From d7695f3c918fa6102d36cf0d46c4cc860fc4053f Mon Sep 17 00:00:00 2001 From: Emily Mabrey Date: Tue, 10 Aug 2021 18:49:56 -0400 Subject: [PATCH] Remove whitespace only changes Signed-off-by: Emily Mabrey --- src/AudioIO.cpp | 137 +++++++++++++++++++------------------- src/AudioIOBase.h | 5 +- src/AudioIOBufferHelper.h | 41 ++++++------ 3 files changed, 92 insertions(+), 91 deletions(-) diff --git a/src/AudioIO.cpp b/src/AudioIO.cpp index c1747c786..7a322c0e1 100644 --- a/src/AudioIO.cpp +++ b/src/AudioIO.cpp @@ -18,8 +18,8 @@ ********************************************************************//** \class AudioIoCallback -\brief AudioIoCallback is a class that implements the callback required -by PortAudio. The callback needs to be responsive, has no GUI, and +\brief AudioIoCallback is a class that implements the callback required +by PortAudio. The callback needs to be responsive, has no GUI, and copies data into and out of the sound card buffers. It also sends data to the meters. @@ -408,7 +408,7 @@ callbacks for these events. *//****************************************************************//** \class AudioIOStartStreamOptions -\brief struct holding stream options, including a pointer to the +\brief struct holding stream options, including a pointer to the time warp info and AudioIOListener and whether the playback is looped. *//*******************************************************************/ @@ -961,7 +961,7 @@ AudioIO::AudioIO() wxASSERT(false); } - // This ASSERT because of casting in the callback + // This ASSERT because of casting in the callback // functions where we cast a tempFloats buffer to a (short*) buffer. // We have to ASSERT in the GUI thread, if we are to see it properly. wxASSERT( sizeof( short ) <= sizeof( float )); @@ -986,8 +986,7 @@ AudioIO::AudioIO() #ifdef EXPERIMENTAL_AUTOMATED_INPUT_LEVEL_ADJUSTMENT mAILAActive = false; #endif - - mStreamToken = 0; + mStreamToken = 0; mLastPaError = paNoError; @@ -1004,18 +1003,18 @@ AudioIO::AudioIO() PaError err = Pa_Initialize(); - if (err != paNoError) { - auto errStr = XO("Could not find any audio devices.\n"); - errStr += XO("You will not be able to play or record audio.\n\n"); - const wxString paErrStr = LAT1CTOWX(Pa_GetErrorText(err)); - if (!paErrStr.empty()) - errStr += XO("Error: %s").Format(paErrStr); - // XXX: we are in libaudacity, popping up dialogs not allowed! A - // long-term solution will probably involve exceptions - AudacityMessageBox( - errStr, - XO("Error Initializing Audio"), - wxICON_ERROR | wxOK); + if (err != paNoError) { + auto errStr = XO("Could not find any audio devices.\n"); + errStr += XO("You will not be able to play or record audio.\n\n"); + const wxString paErrStr = LAT1CTOWX(Pa_GetErrorText(err)); + if (!paErrStr.empty()) + errStr += XO("Error: %s").Format(paErrStr); + // XXX: we are in libaudacity, popping up dialogs not allowed! A + // long-term solution will probably involve exceptions + AudacityMessageBox( + errStr, + XO("Error Initializing Audio"), + wxICON_ERROR|wxOK); // Since PortAudio is not initialized, all calls to PortAudio // functions will fail. This will give reasonable behavior, since @@ -1026,19 +1025,19 @@ AudioIO::AudioIO() #ifdef EXPERIMENTAL_MIDI_OUT PmError pmErr = Pm_Initialize(); - if (pmErr != pmNoError) { - auto errStr = - XO("There was an error initializing the midi i/o layer.\n"); - errStr += XO("You will not be able to play midi.\n\n"); - const wxString pmErrStr = LAT1CTOWX(Pm_GetErrorText(pmErr)); - if (!pmErrStr.empty()) - errStr += XO("Error: %s").Format(pmErrStr); - // XXX: we are in libaudacity, popping up dialogs not allowed! A - // long-term solution will probably involve exceptions - AudacityMessageBox( - errStr, - XO("Error Initializing Midi"), - wxICON_ERROR | wxOK); + if (pmErr != pmNoError) { + auto errStr = + XO("There was an error initializing the midi i/o layer.\n"); + errStr += XO("You will not be able to play midi.\n\n"); + const wxString pmErrStr = LAT1CTOWX(Pm_GetErrorText(pmErr)); + if (!pmErrStr.empty()) + errStr += XO("Error: %s").Format(pmErrStr); + // XXX: we are in libaudacity, popping up dialogs not allowed! A + // long-term solution will probably involve exceptions + AudacityMessageBox( + errStr, + XO("Error Initializing Midi"), + wxICON_ERROR|wxOK); // Same logic for PortMidi as described above for PortAudio } @@ -1253,13 +1252,13 @@ bool AudioIO::StartPortAudioStream(const AudioIOStartStreamOptions &options, mRate = GetBestRate(numCaptureChannels > 0, numPlaybackChannels > 0, sampleRate); // July 2016 (Carsten and Uwe) - // BUG 193: Tell PortAudio sound card will handle 24 bit (under DirectSound) using + // BUG 193: Tell PortAudio sound card will handle 24 bit (under DirectSound) using // userData. int captureFormat_saved = captureFormat; // Special case: Our 24-bit sample format is different from PortAudio's // 3-byte packed format. So just make PortAudio return float samples, // since we need float values anyway to apply the gain. - // ANSWER-ME: So we *never* actually handle 24-bit?! This causes mCapture to + // ANSWER-ME: So we *never* actually handle 24-bit?! This causes mCapture to // be set to floatSample below. // JKC: YES that's right. Internally Audacity uses float, and float has space for // 24 bits as well as exponent. Actual 24 bit would require packing and @@ -1358,7 +1357,7 @@ bool AudioIO::StartPortAudioStream(const AudioIOStartStreamOptions &options, #endif // July 2016 (Carsten and Uwe) - // BUG 193: Possibly tell portAudio to use 24 bit with DirectSound. + // BUG 193: Possibly tell portAudio to use 24 bit with DirectSound. int userData = 24; int* lpUserData = (captureFormat_saved == int24Sample) ? &userData : NULL; @@ -1473,7 +1472,7 @@ void AudioIO::StartMonitoring( const AudioIOStartStreamOptions &options ) // FIXME: TRAP_ERR PaErrorCode 'noted' but not reported in StartMonitoring. // Now start the PortAudio stream! - // TODO: ? Factor out and reuse error reporting code from end of + // TODO: ? Factor out and reuse error reporting code from end of // AudioIO::StartStream? mLastPaError = Pa_StartStream( mPortStreamV19 ); @@ -1706,7 +1705,7 @@ int AudioIO::StartStream(const TransportTracks &tracks, mPlaybackMixers[ii]->Reposition( time ); mPlaybackSchedule.RealTimeInit( time ); } - + // Now that we are done with SetTrackTime(): mTimeQueue.mLastTime = mPlaybackSchedule.GetTrackTime(); if (mTimeQueue.mData) @@ -1857,7 +1856,7 @@ bool AudioIO::AllocateBuffers( // more frequent polling of the mouse playbackTime = lrint(options.pScrubbingOptions->delay * mRate) / mRate; - + wxASSERT( playbackTime >= 0 ); mPlaybackSamplesToCopy = playbackTime * mRate; @@ -2005,7 +2004,7 @@ bool AudioIO::AllocateBuffers( } } } while(!bDone); - + success = true; return true; } @@ -2179,7 +2178,7 @@ void AudioIO::StopStream() // Re-enable system sleep wxPowerResource::Release(wxPOWER_RESOURCE_SCREEN); #endif - + if( mAudioThreadFillBuffersLoopRunning) { // PortAudio callback can use the information that we are stopping to fade @@ -2187,7 +2186,7 @@ void AudioIO::StopStream() mAudioThreadFillBuffersLoopRunning = false; auto latency = static_cast(AudioIOLatencyDuration.Read()); // If we can gracefully fade out in 200ms, with the faded-out play buffers making it through - // the sound card, then do so. If we can't, don't wait around. Just stop quickly and accept + // the sound card, then do so. If we can't, don't wait around. Just stop quickly and accept // there will be a click. if( mbMicroFades && (latency < 150 )) wxMilliSleep( latency + 50); @@ -2305,7 +2304,7 @@ void AudioIO::StopStream() #endif auto pListener = GetListener(); - + // If there's no token, we were just monitoring, so we can // skip this next part... if (mStreamToken > 0) { @@ -2373,7 +2372,7 @@ void AudioIO::StopStream() } ); } - + if (!mLostCaptureIntervals.empty()) { // This scope may combine many splittings of wave tracks @@ -2427,7 +2426,7 @@ void AudioIO::StopStream() e.SetInt(false); wxTheApp->ProcessEvent(e); } - + if (mNumCaptureChannels > 0) { wxCommandEvent e(wasMonitoring ? EVT_AUDIOIO_MONITOR : EVT_AUDIOIO_CAPTURE); @@ -2820,7 +2819,7 @@ void AudioIO::FillBuffers() // wxASSERT(put == frames); // but we can't assert in this thread wxUnusedVar(put); - } + } } available -= frames; @@ -3274,7 +3273,7 @@ void AudioIoCallback::GetNextEvent() if (mNextEvent) { mNextEventTime = (mNextIsNoteOn ? mNextEvent->time : mNextEvent->get_end_time()) + nextOffset;; - } + } if (mNextEventTime > (mPlaybackSchedule.mT1 + midiLoopOffset)){ // terminate playback at mT1 mNextEvent = &gAllNotesOff; mNextEventTime = mPlaybackSchedule.mT1 + midiLoopOffset - ALG_EPS; @@ -3824,7 +3823,7 @@ bool AudioIoCallback::FillOutputBuffers( return false; if( !outputBuffer ) return false; - if(numPlaybackChannels <= 0) + if(numPlaybackChannels <= 0) return false; float *outputFloats = (float *)outputBuffer; @@ -3855,7 +3854,7 @@ bool AudioIoCallback::FillOutputBuffers( // The drop and dropQuickly booleans are so named for historical reasons. // JKC: The original code attempted to be faster by doing nothing on silenced audio. // This, IMHO, is 'premature optimisation'. Instead clearer and cleaner code would - // simply use a gain of 0.0 for silent audio and go on through to the stage of + // simply use a gain of 0.0 for silent audio and go on through to the stage of // applying that 0.0 gain to the data mixed into the buffer. // Then (and only then) we would have if needed fast paths for: // - Applying a uniform gain of 0.0. @@ -3900,7 +3899,7 @@ bool AudioIoCallback::FillOutputBuffers( if (dropQuickly) { len = mPlaybackBuffers[t]->Discard(toGet); - // keep going here. + // keep going here. // we may still need to issue a paComplete. } else { const auto ptrToSample = (samplePtr)bufHelper.get()->tempBufs[chanCnt]; @@ -4000,7 +3999,7 @@ void AudioIoCallback::UpdateTimePosition(unsigned long framesPerBuffer) // Copy from PortAudio to our input buffers. // void AudioIoCallback::FillInputBuffers( - const void *inputBuffer, + const void *inputBuffer, unsigned long framesPerBuffer, const PaStreamCallbackFlags statusFlags, float * tempFloats @@ -4071,10 +4070,10 @@ void AudioIoCallback::FillInputBuffers( wxPrintf(wxT("lost %d samples\n"), (int)(framesPerBuffer - len)); } - if (len <= 0) + if (len <= 0) return; - // We have an ASSERT in the AudioIO constructor to alert us to + // We have an ASSERT in the AudioIO constructor to alert us to // possible issues with the (short*) cast. We'd have a problem if // sizeof(short) > sizeof(float) since our buffers are sized for floats. for(unsigned t = 0; t < numCaptureChannels; t++) { @@ -4111,7 +4110,7 @@ void AudioIoCallback::FillInputBuffers( } // switch // JKC: mCaptureFormat must be for samples with sizeof(float) or - // fewer bytes (because tempFloats is sized for floats). All + // fewer bytes (because tempFloats is sized for floats). All // formats are 2 or 4 bytes, so we are OK. const auto put = mCaptureBuffers[t]->Put( @@ -4157,7 +4156,7 @@ void OldCodeToCalculateLatency() // return true, IFF we have fully handled the callback. // Prime the output buffer with 0's, optionally adding in the playthrough. void AudioIoCallback::DoPlaythrough( - const void *inputBuffer, + const void *inputBuffer, void *outputBuffer, unsigned long framesPerBuffer, float *outputMeterFloats @@ -4194,7 +4193,7 @@ void AudioIoCallback::DoPlaythrough( // Also computes rms void AudioIoCallback::SendVuInputMeterData( float *inputSamples, - unsigned long framesPerBuffer + unsigned long framesPerBuffer ) { const auto numCaptureChannels = mNumCaptureChannels; @@ -4235,7 +4234,7 @@ void AudioIoCallback::SendVuOutputMeterData( return; if( mOutputMeter->IsMeterDisabled() ) return; - if( !outputMeterFloats) + if( !outputMeterFloats) return; // Get here if playback meter is live @@ -4289,7 +4288,7 @@ unsigned AudioIoCallback::CountSoloingTracks(){ // TODO: Consider making the two Track status functions into functions of // WaveTrack. -// true IFF the track should be silent. +// true IFF the track should be silent. // The track may not yet be silent, since it may still be // fading out. bool AudioIoCallback::TrackShouldBeSilent( const WaveTrack &wt ) @@ -4320,8 +4319,8 @@ bool AudioIoCallback::AllTracksAlreadySilent() const bool dropAllQuickly = std::all_of( mPlaybackTracks.begin(), mPlaybackTracks.end(), [&]( const std::shared_ptr< WaveTrack > &vt ) - { return - TrackShouldBeSilent( *vt ) && + { return + TrackShouldBeSilent( *vt ) && TrackHasBeenFadedOut( *vt ); } ); return dropAllQuickly; @@ -4351,9 +4350,9 @@ int AudioIoCallback::AudioCallback(const void *inputBuffer, void *outputBuffer, // but it does nothing unless we have EXPERIMENTAL_MIDI_OUT // TODO: Possibly rename variables to make it clearer which ones are MIDI specific // and which ones affect all audio. - ComputeMidiTimings( - timeInfo, - framesPerBuffer + ComputeMidiTimings( + timeInfo, + framesPerBuffer ); #ifndef USE_MIDI_THREAD if (mMidiStream) @@ -4369,10 +4368,10 @@ int AudioIoCallback::AudioCallback(const void *inputBuffer, void *outputBuffer, float *tempFloats = (float *)alloca(framesPerBuffer*sizeof(float)* MAX(numCaptureChannels,numPlaybackChannels)); - bool bVolEmulationActive = + bool bVolEmulationActive = (outputBuffer && mEmulateMixerOutputVol && mMixerOutputVol != 1.0); - // outputMeterFloats is the scratch pad for the output meter. - // we can often reuse the existing outputBuffer and save on allocating + // outputMeterFloats is the scratch pad for the output meter. + // we can often reuse the existing outputBuffer and save on allocating // something new. float *outputMeterFloats = bVolEmulationActive ? (float *)alloca(framesPerBuffer*numPlaybackChannels * sizeof(float)) : @@ -4397,10 +4396,10 @@ int AudioIoCallback::AudioCallback(const void *inputBuffer, void *outputBuffer, // This function may queue up a pause or resume. // TODO this is a bit dodgy as it toggles the Pause, and - // relies on an idle event to have handled that, so could + // relies on an idle event to have handled that, so could // queue up multiple toggle requests and so do nothing. // Eventually it will sort itself out by random luck, but - // the net effect is a delay in starting/stopping sound activated + // the net effect is a delay in starting/stopping sound activated // recording. CheckSoundActivatedRecordingLevel( inputSamples, @@ -4411,7 +4410,7 @@ int AudioIoCallback::AudioCallback(const void *inputBuffer, void *outputBuffer, // Initialise output buffer to zero or to playthrough data. // Initialise output meter values. DoPlaythrough( - inputBuffer, + inputBuffer, outputBuffer, framesPerBuffer, outputMeterFloats); @@ -4433,7 +4432,7 @@ int AudioIoCallback::AudioCallback(const void *inputBuffer, void *outputBuffer, // To capture input into track (sound from microphone) FillInputBuffers( - inputBuffer, + inputBuffer, framesPerBuffer, statusFlags, tempFloats); @@ -4508,7 +4507,7 @@ void AudioIoCallback::CallbackCheckCompletion( done = mPlaybackSchedule.PlayingAtSpeed() // some leftover length allowed in this case || (mPlaybackSchedule.PlayingStraight() && len == 0); - if(!done) + if(!done) return; #ifdef EXPERIMENTAL_MIDI_OUT diff --git a/src/AudioIOBase.h b/src/AudioIOBase.h index 778c6420d..df9e1c6ea 100644 --- a/src/AudioIOBase.h +++ b/src/AudioIOBase.h @@ -11,6 +11,9 @@ Paul Licameli split from AudioIO.h #ifndef __AUDACITY_AUDIO_IO_BASE__ #define __AUDACITY_AUDIO_IO_BASE__ + + + #include #include #include @@ -35,7 +38,7 @@ class BoundedEnvelope; // Windows build needs complete type for parameter of wxWeakRef // class MeterPanelBase; #include "widgets/MeterPanelBase.h" -using PRCrossfadeData = std::vector< std::vector>; +using PRCrossfadeData = std::vector< std::vector < float > >; #define BAD_STREAM_TIME (-DBL_MAX) diff --git a/src/AudioIOBufferHelper.h b/src/AudioIOBufferHelper.h index 28fd75ec9..46d0a36cc 100644 --- a/src/AudioIOBufferHelper.h +++ b/src/AudioIOBufferHelper.h @@ -10,35 +10,34 @@ class AudioIOBufferHelper private: - unsigned int numPlaybackChannels; - unsigned long framesPerBuffer; + unsigned int numPlaybackChannels; + unsigned long framesPerBuffer; public: - WaveTrack** chans; - float** tempBufs; + WaveTrack** chans; + float** tempBufs; - AudioIOBufferHelper(const unsigned int numPlaybackChannels, const unsigned long framesPerBuffer) { - this->numPlaybackChannels = numPlaybackChannels; - this->framesPerBuffer = framesPerBuffer; + AudioIOBufferHelper(const unsigned int numPlaybackChannels, const unsigned long framesPerBuffer) { + this->numPlaybackChannels = numPlaybackChannels; + this->framesPerBuffer = framesPerBuffer; - this->chans = safenew WaveTrack * [numPlaybackChannels]; - this->tempBufs = safenew float* [numPlaybackChannels]; + this->chans = safenew WaveTrack * [numPlaybackChannels]; + this->tempBufs = safenew float* [numPlaybackChannels]; - tempBufs[0] = safenew float[(size_t)numPlaybackChannels * framesPerBuffer]; - memset(tempBufs[0], 0, (size_t)numPlaybackChannels * (size_t)framesPerBuffer * sizeof(float)); + tempBufs[0] = safenew float[(size_t)numPlaybackChannels * framesPerBuffer]; + memset(tempBufs[0], 0, (size_t)numPlaybackChannels * (size_t)framesPerBuffer * sizeof(float)); - for (unsigned int c = 1; c < numPlaybackChannels; c++) { - tempBufs[c] = tempBufs[c - 1] + framesPerBuffer; - } - } + for (unsigned int c = 1; c < numPlaybackChannels; c++) { + tempBufs[c] = tempBufs[c - 1] + framesPerBuffer; + } + } - ~AudioIOBufferHelper() { - - delete[] tempBufs[0]; - delete[] tempBufs; - delete[] chans; - } + ~AudioIOBufferHelper() { + delete[] tempBufs[0]; + delete[] tempBufs; + delete[] chans; + } }; #endif