diff options
| -rw-r--r-- | xbmc/cores/AudioEngine/Sinks/AESinkDARWINOSX.cpp | 85 | ||||
| -rw-r--r-- | xbmc/cores/AudioEngine/Sinks/AESinkDARWINOSX.h | 2 |
2 files changed, 64 insertions, 23 deletions
diff --git a/xbmc/cores/AudioEngine/Sinks/AESinkDARWINOSX.cpp b/xbmc/cores/AudioEngine/Sinks/AESinkDARWINOSX.cpp index 07a877cc13..cc5889d87a 100644 --- a/xbmc/cores/AudioEngine/Sinks/AESinkDARWINOSX.cpp +++ b/xbmc/cores/AudioEngine/Sinks/AESinkDARWINOSX.cpp @@ -332,7 +332,7 @@ void RegisterDeviceChangedCB(bool bRegister, void *ref) //////////////////////////////////////////////////////////////////////////////////////////// CAESinkDARWINOSX::CAESinkDARWINOSX() -: m_latentFrames(0), m_outputBitstream(false), m_outputBuffer(NULL), m_buffer(NULL) +: m_latentFrames(0), m_outputBitstream(false), m_outputBuffer(NULL), m_planar(false), m_planarBuffer(NULL), m_buffer(NULL) { // By default, kAudioHardwarePropertyRunLoop points at the process's main thread on SnowLeopard, // If your process lacks such a run loop, you can set kAudioHardwarePropertyRunLoop to NULL which @@ -352,6 +352,7 @@ CAESinkDARWINOSX::CAESinkDARWINOSX() } RegisterDeviceChangedCB(true, this); m_started = false; + m_planar = false; } CAESinkDARWINOSX::~CAESinkDARWINOSX() @@ -501,6 +502,14 @@ bool CAESinkDARWINOSX::Initialize(AEAudioFormat &format, std::string &device) index++; } + m_planar = false; + if (streams.size() > 1 && outputFormat.mChannelsPerFrame == 1) + { + CLog::Log(LOGDEBUG, "%s Found planar audio with %u channels?", __FUNCTION__, streams.size()); + outputFormat.mChannelsPerFrame = std::min((size_t)format.m_channelLayout.Count(), streams.size()); + m_planar = true; + } + if (!outputFormat.mFormatID) { CLog::Log(LOGERROR, "%s, Unable to find suitable stream", __FUNCTION__); @@ -553,6 +562,9 @@ bool CAESinkDARWINOSX::Initialize(AEAudioFormat &format, std::string &device) m_device.SetNominalSampleRate(format.m_sampleRate); } + if (m_planar) + m_planarBuffer = new float[format.m_frameSamples]; + unsigned int num_buffers = 4; m_buffer = new AERingBuffer(num_buffers * format.m_frames * format.m_frameSize); CLog::Log(LOGDEBUG, "%s: using buffer size: %u (%f ms)", __FUNCTION__, m_buffer->GetMaxSize(), (float)m_buffer->GetMaxSize() / (format.m_sampleRate * format.m_frameSize)); @@ -611,6 +623,10 @@ void CAESinkDARWINOSX::Deinitialize() delete[] m_outputBuffer; m_outputBuffer = NULL; + m_planar = false; + delete[] m_planarBuffer; + m_planarBuffer = NULL; + m_started = false; } @@ -718,35 +734,58 @@ OSStatus CAESinkDARWINOSX::renderCallback(AudioDeviceID inDevice, const AudioTim CAESinkDARWINOSX *sink = (CAESinkDARWINOSX*)inClientData; sink->m_started = true; - for (unsigned int i = 0; i < outOutputData->mNumberBuffers; i++) + if (sink->m_planar) { - if (sink->m_outputBitstream) + unsigned int channels = std::min((unsigned int)outOutputData->mNumberBuffers, sink->m_format.m_channelLayout.Count()); + unsigned int wanted = outOutputData->mBuffers[0].mDataByteSize; + unsigned int bytes = std::min(sink->m_buffer->GetReadSize() / channels, wanted); + sink->m_buffer->Read((unsigned char *)sink->m_planarBuffer, bytes * channels); + // transform from interleaved to planar + const float *src = sink->m_planarBuffer; + for (unsigned int i = 0; i < bytes / sizeof(float); i++) { - /* HACK for bitstreaming AC3/DTS via PCM. - We reverse the float->S16LE conversion done in the stream or device */ - static const float mul = 1.0f / (INT16_MAX + 1); - - unsigned int wanted = std::min(outOutputData->mBuffers[i].mDataByteSize / sizeof(float), (size_t)sink->m_format.m_frameSamples) * sizeof(int16_t); - if (wanted <= sink->m_buffer->GetReadSize()) + for (unsigned int j = 0; j < channels; j++) { - sink->m_buffer->Read((unsigned char *)sink->m_outputBuffer, wanted); - int16_t *src = sink->m_outputBuffer; - float *dest = (float*)outOutputData->mBuffers[i].mData; - for (unsigned int i = 0; i < wanted / 2; i++) - *dest++ = *src++ * mul; + float *dst = (float *)outOutputData->mBuffers[j].mData; + dst[i] = *src++; } } - else - { - /* buffers appear to come from CA already zero'd, so just copy what is wanted */ - unsigned int wanted = outOutputData->mBuffers[i].mDataByteSize; - unsigned int bytes = std::min(sink->m_buffer->GetReadSize(), wanted); - sink->m_buffer->Read((unsigned char*)outOutputData->mBuffers[i].mData, bytes); - LogLevel(bytes, wanted); - } - + LogLevel(bytes, wanted); // tell the sink we're good for more data condVar.notifyAll(); } + else + { + for (unsigned int i = 0; i < outOutputData->mNumberBuffers; i++) + { + if (sink->m_outputBitstream) + { + /* HACK for bitstreaming AC3/DTS via PCM. + We reverse the float->S16LE conversion done in the stream or device */ + static const float mul = 1.0f / (INT16_MAX + 1); + + unsigned int wanted = std::min(outOutputData->mBuffers[i].mDataByteSize / sizeof(float), (size_t)sink->m_format.m_frameSamples) * sizeof(int16_t); + if (wanted <= sink->m_buffer->GetReadSize()) + { + sink->m_buffer->Read((unsigned char *)sink->m_outputBuffer, wanted); + int16_t *src = sink->m_outputBuffer; + float *dest = (float*)outOutputData->mBuffers[i].mData; + for (unsigned int i = 0; i < wanted / 2; i++) + *dest++ = *src++ * mul; + } + } + else + { + /* buffers appear to come from CA already zero'd, so just copy what is wanted */ + unsigned int wanted = outOutputData->mBuffers[i].mDataByteSize; + unsigned int bytes = std::min(sink->m_buffer->GetReadSize(), wanted); + sink->m_buffer->Read((unsigned char*)outOutputData->mBuffers[i].mData, bytes); + LogLevel(bytes, wanted); + } + + // tell the sink we're good for more data + condVar.notifyAll(); + } + } return noErr; } diff --git a/xbmc/cores/AudioEngine/Sinks/AESinkDARWINOSX.h b/xbmc/cores/AudioEngine/Sinks/AESinkDARWINOSX.h index 5872c17c0d..c85a32bdc8 100644 --- a/xbmc/cores/AudioEngine/Sinks/AESinkDARWINOSX.h +++ b/xbmc/cores/AudioEngine/Sinks/AESinkDARWINOSX.h @@ -59,6 +59,8 @@ private: bool m_outputBitstream; ///< true if we're bistreaming into a LinearPCM stream rather than AC3 stream. int16_t *m_outputBuffer; ///< buffer for bitstreaming + bool m_planar; + float *m_planarBuffer; ///< buffer for planar conversion AERingBuffer *m_buffer; volatile bool m_started; // set once we get a callback from CoreAudio, which can take a little while. |