Mercurial > hg > svapp
diff audioio/AudioCallbackPlaySource.cpp @ 295:14b1768e5406 tonioni
Merge from default branch
| author | Chris Cannam |
|---|---|
| date | Tue, 26 Nov 2013 14:39:57 +0000 |
| parents | cba1e2a3d14b 068235cf5bf7 |
| children | 055ff09f7a08 |
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--- a/audioio/AudioCallbackPlaySource.cpp Fri Nov 22 10:41:36 2013 +0000 +++ b/audioio/AudioCallbackPlaySource.cpp Tue Nov 26 14:39:57 2013 +0000 @@ -107,7 +107,7 @@ if (m_fillThread) { #ifdef DEBUG_AUDIO_PLAY_SOURCE - std::cout << "AudioCallbackPlaySource dtor: awakening thread" << std::endl; + cout << "AudioCallbackPlaySource dtor: awakening thread" << endl; #endif m_condition.wakeAll(); m_fillThread->wait(); @@ -164,7 +164,7 @@ } #ifdef DEBUG_AUDIO_PLAY_SOURCE - std::cout << "AudioCallbackPlaySource.cpp: Adding model with " << modelChannels << " channels at rate " << model->getSampleRate() << std::endl; + cout << "AudioCallbackPlaySource: Adding model with " << modelChannels << " channels at rate " << model->getSampleRate() << endl; #endif if (m_sourceSampleRate == 0) { @@ -245,7 +245,7 @@ } #ifdef DEBUG_AUDIO_PLAY_SOURCE - std::cout << "AudioCallbackPlaySource::addModel: now have " << m_models.size() << " model(s) -- emitting modelReplaced" << std::endl; + cout << "AudioCallbackPlaySource::addModel: now have " << m_models.size() << " model(s) -- emitting modelReplaced" << endl; #endif if (buffersChanged || srChanged) { @@ -256,7 +256,7 @@ this, SLOT(modelChanged(size_t, size_t))); #ifdef DEBUG_AUDIO_PLAY_SOURCE - std::cout << "AudioCallbackPlaySource::addModel: awakening thread" << std::endl; + cout << "AudioCallbackPlaySource::addModel: awakening thread" << endl; #endif m_condition.wakeAll(); @@ -280,7 +280,7 @@ m_mutex.lock(); #ifdef DEBUG_AUDIO_PLAY_SOURCE - std::cout << "AudioCallbackPlaySource::removeModel(" << model << ")" << std::endl; + cout << "AudioCallbackPlaySource::removeModel(" << model << ")" << endl; #endif disconnect(model, SIGNAL(modelChanged(size_t, size_t)), @@ -302,11 +302,11 @@ for (std::set<Model *>::const_iterator i = m_models.begin(); i != m_models.end(); ++i) { #ifdef DEBUG_AUDIO_PLAY_SOURCE - std::cout << "AudioCallbackPlaySource::removeModel(" << model << "): checking end frame on model " << *i << std::endl; + cout << "AudioCallbackPlaySource::removeModel(" << model << "): checking end frame on model " << *i << endl; #endif if ((*i)->getEndFrame() > lastEnd) lastEnd = (*i)->getEndFrame(); #ifdef DEBUG_AUDIO_PLAY_SOURCE - std::cout << "(done, lastEnd now " << lastEnd << ")" << std::endl; + cout << "(done, lastEnd now " << lastEnd << ")" << endl; #endif } m_lastModelEndFrame = lastEnd; @@ -324,7 +324,7 @@ m_mutex.lock(); #ifdef DEBUG_AUDIO_PLAY_SOURCE - std::cout << "AudioCallbackPlaySource::clearModels()" << std::endl; + cout << "AudioCallbackPlaySource::clearModels()" << endl; #endif m_models.clear(); @@ -370,8 +370,8 @@ m_writeBuffers->push_back(new RingBuffer<float>(m_ringBufferSize)); } -// std::cout << "AudioCallbackPlaySource::clearRingBuffers: Created " -// << count << " write buffers" << std::endl; +// cout << "AudioCallbackPlaySource::clearRingBuffers: Created " +// << count << " write buffers" << endl; if (!haveLock) { m_mutex.unlock(); @@ -397,13 +397,13 @@ } #ifdef DEBUG_AUDIO_PLAY_SOURCE - std::cerr << "play(" << startFrame << ") -> playback model "; + cerr << "play(" << startFrame << ") -> playback model "; #endif startFrame = m_viewManager->alignReferenceToPlaybackFrame(startFrame); #ifdef DEBUG_AUDIO_PLAY_SOURCE - std::cerr << startFrame << std::endl; + cerr << startFrame << endl; #endif // The fill thread will automatically empty its buffers before @@ -425,7 +425,7 @@ for (size_t c = 0; c < getTargetChannelCount(); ++c) { RingBuffer<float> *rb = getReadRingBuffer(c); #ifdef DEBUG_AUDIO_PLAY_SOURCE - std::cerr << "reset ring buffer for channel " << c << std::endl; + cerr << "reset ring buffer for channel " << c << endl; #endif if (rb) rb->reset(); } @@ -450,7 +450,7 @@ m_playing = true; #ifdef DEBUG_AUDIO_PLAY_SOURCE - std::cout << "AudioCallbackPlaySource::play: awakening thread" << std::endl; + cout << "AudioCallbackPlaySource::play: awakening thread" << endl; #endif m_condition.wakeAll(); @@ -472,7 +472,7 @@ m_playing = false; #ifdef DEBUG_AUDIO_PLAY_SOURCE - std::cout << "AudioCallbackPlaySource::stop: awakening thread" << std::endl; + cout << "AudioCallbackPlaySource::stop: awakening thread" << endl; #endif m_condition.wakeAll(); @@ -525,7 +525,7 @@ void AudioCallbackPlaySource::audioProcessingOverload() { - std::cerr << "Audio processing overload!" << std::endl; + cerr << "Audio processing overload!" << endl; if (!m_playing) return; @@ -550,7 +550,7 @@ AudioCallbackPlaySource::setTarget(AudioCallbackPlayTarget *target, size_t size) { m_target = target; - std::cout << "AudioCallbackPlaySource::setTarget: Block size -> " << size << std::endl; + cout << "AudioCallbackPlaySource::setTarget: Block size -> " << size << endl; if (size != 0) { m_blockSize = size; } @@ -569,7 +569,7 @@ size_t AudioCallbackPlaySource::getTargetBlockSize() const { -// std::cout << "AudioCallbackPlaySource::getTargetBlockSize() -> " << m_blockSize << std::endl; +// cout << "AudioCallbackPlaySource::getTargetBlockSize() -> " << m_blockSize << endl; return m_blockSize; } @@ -698,7 +698,7 @@ } #ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING - std::cerr << "\nbuffered to: " << bufferedto_t << ", in buffer: " << inbuffer_t << ", time ratio " << timeRatio << "\n stretcher latency: " << stretchlat_t << ", device latency: " << latency_t << "\n since request: " << sincerequest_t << ", last retrieved quantity: " << lastretrieved_t << std::endl; + cerr << "\nbuffered to: " << bufferedto_t << ", in buffer: " << inbuffer_t << ", time ratio " << timeRatio << "\n stretcher latency: " << stretchlat_t << ", device latency: " << latency_t << "\n since request: " << sincerequest_t << ", last retrieved quantity: " << lastretrieved_t << endl; #endif RealTime end = RealTime::frame2RealTime(m_lastModelEndFrame, sourceRate); @@ -755,12 +755,12 @@ RealTime playstart_t = RealTime::frame2RealTime(m_playStartFrame, sourceRate); if (playing_t < playstart_t) { -// std::cerr << "playing_t " << playing_t << " < playstart_t " -// << playstart_t << std::endl; +// cerr << "playing_t " << playing_t << " < playstart_t " +// << playstart_t << endl; if (/*!!! sincerequest_t > RealTime::zeroTime && */ m_playStartedAt + latency_t + stretchlat_t < RealTime::fromSeconds(currentTime)) { -// std::cerr << "but we've been playing for long enough that I think we should disregard it (it probably results from loop wrapping)" << std::endl; +// cerr << "but we've been playing for long enough that I think we should disregard it (it probably results from loop wrapping)" << endl; m_playStartFramePassed = true; } else { playing_t = playstart_t; @@ -771,13 +771,13 @@ } #ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING - std::cerr << "playing_t " << playing_t; + cerr << "playing_t " << playing_t; #endif playing_t = playing_t - m_rangeStarts[inRange]; #ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING - std::cerr << " as offset into range " << inRange << " (start =" << m_rangeStarts[inRange] << " duration =" << m_rangeDurations[inRange] << ") = " << playing_t << std::endl; + cerr << " as offset into range " << inRange << " (start =" << m_rangeStarts[inRange] << " duration =" << m_rangeDurations[inRange] << ") = " << playing_t << endl; #endif while (playing_t < RealTime::zeroTime) { @@ -798,13 +798,13 @@ playing_t = playing_t + m_rangeStarts[inRange]; #ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING - std::cerr << " playing time: " << playing_t << std::endl; + cerr << " playing time: " << playing_t << endl; #endif if (!looping) { if (inRange == m_rangeStarts.size()-1 && playing_t >= m_rangeStarts[inRange] + m_rangeDurations[inRange]) { -std::cerr << "Not looping, inRange " << inRange << " == rangeStarts.size()-1, playing_t " << playing_t << " >= m_rangeStarts[inRange] " << m_rangeStarts[inRange] << " + m_rangeDurations[inRange] " << m_rangeDurations[inRange] << " -- stopping" << std::endl; +cerr << "Not looping, inRange " << inRange << " == rangeStarts.size()-1, playing_t " << playing_t << " >= m_rangeStarts[inRange] " << m_rangeStarts[inRange] << " + m_rangeDurations[inRange] " << m_rangeDurations[inRange] << " -- stopping" << endl; stop(); } } @@ -874,7 +874,7 @@ } #ifdef DEBUG_AUDIO_PLAY_SOURCE - std::cerr << "Now have " << m_rangeStarts.size() << " play ranges" << std::endl; + cerr << "Now have " << m_rangeStarts.size() << " play ranges" << endl; #endif } @@ -937,9 +937,9 @@ } if (!m_converter || !m_crapConverter) { - std::cerr + cerr << "AudioCallbackPlaySource::setModel: ERROR in creating samplerate converter: " - << src_strerror(err) << std::endl; + << src_strerror(err) << endl; if (m_converter) { src_delete(m_converter); @@ -988,7 +988,7 @@ { RealTimePluginInstance *plugin = dynamic_cast<RealTimePluginInstance *>(a); if (a && !plugin) { - std::cerr << "WARNING: AudioCallbackPlaySource::setAuditioningEffect: auditionable object " << a << " is not a real-time plugin instance" << std::endl; + cerr << "WARNING: AudioCallbackPlaySource::setAuditioningEffect: auditionable object " << a << " is not a real-time plugin instance" << endl; } m_mutex.lock(); @@ -1100,9 +1100,9 @@ RingBuffer<float> *rb = getReadRingBuffer(ch); if (!rb) { - std::cerr << "WARNING: AudioCallbackPlaySource::getSourceSamples: " + cerr << "WARNING: AudioCallbackPlaySource::getSourceSamples: " << "No ring buffer available for channel " << ch - << ", returning no data here" << std::endl; + << ", returning no data here" << endl; count = 0; break; } @@ -1110,12 +1110,12 @@ size_t rs = rb->getReadSpace(); if (rs < count) { #ifdef DEBUG_AUDIO_PLAY_SOURCE - std::cerr << "WARNING: AudioCallbackPlaySource::getSourceSamples: " + cerr << "WARNING: AudioCallbackPlaySource::getSourceSamples: " << "Ring buffer for channel " << ch << " has only " << rs << " (of " << count << ") samples available (" << "ring buffer size is " << rb->getSize() << ", write " << "space " << rb->getWriteSpace() << "), " - << "reducing request size" << std::endl; + << "reducing request size" << endl; #endif count = rs; } @@ -1130,7 +1130,7 @@ if (ratio != m_stretchRatio) { if (!ts) { - std::cerr << "WARNING: AudioCallbackPlaySource::getSourceSamples: Time ratio change to " << m_stretchRatio << " is pending, but no stretcher is set" << std::endl; + cerr << "WARNING: AudioCallbackPlaySource::getSourceSamples: Time ratio change to " << m_stretchRatio << " is pending, but no stretcher is set" << endl; m_stretchRatio = 1.f; } else { ts->setTimeRatio(m_stretchRatio); @@ -1172,7 +1172,7 @@ got = rb->read(buffer[ch], request); #ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING - std::cout << "AudioCallbackPlaySource::getSamples: got " << got << " (of " << count << ") samples on channel " << ch << ", signalling for more (possibly)" << std::endl; + cout << "AudioCallbackPlaySource::getSamples: got " << got << " (of " << count << ") samples on channel " << ch << ", signalling for more (possibly)" << endl; #endif } @@ -1186,7 +1186,7 @@ applyAuditioningEffect(count, buffer); #ifdef DEBUG_AUDIO_PLAY_SOURCE - std::cout << "AudioCallbackPlaySource::getSamples: awakening thread" << std::endl; + cout << "AudioCallbackPlaySource::getSamples: awakening thread" << endl; #endif m_condition.wakeAll(); @@ -1211,14 +1211,14 @@ size_t got = reqd; #ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING - std::cerr << "reqd = " <<reqd << ", channels = " << channels << ", ic = " << m_stretcherInputCount << std::endl; + cerr << "reqd = " <<reqd << ", channels = " << channels << ", ic = " << m_stretcherInputCount << endl; #endif for (size_t c = 0; c < channels; ++c) { if (c >= m_stretcherInputCount) continue; if (reqd > m_stretcherInputSizes[c]) { if (c == 0) { - std::cerr << "WARNING: resizing stretcher input buffer from " << m_stretcherInputSizes[c] << " to " << (reqd * 2) << std::endl; + cerr << "WARNING: resizing stretcher input buffer from " << m_stretcherInputSizes[c] << " to " << (reqd * 2) << endl; } delete[] m_stretcherInputs[c]; m_stretcherInputSizes[c] = reqd * 2; @@ -1246,13 +1246,13 @@ #endif } else { - std::cerr << "WARNING: No ring buffer available for channel " << c << " in stretcher input block" << std::endl; + cerr << "WARNING: No ring buffer available for channel " << c << " in stretcher input block" << endl; } } if (got < reqd) { - std::cerr << "WARNING: Read underrun in playback (" - << got << " < " << reqd << ")" << std::endl; + cerr << "WARNING: Read underrun in playback (" + << got << " < " << reqd << ")" << endl; } ts->process(m_stretcherInputs, got, false); @@ -1262,7 +1262,7 @@ if (got == 0) break; if (ts->available() == available) { - std::cerr << "WARNING: AudioCallbackPlaySource::getSamples: Added " << got << " samples to time stretcher, created no new available output samples (warned = " << warned << ")" << std::endl; + cerr << "WARNING: AudioCallbackPlaySource::getSamples: Added " << got << " samples to time stretcher, created no new available output samples (warned = " << warned << ")" << endl; if (++warned == 5) break; } } @@ -1278,7 +1278,7 @@ applyAuditioningEffect(count, buffer); #ifdef DEBUG_AUDIO_PLAY_SOURCE - std::cout << "AudioCallbackPlaySource::getSamples [stretched]: awakening thread" << std::endl; + cout << "AudioCallbackPlaySource::getSamples [stretched]: awakening thread" << endl; #endif m_condition.wakeAll(); @@ -1294,21 +1294,21 @@ if (!plugin) return; if (plugin->getAudioInputCount() != getTargetChannelCount()) { -// std::cerr << "plugin input count " << plugin->getAudioInputCount() +// cerr << "plugin input count " << plugin->getAudioInputCount() // << " != our channel count " << getTargetChannelCount() -// << std::endl; +// << endl; return; } if (plugin->getAudioOutputCount() != getTargetChannelCount()) { -// std::cerr << "plugin output count " << plugin->getAudioOutputCount() +// cerr << "plugin output count " << plugin->getAudioOutputCount() // << " != our channel count " << getTargetChannelCount() -// << std::endl; +// << endl; return; } if (plugin->getBufferSize() < count) { -// std::cerr << "plugin buffer size " << plugin->getBufferSize() +// cerr << "plugin buffer size " << plugin->getBufferSize() // << " < our block size " << count -// << std::endl; +// << endl; return; } @@ -1348,7 +1348,7 @@ if (space == 0) { #ifdef DEBUG_AUDIO_PLAY_SOURCE - std::cout << "AudioCallbackPlaySourceFillThread: no space to fill" << std::endl; + cout << "AudioCallbackPlaySourceFillThread: no space to fill" << endl; #endif return false; } @@ -1359,19 +1359,19 @@ #ifdef DEBUG_AUDIO_PLAY_SOURCE if (!readWriteEqual) { - std::cout << "AudioCallbackPlaySourceFillThread: note read buffers != write buffers" << std::endl; + cout << "AudioCallbackPlaySourceFillThread: note read buffers != write buffers" << endl; } - std::cout << "AudioCallbackPlaySourceFillThread: filling " << space << " frames" << std::endl; + cout << "AudioCallbackPlaySourceFillThread: filling " << space << " frames" << endl; #endif #ifdef DEBUG_AUDIO_PLAY_SOURCE - std::cout << "buffered to " << f << " already" << std::endl; + cout << "buffered to " << f << " already" << endl; #endif bool resample = (getSourceSampleRate() != getTargetSampleRate()); #ifdef DEBUG_AUDIO_PLAY_SOURCE - std::cout << (resample ? "" : "not ") << "resampling (source " << getSourceSampleRate() << ", target " << getTargetSampleRate() << ")" << std::endl; + cout << (resample ? "" : "not ") << "resampling (source " << getSourceSampleRate() << ", target " << getTargetSampleRate() << ")" << endl; #endif size_t channels = getTargetChannelCount(); @@ -1393,7 +1393,7 @@ if (resample && !m_converter) { static bool warned = false; if (!warned) { - std::cerr << "WARNING: sample rates differ, but no converter available!" << std::endl; + cerr << "WARNING: sample rates differ, but no converter available!" << endl; warned = true; } } @@ -1463,7 +1463,7 @@ if (m_timeStretcher && m_timeStretcher->getTimeRatio() < 0.4) { #ifdef DEBUG_AUDIO_PLAY_SOURCE - std::cout << "Using crappy converter" << std::endl; + cout << "Using crappy converter" << endl; #endif err = src_process(m_crapConverter, &data); } else { @@ -1473,15 +1473,15 @@ size_t toCopy = size_t(got * ratio + 0.1); if (err) { - std::cerr + cerr << "AudioCallbackPlaySourceFillThread: ERROR in samplerate conversion: " - << src_strerror(err) << std::endl; + << src_strerror(err) << endl; //!!! Then what? } else { got = data.input_frames_used; toCopy = data.output_frames_gen; #ifdef DEBUG_AUDIO_PLAY_SOURCE - std::cout << "Resampled " << got << " frames to " << toCopy << " frames" << std::endl; + cout << "Resampled " << got << " frames to " << toCopy << " frames" << endl; #endif } @@ -1503,9 +1503,9 @@ space = (reqSpace / generatorBlockSize) * generatorBlockSize; if (space == 0) { #ifdef DEBUG_AUDIO_PLAY_SOURCE - std::cout << "requested fill of " << reqSpace + cout << "requested fill of " << reqSpace << " is less than generator block size of " - << generatorBlockSize << ", leaving it" << std::endl; + << generatorBlockSize << ", leaving it" << endl; #endif return false; } @@ -1533,14 +1533,14 @@ if (wb) { size_t actual = wb->write(bufferPtrs[c], got); #ifdef DEBUG_AUDIO_PLAY_SOURCE - std::cout << "Wrote " << actual << " samples for ch " << c << ", now " + cout << "Wrote " << actual << " samples for ch " << c << ", now " << wb->getReadSpace() << " to read" - << std::endl; + << endl; #endif if (actual < got) { - std::cerr << "WARNING: Buffer overrun in channel " << c + cerr << "WARNING: Buffer overrun in channel " << c << ": wrote " << actual << " of " << got - << " samples" << std::endl; + << " samples" << endl; } } } @@ -1549,7 +1549,7 @@ if (readWriteEqual) m_readBufferFill = f; #ifdef DEBUG_AUDIO_PLAY_SOURCE - std::cout << "Read buffer fill is now " << m_readBufferFill << std::endl; + cout << "Read buffer fill is now " << m_readBufferFill << endl; #endif //!!! how do we know when ended? need to mark up a fully-buffered flag and check this if we find the buffers empty in getSourceSamples @@ -1576,7 +1576,7 @@ size_t channels = getTargetChannelCount(); #ifdef DEBUG_AUDIO_PLAY_SOURCE - std::cout << "Selection playback: start " << frame << ", size " << count <<", channels " << channels << std::endl; + cout << "Selection playback: start " << frame << ", size " << count <<", channels " << channels << endl; #endif if (chunkBufferPtrCount < channels) { @@ -1654,11 +1654,11 @@ nextChunkStart = chunkStart + chunkSize; } -// std::cout << "chunkStart " << chunkStart << ", chunkSize " << chunkSize << ", nextChunkStart " << nextChunkStart << ", frame " << frame << ", count " << count << ", processed " << processed << std::endl; +// cout << "chunkStart " << chunkStart << ", chunkSize " << chunkSize << ", nextChunkStart " << nextChunkStart << ", frame " << frame << ", count " << count << ", processed " << processed << endl; if (!chunkSize) { #ifdef DEBUG_AUDIO_PLAY_SOURCE - std::cout << "Ending selection playback at " << nextChunkStart << std::endl; + cout << "Ending selection playback at " << nextChunkStart << endl; #endif // We need to maintain full buffers so that the other // thread can tell where it's got to in the playback -- so @@ -1668,7 +1668,7 @@ } #ifdef DEBUG_AUDIO_PLAY_SOURCE - std::cout << "Selection playback: chunk at " << chunkStart << " -> " << nextChunkStart << " (size " << chunkSize << ")" << std::endl; + cout << "Selection playback: chunk at " << chunkStart << " -> " << nextChunkStart << " (size " << chunkSize << ")" << endl; #endif size_t got = 0; @@ -1710,7 +1710,7 @@ } #ifdef DEBUG_AUDIO_PLAY_SOURCE - std::cout << "Returning selection playback " << processed << " frames to " << nextChunkStart << std::endl; + cout << "Returning selection playback " << processed << " frames to " << nextChunkStart << endl; #endif frame = nextChunkStart; @@ -1746,7 +1746,7 @@ if (rb) { size_t rs = rb->getReadSpace(); //!!! incorrect when in non-contiguous selection, see comments elsewhere -// std::cout << "rs = " << rs << std::endl; +// cout << "rs = " << rs << endl; if (rs < rf) rf -= rs; else rf = 0; } @@ -1763,17 +1763,17 @@ if (c == 0) { size_t wrs = wb->getReadSpace(); -// std::cout << "wrs = " << wrs << std::endl; +// cout << "wrs = " << wrs << endl; if (wrs < wf) wf -= wrs; else wf = 0; -// std::cout << "wf = " << wf << std::endl; +// cout << "wf = " << wf << endl; if (wf < rf) skip = rf - wf; if (skip == 0) break; } -// std::cout << "skipping " << skip << std::endl; +// cout << "skipping " << skip << endl; wb->skip(skip); } } @@ -1782,7 +1782,7 @@ m_readBuffers = m_writeBuffers; m_readBufferFill = m_writeBufferFill; #ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING - std::cerr << "unified" << std::endl; + cerr << "unified" << endl; #endif } @@ -1792,7 +1792,7 @@ AudioCallbackPlaySource &s(m_source); #ifdef DEBUG_AUDIO_PLAY_SOURCE - std::cout << "AudioCallbackPlaySourceFillThread starting" << std::endl; + cout << "AudioCallbackPlaySourceFillThread starting" << endl; #endif s.m_mutex.lock(); @@ -1809,7 +1809,7 @@ if (work && s.m_playing && s.getSourceSampleRate()) { #ifdef DEBUG_AUDIO_PLAY_SOURCE - std::cout << "AudioCallbackPlaySourceFillThread: not waiting" << std::endl; + cout << "AudioCallbackPlaySourceFillThread: not waiting" << endl; #endif s.m_mutex.unlock(); @@ -1826,22 +1826,22 @@ if (s.m_playing) ms /= 10; #ifdef DEBUG_AUDIO_PLAY_SOURCE - if (!s.m_playing) std::cout << std::endl; - std::cout << "AudioCallbackPlaySourceFillThread: waiting for " << ms << "ms..." << std::endl; + if (!s.m_playing) cout << endl; + cout << "AudioCallbackPlaySourceFillThread: waiting for " << ms << "ms..." << endl; #endif s.m_condition.wait(&s.m_mutex, size_t(ms)); } #ifdef DEBUG_AUDIO_PLAY_SOURCE - std::cout << "AudioCallbackPlaySourceFillThread: awoken" << std::endl; + cout << "AudioCallbackPlaySourceFillThread: awoken" << endl; #endif work = false; if (!s.getSourceSampleRate()) { #ifdef DEBUG_AUDIO_PLAY_SOURCE - std::cout << "AudioCallbackPlaySourceFillThread: source sample rate is zero" << std::endl; + cout << "AudioCallbackPlaySourceFillThread: source sample rate is zero" << endl; #endif continue; } @@ -1850,7 +1850,7 @@ if (playing && !previouslyPlaying) { #ifdef DEBUG_AUDIO_PLAY_SOURCE - std::cout << "AudioCallbackPlaySourceFillThread: playback state changed, resetting" << std::endl; + cout << "AudioCallbackPlaySourceFillThread: playback state changed, resetting" << endl; #endif for (size_t c = 0; c < s.getTargetChannelCount(); ++c) { RingBuffer<float> *rb = s.getReadRingBuffer(c);