DETAILED ACTION
Notice of Pre-AIA or AIA Status
The present application, filed on or after March 16, 2013, is being examined under the first inventor to file provisions of the AIA .
Claim Rejections - 35 USC § 102
The following is a quotation of the appropriate paragraphs of 35 U.S.C. 102 that form the basis for the rejections under this section made in this Office action:
A person shall be entitled to a patent unless –
(a)(1) the claimed invention was patented, described in a printed publication, or in public use, on sale, or otherwise available to the public before the effective filing date of the claimed invention.
Claim(s) 1-4 is/are rejected under 35 U.S.C. 102a1 as being anticipated by Miwa et al (US 20160350064 A1).
As per claim 1, Miwa discloses a computer-implemented method for dynamically clearing a signal processor's audio buffer in a music production environment, the method comprising:
receiving, by a processing unit, an incoming signal (portion of mixer in fig. 1a that receives the audio signal, para 46), wherein the incoming signal comprises at least one of a Musical Instrument Digital Interface (MIDI) signal or an audio signal (the audio signal para 46);
analyzing, by the processing unit, properties of the incoming signal to determine a timing for clearing the audio buffer (portion of the mixer in fig. 1a that performs per para 45: The signal processing in the imparting section 20 is also controlled based on the current value (current data) of various processing parameters. The various processing parameters are also included in the current data, where each parameter, in the context of the digital processing shown in fig, 1a, is associated with a corresponding clock cycle in order to synchronize the cited dsp based functions, where the property of the signal would be its particular amplitude at a particular clock cycle, which is analyzed relative to the other signaling, and modified at each functional step via a subsequent clocking signal, additionally, any user inputs entered in conjunction with an audio signal being played are additional parameters that are analyzed via timing relative to the incoming audio stream as they are used by the same digital audio production dsp; additionally the audio input itself is digital and each sample is times and analyzed relative to all other parameters as per the normal operation of a DSP);
resetting, by the processing unit, the audio buffer of the signal processor based on the determined timing for clearing ( para. 112: the digital mixer 1 detects that the second plug-in E2 in the target slot is not in use in any of all the scenes and the current data, then the digital mixer 1 stops the signal processing of the second plug-in E2 in the target slot and releases the DSP resource assigned to the second plug-in E2. ); and
automatically adjusting, by the processing unit, a volume fade associated with the resetting of the time-based audio effect process to eliminate audio artifacts, wherein the audio buffer is cleared dynamically in response to the incoming signal to allow for the retriggering of the time-based audio effect without sustaining undesired effects from previous signals (the crossfading per para 111 occurs based on bypassing/releasing E2 to prevent a drastic change (noise))
As per claim 2, the method of claim 1, wherein the incoming signal is a Musical Instrument Digital Interface (MIDI) signal, and the properties of the incoming signal include note-on and note-off messages used to determine the timing for clearing the audio buffer. (claim 2 is not mapped as parent claim 1 is drawn to the alternative of the audio).
As per claim 3, the method of claim 1, wherein the incoming signal is an audio signal, and the properties of the incoming signal include the volume of the audio signal used to determine the timing for clearing the audio buffer (all the timing required to perform the functions cited in the claim 1 rejection are read in a digital system via a DSP and as such each signal for each function required the measurable property of volume/amplitude in order to be read and process by the dsp based system of fig. 1a).
As per claim 4, the method of claim 1, further comprising triggering the resetting of the audio buffer based on a predefined threshold of the incoming signal's properties, wherein the threshold is adjustable by a user (para. 106: an instruction from the user to delete Scene 1 and Scene 2, which releases/resets the buffer/resource, based on the threshold of the dsp detecting a valid input form the user).
Claim Rejections - 35 USC § 103
The following is a quotation of 35 U.S.C. 103 which forms the basis for all obviousness rejections set forth in this Office action:
A patent for a claimed invention may not be obtained, notwithstanding that the claimed invention is not identically disclosed as set forth in section 102, if the differences between the claimed invention and the prior art are such that the claimed invention as a whole would have been obvious before the effective filing date of the claimed invention to a person having ordinary skill in the art to which the claimed invention pertains. Patentability shall not be negated by the manner in which the invention was made.
Claim 5-10 is/are rejected under 35 U.S.C. 103 as being unpatentable over Miwa et al (US 20160350064 A1) as applied to claim 1 above, and further in view of Stall et al (US 20180330515 A1).
As per claim 5, Miwa discloses the method of claim 1, but does not specify further comprising hosting the signal processor in a dedicated low-priority CPU thread to reduce processing load and prevent audio artifacts associated with CPU spikes.
Stall discloses a media system using digital signal processing and teaches to host the processor/renderer in a dedicated thread, for the advantage of being able to share the information via a shared interface. It would have been obvious to one skilled at the time of filing to implement the claimed dedicated hosting thread for the purpose of allowing improved information sharing. Implementation in a dedicated thread, by its nature, reduced processing load, and as implemented in the system of Miwa, per para 103 of Miwa is implemented via a high priority DSP/process because it is used first, and also in a lower priority second DSP because it is used after DSP 1 is full.
As per claim 6, the method of claim 1, further comprising alternating between multiple instances of the signal processor to allow for initialization time of the signal processor (per para 92 of Miwa, the multiple instances in the DSP1 and DSP2-4, are used to contribute necessary amounts of resource to initialization of a plugin, where each function, including the dedicated resources from the multiple DSP instances must allow for respective initialization/processing time in order to be performed by the processor ) thereby ensuring timely and accurate retriggering of the time-based audio effect process (the virtual slots described in para 92, in the context of a digital processing system ensure timely and accurate retriggering/mounting of the effect process/plugin.
As per claim 7, the method of claim 1, wherein the step of automatically adjusting a volume fade includes applying a fade-in and fade-out to the time-based audio effect process to smoothly transition the effect on and off, thereby preventing clicks or pops (crossfading per the claim 1 rejection).
As per claim 8, the method of claim 1, further comprising providing a user interface that allows for manual triggering of the resetting of the audio buffer, in addition to automatic retriggering based on the properties of the incoming signal (required to perform the user operations per para 114 relative to the audio being processed, in order to add or replace plugins as described).
As per claim 9, the method of claim 1, further comprising employing a sidechain input to trigger the resetting of the audio buffer based on the audio properties of a different source signal, allowing for creative control over which signals influence the retriggering of the time-based audio effect (the resetting processing as described in the claim 1 rejection, per the sidechain input that detects the function in para. 7: The signal processing apparatus, when a user selects a desired preset and recalls the preset) (the plugin selections by the user are based on the input audio channels, including a different source signal, upon which the selected plugins are to be used).
As per claim 10, the method of claim 1, wherein the time-based audio effect process includes at least one of reverb, delay, or echo (para 42, reverb, delay), and
the method facilitates dynamic management of the effect's duration and intensity in response to the incoming signal (the duration and intensity of a given effect is dynamically managed via being processed per any of the plugin functions in the claim 1 rejection as made by the user).
Claim 11,12,14-19 is/are rejected under 35 U.S.C. 103 as being unpatentable over Miwa et al (US 20160350064 A1) as applied to claim 1 above, and further in view of Sexton et al (US 20150206521 A1)
As per claim 11, Miwa discloses the method of claim 1, but does not specify further comprising utilizing a Hold feature that allows for the time-based effect to sustain for a predetermined duration after retriggering, before the audio buffer is cleared, providing users with control over the sustain duration of the effect.
Sexton discloses a digital audio effect using a button to create sustain (para. 88). It would have been obvious to one skilled in the art at the time of filing to implement a sustain function with a button (hold function) for the purpose of improved musical features. The sustain is performed by for a predetermined duration when the user presses and holds the button after retriggering.
As per claim 12, Miwa discloses the method of claim 1, but does not specify wherein the step of analyzing properties of the incoming signal further includes employing a hysteresis mechanism to determine the timing for clearing the audio buffer, thereby allowing for the adjustment of sensitivity to changes in the incoming signal's properties.
The examiner takes official notice it is well known in the art to implement hysteresis on all digital processes for the purpose of reducing the effect of transient noise. It would have been obvious to one skilled in the art at the time of filing.
As per claim 14, the method of claim 1, implemented as a digital audio workstation (DAW) plugin, wherein the plugin is capable of hosting third-party signal processor plugins for applying and dynamically managing time-based audio effects based on the properties of incoming MIDI or audio signals (the cited plugins include all plugins including first and third party that are dynamically managed via the functions cited in the claim 1 rejection).
As per claim 15, the method of claim 14, wherein the DAW plugin includes a user interface providing real-time visual feedback of the time-based audio effect process, including visual indicators of the retriggering events and the current state of the audio buffer (per manipulations of the plugins via the gui of para 67 and/or fig. 4).
As per claim 16, Miwa discloses the method of claim 14, but does not specify wherein the DAW plugin is compatible with multiple plugin formats, including but not limited to VST3, Audio Units (AU), AAX, and RTAS, facilitating its use across various music production software platforms.
The examiner takes official notice it is well known in the art to use well known audio/file formats including vst3,AU,AAX,RTAS for the purpose of compatibility with well known formats.
As per claim 17, Miwa discloses the method of claim 1, further comprising manually initiating the retriggering of the time-based audio effect process by a user through the actuation of a TRIGGER button provided in a user interface 210, (the effect taught by Sexton per the claim 11 rejection is via a trigger button, and ).
However, Miwa does not specify wherein the manual retriggering can be recorded as automation data within the DAW to replicate the retriggering events during playback.
The examiner takes official notice it is well known in the art to implement automation (recording automation data to trigger events) in audio workstations for the purpose storing the user edits to the signal processing.
As per claim 18, Miwa discloses the method of claim 17, wherein the automation data corresponding to the manual retriggering is programmable by the user in the digital audio workstation (DAW) plugin, allowing for the definition of specific timings for the retriggering of the time-based audio effect process independent of the incoming audio or MIDI signal (the user based functions as applied to the functions of the claim 1 rejection, in view of the automation taught per the claim 17 rejection require defined specific timings relative to each other and the DSPs ((clocking signals)) in order to be digitally implemented).
As per claim 19, the method of claim 17, further comprising the use of the recorded automation data to control the retriggering mechanism in synchronization with either the incoming audio signal or the MIDI signal, or as a standalone retriggering method (per the automation taught per claim 17).
Allowable Subject Matter
Claims 13 objected to as being dependent upon a rejected base claim, but would be allowable over the prior art of record if rewritten in independent form including all of the limitations of the base claim and any intervening claims.
Any inquiry concerning this communication or earlier communications from the examiner should be directed to ALEXANDER KRZYSTAN whose telephone number is 571-272-7498, and whose email address is alexander.krzystan@uspto.gov
The examiner can usually be reached on m-f 7:30-4:00 est.
If attempts to reach the examiner by telephone or email are unsuccessful, the examiner’s supervisor, Fan Tsang can be reached on (571) 272-7547.
The fax phone numbers for the organization where this application or proceeding is assigned are 571-273-8300 for regular communications and 571-273-8300 for After Final communications.
/ALEXANDER KRZYSTAN/Primary Examiner, Art Unit 2653
Examiner Alexander Krzystan
November 24, 2025