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Make many counts of tracks and channels unsigned...

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... And in some places where a library uses signed types, assert that
the reported number is not negative.

What led me to this, is that there are many places where a size_t value for
an allocation is the product of a number of channels and some other number.
This commit is contained in:
Paul Licameli
2016-09-02 15:53:09 -04:00
parent ef23de6a60
commit 67cec5ad83
81 changed files with 272 additions and 262 deletions
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30
src/AudioIO.cpp
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@@ -1948,7 +1948,7 @@ int AudioIO::StartStream(const WaveTrackArray &playbackTracks,
{
WaveTrack *vt = gAudioIO->mPlaybackTracks[i];
int chanCnt = 1;
unsigned chanCnt = 1;
if (vt->GetLinked())
{
i++;
@@ -3634,7 +3634,7 @@ void AudioIO::FillBuffers()
// Append captured samples to the end of the WaveTracks.
// The WaveTracks have their own buffering for efficiency.
AutoSaveFile blockFileLog;
int numChannels = mCaptureTracks.size();
auto numChannels = mCaptureTracks.size();
for( i = 0; (int)i < numChannels; i++ )
{
@@ -3871,9 +3871,8 @@ bool AudioIO::SetHasSolo(bool hasSolo)
void AudioIO::FillMidiBuffers()
{
bool hasSolo = false;
int numPlaybackTracks = gAudioIO->mPlaybackTracks.size();
int t;
for(t = 0; t < numPlaybackTracks; t++ )
auto numPlaybackTracks = gAudioIO->mPlaybackTracks.size();
for(unsigned t = 0; t < numPlaybackTracks; t++ )
if( gAudioIO->mPlaybackTracks[t]->GetSolo() ) {
hasSolo = true;
break;
@@ -4120,7 +4119,7 @@ void AudioIO::AILAProcess(double maxPeak) {
static void DoSoftwarePlaythrough(const void *inputBuffer,
sampleFormat inputFormat,
int inputChannels,
unsigned inputChannels,
float *outputBuffer,
int len)
{
@@ -4151,9 +4150,9 @@ int audacityAudioCallback(const void *inputBuffer, void *outputBuffer,
#endif
const PaStreamCallbackFlags WXUNUSED(statusFlags), void * WXUNUSED(userData) )
{
int numPlaybackChannels = gAudioIO->mNumPlaybackChannels;
int numPlaybackTracks = gAudioIO->mPlaybackTracks.size();
int numCaptureChannels = gAudioIO->mNumCaptureChannels;
auto numPlaybackChannels = gAudioIO->mNumPlaybackChannels;
auto numPlaybackTracks = gAudioIO->mPlaybackTracks.size();
auto numCaptureChannels = gAudioIO->mNumCaptureChannels;
int callbackReturn = paContinue;
void *tempBuffer = alloca(framesPerBuffer*sizeof(float)*
MAX(numCaptureChannels,numPlaybackChannels));
@@ -4178,7 +4177,6 @@ int audacityAudioCallback(const void *inputBuffer, void *outputBuffer,
#endif
unsigned int i;
int t;
/* Send data to recording VU meter if applicable */
@@ -4322,7 +4320,7 @@ int audacityAudioCallback(const void *inputBuffer, void *outputBuffer,
gAudioIO->mWarpedTime = std::abs(gAudioIO->mWarpedTime);
// Reset mixer positions and flush buffers for all tracks
for (i = 0; i < (unsigned int)numPlaybackTracks; i++)
for (i = 0; i < numPlaybackTracks; i++)
{
gAudioIO->mPlaybackMixers[i]->Reposition(gAudioIO->mTime);
const auto toDiscard =
@@ -4347,8 +4345,8 @@ int audacityAudioCallback(const void *inputBuffer, void *outputBuffer,
return paContinue;
}
int numSolo = 0;
for( t = 0; t < numPlaybackTracks; t++ )
unsigned numSolo = 0;
for(unsigned t = 0; t < numPlaybackTracks; t++ )
if( gAudioIO->mPlaybackTracks[t]->GetSolo() )
numSolo++;
#ifdef EXPERIMENTAL_MIDI_OUT
@@ -4372,7 +4370,7 @@ int audacityAudioCallback(const void *inputBuffer, void *outputBuffer,
int group = 0;
int chanCnt = 0;
int maxLen = 0;
for (t = 0; t < numPlaybackTracks; t++)
for (unsigned t = 0; t < numPlaybackTracks; t++)
{
WaveTrack *vt = gAudioIO->mPlaybackTracks[t];
@@ -4573,7 +4571,7 @@ int audacityAudioCallback(const void *inputBuffer, void *outputBuffer,
if( inputBuffer && (numCaptureChannels > 0) )
{
size_t len = framesPerBuffer;
for( t = 0; t < numCaptureChannels; t++) {
for(unsigned t = 0; t < numCaptureChannels; t++) {
len = std::min( len,
gAudioIO->mCaptureBuffers[t]->AvailForPut());
}
@@ -4585,7 +4583,7 @@ int audacityAudioCallback(const void *inputBuffer, void *outputBuffer,
}
if (len > 0) {
for( t = 0; t < numCaptureChannels; t++) {
for(unsigned t = 0; t < numCaptureChannels; t++) {
// dmazzoni:
// Un-interleave. Ugly special-case code required because the