flac: moved code from flacWrite() to _flac_common.c

There is still a lot of duplicated code in flac_plugin.c and
oggflac_plugin.c.  Move code from flac_plugin.c to _flac_common.c, and
use the new function flac_common_write() also in oggflac_plugin.c,
porting lots of optimizations over to it.
This commit is contained in:
Max Kellermann 2008-09-23 23:59:54 +02:00
parent aac5f42698
commit 8bcbe90b25
4 changed files with 97 additions and 129 deletions

View File

@ -196,4 +196,95 @@ void flac_error_common_cb(const char *plugin,
}
}
/* keep this inlined, this is just macro but prettier :) */
static inline int flacSendChunk(FlacData * data)
{
if (decoder_data(data->decoder, data->inStream,
1, data->chunk,
data->chunk_length, data->time,
data->bitRate,
data->replayGainInfo) == DECODE_COMMAND_STOP)
return -1;
return 0;
}
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,
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 < num_channels; c_chan++) {
u16 = buf[c_chan][position];
uc = (unsigned char *)&u16;
for (i = 0; i < bytes_per_sample; i++) {
*dest++ = *uc++;
}
}
}
}
FLAC__StreamDecoderWriteStatus
flac_common_write(FlacData *data, const FLAC__Frame * frame,
const FLAC__int32 *const buf[])
{
unsigned int c_samp;
const unsigned int num_channels = frame->header.channels;
const unsigned int bytes_per_sample =
audio_format_sample_size(&data->audio_format);
const unsigned int bytes_per_channel =
bytes_per_sample * frame->header.channels;
const unsigned int max_samples = FLAC_CHUNK_SIZE / bytes_per_channel;
unsigned int num_samples;
assert(data->audio_format.bits > 0);
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(data->chunk,
buf, c_samp,
c_samp + num_samples);
else
flac_convert(data->chunk,
num_channels, bytes_per_sample, buf,
c_samp, c_samp + num_samples);
data->chunk_length = num_samples * bytes_per_channel;
if (flacSendChunk(data) < 0) {
return
FLAC__STREAM_DECODER_WRITE_STATUS_ABORT;
}
data->chunk_length = 0;
if (decoder_get_command(data->decoder) == DECODE_COMMAND_SEEK) {
return
FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
}
}
return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
}
#endif /* HAVE_FLAC || HAVE_OGGFLAC */

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@ -164,18 +164,9 @@ void flac_error_common_cb(const char *plugin,
struct tag *copyVorbisCommentBlockToMpdTag(const FLAC__StreamMetadata * block,
struct tag *tag);
/* keep this inlined, this is just macro but prettier :) */
static inline int flacSendChunk(FlacData * data)
{
if (decoder_data(data->decoder, data->inStream,
1, data->chunk,
data->chunk_length, data->time,
data->bitRate,
data->replayGainInfo) == DECODE_COMMAND_STOP)
return -1;
return 0;
}
FLAC__StreamDecoderWriteStatus
flac_common_write(FlacData *data, const FLAC__Frame * frame,
const FLAC__int32 *const buf[]);
#endif /* HAVE_FLAC || HAVE_OGGFLAC */

View File

@ -195,59 +195,16 @@ static void flacMetadata(mpd_unused 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,
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 < num_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[],
void *vdata)
{
FlacData *data = (FlacData *) vdata;
FLAC__uint32 samples = frame->header.blocksize;
unsigned int c_samp;
const unsigned int num_channels = frame->header.channels;
const unsigned int bytes_per_sample =
audio_format_sample_size(&data->audio_format);
const unsigned int bytes_per_channel =
bytes_per_sample * frame->header.channels;
const unsigned int max_samples = FLAC_CHUNK_SIZE / bytes_per_channel;
unsigned int num_samples;
FlacData *data = (FlacData *) vdata;
float timeChange;
FLAC__uint64 newPosition = 0;
assert(data->audio_format.bits > 0);
timeChange = ((float)samples) / frame->header.sample_rate;
data->time += timeChange;
@ -261,34 +218,7 @@ static FLAC__StreamDecoderWriteStatus flacWrite(const flac_decoder *dec,
}
data->position = newPosition;
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(data->chunk,
buf, c_samp,
c_samp + num_samples);
else
flac_convert(data->chunk,
num_channels, bytes_per_sample, buf,
c_samp, c_samp + num_samples);
data->chunk_length = num_samples * bytes_per_channel;
if (flacSendChunk(data) < 0) {
return
FLAC__STREAM_DECODER_WRITE_STATUS_ABORT;
}
data->chunk_length = 0;
if (decoder_get_command(data->decoder) == DECODE_COMMAND_SEEK) {
return
FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
}
}
return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
return flac_common_write(data, frame, buf);
}
static struct tag *flacMetadataDup(char *file, int *vorbisCommentFound)

View File

@ -157,50 +157,12 @@ static FLAC__StreamDecoderWriteStatus oggflacWrite(mpd_unused const
{
FlacData *data = (FlacData *) vdata;
FLAC__uint32 samples = frame->header.blocksize;
FLAC__uint16 u16;
unsigned char *uc;
unsigned int c_samp, c_chan;
unsigned int i;
float timeChange;
timeChange = ((float)samples) / frame->header.sample_rate;
data->time += timeChange;
/* ogg123 uses a complicated method of calculating bitrate
* with averaging which I'm not too fond of.
* (waste of memory/CPU cycles, especially given this is _lossless_)
* a get_decode_position() is not available in OggFLAC, either
*
* this does not give an accurate bitrate:
* (bytes_last_read was set in the read callback)
data->bitRate = ((8.0 * data->bytes_last_read *
frame->header.sample_rate)
/((float)samples * 1000)) + 0.5;
*/
for (c_samp = 0; c_samp < frame->header.blocksize; c_samp++) {
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 < audio_format_sample_size(&data->audio_format); i++) {
if (data->chunk_length >= FLAC_CHUNK_SIZE) {
if (flacSendChunk(data) < 0) {
return
FLAC__STREAM_DECODER_WRITE_STATUS_ABORT;
}
data->chunk_length = 0;
if (decoder_get_command(data->decoder) == DECODE_COMMAND_SEEK) {
return
FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
}
}
data->chunk[data->chunk_length++] = *(uc++);
}
}
}
return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
return flac_common_write(data, frame, buf);
}
/* used by TagDup */
@ -372,12 +334,6 @@ static int oggflac_decode(struct decoder * mpd_decoder, InputStream * inStream)
(OggFLAC__seekable_stream_decoder_get_state(decoder));
OggFLAC__seekable_stream_decoder_finish(decoder);
}
/* send last little bit */
if (data.chunk_length > 0 &&
decoder_get_command(mpd_decoder) == DECODE_COMMAND_NONE) {
flacSendChunk(&data);
decoder_flush(mpd_decoder);
}
fail:
oggflac_cleanup(&data, decoder);