mirror of
https://github.com/badaix/snapcast.git
synced 2025-05-24 14:36:15 +02:00
reformatted code
This commit is contained in:
Johannes Pohl
parent
286a107c7c
commit
d9ee0c1f52
5 changed files with 100 additions and 101 deletions
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@ -47,18 +47,19 @@ CoreAudioPlayer::~CoreAudioPlayer()
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void CoreAudioPlayer::playerCallback(AudioQueueRef queue, AudioQueueBufferRef bufferRef)
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{
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/// Estimate the playout delay by checking the number of frames left in the buffer
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/// and add ms_ (= complete buffer size). Based on trying.
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AudioTimeStamp timestamp;
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AudioQueueGetCurrentTime(queue, timeLine, ×tamp, NULL);
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size_t bufferedFrames = (frames_ - ((uint64_t)timestamp.mSampleTime % frames_)) % frames_;
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size_t bufferedMs = bufferedFrames * 1000 / pubStream_->getFormat().rate + (ms_ * (NUM_BUFFERS - 1));
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/// 15ms DAC delay. Based on trying.
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bufferedMs += 15;
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/// Estimate the playout delay by checking the number of frames left in the buffer
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/// and add ms_ (= complete buffer size). Based on trying.
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AudioTimeStamp timestamp;
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AudioQueueGetCurrentTime(queue, timeLine_, ×tamp, NULL);
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size_t bufferedFrames = (frames_ - ((uint64_t)timestamp.mSampleTime % frames_)) % frames_;
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size_t bufferedMs = bufferedFrames * 1000 / pubStream_->getFormat().rate + (ms_ * (NUM_BUFFERS - 1));
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/// 15ms DAC delay. Based on trying.
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bufferedMs += 15;
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// logO << "buffered: " << bufferedFrames << ", ms: " << bufferedMs << ", mSampleTime: " << timestamp.mSampleTime << "\n";
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/// TODO: sometimes this bufferedMS or AudioTimeStamp wraps around 1s (i.e. we're 1s out of sync (behind)) and recovers later on
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chronos::usec delay(bufferedMs * 1000);
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char *buffer = (char*)bufferRef->mAudioData;
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char *buffer = (char*)bufferRef->mAudioData;
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if (!pubStream_->getPlayerChunk(buffer, delay, frames_))
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{
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logO << "Failed to get chunk. Playing silence.\n";
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@ -70,60 +71,60 @@ void CoreAudioPlayer::playerCallback(AudioQueueRef queue, AudioQueueBufferRef bu
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}
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// OSStatus status =
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AudioQueueEnqueueBuffer(queue, bufferRef, 0, NULL);
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AudioQueueEnqueueBuffer(queue, bufferRef, 0, NULL);
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if (!active_)
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{
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AudioQueueStop(queue, false);
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AudioQueueDispose(queue, false);
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CFRunLoopStop(CFRunLoopGetCurrent());
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}
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if (!active_)
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{
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AudioQueueStop(queue, false);
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AudioQueueDispose(queue, false);
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CFRunLoopStop(CFRunLoopGetCurrent());
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}
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}
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void CoreAudioPlayer::worker()
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{
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const SampleFormat& sampleFormat = pubStream_->getFormat();
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const SampleFormat& sampleFormat = pubStream_->getFormat();
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AudioStreamBasicDescription format;
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format.mSampleRate = sampleFormat.rate;
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format.mFormatID = kAudioFormatLinearPCM;
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format.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger;// | kAudioFormatFlagIsPacked;
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format.mBitsPerChannel = sampleFormat.bits;
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format.mChannelsPerFrame = sampleFormat.channels;
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format.mBytesPerFrame = sampleFormat.frameSize;
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format.mFramesPerPacket = 1;
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format.mBytesPerPacket = format.mBytesPerFrame * format.mFramesPerPacket;
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format.mReserved = 0;
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AudioStreamBasicDescription format;
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format.mSampleRate = sampleFormat.rate;
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format.mFormatID = kAudioFormatLinearPCM;
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format.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger;// | kAudioFormatFlagIsPacked;
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format.mBitsPerChannel = sampleFormat.bits;
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format.mChannelsPerFrame = sampleFormat.channels;
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format.mBytesPerFrame = sampleFormat.frameSize;
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format.mFramesPerPacket = 1;
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format.mBytesPerPacket = format.mBytesPerFrame * format.mFramesPerPacket;
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format.mReserved = 0;
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AudioQueueRef queue;
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AudioQueueNewOutput(&format, callback, this, CFRunLoopGetCurrent(), kCFRunLoopCommonModes, 0, &queue);
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AudioQueueCreateTimeline(queue, &timeLine);
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AudioQueueRef queue;
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AudioQueueNewOutput(&format, callback, this, CFRunLoopGetCurrent(), kCFRunLoopCommonModes, 0, &queue);
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AudioQueueCreateTimeline(queue, &timeLine_);
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// Apple recommends this as buffer size:
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// https://developer.apple.com/library/content/documentation/MusicAudio/Conceptual/CoreAudioOverview/CoreAudioEssentials/CoreAudioEssentials.html
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// static const int maxBufferSize = 0x10000; // limit maximum size to 64K
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// static const int minBufferSize = 0x4000; // limit minimum size to 16K
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//
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// For 100ms @ 48000:16:2 we have 19.2K
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// frames: 4800, ms: 100, buffer size: 19200
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// Apple recommends this as buffer size:
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// https://developer.apple.com/library/content/documentation/MusicAudio/Conceptual/CoreAudioOverview/CoreAudioEssentials/CoreAudioEssentials.html
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// static const int maxBufferSize = 0x10000; // limit maximum size to 64K
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// static const int minBufferSize = 0x4000; // limit minimum size to 16K
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//
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// For 100ms @ 48000:16:2 we have 19.2K
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// frames: 4800, ms: 100, buffer size: 19200
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frames_ = (sampleFormat.rate * ms_) / 1000;
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ms_ = frames_ * 1000 / sampleFormat.rate;
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ms_ = frames_ * 1000 / sampleFormat.rate;
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buff_size_ = frames_ * sampleFormat.frameSize;
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logO << "frames: " << frames_ << ", ms: " << ms_ << ", buffer size: " << buff_size_ << "\n";
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logO << "frames: " << frames_ << ", ms: " << ms_ << ", buffer size: " << buff_size_ << "\n";
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AudioQueueBufferRef buffers[NUM_BUFFERS];
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for (int i = 0; i < NUM_BUFFERS; i++)
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{
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AudioQueueAllocateBuffer(queue, buff_size_, &buffers[i]);
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buffers[i]->mAudioDataByteSize = buff_size_;
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callback(this, queue, buffers[i]);
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}
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AudioQueueBufferRef buffers[NUM_BUFFERS];
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for (int i = 0; i < NUM_BUFFERS; i++)
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{
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AudioQueueAllocateBuffer(queue, buff_size_, &buffers[i]);
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buffers[i]->mAudioDataByteSize = buff_size_;
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callback(this, queue, buffers[i]);
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}
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logE << "CoreAudioPlayer::worker\n";
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AudioQueueCreateTimeline(queue, &timeLine);
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AudioQueueStart(queue, NULL);
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CFRunLoopRun();
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logE << "CoreAudioPlayer::worker\n";
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AudioQueueCreateTimeline(queue, &timeLine_);
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AudioQueueStart(queue, NULL);
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CFRunLoopRun();
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}
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@ -45,7 +45,7 @@ public:
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protected:
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virtual void worker();
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AudioQueueTimelineRef timeLine;
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AudioQueueTimelineRef timeLine_;
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size_t ms_;
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size_t frames_;
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size_t buff_size_;
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@ -28,7 +28,7 @@
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using namespace std;
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OggEncoder::OggEncoder(const std::string& codecOptions) : Encoder(codecOptions), lastGranulepos(0)
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OggEncoder::OggEncoder(const std::string& codecOptions) : Encoder(codecOptions), lastGranulepos_(0)
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{
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}
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@ -56,7 +56,7 @@ void OggEncoder::encode(const msg::PcmChunk* chunk)
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double res = 0;
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logD << "payload: " << chunk->payloadSize << "\tframes: " << chunk->getFrameCount() << "\tduration: " << chunk->duration<chronos::msec>().count() << "\n";
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int frames = chunk->getFrameCount();
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float **buffer=vorbis_analysis_buffer(&vd, frames);
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float **buffer=vorbis_analysis_buffer(&vd_, frames);
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/* uninterleave samples */
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for (size_t channel = 0; channel < sampleFormat_.channels; ++channel)
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@ -82,7 +82,7 @@ void OggEncoder::encode(const msg::PcmChunk* chunk)
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}
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/* tell the library how much we actually submitted */
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vorbis_analysis_wrote(&vd, frames);
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vorbis_analysis_wrote(&vd_, frames);
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msg::PcmChunk* oggChunk = new msg::PcmChunk(chunk->format, 0);
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@ -90,36 +90,36 @@ void OggEncoder::encode(const msg::PcmChunk* chunk)
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more involved (potentially parallel) processing. Get a single
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block for encoding now */
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size_t pos = 0;
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while (vorbis_analysis_blockout(&vd, &vb)==1)
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while (vorbis_analysis_blockout(&vd_, &vb_)==1)
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{
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/* analysis, assume we want to use bitrate management */
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vorbis_analysis(&vb, NULL);
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vorbis_bitrate_addblock(&vb);
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vorbis_analysis(&vb_, NULL);
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vorbis_bitrate_addblock(&vb_);
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while (vorbis_bitrate_flushpacket(&vd, &op))
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while (vorbis_bitrate_flushpacket(&vd_, &op_))
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{
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/* weld the packet into the bitstream */
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ogg_stream_packetin(&os, &op);
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ogg_stream_packetin(&os_, &op_);
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/* write out pages (if any) */
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while (true)
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{
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int result = ogg_stream_flush(&os, &og);
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int result = ogg_stream_flush(&os_, &og_);
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if (result == 0)
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break;
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res = os.granulepos - lastGranulepos;
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res = os_.granulepos - lastGranulepos_;
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size_t nextLen = pos + og.header_len + og.body_len;
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size_t nextLen = pos + og_.header_len + og_.body_len;
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// make chunk larger
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if (oggChunk->payloadSize < nextLen)
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oggChunk->payload = (char*)realloc(oggChunk->payload, nextLen);
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memcpy(oggChunk->payload + pos, og.header, og.header_len);
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pos += og.header_len;
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memcpy(oggChunk->payload + pos, og.body, og.body_len);
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pos += og.body_len;
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memcpy(oggChunk->payload + pos, og_.header, og_.header_len);
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pos += og_.header_len;
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memcpy(oggChunk->payload + pos, og_.body, og_.body_len);
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pos += og_.body_len;
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if (ogg_page_eos(&og))
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if (ogg_page_eos(&og_))
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break;
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}
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}
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@ -129,7 +129,7 @@ void OggEncoder::encode(const msg::PcmChunk* chunk)
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{
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res /= (sampleFormat_.rate / 1000.);
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// logO << "res: " << res << "\n";
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lastGranulepos = os.granulepos;
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lastGranulepos_ = os_.granulepos;
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// make oggChunk smaller
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oggChunk->payload = (char*)realloc(oggChunk->payload, pos);
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oggChunk->payloadSize = pos;
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@ -164,7 +164,7 @@ void OggEncoder::initEncoder()
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}
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/********** Encode setup ************/
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vorbis_info_init(&vi);
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vorbis_info_init(&vi_);
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/* choose an encoding mode. A few possibilities commented out, one
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actually used: */
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@ -195,7 +195,7 @@ void OggEncoder::initEncoder()
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*********************************************************************/
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int ret = vorbis_encode_init_vbr(&vi, sampleFormat_.channels, sampleFormat_.rate, quality);
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int ret = vorbis_encode_init_vbr(&vi_, sampleFormat_.channels, sampleFormat_.rate, quality);
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/* do not continue if setup failed; this can happen if we ask for a
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mode that libVorbis does not support (eg, too low a bitrate, etc,
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@ -205,20 +205,20 @@ void OggEncoder::initEncoder()
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throw SnapException("failed to init encoder");
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/* add a comment */
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vorbis_comment_init(&vc);
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vorbis_comment_add_tag(&vc, "TITLE", "SnapStream");
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vorbis_comment_add_tag(&vc, "VERSION", VERSION);
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vorbis_comment_add_tag(&vc, "SAMPLE_FORMAT", sampleFormat_.getFormat().c_str());
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vorbis_comment_init(&vc_);
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vorbis_comment_add_tag(&vc_, "TITLE", "SnapStream");
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vorbis_comment_add_tag(&vc_, "VERSION", VERSION);
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vorbis_comment_add_tag(&vc_, "SAMPLE_FORMAT", sampleFormat_.getFormat().c_str());
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/* set up the analysis state and auxiliary encoding storage */
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vorbis_analysis_init(&vd, &vi);
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vorbis_block_init(&vd, &vb);
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vorbis_analysis_init(&vd_, &vi_);
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vorbis_block_init(&vd_, &vb_);
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/* set up our packet->stream encoder */
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/* pick a random serial number; that way we can more likely build
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chained streams just by concatenation */
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srand(time(NULL));
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ogg_stream_init(&os, rand());
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ogg_stream_init(&os_, rand());
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/* Vorbis streams begin with three headers; the initial header (with
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most of the codec setup parameters) which is mandated by the Ogg
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@ -231,10 +231,10 @@ void OggEncoder::initEncoder()
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ogg_packet header_comm;
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ogg_packet header_code;
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vorbis_analysis_headerout(&vd, &vc, &header, &header_comm, &header_code);
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ogg_stream_packetin(&os, &header);
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ogg_stream_packetin(&os, &header_comm);
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ogg_stream_packetin(&os, &header_code);
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vorbis_analysis_headerout(&vd_, &vc_, &header, &header_comm, &header_code);
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ogg_stream_packetin(&os_, &header);
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ogg_stream_packetin(&os_, &header_comm);
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ogg_stream_packetin(&os_, &header_code);
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/* This ensures the actual
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* audio data will start on a new page, as per spec
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@ -243,16 +243,16 @@ void OggEncoder::initEncoder()
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headerChunk_.reset(new msg::CodecHeader("ogg"));
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while (true)
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{
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int result = ogg_stream_flush(&os, &og);
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int result = ogg_stream_flush(&os_, &og_);
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if (result == 0)
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break;
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headerChunk_->payloadSize += og.header_len + og.body_len;
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headerChunk_->payloadSize += og_.header_len + og_.body_len;
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headerChunk_->payload = (char*)realloc(headerChunk_->payload, headerChunk_->payloadSize);
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logD << "HeadLen: " << og.header_len << ", bodyLen: " << og.body_len << ", result: " << result << "\n";
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memcpy(headerChunk_->payload + pos, og.header, og.header_len);
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pos += og.header_len;
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memcpy(headerChunk_->payload + pos, og.body, og.body_len);
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pos += og.body_len;
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logD << "HeadLen: " << og_.header_len << ", bodyLen: " << og_.body_len << ", result: " << result << "\n";
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memcpy(headerChunk_->payload + pos, og_.header, og_.header_len);
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pos += og_.header_len;
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memcpy(headerChunk_->payload + pos, og_.body, og_.body_len);
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pos += og_.body_len;
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}
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}
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@ -35,19 +35,17 @@ protected:
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virtual void initEncoder();
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private:
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ogg_stream_state os; /* take physical pages, weld into a logical
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stream of packets */
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ogg_page og; /* one Ogg bitstream page. Vorbis packets are inside */
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ogg_packet op; /* one raw packet of data for decode */
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ogg_stream_state os_; /// take physical pages, weld into a logical stream of packets
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ogg_page og_; /// one Ogg bitstream page. Vorbis packets are inside
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ogg_packet op_; /// one raw packet of data for decode
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vorbis_info vi; /* struct that stores all the static vorbis bitstream
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settings */
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vorbis_comment vc; /* struct that stores all the user comments */
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vorbis_info vi_; /// struct that stores all the static vorbis bitstream settings
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vorbis_comment vc_; /// struct that stores all the user comments
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vorbis_dsp_state vd; /* central working state for the packet->PCM decoder */
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vorbis_block vb; /* local working space for packet->PCM decode */
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vorbis_dsp_state vd_; /// central working state for the packet->PCM decoder
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vorbis_block vb_; /// local working space for packet->PCM decode
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ogg_int64_t lastGranulepos;
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ogg_int64_t lastGranulepos_;
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};
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@ -161,7 +161,7 @@ void PublishAvahi::create_services(AvahiClient *c)
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}
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/// Add an additional (hypothetic) subtype
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/* if ((ret = avahi_entry_group_add_service_subtype(group, AVAHI_IF_UNSPEC, AVAHI_PROTO_UNSPEC, AvahiPublishFlags(0), name, "_printer._tcp", NULL, "_magic._sub._printer._tcp") < 0))
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/* if ((ret = avahi_entry_group_add_service_subtype(group, AVAHI_IF_UNSPEC, AVAHI_PROTO_UNSPEC, AvahiPublishFlags(0), name, "_printer._tcp", NULL, "_magic._sub._printer._tcp") < 0))
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{
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fprintf(stderr, "Failed to add subtype _magic._sub._printer._tcp: %s\n", avahi_strerror(ret));
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goto fail;
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