Update to 0.2.6

roc/channel_set.go

@@ -1,5 +1,5 @@
 // DO NOT EDIT! Code generated by generate_enums script from roc-toolkit
-// roc-toolkit git tag: v0.2.5-11-g14d642e9, commit: 14d642e9
+// roc-toolkit git tag: v0.2.6, commit: 3cbb3d13
 
 package roc
 

roc/clock_source.go

@@ -1,5 +1,5 @@
 // DO NOT EDIT! Code generated by generate_enums script from roc-toolkit
-// roc-toolkit git tag: v0.2.5-11-g14d642e9, commit: 14d642e9
+// roc-toolkit git tag: v0.2.6, commit: 3cbb3d13
 
 package roc
 

roc/context_config.go

@@ -5,14 +5,16 @@ package roc
 // See also Context.
 type ContextConfig struct {
 	// Maximum size in bytes of a network packet.
-	// Defines the amount of bytes allocated per network packet.
-	// Sender and receiver won't handle packets larger than this.
-	// If zero, default value is used.
+	//
+	// Defines the amount of bytes allocated per network packet. Sender and
+	// receiver won't handle packets larger than this. If zero, default value is
+	// used.
 	MaxPacketSize uint32
 
 	// Maximum size in bytes of an audio frame.
+	//
 	// Defines the amount of bytes allocated per intermediate internal frame in the
-	// pipeline. Does not limit the size of the frames provided by user.
-	// If zero, default value is used.
+	// pipeline. Does not limit the size of the frames provided by user. If zero,
+	// default value is used.
 	MaxFrameSize uint32
 }

roc/fec_encoding.go

@@ -1,5 +1,5 @@
 // DO NOT EDIT! Code generated by generate_enums script from roc-toolkit
-// roc-toolkit git tag: v0.2.5-11-g14d642e9, commit: 14d642e9
+// roc-toolkit git tag: v0.2.6, commit: 3cbb3d13
 
 package roc
 

roc/frame_encoding.go

@@ -1,5 +1,5 @@
 // DO NOT EDIT! Code generated by generate_enums script from roc-toolkit
-// roc-toolkit git tag: v0.2.5-11-g14d642e9, commit: 14d642e9
+// roc-toolkit git tag: v0.2.6, commit: 3cbb3d13
 
 package roc
 

roc/interface.go

@@ -1,5 +1,5 @@
 // DO NOT EDIT! Code generated by generate_enums script from roc-toolkit
-// roc-toolkit git tag: v0.2.5-11-g14d642e9, commit: 14d642e9
+// roc-toolkit git tag: v0.2.6, commit: 3cbb3d13
 
 package roc
 

roc/packet_encoding.go

@@ -1,5 +1,5 @@
 // DO NOT EDIT! Code generated by generate_enums script from roc-toolkit
-// roc-toolkit git tag: v0.2.5-11-g14d642e9, commit: 14d642e9
+// roc-toolkit git tag: v0.2.6, commit: 3cbb3d13
 
 package roc
 

roc/protocol.go

@@ -1,5 +1,5 @@
 // DO NOT EDIT! Code generated by generate_enums script from roc-toolkit
-// roc-toolkit git tag: v0.2.5-11-g14d642e9, commit: 14d642e9
+// roc-toolkit git tag: v0.2.6, commit: 3cbb3d13
 
 package roc
 

roc/receiver_config.go

@@ -7,71 +7,79 @@ import "time"
 // See also Receiver.
 type ReceiverConfig struct {
 	// The rate of the samples in the frames returned to the user.
-	// Number of samples per channel per second.
-	// Should be set.
+	//
+	// Number of samples per channel per second. Should be set.
 	FrameSampleRate uint32
 
 	// The channel set in the frames returned to the user.
+	//
 	// Should be set.
 	FrameChannels ChannelSet
 
 	// The sample encoding in the frames returned to the user.
+	//
 	// Should be set.
 	FrameEncoding FrameEncoding
 
 	// Clock source to use.
-	// Defines whether read operation will be blocking or non-blocking.
-	// If zero, default value is used.
+	//
+	// Defines whether read operation will be blocking or non-blocking. If zero,
+	// default value is used.
 	ClockSource ClockSource
 
 	// Resampler backend to use.
 	ResamplerBackend ResamplerBackend
 
 	// Resampler profile to use.
+	//
 	// If non-zero, the receiver employs resampler for two purposes:
 	//  - adjust the sender clock to the receiver clock, which may differ a bit
-	//  - convert the packet sample rate to the frame sample rate if they are different
+	//  - convert the packet sample rate to the frame sample rate if they are
+	//    different
 	ResamplerProfile ResamplerProfile
 
 	// Target latency, in nanoseconds.
-	// The session will not start playing until it accumulates the requested latency.
-	// Then, if resampler is enabled, the session will adjust its clock to keep actual
-	// latency as close as possible to the target latency.
-	// If zero, default value is used. Should not be negative, otherwise an error is returned.
+	//
+	// The session will not start playing until it accumulates the requested
+	// latency. Then, if resampler is enabled, the session will adjust its clock to
+	// keep actual latency as close as possible to the target latency. If zero,
+	// default value is used.
 	TargetLatency time.Duration
 
 	// Maximum delta between current and target latency, in nanoseconds.
-	// If current latency becomes larger than the target latency plus this value, the
-	// session is terminated.
-	// If zero, default value is used. Should not be negative, otherwise an error is returned.
+	//
+	// If current latency becomes larger than the target latency plus this value,
+	// the session is terminated. If zero, default value is used.
 	MaxLatencyOverrun time.Duration
 
 	// Maximum delta between target and current latency, in nanoseconds.
-	// If current latency becomes smaller than the target latency minus this value, the
-	// session is terminated.
-	// May be larger than the target latency because current latency may be negative,
-	// which means that the playback run ahead of the last packet received from network.
-	// If zero, default value is used.  Should not be negative, otherwise an error is returned.
+	//
+	// If current latency becomes smaller than the target latency minus this value,
+	// the session is terminated. May be larger than the target latency because
+	// current latency may be negative, which means that the playback run ahead of
+	// the last packet received from network. If zero, default value is used.
 	MaxLatencyUnderrun time.Duration
 
 	// Timeout for the lack of playback, in nanoseconds.
-	// If there is no playback during this period, the session is terminated.
-	// This mechanism allows to detect dead, hanging, or broken clients
-	// generating invalid packets.
-	// If zero, default value is used. If negative, the timeout is disabled.
+	//
+	// If there is no playback during this period, the session is terminated. This
+	// mechanism allows to detect dead, hanging, or broken clients generating
+	// invalid packets. If zero, default value is used. If negative, the timeout is
+	// disabled.
 	NoPlaybackTimeout time.Duration
 
 	// Timeout for broken playback, in nanoseconds.
+	//
 	// If there the playback is considered broken during this period, the session
-	// is terminated. The playback is broken if there is a breakage detected at every
-	// BreakageDetectionWindow during BrokenPlaybackTimeout.
-	// This mechanism allows to detect vicious circles like when all client packets
-	// are a bit late and receiver constantly drops them producing unpleasant noise.
-	// If zero, default value is used. If negative, the timeout is disabled.
+	// is terminated. The playback is broken if there is a breakage detected at
+	// every BreakageDetectionWindow during BrokenPlaybackTimeout. This mechanism
+	// allows to detect vicious circles like when all client packets are a bit late
+	// and receiver constantly drops them producing unpleasant noise. If zero,
+	// default value is used. If negative, the timeout is disabled.
 	BrokenPlaybackTimeout time.Duration
 
 	// Breakage detection window, in nanoseconds.
-	// If zero, default value is used. Should not be negative, otherwise an error is returned.
-	// See BrokenPlaybackTimeout.
+	//
+	// If zero, default value is used. See BrokenPlaybackTimeout.
 	BreakageDetectionWindow time.Duration
 }

roc/resampler_backend.go

@@ -1,5 +1,5 @@
 // DO NOT EDIT! Code generated by generate_enums script from roc-toolkit
-// roc-toolkit git tag: v0.2.5-11-g14d642e9, commit: 14d642e9
+// roc-toolkit git tag: v0.2.6, commit: 3cbb3d13
 
 package roc
 

roc/resampler_profile.go

@@ -1,5 +1,5 @@
 // DO NOT EDIT! Code generated by generate_enums script from roc-toolkit
-// roc-toolkit git tag: v0.2.5-11-g14d642e9, commit: 14d642e9
+// roc-toolkit git tag: v0.2.6, commit: 3cbb3d13
 
 package roc
 

roc/sender_config.go

@@ -7,74 +7,81 @@ import "time"
 // See also Sender.
 type SenderConfig struct {
 	// The rate of the samples in the frames passed to sender.
-	// Number of samples per channel per second.
-	// If FrameSampleRate and PacketSampleRate are different,
-	// resampler should be enabled.
-	// Should be set.
+	//
+	// Number of samples per channel per second. If FrameSampleRate and
+	// PacketSampleRate are different, resampler should be enabled. Should be set.
 	FrameSampleRate uint32
 
 	// The channel set in the frames passed to sender.
+	//
 	// Should be set.
 	FrameChannels ChannelSet
 
 	// The sample encoding in the frames passed to sender.
+	//
 	// Should be set.
 	FrameEncoding FrameEncoding
 
 	// The rate of the samples in the packets generated by sender.
-	// Number of samples per channel per second.
-	// If zero, default value is used.
+	//
+	// Number of samples per channel per second. If zero, default value is used.
 	PacketSampleRate uint32
 
 	// The channel set in the packets generated by sender.
+	//
 	// If zero, default value is used.
 	PacketChannels ChannelSet
 
 	// The sample encoding in the packets generated by sender.
+	//
 	// If zero, default value is used.
 	PacketEncoding PacketEncoding
 
 	// The length of the packets produced by sender, in nanoseconds.
-	// Number of nanoseconds encoded per packet.
-	// The samples written to the sender are buffered until the full packet is
-	// accumulated or the sender is flushed or closed. Larger number reduces
-	// packet overhead but also increases latency.
-	// If zero, default value is used. Should not be negative, otherwise an error is returned.
+	//
+	// Number of nanoseconds encoded per packet. The samples written to the sender
+	// are buffered until the full packet is accumulated or the sender is flushed
+	// or closed. Larger number reduces packet overhead but also increases latency.
+	// If zero, default value is used.
 	PacketLength time.Duration
 
 	// Enable packet interleaving.
-	// If true, the sender shuffles packets before sending them. This
-	// may increase robustness but also increases latency.
+	//
+	// If true, the sender shuffles packets before sending them. This may
+	// increase robustness but also increases latency.
 	PacketInterleaving bool
 
 	// Clock source to use.
-	// Defines whether write operation will be blocking or non-blocking.
-	// If zero, default value is used.
+	//
+	// Defines whether write operation will be blocking or non-blocking. If zero,
+	// default value is used.
 	ClockSource ClockSource
 
 	// Resampler backend to use.
 	ResamplerBackend ResamplerBackend
 
 	// Resampler profile to use.
+	//
 	// If non-zero, the sender employs resampler if the frame sample rate differs
 	// from the packet sample rate.
 	ResamplerProfile ResamplerProfile
 
 	// FEC encoding to use.
-	// If non-zero, the sender employs a FEC codec to generate redundant packets
+	//
+	// If non-zero, the sender employs a FEC encoding to generate redundant packets
 	// which may be used on receiver to restore lost packets. This requires both
-	// sender and receiver to use two separate source and repair ports.
+	// sender and receiver to use two separate source and repair endpoints.
 	FecEncoding FecEncoding
 
 	// Number of source packets per FEC block.
-	// Used if some FEC code is selected.
-	// Larger number increases robustness but also increases latency.
-	// If zero, default value is used.
+	//
+	// Used if some FEC encoding is selected. Larger number increases robustness
+	// but also increases latency. If zero, default value is used.
 	FecBlockSourcePackets uint32
 
 	// Number of repair packets per FEC block.
-	// Used if some FEC code is selected.
-	// Larger number increases robustness but also increases traffic.
-	// If zero, default value is used.
+	//
+	// Used if some FEC encoding is selected. Larger number increases robustness
+	// but also increases traffic. If zero, default value is used.
 	FecBlockRepairPackets uint32
 }