPlugins/_flac_common.c | 58 |
1 files changed, 27 insertions, 31 deletions
diff --git a/src/inputPlugins/_flac_common.c b/src/inputPlugins/_flac_common.c index aedd78396..1ca3aa8c0 100644 --- a/src/inputPlugins/_flac_common.c +++ b/src/inputPlugins/_flac_common.c @@ -262,18 +262,25 @@ static void flac_convert(unsigned char *dest, const FLAC__int32 * const buf[], unsigned int position, unsigned int end) { - unsigned int c_chan, i; - FLAC__uint16 u16; - unsigned char *uc; + switch (bytes_per_sample) { + case 2: + if (num_channels == 2) + flac_convert_stereo16((int16_t*)dest, buf, + position, end); + else + flac_convert_16((int16_t*)dest, num_channels, buf, + position, end); + break; - for (; position < end; ++position) { - for (c_chan = 0; c_chan < num_channels; c_chan++) { - u16 = buf[c_chan][position]; - uc = (unsigned char *)&u16; - for (i = 0; i < bytes_per_sample; i++) { - *dest++ = *uc++; - } - } + case 4: + flac_convert_32((int32_t*)dest, num_channels, buf, + position, end); + break; + + case 1: + flac_convert_8((int8_t*)dest, num_channels, buf, + position, end); + break; } } @@ -291,32 +298,21 @@ flac_common_write(FlacData *data, const FLAC__Frame * frame, assert(dc.audio_format.bits > 0); + if (bytes_per_sample != 1 && bytes_per_sample != 2 && + bytes_per_sample != 4) + /* exotic unsupported bit rate */ + return FLAC__STREAM_DECODER_WRITE_STATUS_ABORT; + for (c_samp = 0; c_samp < frame->header.blocksize; c_samp += num_samples) { num_samples = frame->header.blocksize - c_samp; if (num_samples > max_samples) num_samples = max_samples; - if (num_channels == 2 && bytes_per_sample == 2) - flac_convert_stereo16((int16_t*)data->chunk, - buf, c_samp, - c_samp + num_samples); - else if (bytes_per_sample == 2) - flac_convert_16((int16_t*)data->chunk, - num_channels, buf, c_samp, - c_samp + num_samples); - else if (bytes_per_sample == 4) - flac_convert_32((int32_t*)data->chunk, - num_channels, buf, c_samp, - c_samp + num_samples); - else if (bytes_per_sample == 1) - flac_convert_8((int8_t*)data->chunk, - num_channels, buf, c_samp, - c_samp + num_samples); - else - flac_convert(data->chunk, - num_channels, bytes_per_sample, buf, - c_samp, c_samp + num_samples); + flac_convert(data->chunk, + num_channels, bytes_per_sample, buf, + c_samp, c_samp + num_samples); + data->chunk_length = num_samples * bytes_per_channel; switch (flacSendChunk(data)) { |