[Scummvm-cvs-logs] SF.net SVN: scummvm: [25831] scummvm/trunk/sound/flac.cpp
fingolfin at users.sourceforge.net
fingolfin at users.sourceforge.net
Sat Feb 24 22:13:30 CET 2007
Revision: 25831
http://scummvm.svn.sourceforge.net/scummvm/?rev=25831&view=rev
Author: fingolfin
Date: 2007-02-24 13:13:30 -0800 (Sat, 24 Feb 2007)
Log Message:
-----------
more FLAC code cleanup
Modified Paths:
--------------
scummvm/trunk/sound/flac.cpp
Modified: scummvm/trunk/sound/flac.cpp
===================================================================
--- scummvm/trunk/sound/flac.cpp 2007-02-24 20:50:56 UTC (rev 25830)
+++ scummvm/trunk/sound/flac.cpp 2007-02-24 21:13:30 UTC (rev 25831)
@@ -110,7 +110,7 @@
SampleType *bufReadPos;
uint bufSize;
uint bufFill;
- } _preBuffer;
+ } _sampleCache;
SampleType *_outBuffer;
uint _requestedSamples;
@@ -127,7 +127,11 @@
bool isStereo() const { return _streaminfo.channels >= 2; }
int getRate() const { return _streaminfo.sample_rate; }
- bool endOfData() const { return _streaminfo.channels == 0 || (_lastSampleWritten && _preBuffer.bufFill == 0); }
+ bool endOfData() const {
+ // End of data is reached if there either is no valid stream data available,
+ // or if we reached the last sample and completely emptied the sample cache
+ return _streaminfo.channels == 0 || (_lastSampleWritten && _sampleCache.bufFill == 0);
+ }
bool isStreamDecoderReady() const { return getStreamDecoderState() == FLAC__STREAM_DECODER_SEARCH_FOR_FRAME_SYNC ; }
@@ -179,7 +183,7 @@
_disposeAfterUse(dispose),
_numLoops(numLoops),
_firstSample(0), _lastSample(0),
- _outBuffer(NULL), _requestedSamples(0), _lastSampleWritten(true),
+ _outBuffer(NULL), _requestedSamples(0), _lastSampleWritten(false),
_methodConvertBuffers(&FlacInputStream::convertBuffersGeneric)
{
assert(_inStream);
@@ -187,11 +191,11 @@
// TODO: Implement looping support
- _preBuffer.bufData = NULL;
- _preBuffer.bufFill = 0;
- _preBuffer.bufSize = 0;
+ _sampleCache.bufData = NULL;
+ _sampleCache.bufReadPos = NULL;
+ _sampleCache.bufFill = 0;
+ _sampleCache.bufSize = 0;
- _lastSampleWritten = false;
_methodConvertBuffers = &FlacInputStream::convertBuffersGeneric;
bool success;
@@ -246,7 +250,7 @@
::FLAC__stream_decoder_delete(_decoder);
#endif
}
- delete[] _preBuffer.bufData;
+ delete[] _sampleCache.bufData;
if (_disposeAfterUse)
delete _inStream;
@@ -287,7 +291,7 @@
const bool result = (0 != ::FLAC__stream_decoder_seek_absolute(_decoder, sample));
#endif
if (result) {
- _preBuffer.bufFill = 0;
+ _sampleCache.bufFill = 0;
_lastSampleWritten = (_lastSample != 0 && sample >= _lastSample); // only set if we are SURE
}
return result;
@@ -309,18 +313,20 @@
_outBuffer = buffer;
_requestedSamples = numSamples;
- if (_preBuffer.bufFill > 0) {
- assert(_preBuffer.bufData != NULL && _preBuffer.bufReadPos != NULL && _preBuffer.bufSize > 0);
- assert(_preBuffer.bufReadPos >= _preBuffer.bufData);
- assert(_preBuffer.bufFill % numChannels == 0);
+ // If there is still data in our buffer from the last time around,
+ // copy that first.
+ if (_sampleCache.bufFill > 0) {
+ assert(_sampleCache.bufData != NULL && _sampleCache.bufReadPos != NULL && _sampleCache.bufSize > 0);
+ assert(_sampleCache.bufReadPos >= _sampleCache.bufData);
+ assert(_sampleCache.bufFill % numChannels == 0);
- const uint copySamples = MIN((uint)numSamples, _preBuffer.bufFill);
- memcpy(buffer, _preBuffer.bufReadPos, copySamples*sizeof(buffer[0]));
+ const uint copySamples = MIN((uint)numSamples, _sampleCache.bufFill);
+ memcpy(buffer, _sampleCache.bufReadPos, copySamples*sizeof(buffer[0]));
_outBuffer = buffer + copySamples;
_requestedSamples = numSamples - copySamples;
- _preBuffer.bufReadPos += copySamples;
- _preBuffer.bufFill -= copySamples;
+ _sampleCache.bufReadPos += copySamples;
+ _sampleCache.bufFill -= copySamples;
}
bool decoderOk = true;
@@ -328,32 +334,33 @@
if (!_lastSampleWritten) {
FLAC__StreamDecoderState state = getStreamDecoderState();
+ // Keep poking FLAC to process more samples until we completely satisfied the request
for (; _requestedSamples > 0 && state == FLAC__STREAM_DECODER_SEARCH_FOR_FRAME_SYNC; state = getStreamDecoderState()) {
- assert(_preBuffer.bufFill == 0);
+ assert(_sampleCache.bufFill == 0);
assert(_requestedSamples % numChannels == 0);
processSingleBlock();
}
- if (state != FLAC__STREAM_DECODER_SEARCH_FOR_FRAME_SYNC) {
- switch (state) {
- case FLAC__STREAM_DECODER_END_OF_STREAM :
- _lastSampleWritten = true;
- decoderOk = true; // no REAL error
- break;
-
- default:
- decoderOk = false;
- warning("FlacInputStream: An error occured while decoding. DecoderState is: %s",
- FLAC__StreamDecoderStateString[getStreamDecoderState()]);
- }
+ // Error handling
+ switch (state) {
+ case FLAC__STREAM_DECODER_END_OF_STREAM:
+ _lastSampleWritten = true;
+ break;
+ case FLAC__STREAM_DECODER_SEARCH_FOR_FRAME_SYNC:
+ break;
+ default:
+ decoderOk = false;
+ warning("FlacInputStream: An error occured while decoding. DecoderState is: %s",
+ FLAC__StreamDecoderStateString[getStreamDecoderState()]);
}
}
+ // Compute how many samples we actually produced
const int samples = (int)(_outBuffer - buffer);
assert(samples % numChannels == 0);
- _outBuffer = NULL; // basically unnessecary, only for the purpose of the asserts
- _requestedSamples = 0; // basically unnessecary, only for the purpose of the asserts
+ _outBuffer = NULL; // basically unnecessary, only for the purpose of the asserts
+ _requestedSamples = 0; // basically unnecessary, only for the purpose of the asserts
return decoderOk ? samples : -1;
}
@@ -385,26 +392,25 @@
bool FlacInputStream::allocateBuffer(uint minSamples) {
uint allocateSize = minSamples / getChannels();
- /** insert funky algorythm for optimum buffersize here */
+ // TODO: insert funky algorithm for optimum buffersize here
allocateSize = MIN(_streaminfo.max_blocksize, MAX(_streaminfo.min_blocksize, allocateSize));
- allocateSize += 8 - (allocateSize % 8); // make sure its an nice even amount
+ allocateSize += 8 - (allocateSize % 8); // make sure it's a nice even amount
allocateSize *= getChannels();
- _lastSampleWritten = _lastSampleWritten && _preBuffer.bufFill == 0;
- _preBuffer.bufFill = 0;
- _preBuffer.bufSize = 0;
- delete[] _preBuffer.bufData;
+ _lastSampleWritten = _lastSampleWritten && _sampleCache.bufFill == 0;
+ _sampleCache.bufFill = 0;
+ _sampleCache.bufSize = 0;
+ delete[] _sampleCache.bufData;
- _preBuffer.bufData = new SampleType[allocateSize];
- if (_preBuffer.bufData != NULL) {
- _preBuffer.bufSize = allocateSize;
+ _sampleCache.bufData = new SampleType[allocateSize];
+ if (_sampleCache.bufData != NULL) {
+ _sampleCache.bufSize = allocateSize;
return true;
}
return false;
}
-void FlacInputStream::setBestConvertBufferMethod()
-{
+void FlacInputStream::setBestConvertBufferMethod() {
PFCONVERTBUFFERS tempMethod = &FlacInputStream::convertBuffersGeneric;
const uint numChannels = getChannels();
@@ -429,8 +435,7 @@
}
// 1 channel, no scaling
-void FlacInputStream::convertBuffersMonoNS(SampleType* bufDestination, const FLAC__int32 *inChannels[], uint numSamples, const uint numChannels, const uint8 numBits)
-{
+void FlacInputStream::convertBuffersMonoNS(SampleType* bufDestination, const FLAC__int32 *inChannels[], uint numSamples, const uint numChannels, const uint8 numBits) {
assert(numChannels == 1);
assert(numBits == BUFTYPE_BITS);
@@ -455,8 +460,7 @@
}
// 1 channel, scaling from 8Bit
-void FlacInputStream::convertBuffersMono8Bit(SampleType* bufDestination, const FLAC__int32 *inChannels[], uint numSamples, const uint numChannels, const uint8 numBits)
-{
+void FlacInputStream::convertBuffersMono8Bit(SampleType* bufDestination, const FLAC__int32 *inChannels[], uint numSamples, const uint numChannels, const uint8 numBits) {
assert(numChannels == 1);
assert(numBits == 8);
assert(8 < BUFTYPE_BITS);
@@ -482,8 +486,7 @@
}
// 2 channels, no scaling
-void FlacInputStream::convertBuffersStereoNS(SampleType* bufDestination, const FLAC__int32 *inChannels[], uint numSamples, const uint numChannels, const uint8 numBits)
-{
+void FlacInputStream::convertBuffersStereoNS(SampleType* bufDestination, const FLAC__int32 *inChannels[], uint numSamples, const uint numChannels, const uint8 numBits) {
assert(numChannels == 2);
assert(numBits == BUFTYPE_BITS);
assert(numSamples % 2 == 0); // must be integral multiply of channels
@@ -516,8 +519,7 @@
}
// 2 channels, scaling from 8Bit
-void FlacInputStream::convertBuffersStereo8Bit(SampleType* bufDestination, const FLAC__int32 *inChannels[], uint numSamples, const uint numChannels, const uint8 numBits)
-{
+void FlacInputStream::convertBuffersStereo8Bit(SampleType* bufDestination, const FLAC__int32 *inChannels[], uint numSamples, const uint numChannels, const uint8 numBits) {
assert(numChannels == 2);
assert(numBits == 8);
assert(numSamples % 2 == 0); // must be integral multiply of channels
@@ -550,8 +552,7 @@
}
// all Purpose-conversion - slowest of em all
-void FlacInputStream::convertBuffersGeneric(SampleType* bufDestination, const FLAC__int32 *inChannels[], uint numSamples, const uint numChannels, const uint8 numBits)
-{
+void FlacInputStream::convertBuffersGeneric(SampleType* bufDestination, const FLAC__int32 *inChannels[], uint numSamples, const uint numChannels, const uint8 numBits) {
assert(numSamples % numChannels == 0); // must be integral multiply of channels
if (numBits < BUFTYPE_BITS) {
@@ -584,7 +585,7 @@
assert(frame->header.bits_per_sample == _streaminfo.bits_per_sample);
assert(frame->header.number_type == FLAC__FRAME_NUMBER_TYPE_SAMPLE_NUMBER || _streaminfo.min_blocksize == _streaminfo.max_blocksize);
- assert(_preBuffer.bufFill == 0); // we dont append data
+ assert(_sampleCache.bufFill == 0); // we dont append data
uint numSamples = frame->header.blocksize;
const uint numChannels = getChannels();
@@ -610,10 +611,12 @@
// writing DIRECTLY to the Buffer ScummVM provided
if (_requestedSamples > 0) {
- assert(_requestedSamples % numChannels == 0); // must be integral multiply of channels
+ assert(_requestedSamples % numChannels == 0);
assert(_outBuffer != NULL);
- const uint copySamples = MIN(_requestedSamples,numSamples);
+ // Copy & convert the available samples (limited both by how many we have available, and
+ // by how many are actually needed).
+ const uint copySamples = MIN(_requestedSamples, numSamples);
(*_methodConvertBuffers)(_outBuffer, inChannels, copySamples, numChannels, numBits);
_requestedSamples -= copySamples;
@@ -622,15 +625,15 @@
}
// checking if Buffer fits
- if (_preBuffer.bufSize < numSamples) {
+ if (_sampleCache.bufSize < numSamples) {
if (!allocateBuffer(numSamples))
return FLAC__STREAM_DECODER_WRITE_STATUS_ABORT;
} // optional check if buffer is wasting too much memory ?
- (*_methodConvertBuffers)(_preBuffer.bufData, inChannels, numSamples, numChannels, numBits);
+ (*_methodConvertBuffers)(_sampleCache.bufData, inChannels, numSamples, numChannels, numBits);
- _preBuffer.bufFill = numSamples;
- _preBuffer.bufReadPos = _preBuffer.bufData;
+ _sampleCache.bufFill = numSamples;
+ _sampleCache.bufReadPos = _sampleCache.bufData;
return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
}
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