diff --git a/client/player/coreAudioPlayer.cpp b/client/player/coreAudioPlayer.cpp index a61ca553..e70e5385 100644 --- a/client/player/coreAudioPlayer.cpp +++ b/client/player/coreAudioPlayer.cpp @@ -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, ×tamp, 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_, ×tamp, 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"; @@ -70,60 +71,60 @@ void CoreAudioPlayer::playerCallback(AudioQueueRef queue, AudioQueueBufferRef bu } // OSStatus status = - AudioQueueEnqueueBuffer(queue, bufferRef, 0, NULL); + AudioQueueEnqueueBuffer(queue, bufferRef, 0, NULL); - if (!active_) - { - AudioQueueStop(queue, false); - AudioQueueDispose(queue, false); - CFRunLoopStop(CFRunLoopGetCurrent()); - } + if (!active_) + { + AudioQueueStop(queue, false); + AudioQueueDispose(queue, false); + CFRunLoopStop(CFRunLoopGetCurrent()); + } } void CoreAudioPlayer::worker() { - const SampleFormat& sampleFormat = pubStream_->getFormat(); + 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; + 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 + 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]); - } + 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(); + logE << "CoreAudioPlayer::worker\n"; + AudioQueueCreateTimeline(queue, &timeLine_); + AudioQueueStart(queue, NULL); + CFRunLoopRun(); } diff --git a/client/player/coreAudioPlayer.h b/client/player/coreAudioPlayer.h index 014bc7fa..8098b874 100644 --- a/client/player/coreAudioPlayer.h +++ b/client/player/coreAudioPlayer.h @@ -45,7 +45,7 @@ public: protected: virtual void worker(); - AudioQueueTimelineRef timeLine; + AudioQueueTimelineRef timeLine_; size_t ms_; size_t frames_; size_t buff_size_; diff --git a/server/encoder/oggEncoder.cpp b/server/encoder/oggEncoder.cpp index 229f9080..fd69405a 100644 --- a/server/encoder/oggEncoder.cpp +++ b/server/encoder/oggEncoder.cpp @@ -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) { } @@ -56,7 +56,7 @@ void OggEncoder::encode(const msg::PcmChunk* chunk) double res = 0; logD << "payload: " << chunk->payloadSize << "\tframes: " << chunk->getFrameCount() << "\tduration: " << chunk->duration().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) @@ -82,7 +82,7 @@ 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); @@ -90,36 +90,36 @@ void OggEncoder::encode(const msg::PcmChunk* chunk) 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; } } @@ -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; @@ -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: */ @@ -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, @@ -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 @@ -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 @@ -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; } } diff --git a/server/encoder/oggEncoder.h b/server/encoder/oggEncoder.h index 19393f62..8c8412f4 100644 --- a/server/encoder/oggEncoder.h +++ b/server/encoder/oggEncoder.h @@ -35,19 +35,17 @@ protected: 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_; }; diff --git a/server/publishZeroConf/publishAvahi.cpp b/server/publishZeroConf/publishAvahi.cpp index acbfc71d..f6a75193 100644 --- a/server/publishZeroConf/publishAvahi.cpp +++ b/server/publishZeroConf/publishAvahi.cpp @@ -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;