3pp-mirror/snapcast
1
0
Fork 0
mirror of https://github.com/badaix/snapcast.git synced 2025-05-25 06:56:15 +02:00
Code Issues Projects Releases Packages Wiki Activity

reformatted code

Browse source
This commit is contained in:
Johannes Pohl 2016-10-20 11:00:24 +02:00
parent 286a107c7c
commit d9ee0c1f52
5 changed files with 100 additions and 101 deletions
Download patch file Download diff file Expand all files Collapse all files

101
client/player/coreAudioPlayer.cpp
View file

@ -47,18 +47,19 @@ CoreAudioPlayer::~CoreAudioPlayer()
void CoreAudioPlayer::playerCallback(AudioQueueRef queue, AudioQueueBufferRef bufferRef) void CoreAudioPlayer::playerCallback(AudioQueueRef queue, AudioQueueBufferRef bufferRef)
{ {
/// Estimate the playout delay by checking the number of frames left in the buffer /// Estimate the playout delay by checking the number of frames left in the buffer
/// and add ms_ (= complete buffer size). Based on trying. /// and add ms_ (= complete buffer size). Based on trying.
AudioTimeStamp timestamp; AudioTimeStamp timestamp;
AudioQueueGetCurrentTime(queue, timeLine, &timestamp, NULL); AudioQueueGetCurrentTime(queue, timeLine_, &timestamp, NULL);
size_t bufferedFrames = (frames_ - ((uint64_t)timestamp.mSampleTime % frames_)) % frames_; size_t bufferedFrames = (frames_ - ((uint64_t)timestamp.mSampleTime % frames_)) % frames_;
size_t bufferedMs = bufferedFrames * 1000 / pubStream_->getFormat().rate + (ms_ * (NUM_BUFFERS - 1)); size_t bufferedMs = bufferedFrames * 1000 / pubStream_->getFormat().rate + (ms_ * (NUM_BUFFERS - 1));
/// 15ms DAC delay. Based on trying. /// 15ms DAC delay. Based on trying.
bufferedMs += 15; bufferedMs += 15;
// logO << "buffered: " << bufferedFrames << ", ms: " << bufferedMs << ", mSampleTime: " << timestamp.mSampleTime << "\n"; // logO << "buffered: " << bufferedFrames << ", ms: " << bufferedMs << ", mSampleTime: " << timestamp.mSampleTime << "\n";
/// TODO: sometimes this bufferedMS or AudioTimeStamp wraps around 1s (i.e. we're 1s out of sync (behind)) and recovers later on
chronos::usec delay(bufferedMs * 1000); chronos::usec delay(bufferedMs * 1000);
char *buffer = (char*)bufferRef->mAudioData; char *buffer = (char*)bufferRef->mAudioData;
if (!pubStream_->getPlayerChunk(buffer, delay, frames_)) if (!pubStream_->getPlayerChunk(buffer, delay, frames_))
{ {
logO << "Failed to get chunk. Playing silence.\n"; logO << "Failed to get chunk. Playing silence.\n";
@ -70,60 +71,60 @@ void CoreAudioPlayer::playerCallback(AudioQueueRef queue, AudioQueueBufferRef bu
} }
// OSStatus status = // OSStatus status =
AudioQueueEnqueueBuffer(queue, bufferRef, 0, NULL); AudioQueueEnqueueBuffer(queue, bufferRef, 0, NULL);
if (!active_) if (!active_)
{ {
AudioQueueStop(queue, false); AudioQueueStop(queue, false);
AudioQueueDispose(queue, false); AudioQueueDispose(queue, false);
CFRunLoopStop(CFRunLoopGetCurrent()); CFRunLoopStop(CFRunLoopGetCurrent());
} }
} }
void CoreAudioPlayer::worker() void CoreAudioPlayer::worker()
{ {
const SampleFormat& sampleFormat = pubStream_->getFormat(); const SampleFormat& sampleFormat = pubStream_->getFormat();
AudioStreamBasicDescription format; AudioStreamBasicDescription format;
format.mSampleRate = sampleFormat.rate; format.mSampleRate = sampleFormat.rate;
format.mFormatID = kAudioFormatLinearPCM; format.mFormatID = kAudioFormatLinearPCM;
format.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger;// | kAudioFormatFlagIsPacked; format.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger;// | kAudioFormatFlagIsPacked;
format.mBitsPerChannel = sampleFormat.bits; format.mBitsPerChannel = sampleFormat.bits;
format.mChannelsPerFrame = sampleFormat.channels; format.mChannelsPerFrame = sampleFormat.channels;
format.mBytesPerFrame = sampleFormat.frameSize; format.mBytesPerFrame = sampleFormat.frameSize;
format.mFramesPerPacket = 1; format.mFramesPerPacket = 1;
format.mBytesPerPacket = format.mBytesPerFrame * format.mFramesPerPacket; format.mBytesPerPacket = format.mBytesPerFrame * format.mFramesPerPacket;
format.mReserved = 0; format.mReserved = 0;
AudioQueueRef queue; AudioQueueRef queue;
AudioQueueNewOutput(&format, callback, this, CFRunLoopGetCurrent(), kCFRunLoopCommonModes, 0, &queue); AudioQueueNewOutput(&format, callback, this, CFRunLoopGetCurrent(), kCFRunLoopCommonModes, 0, &queue);
AudioQueueCreateTimeline(queue, &timeLine); AudioQueueCreateTimeline(queue, &timeLine_);
// Apple recommends this as buffer size: // Apple recommends this as buffer size:
// https://developer.apple.com/library/content/documentation/MusicAudio/Conceptual/CoreAudioOverview/CoreAudioEssentials/CoreAudioEssentials.html // https://developer.apple.com/library/content/documentation/MusicAudio/Conceptual/CoreAudioOverview/CoreAudioEssentials/CoreAudioEssentials.html
// static const int maxBufferSize = 0x10000; // limit maximum size to 64K // static const int maxBufferSize = 0x10000; // limit maximum size to 64K
// static const int minBufferSize = 0x4000; // limit minimum size to 16K // static const int minBufferSize = 0x4000; // limit minimum size to 16K
// //
// For 100ms @ 48000:16:2 we have 19.2K // For 100ms @ 48000:16:2 we have 19.2K
// frames: 4800, ms: 100, buffer size: 19200 // frames: 4800, ms: 100, buffer size: 19200
frames_ = (sampleFormat.rate * ms_) / 1000; frames_ = (sampleFormat.rate * ms_) / 1000;
ms_ = frames_ * 1000 / sampleFormat.rate; ms_ = frames_ * 1000 / sampleFormat.rate;
buff_size_ = frames_ * sampleFormat.frameSize; buff_size_ = frames_ * sampleFormat.frameSize;
logO << "frames: " << frames_ << ", ms: " << ms_ << ", buffer size: " << buff_size_ << "\n"; logO << "frames: " << frames_ << ", ms: " << ms_ << ", buffer size: " << buff_size_ << "\n";
AudioQueueBufferRef buffers[NUM_BUFFERS]; AudioQueueBufferRef buffers[NUM_BUFFERS];
for (int i = 0; i < NUM_BUFFERS; i++) for (int i = 0; i < NUM_BUFFERS; i++)
{ {
AudioQueueAllocateBuffer(queue, buff_size_, &buffers[i]); AudioQueueAllocateBuffer(queue, buff_size_, &buffers[i]);
buffers[i]->mAudioDataByteSize = buff_size_; buffers[i]->mAudioDataByteSize = buff_size_;
callback(this, queue, buffers[i]); callback(this, queue, buffers[i]);
} }
logE << "CoreAudioPlayer::worker\n"; logE << "CoreAudioPlayer::worker\n";
AudioQueueCreateTimeline(queue, &timeLine); AudioQueueCreateTimeline(queue, &timeLine_);
AudioQueueStart(queue, NULL); AudioQueueStart(queue, NULL);
CFRunLoopRun(); CFRunLoopRun();
} }

2
client/player/coreAudioPlayer.h
View file

@ -45,7 +45,7 @@ public:
protected: protected:
virtual void worker(); virtual void worker();
AudioQueueTimelineRef timeLine; AudioQueueTimelineRef timeLine_;
size_t ms_; size_t ms_;
size_t frames_; size_t frames_;
size_t buff_size_; size_t buff_size_;

74
server/encoder/oggEncoder.cpp
View file

@ -28,7 +28,7 @@
using namespace std; using namespace std;
OggEncoder::OggEncoder(const std::string& codecOptions) : Encoder(codecOptions), lastGranulepos(0) OggEncoder::OggEncoder(const std::string& codecOptions) : Encoder(codecOptions), lastGranulepos_(0)
{ {
} }
@ -56,7 +56,7 @@ void OggEncoder::encode(const msg::PcmChunk* chunk)
double res = 0; double res = 0;
logD << "payload: " << chunk->payloadSize << "\tframes: " << chunk->getFrameCount() << "\tduration: " << chunk->duration<chronos::msec>().count() << "\n"; logD << "payload: " << chunk->payloadSize << "\tframes: " << chunk->getFrameCount() << "\tduration: " << chunk->duration<chronos::msec>().count() << "\n";
int frames = chunk->getFrameCount(); int frames = chunk->getFrameCount();
float **buffer=vorbis_analysis_buffer(&vd, frames); float **buffer=vorbis_analysis_buffer(&vd_, frames);
/* uninterleave samples */ /* uninterleave samples */
for (size_t channel = 0; channel < sampleFormat_.channels; ++channel) for (size_t channel = 0; channel < sampleFormat_.channels; ++channel)
@ -82,7 +82,7 @@ void OggEncoder::encode(const msg::PcmChunk* chunk)
} }
/* tell the library how much we actually submitted */ /* tell the library how much we actually submitted */
vorbis_analysis_wrote(&vd, frames); vorbis_analysis_wrote(&vd_, frames);
msg::PcmChunk* oggChunk = new msg::PcmChunk(chunk->format, 0); msg::PcmChunk* oggChunk = new msg::PcmChunk(chunk->format, 0);
@ -90,36 +90,36 @@ void OggEncoder::encode(const msg::PcmChunk* chunk)
more involved (potentially parallel) processing. Get a single more involved (potentially parallel) processing. Get a single
block for encoding now */ block for encoding now */
size_t pos = 0; size_t pos = 0;
while (vorbis_analysis_blockout(&vd, &vb)==1) while (vorbis_analysis_blockout(&vd_, &vb_)==1)
{ {
/* analysis, assume we want to use bitrate management */ /* analysis, assume we want to use bitrate management */
vorbis_analysis(&vb, NULL); vorbis_analysis(&vb_, NULL);
vorbis_bitrate_addblock(&vb); vorbis_bitrate_addblock(&vb_);
while (vorbis_bitrate_flushpacket(&vd, &op)) while (vorbis_bitrate_flushpacket(&vd_, &op_))
{ {
/* weld the packet into the bitstream */ /* weld the packet into the bitstream */
ogg_stream_packetin(&os, &op); ogg_stream_packetin(&os_, &op_);
/* write out pages (if any) */ /* write out pages (if any) */
while (true) while (true)
{ {
int result = ogg_stream_flush(&os, &og); int result = ogg_stream_flush(&os_, &og_);
if (result == 0) if (result == 0)
break; break;
res = os.granulepos - lastGranulepos; res = os_.granulepos - lastGranulepos_;
size_t nextLen = pos + og.header_len + og.body_len; size_t nextLen = pos + og_.header_len + og_.body_len;
// make chunk larger // make chunk larger
if (oggChunk->payloadSize < nextLen) if (oggChunk->payloadSize < nextLen)
oggChunk->payload = (char*)realloc(oggChunk->payload, nextLen); oggChunk->payload = (char*)realloc(oggChunk->payload, nextLen);
memcpy(oggChunk->payload + pos, og.header, og.header_len); memcpy(oggChunk->payload + pos, og_.header, og_.header_len);
pos += og.header_len; pos += og_.header_len;
memcpy(oggChunk->payload + pos, og.body, og.body_len); memcpy(oggChunk->payload + pos, og_.body, og_.body_len);
pos += og.body_len; pos += og_.body_len;
if (ogg_page_eos(&og)) if (ogg_page_eos(&og_))
break; break;
} }
} }
@ -129,7 +129,7 @@ void OggEncoder::encode(const msg::PcmChunk* chunk)
{ {
res /= (sampleFormat_.rate / 1000.); res /= (sampleFormat_.rate / 1000.);
// logO << "res: " << res << "\n"; // logO << "res: " << res << "\n";
lastGranulepos = os.granulepos; lastGranulepos_ = os_.granulepos;
// make oggChunk smaller // make oggChunk smaller
oggChunk->payload = (char*)realloc(oggChunk->payload, pos); oggChunk->payload = (char*)realloc(oggChunk->payload, pos);
oggChunk->payloadSize = pos; oggChunk->payloadSize = pos;
@ -164,7 +164,7 @@ void OggEncoder::initEncoder()
} }
/********** Encode setup ************/ /********** Encode setup ************/
vorbis_info_init(&vi); vorbis_info_init(&vi_);
/* choose an encoding mode. A few possibilities commented out, one /* choose an encoding mode. A few possibilities commented out, one
actually used: */ actually used: */
@ -195,7 +195,7 @@ void OggEncoder::initEncoder()
*********************************************************************/ *********************************************************************/
int ret = vorbis_encode_init_vbr(&vi, sampleFormat_.channels, sampleFormat_.rate, quality); int ret = vorbis_encode_init_vbr(&vi_, sampleFormat_.channels, sampleFormat_.rate, quality);
/* do not continue if setup failed; this can happen if we ask for a /* do not continue if setup failed; this can happen if we ask for a
mode that libVorbis does not support (eg, too low a bitrate, etc, mode that libVorbis does not support (eg, too low a bitrate, etc,
@ -205,20 +205,20 @@ void OggEncoder::initEncoder()
throw SnapException("failed to init encoder"); throw SnapException("failed to init encoder");
/* add a comment */ /* add a comment */
vorbis_comment_init(&vc); vorbis_comment_init(&vc_);
vorbis_comment_add_tag(&vc, "TITLE", "SnapStream"); vorbis_comment_add_tag(&vc_, "TITLE", "SnapStream");
vorbis_comment_add_tag(&vc, "VERSION", VERSION); vorbis_comment_add_tag(&vc_, "VERSION", VERSION);
vorbis_comment_add_tag(&vc, "SAMPLE_FORMAT", sampleFormat_.getFormat().c_str()); vorbis_comment_add_tag(&vc_, "SAMPLE_FORMAT", sampleFormat_.getFormat().c_str());
/* set up the analysis state and auxiliary encoding storage */ /* set up the analysis state and auxiliary encoding storage */
vorbis_analysis_init(&vd, &vi); vorbis_analysis_init(&vd_, &vi_);
vorbis_block_init(&vd, &vb); vorbis_block_init(&vd_, &vb_);
/* set up our packet->stream encoder */ /* set up our packet->stream encoder */
/* pick a random serial number; that way we can more likely build /* pick a random serial number; that way we can more likely build
chained streams just by concatenation */ chained streams just by concatenation */
srand(time(NULL)); srand(time(NULL));
ogg_stream_init(&os, rand()); ogg_stream_init(&os_, rand());
/* Vorbis streams begin with three headers; the initial header (with /* Vorbis streams begin with three headers; the initial header (with
most of the codec setup parameters) which is mandated by the Ogg most of the codec setup parameters) which is mandated by the Ogg
@ -231,10 +231,10 @@ void OggEncoder::initEncoder()
ogg_packet header_comm; ogg_packet header_comm;
ogg_packet header_code; ogg_packet header_code;
vorbis_analysis_headerout(&vd, &vc, &header, &header_comm, &header_code); vorbis_analysis_headerout(&vd_, &vc_, &header, &header_comm, &header_code);
ogg_stream_packetin(&os, &header); ogg_stream_packetin(&os_, &header);
ogg_stream_packetin(&os, &header_comm); ogg_stream_packetin(&os_, &header_comm);
ogg_stream_packetin(&os, &header_code); ogg_stream_packetin(&os_, &header_code);
/* This ensures the actual /* This ensures the actual
* audio data will start on a new page, as per spec * audio data will start on a new page, as per spec
@ -243,16 +243,16 @@ void OggEncoder::initEncoder()
headerChunk_.reset(new msg::CodecHeader("ogg")); headerChunk_.reset(new msg::CodecHeader("ogg"));
while (true) while (true)
{ {
int result = ogg_stream_flush(&os, &og); int result = ogg_stream_flush(&os_, &og_);
if (result == 0) if (result == 0)
break; break;
headerChunk_->payloadSize += og.header_len + og.body_len; headerChunk_->payloadSize += og_.header_len + og_.body_len;
headerChunk_->payload = (char*)realloc(headerChunk_->payload, headerChunk_->payloadSize); headerChunk_->payload = (char*)realloc(headerChunk_->payload, headerChunk_->payloadSize);
logD << "HeadLen: " << og.header_len << ", bodyLen: " << og.body_len << ", result: " << result << "\n"; logD << "HeadLen: " << og_.header_len << ", bodyLen: " << og_.body_len << ", result: " << result << "\n";
memcpy(headerChunk_->payload + pos, og.header, og.header_len); memcpy(headerChunk_->payload + pos, og_.header, og_.header_len);
pos += og.header_len; pos += og_.header_len;
memcpy(headerChunk_->payload + pos, og.body, og.body_len); memcpy(headerChunk_->payload + pos, og_.body, og_.body_len);
pos += og.body_len; pos += og_.body_len;
} }
} }

18
server/encoder/oggEncoder.h
View file

@ -35,19 +35,17 @@ protected:
virtual void initEncoder(); virtual void initEncoder();
private: private:
ogg_stream_state os; /* take physical pages, weld into a logical ogg_stream_state os_; /// take physical pages, weld into a logical stream of packets
stream of packets */ ogg_page og_; /// one Ogg bitstream page. Vorbis packets are inside
ogg_page og; /* one Ogg bitstream page. Vorbis packets are inside */ ogg_packet op_; /// one raw packet of data for decode
ogg_packet op; /* one raw packet of data for decode */
vorbis_info vi; /* struct that stores all the static vorbis bitstream vorbis_info vi_; /// struct that stores all the static vorbis bitstream settings
settings */ vorbis_comment vc_; /// struct that stores all the user comments
vorbis_comment vc; /* struct that stores all the user comments */
vorbis_dsp_state vd; /* central working state for the packet->PCM decoder */ vorbis_dsp_state vd_; /// central working state for the packet->PCM decoder
vorbis_block vb; /* local working space for packet->PCM decode */ vorbis_block vb_; /// local working space for packet->PCM decode
ogg_int64_t lastGranulepos; ogg_int64_t lastGranulepos_;
}; };

2
server/publishZeroConf/publishAvahi.cpp
View file

@ -161,7 +161,7 @@ void PublishAvahi::create_services(AvahiClient *c)
} }
/// Add an additional (hypothetic) subtype /// Add an additional (hypothetic) subtype
/* 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)) /* 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))
{ {
fprintf(stderr, "Failed to add subtype _magic._sub._printer._tcp: %s\n", avahi_strerror(ret)); fprintf(stderr, "Failed to add subtype _magic._sub._printer._tcp: %s\n", avahi_strerror(ret));
goto fail; goto fail;