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reformatted code
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Johannes Pohl committed Oct 20, 2016
1 parent 286a107 commit d9ee0c1
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Showing 5 changed files with 104 additions and 105 deletions.
113 changes: 57 additions & 56 deletions client/player/coreAudioPlayer.cpp
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Original file line number Diff line number Diff line change
Expand Up @@ -47,18 +47,19 @@ CoreAudioPlayer::~CoreAudioPlayer()

void CoreAudioPlayer::playerCallback(AudioQueueRef queue, AudioQueueBufferRef bufferRef)
{
/// Estimate the playout delay by checking the number of frames left in the buffer
/// and add ms_ (= complete buffer size). Based on trying.
AudioTimeStamp timestamp;
AudioQueueGetCurrentTime(queue, timeLine, &timestamp, NULL);
size_t bufferedFrames = (frames_ - ((uint64_t)timestamp.mSampleTime % frames_)) % frames_;
size_t bufferedMs = bufferedFrames * 1000 / pubStream_->getFormat().rate + (ms_ * (NUM_BUFFERS - 1));
/// 15ms DAC delay. Based on trying.
bufferedMs += 15;
/// Estimate the playout delay by checking the number of frames left in the buffer
/// and add ms_ (= complete buffer size). Based on trying.
AudioTimeStamp timestamp;
AudioQueueGetCurrentTime(queue, timeLine_, &timestamp, NULL);
size_t bufferedFrames = (frames_ - ((uint64_t)timestamp.mSampleTime % frames_)) % frames_;
size_t bufferedMs = bufferedFrames * 1000 / pubStream_->getFormat().rate + (ms_ * (NUM_BUFFERS - 1));
/// 15ms DAC delay. Based on trying.
bufferedMs += 15;
// 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);
char *buffer = (char*)bufferRef->mAudioData;
char *buffer = (char*)bufferRef->mAudioData;
if (!pubStream_->getPlayerChunk(buffer, delay, frames_))
{
logO << "Failed to get chunk. Playing silence.\n";
Expand All @@ -70,60 +71,60 @@ void CoreAudioPlayer::playerCallback(AudioQueueRef queue, AudioQueueBufferRef bu
}

// OSStatus status =
AudioQueueEnqueueBuffer(queue, bufferRef, 0, NULL);

if (!active_)
{
AudioQueueStop(queue, false);
AudioQueueDispose(queue, false);
CFRunLoopStop(CFRunLoopGetCurrent());
}
AudioQueueEnqueueBuffer(queue, bufferRef, 0, NULL);

if (!active_)
{
AudioQueueStop(queue, false);
AudioQueueDispose(queue, false);
CFRunLoopStop(CFRunLoopGetCurrent());
}
}


void CoreAudioPlayer::worker()
{
const SampleFormat& sampleFormat = pubStream_->getFormat();

AudioStreamBasicDescription format;
format.mSampleRate = sampleFormat.rate;
format.mFormatID = kAudioFormatLinearPCM;
format.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger;// | kAudioFormatFlagIsPacked;
format.mBitsPerChannel = sampleFormat.bits;
format.mChannelsPerFrame = sampleFormat.channels;
format.mBytesPerFrame = sampleFormat.frameSize;
format.mFramesPerPacket = 1;
format.mBytesPerPacket = format.mBytesPerFrame * format.mFramesPerPacket;
format.mReserved = 0;

AudioQueueRef queue;
AudioQueueNewOutput(&format, callback, this, CFRunLoopGetCurrent(), kCFRunLoopCommonModes, 0, &queue);
AudioQueueCreateTimeline(queue, &timeLine);
// Apple recommends this as buffer size:
// 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 minBufferSize = 0x4000; // limit minimum size to 16K
//
// For 100ms @ 48000:16:2 we have 19.2K
// frames: 4800, ms: 100, buffer size: 19200
const SampleFormat& sampleFormat = pubStream_->getFormat();

AudioStreamBasicDescription format;
format.mSampleRate = sampleFormat.rate;
format.mFormatID = kAudioFormatLinearPCM;
format.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger;// | kAudioFormatFlagIsPacked;
format.mBitsPerChannel = sampleFormat.bits;
format.mChannelsPerFrame = sampleFormat.channels;
format.mBytesPerFrame = sampleFormat.frameSize;
format.mFramesPerPacket = 1;
format.mBytesPerPacket = format.mBytesPerFrame * format.mFramesPerPacket;
format.mReserved = 0;

AudioQueueRef queue;
AudioQueueNewOutput(&format, callback, this, CFRunLoopGetCurrent(), kCFRunLoopCommonModes, 0, &queue);
AudioQueueCreateTimeline(queue, &timeLine_);
// Apple recommends this as buffer size:
// 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 minBufferSize = 0x4000; // limit minimum size to 16K
//
// For 100ms @ 48000:16:2 we have 19.2K
// frames: 4800, ms: 100, buffer size: 19200
frames_ = (sampleFormat.rate * ms_) / 1000;
ms_ = frames_ * 1000 / sampleFormat.rate;
ms_ = frames_ * 1000 / sampleFormat.rate;
buff_size_ = frames_ * sampleFormat.frameSize;
logO << "frames: " << frames_ << ", ms: " << ms_ << ", buffer size: " << buff_size_ << "\n";
AudioQueueBufferRef buffers[NUM_BUFFERS];
for (int i = 0; i < NUM_BUFFERS; i++)
{
AudioQueueAllocateBuffer(queue, buff_size_, &buffers[i]);
buffers[i]->mAudioDataByteSize = buff_size_;
callback(this, queue, buffers[i]);
}

logE << "CoreAudioPlayer::worker\n";
AudioQueueCreateTimeline(queue, &timeLine);
AudioQueueStart(queue, NULL);
CFRunLoopRun();
logO << "frames: " << frames_ << ", ms: " << ms_ << ", buffer size: " << buff_size_ << "\n";
AudioQueueBufferRef buffers[NUM_BUFFERS];
for (int i = 0; i < NUM_BUFFERS; i++)
{
AudioQueueAllocateBuffer(queue, buff_size_, &buffers[i]);
buffers[i]->mAudioDataByteSize = buff_size_;
callback(this, queue, buffers[i]);
}

logE << "CoreAudioPlayer::worker\n";
AudioQueueCreateTimeline(queue, &timeLine_);
AudioQueueStart(queue, NULL);
CFRunLoopRun();
}


2 changes: 1 addition & 1 deletion client/player/coreAudioPlayer.h
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Original file line number Diff line number Diff line change
Expand Up @@ -45,7 +45,7 @@ class CoreAudioPlayer : public Player
protected:
virtual void worker();

AudioQueueTimelineRef timeLine;
AudioQueueTimelineRef timeLine_;
size_t ms_;
size_t frames_;
size_t buff_size_;
Expand Down
74 changes: 37 additions & 37 deletions server/encoder/oggEncoder.cpp
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Original file line number Diff line number Diff line change
Expand Up @@ -28,7 +28,7 @@
using namespace std;


OggEncoder::OggEncoder(const std::string& codecOptions) : Encoder(codecOptions), lastGranulepos(0)
OggEncoder::OggEncoder(const std::string& codecOptions) : Encoder(codecOptions), lastGranulepos_(0)
{
}

Expand Down Expand Up @@ -56,7 +56,7 @@ void OggEncoder::encode(const msg::PcmChunk* chunk)
double res = 0;
logD << "payload: " << chunk->payloadSize << "\tframes: " << chunk->getFrameCount() << "\tduration: " << chunk->duration<chronos::msec>().count() << "\n";
int frames = chunk->getFrameCount();
float **buffer=vorbis_analysis_buffer(&vd, frames);
float **buffer=vorbis_analysis_buffer(&vd_, frames);

/* uninterleave samples */
for (size_t channel = 0; channel < sampleFormat_.channels; ++channel)
Expand All @@ -82,44 +82,44 @@ void OggEncoder::encode(const msg::PcmChunk* chunk)
}

/* 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);

/* vorbis does some data preanalysis, then divvies up blocks for
more involved (potentially parallel) processing. Get a single
block for encoding now */
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 */
vorbis_analysis(&vb, NULL);
vorbis_bitrate_addblock(&vb);
vorbis_analysis(&vb_, NULL);
vorbis_bitrate_addblock(&vb_);

while (vorbis_bitrate_flushpacket(&vd, &op))
while (vorbis_bitrate_flushpacket(&vd_, &op_))
{
/* weld the packet into the bitstream */
ogg_stream_packetin(&os, &op);
ogg_stream_packetin(&os_, &op_);

/* write out pages (if any) */
while (true)
{
int result = ogg_stream_flush(&os, &og);
int result = ogg_stream_flush(&os_, &og_);
if (result == 0)
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
if (oggChunk->payloadSize < nextLen)
oggChunk->payload = (char*)realloc(oggChunk->payload, nextLen);

memcpy(oggChunk->payload + pos, og.header, og.header_len);
pos += og.header_len;
memcpy(oggChunk->payload + pos, og.body, og.body_len);
pos += og.body_len;
memcpy(oggChunk->payload + pos, og_.header, og_.header_len);
pos += og_.header_len;
memcpy(oggChunk->payload + pos, og_.body, og_.body_len);
pos += og_.body_len;

if (ogg_page_eos(&og))
if (ogg_page_eos(&og_))
break;
}
}
Expand All @@ -129,7 +129,7 @@ void OggEncoder::encode(const msg::PcmChunk* chunk)
{
res /= (sampleFormat_.rate / 1000.);
// logO << "res: " << res << "\n";
lastGranulepos = os.granulepos;
lastGranulepos_ = os_.granulepos;
// make oggChunk smaller
oggChunk->payload = (char*)realloc(oggChunk->payload, pos);
oggChunk->payloadSize = pos;
Expand Down Expand Up @@ -164,7 +164,7 @@ void OggEncoder::initEncoder()
}

/********** Encode setup ************/
vorbis_info_init(&vi);
vorbis_info_init(&vi_);

/* choose an encoding mode. A few possibilities commented out, one
actually used: */
Expand Down Expand Up @@ -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
mode that libVorbis does not support (eg, too low a bitrate, etc,
Expand All @@ -205,20 +205,20 @@ void OggEncoder::initEncoder()
throw SnapException("failed to init encoder");

/* add a comment */
vorbis_comment_init(&vc);
vorbis_comment_add_tag(&vc, "TITLE", "SnapStream");
vorbis_comment_add_tag(&vc, "VERSION", VERSION);
vorbis_comment_add_tag(&vc, "SAMPLE_FORMAT", sampleFormat_.getFormat().c_str());
vorbis_comment_init(&vc_);
vorbis_comment_add_tag(&vc_, "TITLE", "SnapStream");
vorbis_comment_add_tag(&vc_, "VERSION", VERSION);
vorbis_comment_add_tag(&vc_, "SAMPLE_FORMAT", sampleFormat_.getFormat().c_str());

/* set up the analysis state and auxiliary encoding storage */
vorbis_analysis_init(&vd, &vi);
vorbis_block_init(&vd, &vb);
vorbis_analysis_init(&vd_, &vi_);
vorbis_block_init(&vd_, &vb_);

/* set up our packet->stream encoder */
/* pick a random serial number; that way we can more likely build
chained streams just by concatenation */
srand(time(NULL));
ogg_stream_init(&os, rand());
ogg_stream_init(&os_, rand());

/* Vorbis streams begin with three headers; the initial header (with
most of the codec setup parameters) which is mandated by the Ogg
Expand All @@ -231,10 +231,10 @@ void OggEncoder::initEncoder()
ogg_packet header_comm;
ogg_packet header_code;

vorbis_analysis_headerout(&vd, &vc, &header, &header_comm, &header_code);
ogg_stream_packetin(&os, &header);
ogg_stream_packetin(&os, &header_comm);
ogg_stream_packetin(&os, &header_code);
vorbis_analysis_headerout(&vd_, &vc_, &header, &header_comm, &header_code);
ogg_stream_packetin(&os_, &header);
ogg_stream_packetin(&os_, &header_comm);
ogg_stream_packetin(&os_, &header_code);

/* This ensures the actual
* audio data will start on a new page, as per spec
Expand All @@ -243,16 +243,16 @@ void OggEncoder::initEncoder()
headerChunk_.reset(new msg::CodecHeader("ogg"));
while (true)
{
int result = ogg_stream_flush(&os, &og);
int result = ogg_stream_flush(&os_, &og_);
if (result == 0)
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);
logD << "HeadLen: " << og.header_len << ", bodyLen: " << og.body_len << ", result: " << result << "\n";
memcpy(headerChunk_->payload + pos, og.header, og.header_len);
pos += og.header_len;
memcpy(headerChunk_->payload + pos, og.body, og.body_len);
pos += og.body_len;
logD << "HeadLen: " << og_.header_len << ", bodyLen: " << og_.body_len << ", result: " << result << "\n";
memcpy(headerChunk_->payload + pos, og_.header, og_.header_len);
pos += og_.header_len;
memcpy(headerChunk_->payload + pos, og_.body, og_.body_len);
pos += og_.body_len;
}
}

Expand Down
18 changes: 8 additions & 10 deletions server/encoder/oggEncoder.h
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Original file line number Diff line number Diff line change
Expand Up @@ -35,19 +35,17 @@ class OggEncoder : public Encoder
virtual void initEncoder();

private:
ogg_stream_state os; /* take physical pages, weld into a logical
stream of packets */
ogg_page og; /* one Ogg bitstream page. Vorbis packets are inside */
ogg_packet op; /* one raw packet of data for decode */
ogg_stream_state os_; /// take physical pages, weld into a logical stream of packets
ogg_page og_; /// one Ogg bitstream page. Vorbis packets are inside
ogg_packet op_; /// one raw packet of data for decode

vorbis_info vi; /* struct that stores all the static vorbis bitstream
settings */
vorbis_comment vc; /* struct that stores all the user comments */
vorbis_info vi_; /// struct that stores all the static vorbis bitstream settings
vorbis_comment vc_; /// struct that stores all the user comments

vorbis_dsp_state vd; /* central working state for the packet->PCM decoder */
vorbis_block vb; /* local working space for packet->PCM decode */
vorbis_dsp_state vd_; /// central working state for the packet->PCM decoder
vorbis_block vb_; /// local working space for packet->PCM decode

ogg_int64_t lastGranulepos;
ogg_int64_t lastGranulepos_;
};


Expand Down
2 changes: 1 addition & 1 deletion server/publishZeroConf/publishAvahi.cpp
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Original file line number Diff line number Diff line change
Expand Up @@ -161,7 +161,7 @@ void PublishAvahi::create_services(AvahiClient *c)
}

/// 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));
goto fail;
Expand Down

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