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AudioLib: When handling a LOOP_END tag in FLAC or Ogg, truncate any blocks that read past it so that the specified value is respected precisely, instead of rounding up to the nearest block size.

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DONT_BUILD.

git-svn-id: https://svn.eduke32.com/eduke32@5163 1a8010ca-5511-0410-912e-c29ae57300e0
This commit is contained in:
hendricks266 2015-04-24 00:09:42 +00:00
parent 940bef522d
commit c47ebac6c1
2 changed files with 17 additions and 7 deletions
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20
polymer/eduke32/source/jaudiolib/src/flac.c
View file

@ -208,7 +208,7 @@ FLAC__StreamDecoderWriteStatus write_flac_stream(const FLAC__StreamDecoder *deco
{ {
flac_data * fd = (flac_data *) client_data; flac_data * fd = (flac_data *) client_data;
VoiceNode *voice = fd->owner; VoiceNode *voice = fd->owner;
const FLAC__uint64 samples = frame->header.blocksize; FLAC__uint64 samples = frame->header.blocksize;
UNREFERENCED_PARAMETER(decoder); UNREFERENCED_PARAMETER(decoder);
@ -216,6 +216,19 @@ FLAC__StreamDecoderWriteStatus write_flac_stream(const FLAC__StreamDecoder *deco
voice->bits = frame->header.bits_per_sample; voice->bits = frame->header.bits_per_sample;
voice->SamplingRate = frame->header.sample_rate; voice->SamplingRate = frame->header.sample_rate;
if (frame->header.number_type == FLAC__FRAME_NUMBER_TYPE_FRAME_NUMBER)
fd->sample_pos = frame->header.number.frame_number;
else if (frame->header.number_type == FLAC__FRAME_NUMBER_TYPE_SAMPLE_NUMBER)
fd->sample_pos = frame->header.number.sample_number;
if ((FLAC__uint64)(uintptr_t)voice->LoopEnd > 0 && fd->sample_pos + samples >= (FLAC__uint64)(uintptr_t)voice->LoopEnd)
{
samples = (FLAC__uint64)(uintptr_t)voice->LoopEnd - fd->sample_pos;
if (!FLAC__stream_decoder_seek_absolute(fd->stream, (FLAC__uint64)(uintptr_t)voice->LoopStart))
MV_Printf("MV_GetNextFLACBlock FLAC__stream_decoder_seek_absolute: LOOP_START %ul, LOOP_END %ul\n",
(FLAC__uint64)(uintptr_t)voice->LoopStart, (FLAC__uint64)(uintptr_t)voice->LoopEnd);
}
voice->length = ((samples * (voice->bits/8))/2)<<voice->bits; voice->length = ((samples * (voice->bits/8))/2)<<voice->bits;
voice->position = 0; voice->position = 0;
voice->BlockLength = 0; voice->BlockLength = 0;
@ -224,11 +237,6 @@ FLAC__StreamDecoderWriteStatus write_flac_stream(const FLAC__StreamDecoder *deco
voice->FixedPointBufferSize = ( voice->RateScale * MV_MIXBUFFERSIZE ) - voice->RateScale; voice->FixedPointBufferSize = ( voice->RateScale * MV_MIXBUFFERSIZE ) - voice->RateScale;
MV_SetVoiceMixMode( voice ); MV_SetVoiceMixMode( voice );
if (frame->header.number_type == FLAC__FRAME_NUMBER_TYPE_FRAME_NUMBER)
fd->sample_pos = frame->header.number.frame_number;
else if (frame->header.number_type == FLAC__FRAME_NUMBER_TYPE_SAMPLE_NUMBER)
fd->sample_pos = frame->header.number.sample_number;
{ {
const size_t size = samples * voice->channels * (voice->bits/8); const size_t size = samples * voice->channels * (voice->bits/8);
if (size > fd->blocksize) if (size > fd->blocksize)

4
polymer/eduke32/source/jaudiolib/src/vorbis.c
View file

@ -242,9 +242,11 @@ static playbackstatus MV_GetNextVorbisBlock
#endif #endif
//fprintf(stderr, "ov_read = %d\n", bytes); //fprintf(stderr, "ov_read = %d\n", bytes);
if (bytes > 0) { if (bytes > 0) {
ogg_int64_t currentPosition;
bytesread += bytes; bytesread += bytes;
if ((ogg_int64_t)(intptr_t)voice->LoopEnd > 0 if ((ogg_int64_t)(intptr_t)voice->LoopEnd > 0
&& ov_pcm_tell(&vd->vf) >= (ogg_int64_t)(intptr_t)voice->LoopEnd) { && (currentPosition = ov_pcm_tell(&vd->vf)) >= (ogg_int64_t)(intptr_t)voice->LoopEnd) {
bytesread -= (currentPosition - (ogg_int64_t)(intptr_t)voice->LoopEnd) * voice->channels * 2; // (voice->bits>>3)
err = ov_pcm_seek(&vd->vf,(ogg_int64_t)(intptr_t)voice->LoopStart); err = ov_pcm_seek(&vd->vf,(ogg_int64_t)(intptr_t)voice->LoopStart);
if (err != 0) { if (err != 0) {
MV_Printf("MV_GetNextVorbisBlock ov_pcm_seek: LOOP_START %l, LOOP_END %l, err %d\n", MV_Printf("MV_GetNextVorbisBlock ov_pcm_seek: LOOP_START %l, LOOP_END %l, err %d\n",