From de15abf3ee0bc70d15370b2e58d559ca43ea21ed Mon Sep 17 00:00:00 2001 From: =?utf8?q?J=C3=B6rn=20Nettingsmeier?= Date: Tue, 18 Feb 2014 23:13:02 +0100 Subject: [PATCH] copy-edit chapter 19 --- _manual/19_synchronization.html | 3 - .../01_on-clock-and-time.html | 62 ++-- .../02_latency-and-latency-compensation.html | 256 +++++++++++----- .../03_timecode-generators-and-slaves.html | 283 +++++++++++------- ...overview-of-timecode-related-settings.html | 120 ++++++-- 5 files changed, 478 insertions(+), 246 deletions(-) diff --git a/_manual/19_synchronization.html b/_manual/19_synchronization.html index 3e9a735..752f2b8 100644 --- a/_manual/19_synchronization.html +++ b/_manual/19_synchronization.html @@ -3,9 +3,6 @@ layout: default title: Synchronization --- - - -

Ardour can be synchronized with a variety of external devices and other software.

{% children %} diff --git a/_manual/19_synchronization/01_on-clock-and-time.html b/_manual/19_synchronization/01_on-clock-and-time.html index 30861ff..c64a5d8 100644 --- a/_manual/19_synchronization/01_on-clock-and-time.html +++ b/_manual/19_synchronization/01_on-clock-and-time.html @@ -4,42 +4,70 @@ title: On Clock and Time ---

-Synchronization in multimedia involves two concepts which are often confused: clock (or speed) and time (location in time). + Synchronization in multimedia involves two concepts which are + often confused: clock (or speed) and time (location + in time).

-A clock is the mechanism by which two systems tick simultaneously. -In the audio world this is generally referred to as Word Clock. -It does not carry any absolute reference to a point in time: A clock is used to keep a systems sample rate constant, regular and accurate. -Word clock is usually at the frequency of the sample rate - ie at 48KHz, its period is about 20μs. Word Clock is the most common 'sample rate' based clock but other clocks do exist such as Black and Burst, Tri-Level and DARS. Sample rates can also be derived from these clocks as well. + A clock determines the speet at which one or more systems + operate. In the audio world this is generally referred to as + Word Clock. + It does not carry any absolute reference to a point in time: A clock is + used to keep a system's sample rate regular and accurate. + Word clock is usually at the frequency of the sample rate — + at 48 kHz, its period is about 20 Î¼s. Word Clock is the most + common sample rate based clock but other clocks do exist such as Black and + Burst, Tri-Level and DARS. Sample rates can be derived from these clocks as well.

-Time – or timecode – on the other hand specifies an absolute relationship or position on a timeline e.g. 01:02:03:04 (expressed as Hours:Mins:Secs:Frames). It is actual data and not a clock signal per se. -The granularity of timecode is Video Frames and is an order of magnitude lower than, say, Word Clock which is counted in samples. A typical frame rate is 25 fps with a period of 40ms. -In the case of 48KHz and 25fps, there are 1920 audio samples per video frame. + Time or timecode specifies an absolute position on a timeline, + such as 01:02:03:04 (expressed as Hours:Mins:Secs:Frames). It is + actual data and not a clock signal per se. + The granularity of timecode is Video Frames and is an order of + magnitude lower than, say, Word Clock which is counted in + samples. A typical frame rate is 25 fps with a period of + 40 ms. + In the case of 48 kHz and 25 fps, there are 1920 audio samples + per video frame.

-The concept of clock and timecode is reflected in JACK and Ardour: + The concepts of clock and timecode are reflected in JACK and Ardour:

-JACK provides clock synchronization and is not concerned with time code (this is not entirely true, more on jack-transport later). -Within software, jackd provides sample-accurate synchronization between all JACK applications. -On the hardware side JACK uses the clock of the audio-interface. Synchronization of multiple interfaces requires hardware support to sync the clocks. -If two interfaces run at different clocks the only way to align the signals is via re-sampling (SRC - Sample Rate Conversion) - which decreases fidelity. + JACK provides clock synchronization and is not concerned with time code + (this is not entirely true, more on jack-transport later). + On the software side, jackd provides sample-accurate synchronization + between all JACK applications. + On the hardware side, JACK uses the clock of the audio-interface. + Synchronization of multiple interfaces requires hardware support to sync + the clocks. + If two interfaces run at different clocks the only way to align the + signals is via re-sampling (SRC - Sample Rate Conversion), which is + expensive in terms of CPU usage and may decreases fidelity if done + incorrectly.

-Timecode is used to align systems already synchronized by a clock to a common point in time, this is application specific and various standards and methods exist to do this. + Timecode is used to align systems already synchronized by a clock to + a common point in time, this is application specific and various + standards and methods exist to do this.

-

-NB. to make things confusing, there are possibilities to synchronize clocks using timecode. e.g. using mechanism called jam-sync and a Phase-Locked-Loop. +

+ To make things confusing, there are possibilities to synchronize clocks + using timecode. e.g. using mechanism called jam-sync and a + phase-locked loop.

-An interesting point to note is that LTC (Linear Time Code) is a Manchester encoded, frequency modulated signal that carries both 'Clock' and 'Time'. It is possible to extract absolute position data and speed from it. + An interesting point to note is that LTC (Linear Time Code) is a + Manchester encoded, frequency modulated signal that carries both + clock and time. It is possible to extract absolute position data + and speed from it.

diff --git a/_manual/19_synchronization/02_latency-and-latency-compensation.html b/_manual/19_synchronization/02_latency-and-latency-compensation.html index 9954e0d..6b11a6b 100644 --- a/_manual/19_synchronization/02_latency-and-latency-compensation.html +++ b/_manual/19_synchronization/02_latency-and-latency-compensation.html @@ -1,152 +1,248 @@ --- layout: default title: Latency and Latency-Compensation -menu_title: About Latency +menu_title: Latency --- -

Latency

-

-When speaking about synchronization, it is also necessary to speak of latency. -Latency is a system's reaction time to a given stimulus. There are many factors that contribute to the total latency of a system. -In order to achieve exact time synchronization all sources of latency need to be taken into account and compensated for. + Latency + is a system's reaction time to a given stimulus. There are many factors that + contribute to the total latency of a system. In order to achieve exact time + synchronization all sources of latency need to be taken into account and + compensated for.

- -

Latency chain

-

Figure: Latency chain. The numbers are an example for a typical PC. With professional gear and an optimized system the total roundtrip latency is usually lower. The important point is that latency is always additive and a sum of many independent factors.

+

Sources of Latency

+

Sound propagation through the air

-There is not much that can done about the first two other than using headphones or sitting near the loudspeaker and buying quality gear. + Since sound is a mechanical perturbation in a fluid, it travels at + comparatively slow speed + of about 340 m/s. As a consequence, your acoustic guitar or piano has a + latency of about 1–2 ms, due to the propagation time of the sound + between your instrument and your ear.

- +

Digital-to-Analog and Analog-to-Digital conversion

-Processing latency is usually divided into capture latency and playback latency: + Electric signals travel quite fast (on the order of the speed of light), + so their propagation time is negligible in this context. But the conversions + between the analog and digital domain take a comparatively long time to perform, + so their contribution to the total latency may be considerable on + otherwise very low-latency systems. Conversion delay is usually below 1 ms.

- - - +

Digital Signal Processing

-But this division is an implementation detail of no great interest. What really matters is the combination of both. It is called processing roundtrip latency: the time necessary for a certain audio event to be captured, processed and played back. + Digital processors tend to process audio in chunks, and the size of that chunk + depends on the needs of the algorithm and performance/cost considerations. + This is usually the main cause of latency when you use a computer and one you + can try to predict and optimize.

- +

Computer I/O Architecture

-It is important to note that processing latency in a jackd is a matter of choice: It can be lowered within the limits imposed only by the hardware (audio-device, CPU and bus-speed) and audio driver. Lower latencies increase the load on the computer-system because it needs to process the audio in smaller chunks which arrive much more frequently. The lower the latency, the more likely the system will fail to meet its processing deadline and the dreaded xrun (short for buffer over-run and buffer under-run) will make its appearance more often, leaving its merry trail of clicks, pops and crackles. + A computer is a general purpose processor, not a digital audio processor. + This means our audio data has to jump a lot of fences in its path from the + outside to the CPU and back, contending in the process with some other parts + of the system vying for the same resources (CPU time, bus bandwidth, etc.)

+

The Latency chain

+ +Latency chain

-The digital I/O latency is usually negligible for integrated or PCI audio devices but for USB or FireWire interfaces the bus clocking and buffering can add some milliseconds. + Figure: Latency chain. + The numbers are an example for a typical PC. With professional gear and an + optimized system the total roundtrip latency is usually lower. The important + point is that latency is always additive and a sum of many independent factors.

-Low-latency is not always a feature you want to have. It comes with a couple of drawbacks: the most prominent is increased power-consumption because the CPU needs to process many small chunks of audio-data, it is constantly active and can not enter power-saving mode (think fan-noise). Furthermore, if more than one application (sound-processor) is involved in processing the sound, each of these needs to run for a short, well defined time for each audio-cycle which results in a much higher system-load and an increased chance of x-runs. Reliable low-latency (≤10ms) on GNU/Linux can usually only be achieved by running a realtime-kernel. + Processing latency is usually divided into capture latency (the time + it takes for the digitized audio to be available for digital processing, usually + one audio period), and playback latency (the time it takes for + In practice, the combination of both matters. It is called roundtrip + latency: the time necessary for a certain audio event to be captured, + processed and played back. +

+

+ It is important to note that processing latency in a jackd is a matter of + choice. It can be lowered within the limits imposed by the hardware (audio + device, CPU and bus speed) and audio driver. Lower latencies increase the + load on the system because it needs to process the audio in smaller chunks + which arrive much more frequently. The lower the latency, the more likely + the system will fail to meet its processing deadline and the dreaded + xrun (short for buffer over- or under-run) will make its + appearance more often, leaving its merry trail of clicks, pops and crackles.

-Yet there are a few situations where a low-latency is really important, because they require very quick response from the computer. + The digital I/O latency is usually negligible for integrated or + PCI audio devices, but + for USB or FireWire interfaces the bus clocking and buffering can add some + milliseconds.

- + +

Low Latency usecases

-In many other cases - such as playback, recording, overdubbing, mixing, mastering, etc. latency is not important, It can be relatively large and easily be compensated for. + Low latency is not always a feature you want to have. It + comes with a couple of drawbacks: the most prominent is increased power + consumption because the CPU needs to process many small chunks of audio data, + it is constantly active and can not enter power-saving mode (think fan-noise). + Since each application that is part of the signal chain must run in every + audio cycle, low-latency systems will undergocontext switches + between applications more often, which incur a significant overhead. + This results in a much higher system load and an increased chance of xruns.

-

-To explain that statement: During mixing or mastering you don't care if it takes 10ms or 100ms between the instant you press the play button and sound coming from the speaker. The same is true when recording with a count in. + For a few applications, low latency is critical:

- +

Playing virtual instruments

-During tracking it is important that the sound that is currently being played back is internally aligned with the sound that is being recorded. + A large delay between the pressing of the keys and the sound the instrument + produces will throw-off the timing of most instrumentalists (save church + organists, whom we believe to be awesome latency-compensation organic systems.)

- +

Software audio monitoring

-This is where latency-compensation comes into play. There are two possibilities to compensate for latency in a DAW: read-ahead the DAW starts playing a bit early (relative to the playhead), so that when the sound arrives at the speakers a short time later, it is exactly aligned with the material that is being recorded. -And write-behind; since we know that play-back has latency, the incoming audio can be delayed by the same amount to line things up again. + If a singer is hearing her own voice through two different paths, her head + bones and headphones, even small latencies can be very disturbing and + manifest as a tinny, irritating sound.

- +

Live effects

-As you may see, the second approach is prone to various implementation issues regarding timecode and transport synchronization. Ardour uses read-ahead to compensate for latency. The time displayed in the Ardour clock corresponds to the audio-signal that you hear on the speakers (and is not where Ardour reads files from disk). + Low latency is important when using the computer as an effect rack for + inline effects such as compression or EQ. For reverbs, slightly higher + latency might be tolerable, if the direct sound is not routed through the + computer. +

+

Live mixing

+

+ Some sound engineers use a computer for mixing live performances. + Basically that is a combination of the above: monitoring on stage, + effects processing and EQ.

-

-As a side note, this is also one of the reasons why many projects start at timecode 01:00:00:00. When compensating for output-latency the DAW will need to read data from before the start of the session so that the audio arrives in time at the output when the timecode hits 01:00:00:00. Ardour3 does handle the case of 00:00:00:00 properly but not all systems/software/hardware that you may inter-operate with may behave the same. + In many other cases, such as playback, recording, overdubbing, mixing, + mastering, etc. latency is not important, since it can easily be + compensated for.
+ To explain that statement: During mixing or mastering you don't care + if it takes 10ms or 100ms between the instant you press the play button + and sound coming from the speaker. The same is true when recording with a count in.

+

Latency compensation

+

+ During tracking it is important that the sound that is currently being + played back is internally aligned with the sound that is being recorded. +

+

+ This is where latency-compensation comes into play. There are two ways to + compensate for latency in a DAW, read-ahead and + write-behind. The DAW starts playing a bit early (relative to + the playhead), so that when the sound arrives at the speakers a short time + later, it is exactly aligned with the material that is being recorded. + Since we know that play-back has latency, the incoming audio can be delayed + by the same amount to line things up again. +

+

+ As you may see, the second approach is prone to various implementation + issues regarding timecode and transport synchronization. Ardour uses read-ahead + to compensate for latency. The time displayed in the Ardour clock corresponds + to the audio-signal that you hear on the speakers (and is not where Ardour + reads files from disk). +

+

+ As a side note, this is also one of the reasons why many projects start at + timecode 01:00:00:00. When compensating for output latency the + DAW will need to read data from before the start of the session, so that the + audio arrives in time at the output when the timecode hits 01:00:00:00. + Ardour3 does handle the case of 00:00:00:00 properly but not all + systems/software/hardware that you may inter-operate with may behave the same. +

Latency Compensation And Clock Sync

-To achieve sample accurate timecode synchronization, the latency introduced by the audio-setup needs to be known and compensated for. + To achieve sample accurate timecode synchronization, the latency introduced + by the audio setup needs to be known and compensated for.

-

-In order to compensate for Latency, JACK or JACK applications need to know exactly how long a certain signal needs to be read-ahead or delayed: + In order to compensate for latency, JACK or JACK applications need to know + exactly how long a certain signal needs to be read-ahead or delayed: +

+Jack Latency Compensation +

+ Figure: Jack Latency Compensation.

- -

Jack Latency Compensation

-

Figure: Jack Latency Compensation. This figure outlines the jack latency API. -- excerpt from http://jackaudio.org/files/jack-latency.png

-

-In the figure above, clients A and B need to be able to answer the following two questions: + In the figure above, clients A and B need to be able to answer the following + two questions:

-JACK features an API that allows applications to determine the answers to above questions. However JACK can not know about the additional latency that is introduced by the computer architecture, operating system and soundcard. These values are indicated by -I and -O and vary from system to system but are constant on each. On a general purpose computer system the only way to accurately learn about the total (additional) latency is to measure it. + JACK features an API + that allows applications to determine the answers to above questions. + However JACK can not know about the additional latency that is introduced + by the computer architecture, operating system and soundcard. These values + can be specified by the JACK command line parameters -I + and -O and vary from system + to system but are constant on each. On a general purpose computer system + the only way to accurately learn about the total (additional) latency is to + measure it.

Calibrating JACK Latency

-

-Linux DSP guru Fons Adriaensen wrote a tool called jack_delay to accurately measure the roundtrip latency of a closed loop audio chain, with sub-sample accuracy. JACK itself includes a variant of this tool called jack_iodelay. + Linux DSP guru Fons Adriaensen wrote a tool called jack_delay + to accurately measure the roundtrip latency of a closed loop audio chain, + with sub-sample accuracy. JACK itself includes a variant of this tool + called jack_iodelay.

-

-Jack_iodelay allows you to measure the total latency of the system, subtracts the known latency of JACK itself and suggests parameters for jackd's audio-backend -I and -O options. + Jack_iodelay allows you to measure the total latency of the system, + subtracts the known latency of JACK itself and suggests values for + jackd's audio-backend parameters.

-

-jack_[io]delay works by emitting some rather annoying tones, capturing them again after a round trip through the whole chain, and measuring the difference in phase so it can estimate with great accuracy the time taken. This is not a theoretical estimation, jack_delay is a measuring tool that provides very accurate answers. + jack_[io]delay works by emitting some rather annoying tones, capturing + them again after a round trip through the whole chain, and measuring the + difference in phase so it can estimate with great accuracy the time taken.

-

-You can close the loop in a number of ways: + You can close the loop in a number of ways:

-

-Once you have closed the loop you have to: + Once you have closed the loop you have to:

    -
  1. Launch jackd with the configuration you want to test.
    2. -
  2. Launch jack_delay on the commandline.
    3. -
  3. Make the appropriate connections between your jack ports so the loop is closed.
    4. -
  4. Adjust the playback and capture levels in your mixer.
    5. +
  5. Launch jackd with the configuration you want to test.
    6. +
  6. Launch jack_delay on the commandline.
    7. +
  7. Make the appropriate connections between your jack ports so the loop is closed.
    8. +
  8. Adjust the playback and capture levels in your mixer.
diff --git a/_manual/19_synchronization/03_timecode-generators-and-slaves.html b/_manual/19_synchronization/03_timecode-generators-and-slaves.html index babca14..dfc3041 100644 --- a/_manual/19_synchronization/03_timecode-generators-and-slaves.html +++ b/_manual/19_synchronization/03_timecode-generators-and-slaves.html @@ -3,108 +3,131 @@ layout: default title: Timecode Generators and Slaves --- -

Ardour Timecode Generators and Slaves

-There are three common timecode formats: + Ardour supports three common timecode formats: + LTC, + MTC, and + MIDI Clock, as well as + JACK-transport, a JACK-specific timecode implementation.

- - -

-As well as a JACK specific timecode implementation: -

- -

-Ardour supports all of these standards. -It can generate timecode and thus act as timecode master providing timecode information to other applications. -Ardour can also be slaved to some external source in which case the playhead follows the incoming timecode. -

- -

-Combining the timecode slave and generator modes, Ardour can also translate timecode. e.g create LTC timecode from incoming MTC. + Ardour can generate timecode and thus act as timecode master, + providing timecode information to other applications. Ardour can also be + slaved to some external source in which case the playhead + follows the incoming timecode.
+ Combining the timecode slave and generator modes, Ardour can also + translate timecode. e.g create LTC timecode from incoming MTC.

Ardour Timecode Configuration

-Each Ardour session has a specific timecode frames-per-second setting which is configured in session > properties > timecode. The selected timecode affects the timecode-ruler in the main window as well as the clock itself. + Each Ardour session has a specific timecode frames-per-second setting which + is configured in session > properties > + timecode. The selected timecode affects the timecoderuler in the main + window as well as the clock itself.

-Note that some timecode formats are limited to a subset of Ardour's available fps. e.g. MTC is limited to 24, 25, 29.97 and 30 fps. + Note that some timecode formats do not support all of Ardour's available + fps settings. MTC is limited to 24, 25, 29.97 and 30 fps.

-The video-pullup modes change the effective samplerate of Ardour to allows for changing a film soundtrack from one frame rate to another. The concept is beyond the scope of this manual, but wikipedia's entry on Telecine may get you started. + The video pull-up modes change the effective samplerate of Ardour to allow + for changing a film soundtrack from one frame rate to another. The concept is + beyond the scope of this manual, but Wikipedia's entry on + Telecine + may get you started.

Ardour Timecode Generator Configuration

-This is pretty straight forward: simply turn it on. The MTC and MIDI-Clock generator do not have any options. -For the LTC generator the volume of the generated LTC can be configured. JACK-transport can not be generated. Jack itself is always sample-sync to the jack-cycle and does not slave to anything. + This is pretty straightforward: simply turn it on. The MTC and MIDI-Clock + generator do not have any options. The LTC generator has a configurable + output level. JACK-transport cannot be generated. Jack itself is + always synced to its own cycle and cannot do varispeed — it will + always be synced to a hardware clock or another JACK master.

-The relevant settings for timecode generator can be found in the Preferences dialog: "MIDI Preferences" (for MTC, MClk) and "Transport Preferences" respectively. + The relevant settings for timecode generator can be found in + Edit > Preferences > MIDI Preferences (for MTC, + MC) and + Edit > Preferences > Transport Preferences + (for LTC).

-The timecode is sent to jack-ports ardour:MTC out, ardour:MIDI clock out and ardour:LTC-out. Multiple generators can be active simultaneously. + The timecode is sent to jack-ports ardour:MTC out, + ardour:MIDI clock out and ardour:LTC-out. Multiple + generators can be active simultaneously.

-

-Note that - at the time of writing this - only the LTC generator supports latency compensation. This is due to the fact the Ardour MIDI ports are not yet latency compensated. +

+ Note that, as of Jan 2014, only the LTC generator supports latency + compensation. This is due to the fact the Ardour MIDI ports are not + yet latency compensated.

-In session > properties it is possible to define an offset between Ardour's internal time and the timecode sent. Currently only the LTC generator honors this offset. + In Session > Properties, it is possible to + define an offset between Ardour's internal time and the timecode sent. + Currently only the LTC generator honors this offset.

-

-Both LTC and MTC are limited to max of 30fps. Using frame-rates larger than that will disable the generator. In both cases also only 24, 25, 29.97df and 30fps are well defined by specifications (such as SMPTE-12M, EU and the MIDI standard). + Both LTC and MTC are limited to 30 fps. Using frame rates larger + than that will disable the generator. In both cases also only 24, 25, + 29.97df (drop-frame) and 30 fps are well defined by specifications (such as + SMPTE-12M, EU and the MIDI standard).

MTC Generator

-

-There are no options. Ardour sends full MTC frames whenever the transport is relocated or changes state (start/stop). MTC quarter frames are sent when the transport is rolling and the transport speed is within 93% and 107%. + The MTC generator has no options. Ardour sends full MTC + frames whenever the transport is relocated or changes state (start/stop). + MTC quarter frames are sent when the transport is rolling and + the transport speed is within 93% and 107%.

- -

LTC Generator

-

-The volume of the LTC signal can be configured in in the Preferences > Transport dialog. By default it is set to -18dBFS which corresponds to 0dBu in an EBU calibrated system. + The level of the LTC generator output signal can be configured + in in the Preferences > Transport dialog. By + default it is set to -18 dBFS, which corresponds to 0dBu in an EBU + calibrated system.

-

-The LTC generator has an additional option to keep sending timecode even when the transport is stopped. This mode is intended to drive analog tape machines which unspool the tape if no LTC timecode is received. + The LTC generator has an additional option to keep sending timecode even + when the transport is stopped. This mode is intended to drive analog tape + machines which unspool the tape if no LTC timecode is received.

-

-LTC is send regardless of Ardour's transport-speed. It is accurately generated even for very slow speeds (<5%) and only limited by the soundcard's sampling-rate and filter (see Gibbs phenomenon) for high speeds. + LTC is send regardless of Ardour's transport speed. It is accurately + generated even for very slow speeds (<5%) and only limited by the + soundcard's sampling-rate and filter (see + Gibbs phenomenon) + for high speeds.

Ardour Slave Configuration

- -

-Switching the timecode-source can be done via the button just right of Ardour's main clock. By default it is set to Internal in which case Ardour will ignore any external timecode. The button allows to toggle between Internal and the configured timecode source which is chosen in Edit > Preferences > Transport. +

+ The timecode source can be switched with the button just right of + Ardour's main clock. By default it is set to Internal in which case Ardour will ignore any external + timecode. The button allows to toggle between Internal and the configured + timecode source which is chosen in Edit > Preferences + > Transport.

-

-When Ardour is chasing an external timecode source the following cases need to be distinguished: + When Ardour is chasing (synchronizing to) an external timecode + source, the following cases need to be distinguished:

  1. the timecode source shares the clock
    2. @@ -115,124 +138,156 @@ When Ardour is chasing an external timecode source the following cases need to b
  2. the timecode source uses the same FPS setting as Ardour
  3. the timecode source runs at different frames-per-second
-

-In both cases the first option is preferred: clock sync + same FPS setting. + In both cases the first option is preferred: clock sync + same FPS setting.

- -

Frames-per-second

-

-If the frames-per-second don't match, Ardour can either re-calculate (map) the frames or the configured FPS (session > properties) can be changed automatically while the slave is active. The behavior is configured with the checkbox in Edit > Preferences > Transport labeled Match session video frame rate to external timecode: When enabled the session video frame rate will be changed to match that of the selected external timecode source. When disabled the session video frame rate will not be changed to match that of the selected external timecode source. Instead the frame rate indication in the main clock will flash red and Ardour will convert between the external timecode standard and the session standard. + If the frames-per-second do not match, Ardour can either re-calculate + and map the frames, or the configured FPS (Session > + Properties) can be changed automatically while the slave is active. + The behavior is configured with the checkbox Edit + > Preferences > Transport > Match session video frame rate to + external timecode. +

+

+ When enabled, the session video frame rate will be changed to match that + of the selected external timecode source. When disabled, the session video + frame rate will not be changed to match that of the selected external + timecode source. Instead the frame rate indication in the main clock will + flash red, and Ardour will convert between the external timecode standard + and the session standard. +

+

+ 29.97 drop-frame timecode is another corner case. While the SMPTE 12M-1999 + specifies 29.97df as 30000/1001 frames per second, not all hardware devices + follow that standard. The checkbox + Lock to 29.9700 fps instead of 30000/1001 allows + to use a compatibility mode for those devices.
+ When enabled, the external timecode source is assumed to use 29.970000 fps + instead of 30000/1001. SMPTE 12M-1999 specifies 29.97df as 30000/1001. The + spec further mentions that drop-frame + timecode has an accumulated error of -86 ms over a 24-hour period. + Drop-frame timecode would compensate exactly for a NTSC color frame rate + of 30 * 0.9990 (ie 29.970000). That is not the actual rate. However, + some vendors use that rate — despite it being against the specs + — because the variant of using exactly 29.97 fps yields zero timecode + drift.

-

-An edge case can also occur with 29.97 drop-frame timecode. While the SMPTE 12M-1999 specifies 29.97df as 30000/1001 frames per second, not all hardware devices follow that standard. The checkbox Lock to 29.9700 fps instead of 30000/1001 allows to use a compatibility mode for those devices: -

- -

-When enabled the external timecode source is assumed to use 29.970000 fps instead of 30000/1001. SMPTE 12M-1999 specifies 29.97df as 30000/1001. The spec further mentions that drop-frame timecode has an accumulated error of -86ms over a 24-hour period. Drop-frame timecode would compensate exactly for a NTSC color frame rate of 30 * 0.9990 (ie 29.970000). That is not the actual rate. However, some vendors use that rate - despite it being against the specs - because the variant of using exactly 29.97 fps yields zero timecode drift. -

- - -

Clock Sync Lock

-

-As described in the On Clock and Time Section, timecode and clock are independent. If the external timecode-source is not sample-sync with the audio-hardware (and jack), ardour needs to vari-speed to adjust for the discrepancy. + As described in the + On Clock and Time + chapter, timecode and clock are independent. If the external timecode + source is not in sample-sync with the audio hardware (and JACK), Ardour + needs to run at varispeed to adjust for the discrepancy.

-

-The checkbox External timecode is sync locked allows to select the behavior according to your setup. When enabled indicates that the selected external timecode source shares sync (Black & Burst, Wordclock, etc) with the audio interface. + The checkbox External timecode is sync locked + allows to select the behavior according to your setup. When enabled, it + indicates that the selected external timecode source shares sync (Black + & Burst, Wordclock, etc) with the audio interface.

-

-In other words: if enabled, Ardour will only use perform initial synchronization and keep playing at speed 1.0 instead of vari-speed adjusting to compensate for drift. + In other words: if enabled, Ardour will only perform initial + synchronization and keep playing at speed 1.0 instead of vari-speed + adjusting to compensate for drift.

- -

-Note that vari-speed is unavailable when recording in Ardour and all tracking happens at speed 1.0. So if you want to record in sync with external timecode it must be sample-locked or it will drift over time. +

+ Note that vari-speed is unavailable when recording in Ardour, and all + tracking happens at speed 1.0. So if you want to record in sync with + external timecode it must be sample-locked or it will drift over time.

- - -

MClk - MIDI Clock

- +

MIDI Clock

-MIDI Clock is not a timecode format but tempo-based time. The absolute reference point is expressed as beats-per-minute and Bar, Beat and Tick. There is no concept of sample-locking for Midi clock signals. Ardour will vari-speed if necessary to chase the incoming signal. + MIDI Clock is not a timecode format but tempo-based time. The + absolute reference point is expressed as beats-per-minute and Bar, Beat + and Tick. There is no concept of sample-locking for MIDI clock signals. + Ardour will vari-speed if necessary to chase the incoming signal.

-

-Note that the MIDI Clock source must be connected to ardour:MIDI clock in port. + Note that the MIDI Clock source must be connected to the + ardour:MIDI clock in port.

- -

LTC - Linear Timecode

-

-The LTC slave decodes an incoming LTC signal on a jack-audio port. It will auto-detect the frame-rate and start locking to the signal once two consecutive LTC frames have been received. + The LTC slave decodes an incoming LTC signal on a JACK audio + port. It will auto-detect the frame rate and start locking to the signal + once two consecutive LTC frames have been received.

-

-The incoming timecode signal needs to arrive at the ardour:LTC-in port. Port-connections are restored for each session and the preference dialog offers an option to select it for all sessions. + The incoming timecode signal needs to arrive at the + ardour:LTC-in port. Port-connections are restored for each + session and the preference dialog offers an option to select it for all + sessions.

-

-Ardour's transport is aligned to LTC-frame start/end positions according to the SMPTE 12M-1999 spec which means that the first bit of an LTC-Frame is aligned to different Lines of a Video-Frame, depending on the TV standard used. Only for Film (24fps) does the LTC-Frame directly match the video Frame boundaries. + Ardour's transport is aligned to LTC-frame start/end positions according + to the SMPTE 12M-1999 specification, which means that the first bit of an + LTC-Frame is aligned to different Lines of a Video-Frame, depending on the + TV standard used. Only for Film (24fps) does the LTC-Frame directly match + the video Frame boundaries.

-

LTC frame alignment

+LTC frame alignment

Figure: LTC frame alignment for the 525/60 TV standard

-

-Ardour supports vari-speed and backwards playback but will only follow speed changes if the sync locked configuration option is disabled. + Ardour supports vari-speed and backwards playback but will only follow + speed changes if the sync locked option is + disabled.

-

-While Ardour is chasing LTC, the main transport clock will display the received Timecode as well as the delta between the incoming signal and Ardour's transport position. + While Ardour is chasing LTC, the main transport clock will display the + received Timecode as well as the delta between the incoming signal and + Ardour's transport position.

-

-A global offset between incoming timecode and Ardour's transport can be configured in Session > Properties. + A global offset between incoming timecode and Ardour's transport can be + configured in Session > Properties.

-

-The user-bits in the received LTC frame are ignored. + The user-bits in the received LTC frame are ignored.

- -

MTC - MIDI Timecode

-

-Ardour's MTC slave parses full timecode (sysex messages) as well as MTC quarter-frames arriving on the ardour:MTC in port. The transport will only start rolling once a complete sequence of 8 quarter frames has been received. + Ardour's MTC slave parses full timecode messages as well as + MTC quarter-frame messages arriving on the + ardour:MTC in port. The transport will only start rolling + once a complete sequence of 8 quarter frames has been received.

-

-Ardour supports vari-speed and backwards playback but will only follow MTC speed changes if the sync locked configuration option is disabled. + Ardour supports vari-speed and backwards playback but will only follow + MTC speed changes if the sync locked option + is disabled.

-

-When Ardour is chasing MTC, the main transport clock will display the received Timecode as well as the delta between the incoming signal and Ardour's transport position. + When Ardour is chasing MTC, the main transport clock will display the + received Timecode as well as the delta between the incoming signal and + Ardour's transport position.

- -

JACK Transport

-

-When slaved to jack, Ardour's transport will be identical to JACK-transport. As opposed to other slaves, Ardour can be used to control the JACK transport states (stopped/rolling). No port-connections need to be made for jack-transport to work. + When slaved to jack, Ardour's transport will be identical to + JACK-transport. As opposed to other slaves, Ardour can be used to control + the JACK transport states (stopped/rolling). No port connections need to + be made for jack-transport to work.

-

-JACK-transport does not support vari-speed, nor offsets. Ardour does not chase the timecode but is always in perfect sample-sync with it. + JACK-transport does not support vari-speed, nor offsets. Ardour does not + chase the timecode but is always in perfect sample-sync with it.

-

-JACK-transport also includes temp-based-time information ie. Bar:Beats:Ticks and beats-per-minute. However, only one JACK application can provide this information at a given time. The checkbox JACK Time Master in the Session > Properties dialog allows to configure Ardour to act as translator from timecode to BBT information. + JACK-transport also includes temp-based-time information in Bar:Beats:Ticks + and beats-per-minute. However, only one JACK application can provide this + information at a given time. The checkbox + Session > Properties > JACK Time Master + configures Ardour to act as translator from timecode to BBT information.

diff --git a/_manual/19_synchronization/04_overview-of-timecode-related-settings.html b/_manual/19_synchronization/04_overview-of-timecode-related-settings.html index b7ff58e..03c6945 100644 --- a/_manual/19_synchronization/04_overview-of-timecode-related-settings.html +++ b/_manual/19_synchronization/04_overview-of-timecode-related-settings.html @@ -4,49 +4,105 @@ title: Overview of all Timecode related settings menu_title: Overview of Timecode settings --- -

Accessing the Settings and Preferences

-

-Timecode related settings are accessed from the menu: + Timecode settings are accessed from the menu in three places:

- -

Timecode Settings

-

Thes settings are session specific:

- - +
+
Timecode frames-per-second
+
+ Configure timecode frames-per-second (23.976, 24, 24.975, 25, 29.97, + 29.97 drop, 30, 30 drop, 59.94, 60). Note that all fractional + framerates are actually fps*(1000.0/1001.0). +
+
Pull up/down
+
+ Video pull-up modes change the effective samplerate of Ardour to + allow for changing a film soundtrack from one frame rate to another. + See Telecine +
+
Slave Timecode offset
+
+ The specified offset is added to the received timecode (MTC or + LTC). +
+
Timecode Generator offset
+
+ Specify an offset which is added to the generated timecode (so far only LTC). +
+
JACK Time Master
+
+ Provide Bar|Beat|Tick and other information to JACK. +
+
+

These settings are session specific.

Transport Preferences

- - +
+
External timecode source
+
+ Select timecode source: JACK, LTC, MTC, MIDI Clock +
+
Match session video frame rate to external timecode
+
+ This option controls the value of the video frame rate while + chasing an external timecode source. When enabled, the + session video frame rate will be changed to match that of the selected + external timecode source. When disabled, the session video frame rate + will not be changed to match that of the selected external timecode + source. Instead the frame rate indication in the main clock will flash + red and Ardour will convert between the external timecode standard and + the session standard. +
+
External timecode is sync locked
+
+ Indicates that the selected external timecode source shares sync (Black + & Burst, Wordclock, etc) with the audio interface. +
+
Lock to 29.9700 fps instead of 30000/1001
+
+ The external timecode source is assumed to use 29.97 fps instead of + 30000/1001. SMPTE 12M-1999 specifies 29.97df as 30000/1001. The spec + further mentions that drop-frame timecode has an accumulated error of -86ms + over a 24-hour period. Drop-frame timecode would compensate exactly for a + NTSC color frame rate of 30 * 0.9990 (ie 29.970000). That is not the actual + rate. However, some vendors use that rate — despite it being against + the specs — because the variant of using exactly 29.97 fps has zero + timecode drift. +
+
LTC incoming port
+
+ Offers a session agnostic way to retain the LTC port connection. +
+
Enable LTC generator
+
Does just what it says.
+
Send LTC while stopped
+
+ Enable to continue to send LTC information even when the transport + (playhead) is not moving. This mode is intended to drive analog tape + machines which unspool the tape if no LTC timecode is received. +
+
LTC generator level
+
+ Specify the Peak Volume of the generated LTC signal in dbFS. A good value + is 0 dBu (which is -18 dbFS in an EBU calibrated system). +
+
+

These settings are common to all sessions.

MIDI Preferences

- +
+
Send MIDI Timecode
Enable MTC generator
+
Send MIDI Clock
Enable MIDI Clock generator
+
+

These settings are also common to all sessions.

-- 2.37.2