mirror of
https://github.com/cookiengineer/audacity
synced 2025-07-25 08:58:06 +02:00
Remove whitespace only changes
Signed-off-by: Emily Mabrey <emabrey@tenacityaudio.org>
This commit is contained in:
Emily Mabrey
parent
d4a560f6f3
commit
d7695f3c91
137
src/AudioIO.cpp
137
src/AudioIO.cpp
@ -18,8 +18,8 @@
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********************************************************************//**
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\class AudioIoCallback
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\brief AudioIoCallback is a class that implements the callback required
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by PortAudio. The callback needs to be responsive, has no GUI, and
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\brief AudioIoCallback is a class that implements the callback required
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by PortAudio. The callback needs to be responsive, has no GUI, and
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copies data into and out of the sound card buffers. It also sends data
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to the meters.
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@ -408,7 +408,7 @@ callbacks for these events.
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*//****************************************************************//**
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\class AudioIOStartStreamOptions
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\brief struct holding stream options, including a pointer to the
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\brief struct holding stream options, including a pointer to the
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time warp info and AudioIOListener and whether the playback is looped.
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*//*******************************************************************/
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@ -961,7 +961,7 @@ AudioIO::AudioIO()
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wxASSERT(false);
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}
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// This ASSERT because of casting in the callback
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// This ASSERT because of casting in the callback
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// functions where we cast a tempFloats buffer to a (short*) buffer.
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// We have to ASSERT in the GUI thread, if we are to see it properly.
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wxASSERT( sizeof( short ) <= sizeof( float ));
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@ -986,8 +986,7 @@ AudioIO::AudioIO()
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#ifdef EXPERIMENTAL_AUTOMATED_INPUT_LEVEL_ADJUSTMENT
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mAILAActive = false;
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#endif
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mStreamToken = 0;
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mStreamToken = 0;
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mLastPaError = paNoError;
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@ -1004,18 +1003,18 @@ AudioIO::AudioIO()
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PaError err = Pa_Initialize();
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if (err != paNoError) {
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auto errStr = XO("Could not find any audio devices.\n");
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errStr += XO("You will not be able to play or record audio.\n\n");
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const wxString paErrStr = LAT1CTOWX(Pa_GetErrorText(err));
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if (!paErrStr.empty())
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errStr += XO("Error: %s").Format(paErrStr);
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// XXX: we are in libaudacity, popping up dialogs not allowed! A
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// long-term solution will probably involve exceptions
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AudacityMessageBox(
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errStr,
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XO("Error Initializing Audio"),
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wxICON_ERROR | wxOK);
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if (err != paNoError) {
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auto errStr = XO("Could not find any audio devices.\n");
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errStr += XO("You will not be able to play or record audio.\n\n");
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const wxString paErrStr = LAT1CTOWX(Pa_GetErrorText(err));
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if (!paErrStr.empty())
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errStr += XO("Error: %s").Format(paErrStr);
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// XXX: we are in libaudacity, popping up dialogs not allowed! A
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// long-term solution will probably involve exceptions
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AudacityMessageBox(
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errStr,
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XO("Error Initializing Audio"),
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wxICON_ERROR|wxOK);
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// Since PortAudio is not initialized, all calls to PortAudio
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// functions will fail. This will give reasonable behavior, since
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@ -1026,19 +1025,19 @@ AudioIO::AudioIO()
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#ifdef EXPERIMENTAL_MIDI_OUT
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PmError pmErr = Pm_Initialize();
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if (pmErr != pmNoError) {
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auto errStr =
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XO("There was an error initializing the midi i/o layer.\n");
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errStr += XO("You will not be able to play midi.\n\n");
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const wxString pmErrStr = LAT1CTOWX(Pm_GetErrorText(pmErr));
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if (!pmErrStr.empty())
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errStr += XO("Error: %s").Format(pmErrStr);
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// XXX: we are in libaudacity, popping up dialogs not allowed! A
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// long-term solution will probably involve exceptions
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AudacityMessageBox(
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errStr,
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XO("Error Initializing Midi"),
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wxICON_ERROR | wxOK);
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if (pmErr != pmNoError) {
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auto errStr =
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XO("There was an error initializing the midi i/o layer.\n");
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errStr += XO("You will not be able to play midi.\n\n");
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const wxString pmErrStr = LAT1CTOWX(Pm_GetErrorText(pmErr));
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if (!pmErrStr.empty())
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errStr += XO("Error: %s").Format(pmErrStr);
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// XXX: we are in libaudacity, popping up dialogs not allowed! A
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// long-term solution will probably involve exceptions
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AudacityMessageBox(
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errStr,
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XO("Error Initializing Midi"),
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wxICON_ERROR|wxOK);
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// Same logic for PortMidi as described above for PortAudio
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}
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@ -1253,13 +1252,13 @@ bool AudioIO::StartPortAudioStream(const AudioIOStartStreamOptions &options,
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mRate = GetBestRate(numCaptureChannels > 0, numPlaybackChannels > 0, sampleRate);
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// July 2016 (Carsten and Uwe)
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// BUG 193: Tell PortAudio sound card will handle 24 bit (under DirectSound) using
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// BUG 193: Tell PortAudio sound card will handle 24 bit (under DirectSound) using
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// userData.
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int captureFormat_saved = captureFormat;
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// Special case: Our 24-bit sample format is different from PortAudio's
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// 3-byte packed format. So just make PortAudio return float samples,
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// since we need float values anyway to apply the gain.
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// ANSWER-ME: So we *never* actually handle 24-bit?! This causes mCapture to
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// ANSWER-ME: So we *never* actually handle 24-bit?! This causes mCapture to
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// be set to floatSample below.
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// JKC: YES that's right. Internally Audacity uses float, and float has space for
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// 24 bits as well as exponent. Actual 24 bit would require packing and
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@ -1358,7 +1357,7 @@ bool AudioIO::StartPortAudioStream(const AudioIOStartStreamOptions &options,
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#endif
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// July 2016 (Carsten and Uwe)
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// BUG 193: Possibly tell portAudio to use 24 bit with DirectSound.
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// BUG 193: Possibly tell portAudio to use 24 bit with DirectSound.
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int userData = 24;
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int* lpUserData = (captureFormat_saved == int24Sample) ? &userData : NULL;
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@ -1473,7 +1472,7 @@ void AudioIO::StartMonitoring( const AudioIOStartStreamOptions &options )
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// FIXME: TRAP_ERR PaErrorCode 'noted' but not reported in StartMonitoring.
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// Now start the PortAudio stream!
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// TODO: ? Factor out and reuse error reporting code from end of
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// TODO: ? Factor out and reuse error reporting code from end of
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// AudioIO::StartStream?
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mLastPaError = Pa_StartStream( mPortStreamV19 );
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@ -1706,7 +1705,7 @@ int AudioIO::StartStream(const TransportTracks &tracks,
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mPlaybackMixers[ii]->Reposition( time );
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mPlaybackSchedule.RealTimeInit( time );
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}
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// Now that we are done with SetTrackTime():
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mTimeQueue.mLastTime = mPlaybackSchedule.GetTrackTime();
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if (mTimeQueue.mData)
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@ -1857,7 +1856,7 @@ bool AudioIO::AllocateBuffers(
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// more frequent polling of the mouse
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playbackTime =
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lrint(options.pScrubbingOptions->delay * mRate) / mRate;
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wxASSERT( playbackTime >= 0 );
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mPlaybackSamplesToCopy = playbackTime * mRate;
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@ -2005,7 +2004,7 @@ bool AudioIO::AllocateBuffers(
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}
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}
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} while(!bDone);
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success = true;
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return true;
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}
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@ -2179,7 +2178,7 @@ void AudioIO::StopStream()
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// Re-enable system sleep
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wxPowerResource::Release(wxPOWER_RESOURCE_SCREEN);
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#endif
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if( mAudioThreadFillBuffersLoopRunning)
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{
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// PortAudio callback can use the information that we are stopping to fade
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@ -2187,7 +2186,7 @@ void AudioIO::StopStream()
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mAudioThreadFillBuffersLoopRunning = false;
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auto latency = static_cast<long>(AudioIOLatencyDuration.Read());
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// If we can gracefully fade out in 200ms, with the faded-out play buffers making it through
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// the sound card, then do so. If we can't, don't wait around. Just stop quickly and accept
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// the sound card, then do so. If we can't, don't wait around. Just stop quickly and accept
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// there will be a click.
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if( mbMicroFades && (latency < 150 ))
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wxMilliSleep( latency + 50);
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@ -2305,7 +2304,7 @@ void AudioIO::StopStream()
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#endif
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auto pListener = GetListener();
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// If there's no token, we were just monitoring, so we can
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// skip this next part...
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if (mStreamToken > 0) {
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@ -2373,7 +2372,7 @@ void AudioIO::StopStream()
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} );
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}
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if (!mLostCaptureIntervals.empty())
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{
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// This scope may combine many splittings of wave tracks
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@ -2427,7 +2426,7 @@ void AudioIO::StopStream()
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e.SetInt(false);
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wxTheApp->ProcessEvent(e);
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}
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if (mNumCaptureChannels > 0)
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{
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wxCommandEvent e(wasMonitoring ? EVT_AUDIOIO_MONITOR : EVT_AUDIOIO_CAPTURE);
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@ -2820,7 +2819,7 @@ void AudioIO::FillBuffers()
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// wxASSERT(put == frames);
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// but we can't assert in this thread
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wxUnusedVar(put);
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}
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}
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}
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available -= frames;
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@ -3274,7 +3273,7 @@ void AudioIoCallback::GetNextEvent()
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if (mNextEvent) {
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mNextEventTime = (mNextIsNoteOn ? mNextEvent->time :
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mNextEvent->get_end_time()) + nextOffset;;
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}
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}
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if (mNextEventTime > (mPlaybackSchedule.mT1 + midiLoopOffset)){ // terminate playback at mT1
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mNextEvent = &gAllNotesOff;
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mNextEventTime = mPlaybackSchedule.mT1 + midiLoopOffset - ALG_EPS;
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@ -3824,7 +3823,7 @@ bool AudioIoCallback::FillOutputBuffers(
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return false;
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if( !outputBuffer )
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return false;
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if(numPlaybackChannels <= 0)
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if(numPlaybackChannels <= 0)
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return false;
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float *outputFloats = (float *)outputBuffer;
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@ -3855,7 +3854,7 @@ bool AudioIoCallback::FillOutputBuffers(
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// The drop and dropQuickly booleans are so named for historical reasons.
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// JKC: The original code attempted to be faster by doing nothing on silenced audio.
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// This, IMHO, is 'premature optimisation'. Instead clearer and cleaner code would
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// simply use a gain of 0.0 for silent audio and go on through to the stage of
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// simply use a gain of 0.0 for silent audio and go on through to the stage of
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// applying that 0.0 gain to the data mixed into the buffer.
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// Then (and only then) we would have if needed fast paths for:
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// - Applying a uniform gain of 0.0.
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@ -3900,7 +3899,7 @@ bool AudioIoCallback::FillOutputBuffers(
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if (dropQuickly) {
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len = mPlaybackBuffers[t]->Discard(toGet);
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// keep going here.
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// keep going here.
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// we may still need to issue a paComplete.
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} else {
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const auto ptrToSample = (samplePtr)bufHelper.get()->tempBufs[chanCnt];
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@ -4000,7 +3999,7 @@ void AudioIoCallback::UpdateTimePosition(unsigned long framesPerBuffer)
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// Copy from PortAudio to our input buffers.
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//
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void AudioIoCallback::FillInputBuffers(
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const void *inputBuffer,
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const void *inputBuffer,
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unsigned long framesPerBuffer,
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const PaStreamCallbackFlags statusFlags,
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float * tempFloats
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@ -4071,10 +4070,10 @@ void AudioIoCallback::FillInputBuffers(
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wxPrintf(wxT("lost %d samples\n"), (int)(framesPerBuffer - len));
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}
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if (len <= 0)
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if (len <= 0)
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return;
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// We have an ASSERT in the AudioIO constructor to alert us to
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// We have an ASSERT in the AudioIO constructor to alert us to
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// possible issues with the (short*) cast. We'd have a problem if
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// sizeof(short) > sizeof(float) since our buffers are sized for floats.
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for(unsigned t = 0; t < numCaptureChannels; t++) {
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@ -4111,7 +4110,7 @@ void AudioIoCallback::FillInputBuffers(
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} // switch
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// JKC: mCaptureFormat must be for samples with sizeof(float) or
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// fewer bytes (because tempFloats is sized for floats). All
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// fewer bytes (because tempFloats is sized for floats). All
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// formats are 2 or 4 bytes, so we are OK.
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const auto put =
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mCaptureBuffers[t]->Put(
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@ -4157,7 +4156,7 @@ void OldCodeToCalculateLatency()
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// return true, IFF we have fully handled the callback.
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// Prime the output buffer with 0's, optionally adding in the playthrough.
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void AudioIoCallback::DoPlaythrough(
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const void *inputBuffer,
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const void *inputBuffer,
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void *outputBuffer,
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unsigned long framesPerBuffer,
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float *outputMeterFloats
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@ -4194,7 +4193,7 @@ void AudioIoCallback::DoPlaythrough(
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// Also computes rms
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void AudioIoCallback::SendVuInputMeterData(
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float *inputSamples,
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unsigned long framesPerBuffer
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unsigned long framesPerBuffer
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)
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{
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const auto numCaptureChannels = mNumCaptureChannels;
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@ -4235,7 +4234,7 @@ void AudioIoCallback::SendVuOutputMeterData(
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return;
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if( mOutputMeter->IsMeterDisabled() )
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return;
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if( !outputMeterFloats)
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if( !outputMeterFloats)
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return;
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// Get here if playback meter is live
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@ -4289,7 +4288,7 @@ unsigned AudioIoCallback::CountSoloingTracks(){
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// TODO: Consider making the two Track status functions into functions of
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// WaveTrack.
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// true IFF the track should be silent.
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// true IFF the track should be silent.
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// The track may not yet be silent, since it may still be
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// fading out.
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bool AudioIoCallback::TrackShouldBeSilent( const WaveTrack &wt )
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@ -4320,8 +4319,8 @@ bool AudioIoCallback::AllTracksAlreadySilent()
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const bool dropAllQuickly = std::all_of(
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mPlaybackTracks.begin(), mPlaybackTracks.end(),
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[&]( const std::shared_ptr< WaveTrack > &vt )
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{ return
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TrackShouldBeSilent( *vt ) &&
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{ return
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TrackShouldBeSilent( *vt ) &&
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TrackHasBeenFadedOut( *vt ); }
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);
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return dropAllQuickly;
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@ -4351,9 +4350,9 @@ int AudioIoCallback::AudioCallback(const void *inputBuffer, void *outputBuffer,
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// but it does nothing unless we have EXPERIMENTAL_MIDI_OUT
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// TODO: Possibly rename variables to make it clearer which ones are MIDI specific
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// and which ones affect all audio.
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ComputeMidiTimings(
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timeInfo,
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framesPerBuffer
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ComputeMidiTimings(
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timeInfo,
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framesPerBuffer
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);
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#ifndef USE_MIDI_THREAD
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if (mMidiStream)
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@ -4369,10 +4368,10 @@ int AudioIoCallback::AudioCallback(const void *inputBuffer, void *outputBuffer,
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float *tempFloats = (float *)alloca(framesPerBuffer*sizeof(float)*
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MAX(numCaptureChannels,numPlaybackChannels));
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bool bVolEmulationActive =
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bool bVolEmulationActive =
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(outputBuffer && mEmulateMixerOutputVol && mMixerOutputVol != 1.0);
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// outputMeterFloats is the scratch pad for the output meter.
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// we can often reuse the existing outputBuffer and save on allocating
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// outputMeterFloats is the scratch pad for the output meter.
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// we can often reuse the existing outputBuffer and save on allocating
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// something new.
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float *outputMeterFloats = bVolEmulationActive ?
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(float *)alloca(framesPerBuffer*numPlaybackChannels * sizeof(float)) :
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@ -4397,10 +4396,10 @@ int AudioIoCallback::AudioCallback(const void *inputBuffer, void *outputBuffer,
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// This function may queue up a pause or resume.
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// TODO this is a bit dodgy as it toggles the Pause, and
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// relies on an idle event to have handled that, so could
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// relies on an idle event to have handled that, so could
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// queue up multiple toggle requests and so do nothing.
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// Eventually it will sort itself out by random luck, but
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// the net effect is a delay in starting/stopping sound activated
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// the net effect is a delay in starting/stopping sound activated
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// recording.
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CheckSoundActivatedRecordingLevel(
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inputSamples,
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@ -4411,7 +4410,7 @@ int AudioIoCallback::AudioCallback(const void *inputBuffer, void *outputBuffer,
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// Initialise output buffer to zero or to playthrough data.
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// Initialise output meter values.
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DoPlaythrough(
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inputBuffer,
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inputBuffer,
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outputBuffer,
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framesPerBuffer,
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outputMeterFloats);
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@ -4433,7 +4432,7 @@ int AudioIoCallback::AudioCallback(const void *inputBuffer, void *outputBuffer,
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// To capture input into track (sound from microphone)
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FillInputBuffers(
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inputBuffer,
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inputBuffer,
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framesPerBuffer,
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statusFlags,
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tempFloats);
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@ -4508,7 +4507,7 @@ void AudioIoCallback::CallbackCheckCompletion(
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done = mPlaybackSchedule.PlayingAtSpeed()
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// some leftover length allowed in this case
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|| (mPlaybackSchedule.PlayingStraight() && len == 0);
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if(!done)
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if(!done)
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return;
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#ifdef EXPERIMENTAL_MIDI_OUT
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@ -11,6 +11,9 @@ Paul Licameli split from AudioIO.h
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#ifndef __AUDACITY_AUDIO_IO_BASE__
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#define __AUDACITY_AUDIO_IO_BASE__
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#include <cfloat>
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#include <functional>
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#include <vector>
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@ -35,7 +38,7 @@ class BoundedEnvelope;
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// Windows build needs complete type for parameter of wxWeakRef
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// class MeterPanelBase;
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#include "widgets/MeterPanelBase.h"
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using PRCrossfadeData = std::vector< std::vector<float>>;
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using PRCrossfadeData = std::vector< std::vector < float > >;
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#define BAD_STREAM_TIME (-DBL_MAX)
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@ -10,35 +10,34 @@ class AudioIOBufferHelper
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private:
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unsigned int numPlaybackChannels;
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unsigned long framesPerBuffer;
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unsigned int numPlaybackChannels;
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unsigned long framesPerBuffer;
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public:
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WaveTrack** chans;
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float** tempBufs;
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WaveTrack** chans;
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float** tempBufs;
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||||
|
||||
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
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user