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Remove naked new[] in: AudioIO

This commit is contained in:
Paul Licameli 2016-04-14 12:08:36 -04:00
parent e6db1a57a3
commit 2b23667211
2 changed files with 30 additions and 71 deletions
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93
src/AudioIO.cpp
View File

@ -1705,10 +1705,11 @@ int AudioIO::StartStream(const ConstWaveTrackArray &playbackTracks,
mPlayMode = options.playLooped ? PLAY_LOOPED : PLAY_STRAIGHT;
mCutPreviewGapStart = options.cutPreviewGapStart;
mCutPreviewGapLen = options.cutPreviewGapLen;
mPlaybackBuffers = NULL;
mPlaybackMixers = NULL;
mCaptureBuffers = NULL;
mResample = NULL;
mPlaybackBuffers.reset();
mPlaybackMixers.reset();
mCaptureBuffers.reset();
mResample.reset();
double playbackTime = 4.0;
@ -1859,12 +1860,8 @@ int AudioIO::StartStream(const ConstWaveTrackArray &playbackTracks,
auto playbackMixBufferSize =
mPlaybackSamplesToCopy;
mPlaybackBuffers = new RingBuffer* [mPlaybackTracks.size()];
mPlaybackMixers = new Mixer* [mPlaybackTracks.size()];
// Set everything to zero in case we have to DELETE these due to a memory exception.
memset(mPlaybackBuffers, 0, sizeof(RingBuffer*)*mPlaybackTracks.size());
memset(mPlaybackMixers, 0, sizeof(Mixer*)*mPlaybackTracks.size());
mPlaybackBuffers.reinit(mPlaybackTracks.size());
mPlaybackMixers.reinit(mPlaybackTracks.size());
const Mixer::WarpOptions &warpOptions =
#ifdef EXPERIMENTAL_SCRUBBING_SUPPORT
@ -1878,10 +1875,11 @@ int AudioIO::StartStream(const ConstWaveTrackArray &playbackTracks,
for (unsigned int i = 0; i < mPlaybackTracks.size(); i++)
{
mPlaybackBuffers[i] = new RingBuffer(floatSample, playbackBufferSize);
mPlaybackBuffers[i] = std::make_unique<RingBuffer>(floatSample, playbackBufferSize);
// MB: use normal time for the end time, not warped time!
mPlaybackMixers[i] = new Mixer(WaveTrackConstArray{ mPlaybackTracks[i] },
mPlaybackMixers[i] = std::make_unique<Mixer>
(WaveTrackConstArray{ mPlaybackTracks[i] },
warpOptions,
mT0, mT1, 1,
playbackMixBufferSize, false,
@ -1904,19 +1902,16 @@ int AudioIO::StartStream(const ConstWaveTrackArray &playbackTracks,
return 0;
}
mCaptureBuffers = new RingBuffer* [mCaptureTracks.size()];
mResample = new Resample* [mCaptureTracks.size()];
mCaptureBuffers.reinit(mCaptureTracks.size());
mResample.reinit(mCaptureTracks.size());
mFactor = sampleRate / mRate;
// Set everything to zero in case we have to DELETE these due to a memory exception.
memset(mCaptureBuffers, 0, sizeof(RingBuffer*)*mCaptureTracks.size());
memset(mResample, 0, sizeof(Resample*)*mCaptureTracks.size());
for( unsigned int i = 0; i < mCaptureTracks.size(); i++ )
{
mCaptureBuffers[i] = new RingBuffer( mCaptureTracks[i]->GetSampleFormat(),
mCaptureBuffers[i] = std::make_unique<RingBuffer>
( mCaptureTracks[i]->GetSampleFormat(),
captureBufferSize );
mResample[i] = new Resample(true, mFactor, mFactor); // constant rate resampling
mResample[i] = std::make_unique<Resample>(true, mFactor, mFactor); // constant rate resampling
}
}
}
@ -2087,37 +2082,10 @@ void AudioIO::StartStreamCleanup(bool bOnlyBuffers)
EffectManager::Get().RealtimeFinalize();
}
if(mPlaybackBuffers)
{
for (unsigned int i = 0; i < mPlaybackTracks.size(); i++)
delete mPlaybackBuffers[i];
delete [] mPlaybackBuffers;
mPlaybackBuffers = NULL;
}
if(mPlaybackMixers)
{
for (unsigned int i = 0; i < mPlaybackTracks.size(); i++)
delete mPlaybackMixers[i];
delete [] mPlaybackMixers;
mPlaybackMixers = NULL;
}
if(mCaptureBuffers)
{
for (unsigned int i = 0; i < mCaptureTracks.size(); i++)
delete mCaptureBuffers[i];
delete [] mCaptureBuffers;
mCaptureBuffers = NULL;
}
if(mResample)
{
for (unsigned int i = 0; i < mCaptureTracks.size(); i++)
delete mResample[i];
delete [] mResample;
mResample = NULL;
}
mPlaybackBuffers.reset();
mPlaybackMixers.reset();
mCaptureBuffers.reset();
mResample.reset();
if(!bOnlyBuffers)
{
@ -2443,14 +2411,8 @@ void AudioIO::StopStream()
if (mPlaybackTracks.size() > 0)
{
for (unsigned int i = 0; i < mPlaybackTracks.size(); i++)
{
delete mPlaybackBuffers[i];
delete mPlaybackMixers[i];
}
delete[] mPlaybackBuffers;
delete[] mPlaybackMixers;
mPlaybackBuffers.reset();
mPlaybackMixers.reset();
}
//
@ -2458,6 +2420,9 @@ void AudioIO::StopStream()
//
if (mCaptureTracks.size() > 0)
{
mCaptureBuffers.reset();
mResample.reset();
//
// We only apply latency correction when we actually played back
// tracks during the recording. If we did not play back tracks,
@ -2473,9 +2438,6 @@ void AudioIO::StopStream()
for (unsigned int i = 0; i < mCaptureTracks.size(); i++)
{
delete mCaptureBuffers[i];
delete mResample[i];
WaveTrack* track = mCaptureTracks[i];
track->Flush();
@ -2521,9 +2483,6 @@ void AudioIO::StopStream()
}
}
}
delete[] mCaptureBuffers;
delete[] mResample;
}
}
@ -4378,7 +4337,7 @@ int audacityAudioCallback(const void *inputBuffer, void *outputBuffer,
bool selected = false;
int group = 0;
int chanCnt = 0;
int maxLen = 0;
decltype(framesPerBuffer) maxLen = 0;
for (unsigned t = 0; t < numPlaybackTracks; t++)
{
const WaveTrack *vt = gAudioIO->mPlaybackTracks[t];
@ -4410,7 +4369,7 @@ int audacityAudioCallback(const void *inputBuffer, void *outputBuffer,
#define ORIGINAL_DO_NOT_PLAY_ALL_MUTED_TRACKS_TO_END
#ifdef ORIGINAL_DO_NOT_PLAY_ALL_MUTED_TRACKS_TO_END
int len = 0;
decltype(framesPerBuffer) len = 0;
// this is original code prior to r10680 -RBD
if (cut)
{

8
src/AudioIO.h
View File

@ -555,13 +555,13 @@ private:
#ifdef EXPERIMENTAL_MIDI_OUT
std::unique_ptr<AudioThread> mMidiThread;
#endif
Resample **mResample;
RingBuffer **mCaptureBuffers;
ArrayOf<std::unique_ptr<Resample>> mResample;
ArrayOf<std::unique_ptr<RingBuffer>> mCaptureBuffers;
WaveTrackArray mCaptureTracks;
RingBuffer **mPlaybackBuffers;
ArrayOf<std::unique_ptr<RingBuffer>> mPlaybackBuffers;
ConstWaveTrackArray mPlaybackTracks;
Mixer **mPlaybackMixers;
ArrayOf<std::unique_ptr<Mixer>> mPlaybackMixers;
volatile int mStreamToken;
static int mNextStreamToken;
double mFactor;