--- pyepl-1.1.0.orig/changelog.txt +++ pyepl-1.1.0/changelog.txt @@ -1,4 +1,9 @@ +Version ?.?.? +------------- + - Fixed a sound bug that cleared the recording buffer when you played + a sound while trying to record. + - Added ability to flush the recording buffer. Version 1.1.0 --- pyepl-1.1.0.orig/code/sound.py +++ pyepl-1.1.0/code/sound.py @@ -532,6 +532,7 @@ Clean up the sound system. """ self.playStop() + self.stopRecording() self.eplsound.stopstream() def play(self, soundClip, t = None, ampFactor=1.0, doDelay=True): @@ -630,7 +631,7 @@ # see if stop the time if self.startInd < self.endInd: - # determin how much to append + # determine how much to append actualInd = self.startInd + self.bytes_per_append # make sure it's not beyond the end @@ -639,7 +640,8 @@ actualInd = self.endInd # append the sound - appended = self.eplsound.append(s[self.startInd:actualInd], len(s[self.startInd:actualInd])/self.eplsound.FORMAT_SIZE, 0, ampFactor) + appended = self.eplsound.append(s[self.startInd:actualInd], + len(s[self.startInd:actualInd])/self.eplsound.FORMAT_SIZE, 0, ampFactor) self.last_play = currentTime @@ -653,26 +655,16 @@ self.playStop() def playLoopStop(self, doUnload=True): - if self.playing: - self.playing = False - removePollCallback(self.__playLoopCallback__) - - # clear the sound buffer to stop playing - self.eplsound.clear() - - if isinstance(self.currentClip, FileAudioClip) and self.currentClip.isLoaded() and doUnload: - self.currentClip.unload() - - return self.eplsound.getSamplesPlayed() - + self.playStop(doUnload) def playStop(self, doUnload=True): if self.playing: self.playing = False removePollCallback(self.__playCallback__) + removePollCallback(self.__playLoopCallback__) # clear the sound buffer to stop playing - self.eplsound.clear() + self.eplsound.clearPlayBuffer() if isinstance(self.currentClip, FileAudioClip) and self.currentClip.isLoaded() and doUnload: self.currentClip.unload() @@ -849,6 +841,20 @@ return (self.recClip,timeInterval) + def flush(self): + """ + Flush the recording buffer. + """ + currentTime = timing.now() + newstuff = self.getBuffData() + + # Update the last time + self.last_rec = currentTime + + if len(newstuff) > 0: + # append the data to the clip + self.recClip.append(newstuff, self.eplsound.getRecChans()) + def __recCallback__(self): """ --- pyepl-1.1.0.orig/code/display.py +++ pyepl-1.1.0/code/display.py @@ -62,7 +62,9 @@ self.loadRequireCount -= 1 if self.loadRequireCount < 0: self.loadRequireCount = 0 - if self.loadRequireCount == 0 and self.isLoaded(): + if self.loadRequireCount == 0 and \ + self.isLoaded() and \ + (hasattr(self,'filename') and not self.filename is None): self.unload() def show(self, x, y): # to be overridden """ @@ -915,7 +917,7 @@ return result # create the initial font defaults... -defaultFont = Font(os.path.join(pyepldir, "resources", "vera.ttf")) +defaultFont = Font("/usr/share/fonts/truetype/ttf-dejavu/DejaVuSans.ttf") defaultFontSize = 0.0625 defaultFontColor = Color("white") --- pyepl-1.1.0.orig/code/hardware/sound/RtAudio.cpp +++ pyepl-1.1.0/code/hardware/sound/RtAudio.cpp @@ -38,7 +38,7 @@ */ /************************************************************************/ -// RtAudio: Version 4.0.5 +// RtAudio: Version 4.0.6 #include "RtAudio.h" #include @@ -403,7 +403,9 @@ // implementation. struct CoreHandle { AudioDeviceID id[2]; // device ids +#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) AudioDeviceIOProcID procId[2]; +#endif UInt32 iStream[2]; // device stream index (or first if using multiple) UInt32 nStreams[2]; // number of streams to use bool xrun[2]; @@ -539,7 +541,7 @@ // Get the device name. info.name.erase(); - char name[256]; + char name[256]; dataSize = 256; result = AudioDeviceGetProperty( id, 0, false, kAudioDevicePropertyDeviceManufacturer, @@ -567,7 +569,7 @@ info.name.append( (const char *)name, strlen(name) ); // Get the output stream "configuration". - AudioBufferList *bufferList = nil; + AudioBufferList *bufferList = nil; result = AudioDeviceGetPropertyInfo( id, 0, false, kAudioDevicePropertyStreamConfiguration, &dataSize, NULL ); @@ -790,7 +792,7 @@ } // Get the stream "configuration". - AudioBufferList *bufferList; + AudioBufferList *bufferList; result = AudioDeviceGetPropertyInfo( id, 0, isInput, kAudioDevicePropertyStreamConfiguration, &dataSize, NULL ); @@ -884,7 +886,7 @@ free( bufferList ); // Determine the buffer size. - AudioValueRange bufferRange; + AudioValueRange bufferRange; dataSize = sizeof( AudioValueRange ); result = AudioDeviceGetProperty( id, 0, isInput, kAudioDevicePropertyBufferFrameSizeRange, @@ -940,7 +942,7 @@ // Now set the stream format. Also, check the physical format of the // device and change that if necessary. - AudioStreamBasicDescription description; + AudioStreamBasicDescription description; dataSize = sizeof( AudioStreamBasicDescription ); bool updateFormat; @@ -991,20 +993,31 @@ return FAILURE; } +// Snow Leopard / Leopard Fix ================================================= + +/**/description.mSampleRate = sampleRate; +/**/result = AudioDeviceSetProperty( streamIDs[firstStream+i], NULL, 0, isInput, +/**/ kAudioDevicePropertyStreamFormat, +/**/ dataSize, &description ); + if ( description.mFormatID != kAudioFormatLinearPCM || description.mBitsPerChannel < 24 ) { description.mFormatID = kAudioFormatLinearPCM; - AudioStreamBasicDescription testDescription = description; + AudioStreamBasicDescription testDescription = description; unsigned long formatFlags; // We'll try higher bit rates first and then work our way down. testDescription.mBitsPerChannel = 32; - formatFlags = description.mFormatFlags | kLinearPCMFormatFlagIsFloat & ~kLinearPCMFormatFlagIsSignedInteger; +/**/ testDescription.mBytesPerFrame = testDescription.mBitsPerChannel/8 * testDescription.mChannelsPerFrame; +/**/ testDescription.mBytesPerPacket = testDescription.mBytesPerFrame * testDescription.mFramesPerPacket; + formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsFloat) & ~kLinearPCMFormatFlagIsSignedInteger; testDescription.mFormatFlags = formatFlags; result = AudioStreamSetProperty( streamIDs[firstStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription ); if ( result == noErr ) continue; testDescription = description; testDescription.mBitsPerChannel = 32; +/**/ testDescription.mBytesPerFrame = testDescription.mBitsPerChannel/8 * testDescription.mChannelsPerFrame; +/**/ testDescription.mBytesPerPacket = testDescription.mBytesPerFrame * testDescription.mFramesPerPacket; formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger) & ~kLinearPCMFormatFlagIsFloat; testDescription.mFormatFlags = formatFlags; result = AudioStreamSetProperty( streamIDs[firstStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription ); @@ -1012,24 +1025,33 @@ testDescription = description; testDescription.mBitsPerChannel = 24; +/**/ testDescription.mBytesPerFrame = testDescription.mBitsPerChannel/8 * testDescription.mChannelsPerFrame; +/**/ testDescription.mBytesPerPacket = testDescription.mBytesPerFrame * testDescription.mFramesPerPacket; testDescription.mFormatFlags = formatFlags; result = AudioStreamSetProperty( streamIDs[firstStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription ); if ( result == noErr ) continue; testDescription = description; testDescription.mBitsPerChannel = 16; +/**/ testDescription.mBytesPerFrame = testDescription.mBitsPerChannel/8 * testDescription.mChannelsPerFrame; +/**/ testDescription.mBytesPerPacket = testDescription.mBytesPerFrame * testDescription.mFramesPerPacket; testDescription.mFormatFlags = formatFlags; result = AudioStreamSetProperty( streamIDs[firstStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription ); if ( result == noErr ) continue; testDescription = description; testDescription.mBitsPerChannel = 8; +/**/ testDescription.mBytesPerFrame = testDescription.mBitsPerChannel/8 * testDescription.mChannelsPerFrame; +/**/ testDescription.mBytesPerPacket = testDescription.mBytesPerFrame * testDescription.mFramesPerPacket; testDescription.mFormatFlags = formatFlags; result = AudioStreamSetProperty( streamIDs[firstStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription ); if ( result != noErr ) { errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting physical data format for device (" << device << ")."; errorText_ = errorStream_.str(); return FAILURE; + +// End Snow Leopard / Leopard Fix ============================================= + } } } @@ -5646,7 +5668,7 @@ } // Set the buffer number, which in ALSA is referred to as the "period". - int totalSize, dir; + int totalSize, dir = 0; unsigned int periods = 0; if ( options ) periods = options->numberOfBuffers; totalSize = *bufferSize * periods; --- pyepl-1.1.0.orig/code/hardware/sound/eplSound.cpp +++ pyepl-1.1.0/code/hardware/sound/eplSound.cpp @@ -19,6 +19,9 @@ unsigned int sampRate, unsigned int bufSize) { // get audio instance + RtAudio audio; + playAudio = NULL; + recAudio = NULL; // let's see warnings audio.showWarnings( true ); @@ -84,37 +87,53 @@ try { // open the proper stream - if ((playChans>0) && (recChans>0)) + if ((playChans>0) && (recChans>0) && (iParams.deviceId == oParams.deviceId)) { + // it's duplex + isDuplex = 1; + // open both - audio.openStream( &oParams, &iParams, FORMAT, sampleRate, - &bufferSize, &inout, (void *)data ); + playAudio = new RtAudio(); + playAudio->showWarnings( true ); + playAudio->openStream( &oParams, &iParams, FORMAT, sampleRate, + &bufferSize, &inout, (void *)data ); + recAudio = playAudio; } - else if (playChans>0) + else + { + // not full duplex + isDuplex = 0; + + if (playChans>0) + { + // open output + playAudio = new RtAudio(); + playAudio->showWarnings( true ); + playAudio->openStream( &oParams, NULL, FORMAT, sampleRate, + &bufferSize, &inout, (void *)data ); + } + if (recChans>0) + { + // open input + recAudio = new RtAudio(); + recAudio->showWarnings( true ); + recAudio->openStream( NULL, &iParams, FORMAT, sampleRate, + &bufferSize, &inout, (void *)data ); + } + } + + if (playChans==0) { - // open only output - audio.openStream( &oParams, NULL, FORMAT, sampleRate, - &bufferSize, &inout, (void *)data ); // No input chans discovered cerr << "No default input device with correct channel info was found!" << endl; cerr << "You will only be able to record sound." << endl; } - else if (recChans>0) + if (recChans==0) { - // open only input - audio.openStream( NULL, &iParams, FORMAT, sampleRate, - &bufferSize, &inout, (void *)data ); - // No output chans discovered cerr << "No default output device with correct channel info was found!" << endl; cerr << "You will not be able to play sound." << endl; } - else - { - cerr << "No default i/o device with correct channel info was found!" << endl; - cerr << "You will not be able to play or record sound." << endl; - //exit( EXIT_FAILURE ); - } } catch ( RtError& e ) { e.printMessage(); @@ -127,10 +146,27 @@ eplSound::~eplSound() { - if (audio.isStreamOpen() > 0) + // stop streaming + stopstream(1); + + // close the play stream + if (playAudio != NULL) { - stopstream(); - audio.closeStream(); + if (playAudio->isStreamOpen() > 0) + playAudio->closeStream(); + + // clean it up + delete playAudio; + } + + // see if should clean up record stream + if ((!isDuplex) && (recAudio != NULL)) + { + if (recAudio->isStreamOpen() > 0) + recAudio->closeStream(); + + // clean it up + delete recAudio; } delete data; @@ -143,6 +179,18 @@ return; } +void eplSound::clearPlayBuffer() +{ + data->playBuf->clear(); + return; +} + +void eplSound::clearRecBuffer() +{ + data->recBuf->clear(); + return; +} + int eplSound::recstart() { @@ -171,31 +219,71 @@ //if (!(audio.isStreamOpen()) && (streaming == 0)) if (streaming == 0) { - try { - audio.startStream(); + // start playing + if (playAudio != NULL) + { + try { + playAudio->startStream(); + } + catch (RtError &error) { + error.printMessage(); + exit(EXIT_FAILURE); + } } - catch (RtError &error) { - error.printMessage(); - exit(EXIT_FAILURE); + + if ((recAudio != NULL) && (recAudio != playAudio)) + { + try { + recAudio->startStream(); + } + catch (RtError &error) { + error.printMessage(); + exit(EXIT_FAILURE); + } } + + // say we are streaming streaming = 1; } return 0; } -int eplSound::stopstream() +int eplSound::stopstream(int abort) { //if (audio.isStreamOpen() && streaming == 1)) if (streaming == 1) { - try { - audio.stopStream(); + // stop playing + if (playAudio != NULL) + { + try { + if (abort) + playAudio->abortStream(); + else + playAudio->stopStream(); + } + catch (RtError &error) { + error.printMessage(); + exit(EXIT_FAILURE); + } } - catch (RtError &error) { - error.printMessage(); - exit(EXIT_FAILURE); + + // stop playing + if ((recAudio != NULL) && (recAudio != playAudio)) + { + try { + if (abort) + recAudio->abortStream(); + else + recAudio->stopStream(); + } + catch (RtError &error) { + error.printMessage(); + exit(EXIT_FAILURE); + } } + streaming = 0; data->recording = 0; } @@ -256,11 +344,31 @@ return sampleRate; } +unsigned int eplSound::getPlayStreamSampleRate() +{ + return playAudio->getStreamSampleRate(); +} + +unsigned int eplSound::getRecStreamSampleRate() +{ + return recAudio->getStreamSampleRate(); +} + long eplSound::getBufferUsed() { return data->playBuf->getUsed(); } +long eplSound::getPlayStreamLatency() +{ + return playAudio->getStreamLatency(); +} + +long eplSound::getRecStreamLatency() +{ + return recAudio->getStreamLatency(); +} + int inout( void *outputBuffer, void *inputBuffer, unsigned int nBufferFrames, double streamTime, RtAudioStreamStatus status, void *data ) { @@ -297,7 +405,7 @@ sizeof(MY_TYPE)*((nBufferFrames*mydata->playChans)-written)); } } - + return 0; } --- pyepl-1.1.0.orig/code/hardware/sound/RtAudio.h +++ pyepl-1.1.0/code/hardware/sound/RtAudio.h @@ -42,7 +42,7 @@ \file RtAudio.h */ -// RtAudio: Version 4.0.5 +// RtAudio: Version 4.0.6 #ifndef __RTAUDIO_H #define __RTAUDIO_H --- pyepl-1.1.0.orig/code/hardware/sound/eplSound.h +++ pyepl-1.1.0/code/hardware/sound/eplSound.h @@ -49,17 +49,23 @@ long append(MY_TYPE *newdata, long length, int overwrite, float ampFactor); long consume(MY_TYPE *newdata, long length); void clear(); + void clearPlayBuffer(); + void clearRecBuffer(); int recstart(); int recstop(); int startstream(); - int stopstream(); + int stopstream(int abort=0); int getBufferSize(); long getSamplesPlayed(); void resetSamplesPlayed(); int getRecChans(); int getPlayChans(); int getSampleRate(); + unsigned int getPlayStreamSampleRate(); + unsigned int getRecStreamSampleRate(); long getBufferUsed(); + long getPlayStreamLatency(); + long getRecStreamLatency(); static const int SCALE=32767; static const unsigned int SAMPLE_RATE=44100; static const int SAMPLE_SILENCE=0; @@ -71,10 +77,10 @@ static const long REC_BUF_LEN=60; static const int NUM_CHANNELS=2; private: - RtAudio audio; - //RtAudio *playaudio; - //RtAudio *recaudio; - //int isDuplex; + //RtAudio audio; + RtAudio *playAudio; + RtAudio *recAudio; + int isDuplex; unsigned int playChans; unsigned int recChans; //int rate; --- pyepl-1.1.0.orig/code/hardware/vr/Makefile +++ pyepl-1.1.0/code/hardware/vr/Makefile @@ -18,11 +18,14 @@ ifeq ($(PLATFORM), Linux) +ODE_CFLAGS = $(shell ode-config --cflags) +ODE_LDFLAGS = $(shell ode-config --libs) + avatar.so: avatar.o - gcc -shared avatar.o -o avatar.so + gcc -shared avatar.o -o avatar.so $(ODE_LDFLAGS) avatar.o: avatar.c - gcc -c -fPIC -I$(INCLUDEPY) avatar.c + gcc -c -fPIC -I$(INCLUDEPY) $(ODE_CFLAGS) avatar.c avatar.c: avatar.pyx pyrexc avatar.pyx