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 .
This office action is in response to applicant’s amendment dated 8/14/2024, claims 1-20 were amended, claims 21-22 were cancelled and no claims were newly introduced. Accordingly claims 1-20 are currently pending in the application.
Claim Rejections - 35 USC § 102
In the event the determination of the status of the application as subject to AIA 35 U.S.C. 102 and 103 (or as subject to pre-AIA 35 U.S.C. 102 and 103) is incorrect, any correction of the statutory basis (i.e., changing from AIA to pre-AIA ) for the rejection will not be considered a new ground of rejection if the prior art relied upon, and the rationale supporting the rejection, would be the same under either status.
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, 8, 19 is/are rejected under 35 U.S.C. 102(a)(1) as being anticipated by “Analysis and synthesis of room reverberation based on statistical time-frequency model” by Jean-Marc Jot et al. AES September 2007 103rd Convention Hereinafter Jean-Marc.
Regarding claim 1, Jean-Marc teaches An apparatus for assisting spatial rendering in at least one acoustic environment (“the Room module can be directly used as a stereo or binaural reverberator providing individual adjustment of each early reflection” in Section 5.5 second paragraph), the apparatus comprising at least one processor (“(bottom) Structure of a spatial sound processor adaptable to various reproduction formats over loudspeakers or headphones” in Fig. 14 on Page 43); and at least one memory storing instructions that, when executed with the at least one processor, cause the apparatus at least to (implied since the processor is running software in “as implemented in the IRCAM Spat software” in Fig. 14, Page 43): determine a source-listener distance (“the position of each source relative to the listener its distance r” in paragraph 4, page 24) ; determine an attenuation parameter value, the attenuation parameter associated with the source-listener distance (“Depending on the application these temporal parameters can be fixed according to the size of the virtual room or variable with the source-receiver distance r (for instance via a simplified image source model).” in first paragraph page 25); determine an average attenuation parameter value associated with at least one acoustic environment (“the room, represented by its volume V and its reverberation time Tr(j) (which can be expressed in terms of the absorption coefficient of the air and walls and the surface of the walls, according to classical formulas of acoustics such as the Sabine or Eyring formulas [13, 141)” in paragraph 4, page 24 and “Considering an impulsive acoustic excitation emitted at time t = 0 from any point in the room, we assume that at any time later than the room's mixing time, the acoustical energy is uniformly distributed across the room. The reverberation then dies out simultaneously for all receiver positions in the room according to the reverberation time Tr(j), which is assumed to be independent of position. This implies that a unique time4requency envelope characterizes the late reverberation decay across the room, irrespective of source and receiver positions or orientations” in Section 4.1 first paragraph, page 14, and “This physical interpretation of the stochastic model of reverberation decays offers a separation of room characteristics and transducer characteristics: the temporal decay rate characterizes of the room. while the initial spectrum characterizes the source receiver pair (irrespective of position)” in first paragraph of Page 16); determine a compensated attenuation parameter value based on the average attenuation parameter value and the attenuation parameter value (“ In this model, the complete description of a sound scene is given by the following control interface parameters: the position of each source relative to the listener: its distance r and orientation phi, its angular localization, which controls the directional panning of the direct sound); the radiation characteristics of each sound source, represented by its directivity S<t,(f) and its diffuse-field transfer function Sd(f), the room, represented by its volume V and its reverberation time Tr(j) (which can be expressed in terms of the absorption coefficient of the air and walls and the surface of the walls, according to classical formulas of acoustics such as the Sabine or Eyring formulas [13, 141)“ in Paragraph 4, Page 24); obtain an input signal (“As illustrated by the structure of the Room module shown on Fig. 14, the basic FDN (denoted reverb) can be extended to allow separate control of early reflections and later reverberation [S. 7]. On this example, the early module is a delay line producing several delayed copies of the mono input signal, which are used both to render the early reflections R1 and feed the subsequent stages of the reverberator” in Section 5.5 in Page 22); and generate a late reverberation audio signal part with applying a reverberator to the input signal (R3 in Fig. 14 is the late reverberation produced from Input signal “the early module is a delay line producing several delayed copies of the mono input signal,” in Section 5.5. page 22), the reverberator configured with the compensated attenuation parameter value to compensate within the reverberation level for the attenuation parameter value associated with the source-listener distance (“In this model, the complete description of a sound scene is given by the following control interface parameters: the position of each source relative to the listener: its distance r and orientation phi, its angular localization, which controls the directional panning of the direct sound); the radiation characteristics of each sound source, represented by its directivity S<t,(f) and its diffuse-field transfer function Sd(f), the room, represented by its volume V and its reverberation time Tr(j) (which can be expressed in terms of the absorption coefficient of the air and walls and the surface of the walls, according to classical formulas of acoustics such as the Sabine or Eyring formulas [13, 141)“ in Paragraph 4, Page 24).
Regarding claim 8, Jean-Marc teaches the apparatus of claim 1, Jean-Marc further teaches the apparatus further comprising wherein the instructions, when executed with the at least one processor, cause the apparatus to determine the attenuation parameter value based on the source-listener distance (“Depending on the application these temporal parameters can be fixed according to the size of the virtual room or variable with the source-receiver distance r (for instance via a simplified image source model).” in first paragraph page 25).
Regarding claim 19, claim is rejected for being the method comprising at least the same elements and performing at least the same functions performed by the apparatus of rejected claim 1 (see rejection of claim 1 above).
Claim(s) 13-17, 20 is/are rejected under 35 U.S.C. 102(a)(1) as being anticipated by Eronen et al. (WO 2021186102 A1) hereinafter Eronen.
Regarding claim 13, Eronen teaches an apparatus for assisting spatial rendering in at least one acoustic environment (system in Fig. 2), the apparatus comprising: at least one processor ; and at least one memory storing instructions that, when executed with the at least one processor, cause the apparatus at least to (“2007 and 2011 in Fig. 20): obtain acoustic environment geometry associated with the at least one acoustic environment (203 in Fig. 2); determine an average attenuation parameter value associated with the at least one acoustic environment (“renderer is configured to extract a direct path audio signal from the delay line 1803 and apply a filter To 1805 that contains such room simulation dependent effects such as: distance-based attenuation, air absorption, and source directivity. This filter can be a single filter or multiple cascaded modifications” in last paragraph of page 38); and generate a bitstream associated with the average attenuation parameter value (bitstream 214 in Fig. 2 and “in the MPEG-I scenario for virtual reality audio rendering the encoder device can select one or more recorded room impulse responses to be used for rendering an acoustic scene. These selected impulse responses are then sent in the audio bitstream to the Tenderer device” in second paragraph page 41), wherein the bitstream is to be employed in assisting a configuration of a reverberator within the spatial rendering (“The individual reflection filters obtained as the result of the individual reflection filter extraction process are therefore included in the audio bitstream 1920 to be communicated to the Tenderer 1921. The encoder includes the necessary individual reflection filters based on materials found in the encoder input format (EIF) scene description for the scene geometry” in paragraph 5, page 50).
Regarding claim 14, Eronen teaches the apparatus of claim 13, Eronen further teaches the apparatus further comprising wherein the instructions, when executed with the at least one processor, cause the apparatus to determine an average distance associated with at least one audio environment geometry (205 in Fig. 2), and wherein the instructions, when executed with the at least one processor, cause the apparatus to determine the average attenuation parameter value based on the average distance (“The simulated room reverberation generator 407 is configured to receive the obtained database 406, either directly from the generator 403 or from storage 405. Furthermore, the simulated room reverberation generator 407 is configured to receive the audio scene signals (for example the audio objects or MPEG- H 3D audio) and generate simulated room reverberation audio signals. In other words the simulated room reverberation generator 407 is configured to receive direct audio and output both direct audio and reverberation audio as the reverberation generator provides the modelled delay and attenuation (due to distance).” in last Paragraph, page 29 to first paragraph page 30).
Regarding claim 15, Eronen teaches the apparatus of claim 14, Eronen further teaches the apparatus further comprising wherein the instructions, when executed with the at least one processor cause the apparatus to apply one of: a closed form expression to calculate the average distance of two points in a geometric shape associated with the at least one audio environment geometry; or a sampling procedure to simulate possible source and listener positions in the at least one audio environment geometry (“These embodiments can be summarized as: receiving a spatial room impulse response (RIR) containing at least one clean individual reflection; performing spatial decomposition to determine the direction of arrival (DOA) for time samples in the spatial RIR; using the determined DOA and a sound pressure level of the spatial RIR to determine the position of at least one clean individual reflection” in last paragraph , page 24 to first paragraph page 25).
Regarding claim 16, Eronen teaches the apparatus of claim 14, Eronen further teaches the apparatus further comprising wherein the instructions, when executed with the at least one processor, cause the apparatus to select between a three dimensional (“The additional degrees of freedom in six degrees of freedom (6D0F) audio rendering enable the listener to move in the audio scene along the three cartesian dimensions x, y, and z” in paragraph 4, page 1) or two dimensional geometry (“The standard will be based on MPEG-FI 3D Audio, which supports three degrees of freedom (3DoF) based rendering of object, channel, and FIOA content. In 3DoF rendering, the listener is able to listen to the audio scene at a single location while rotating their head in three dimensions (yaw, pitch, roll)” in paragraph 3, page 1) when calculating the average distance between two points in the at least one audio environment geometry (“The audio scene 201 may furthermore comprise the audio signal information 205. The audio signal information 205 can comprise audio elements as objects, channels, HOA and metadata parameters such as source position, orientation, directivity, size etc.” in last paragraph, page 21).
Regarding claim 17, Eronen teaches the apparatus of claim 13, Eronen further teaches the apparatus further comprising wherein the bitstream comprises the average attenuation parameter value (“generate simulated room reverberation audio signals. In other words the simulated room reverberation generator 407 is configured to receive direct audio and output both direct audio and reverberation audio as the reverberation generator provides the modelled delay and attenuation (due to distance)” in first paragraph, page 30).
Regarding claim 20, claim is rejected for being the method comprising at least the same elements and performing at least the same functions performed by the apparatus of rejected claim 13 (see rejection of claim 13 above).
Claim Rejections - 35 USC § 103
In the event the determination of the status of the application as subject to AIA 35 U.S.C. 102 and 103 (or as subject to pre-AIA 35 U.S.C. 102 and 103) is incorrect, any correction of the statutory basis (i.e., changing from AIA to pre-AIA ) for the rejection will not be considered a new ground of rejection if the prior art relied upon, and the rationale supporting the rejection, would be the same under either status.
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.
The factual inquiries for establishing a background for determining obviousness under 35 U.S.C. 103 are summarized as follows:
1. Determining the scope and contents of the prior art.
2. Ascertaining the differences between the prior art and the claims at issue.
3. Resolving the level of ordinary skill in the pertinent art.
4. Considering objective evidence present in the application indicating obviousness or nonobviousness.
Claim(s) 2-3, 5-7 is/are rejected under 35 U.S.C. 103 as being unpatentable over “Analysis and synthesis of room reverberation based on statistical time-frequency model” by Jean-Marc Jot et al. AES September 1007 103rd Convention, Hereinafter Jean-Marc.
Regarding claim 2, Jean-Marc teaches the apparatus of claim 1, Jean-Marc further teaches the apparatus further comprising wherein the instructions, when executed with the at least one processor, cause the apparatus to obtain acoustic environment geometry associated with the at least one acoustic environment (“In this statistical approach, the roam is fully parametrized by its volume and its reverberation time Tr(f).” in Paragraph 1, page 25), wherein the instructions, when executed with the at least one processor, cause the apparatus to determine the attenuation parameter value based on the at least one audio environment geometry ( the model uses control parameters like source listener position and room geometry in “the room, represented by its volume V and its reverberation time Tr(j) (which can be expressed in terms of the absorption coefficient of the air and walls and the surface of the walls, according to classical formulas of acoustics such as the Sabine or Eyring formulas [13, 141).” in paragraph 4 page 24),
Jean-Marc does not specifically disclose the attenuation parameter being an average attenuation parameter however,
An ordinary skilled in the art would be motivated to use an average value for the benefit of reducing processing complexity load on the apparatus, it would have been obvious to a person of ordinary skill in the art prior to the effective filing date of the claimed invention to modify Jean-Marc to use an average value.
Regarding claim 3, Jean-Marc as modified teaches the apparatus of claim 2, Jean-Marc further teaches the apparatus further comprising wherein the instructions, when executed with the at least one processor, cause the apparatus to: determine an distance between two points in the at least one audio environment geometry; and determine the attenuation parameter value based on the at least one audio environment geometry based on the average distance (“Depending on the application these temporal parameters can be fixed according to the size of the virtual room or variable with the source-receiver distance r (for instance via a simplified image source model).” in first paragraph page 25),
Jean-Marc does not specifically disclose the distance or the attenuation parameter being an average value however,
An ordinary skilled in the art would be motivated to use an average value for both the distance and the attenuation parameter for the benefit of reducing processing complexity load on the apparatus, it would have been obvious to a person of ordinary skill in the art prior to the effective filing date of the claimed invention to modify Jean-Marc to use an average value.
Regarding claim 5, Jean-Marc teaches the apparatus of claim 1, Jean-Marc further teaches the apparatus further comprising wherein the instructions, when executed with the at least one processor, cause the apparatus to receive the attenuation parameter value (“Depending on the application these temporal parameters can be fixed according to the size of the virtual room or variable with the source-receiver distance r (for instance via a simplified image source model).” in first paragraph page 25),
Jean-Marc does not specifically disclose the apparatus further comprising the attenuation parameter being an average attenuation parameter however,
An ordinary skilled in the art would be motivated to use an average value for the benefit of reducing processing complexity load on the apparatus, it would have been obvious to a person of ordinary skill in the art prior to the effective filing date of the claimed invention to modify Jean-Marc to use an average value.
Regarding claim 6, Jean-Marc teaches the apparatus of claim 1, Jean-Marc further teaches the apparatus further comprising wherein the instructions, when executed with the at least one processor, cause the apparatus to: receive the distance of the at least one acoustic environment; and determine the attenuation parameter value associated with the at least one acoustic environment based on the received distance (“ In this model, the complete description of a sound scene is given by the following control interface parameters: the position of each source relative to the listener: its distance r and orientation phi, its angular localization, which controls the directional panning of the direct sound); the radiation characteristics of each sound source, represented by its directivity S<t,(f) in paragraph 4, page 24),
Jean-Marc does not specifically disclose the distance and the attenuation parameter being an average value however,
An ordinary skilled in the art would be motivated to use an average value for both the distance and the attenuation parameter for the benefit of reducing processing complexity load on the apparatus, it would have been obvious to a person of ordinary skill in the art prior to the effective filing date of the claimed invention to modify Jean-Marc to use an average value.
Regarding claim 7, Jean-Marc teaches the apparatus of claim 1, Jean-Marc further teaches the apparatus further comprising wherein the instructions, when executed with the at least one processor, cause the apparatus to: receive attenuation parameter values (“Depending on the application these temporal parameters can be fixed according to the size of the virtual room or variable with the source-receiver distance r within the at least one acoustic environment (for instance via a simplified image source model).” in first paragraph page 25); Jean -Marc does not specifically disclose the apparatus further comprising associated with the at least one acoustic environment based on an arithmetic or geometric mean of the received attenuation parameter values however,
Since the advantage of determine the average attenuation parameter value based an arithmetic/geometric mean of the received attenuation parameter values is to reduce processing complexity load on the apparatus, it would have been obvious to a person of ordinary skills in the art prior to the effective filing date of the claimed invention to modify Jean-Marc.
Claim(s) 9-12 is/are rejected under 35 U.S.C. 103 as being unpatentable over “Analysis and synthesis of room reverberation based on statistical time-frequency model” by Jean-Marc Jot et al. AES September 1007 103rd Convention, Hereinafter Jean-Marc in view of Lee et al. (US 20230224661 A1) hereinafter Lee.
Regarding claim 9, Jean-Marc teaches the apparatus of claim 8, Jean-Marc does not specifically disclose the apparatus further comprising wherein the instructions, when executed with the at least one processor, cause the apparatus to: obtain at least one of the source or the listener position from metadata associated with the at least one audio environment; and select between three dimension or two dimension at least one audio environment geometry when calculating the source-listener distance however,
Since it is known in the art as evidenced by Lee for an apparatus to further comprise wherein the instructions, when executed with the at least one processor, cause the apparatus to: obtain at least one of the source or the listener position from metadata associated with the at least one audio environment; and select between three dimension or two dimension at least one audio environment geometry when calculating the source-listener distance in (“The determining of whether the obstacle is present may include determining, based on a location of the listener, and a location of a sound source and acoustic geometry information included in the metadata” in ¶[0009]),
An ordinary skilled in the art would be motivated to modify the invention of Jean-Marc with the teachings of Lee for the benefit of saving processing resources of the device, therefore it would have been obvious to a person of ordinary skill in the art prior to the effective filing date of the claimed invention to modify Jean-Marc with Lee.
Regarding claim 10, Jean-Marc teaches the apparatus of claim 1, Jean-Marc does not specifically disclose the apparatus further comprising wherein the instructions, when executed with the at least one processor, cause the apparatus to determine a source-listener distance dependent gain for the reverberator for compensating the average attenuation parameter value however,
Since it is known int the art as evidenced by Lee for an apparatus to further comprise wherein the instructions, when executed with the at least one processor, cause the apparatus to determine a source-listener distance dependent gain for the reverberator for compensating the average attenuation parameter value (“the rendering apparatus 101 may modify the direct sound control information so as to reduce a gain of the direct sound. For example, when it is determined that the obstacle is present in the direct sound transmission path, the rendering apparatus 101 may modify the direct sound control information using a low pass filter” in ¶[0057]),
An ordinary skilled in the art would be motivated to modify the invention of Jean-Marc with the teachings of Lee for the benefit of saving processing resources of the device, therefore it would have been obvious to a person of ordinary skill in the art prior to the effective filing date of the claimed invention to modify Jean-Marc with Lee.
Regarding claim 11, Jean-Marc teaches the apparatus of claim 1, Jean-Marc does not specifically disclose the apparatus further comprising wherein the instructions, when executed with the at least one processor, cause the apparatus to: adjust a ratio parameter based on the average attenuation parameter value in the at least one acoustic environment; and apply the reverberator to a part of the input signal, the part of the input signal based on a filter applied to the input signal configured with the ratio parameter or based on the ratio parameter however,
Since it is known int the art as evidenced by Lee for an apparatus to further comprise wherein the instructions, when executed with the at least one processor, cause the apparatus to: adjust a ratio parameter based on the average attenuation parameter value in the at least one acoustic environment; and apply the reverberator to a part of the input signal, the part of the input signal based on a filter applied to the input signal configured with the ratio parameter or based on the ratio parameter in (“The rendering apparatus 101 may reduce transformation of the BRIR by lowering the transformation intensity of the binaural filter as the determined ratio is lower” in ¶[0058]),
An ordinary skilled in the art would be motivated to modify the invention of Jean-Marc with the teachings of Lee for the benefit of saving processing resources of the device, therefore it would have been obvious to a person of ordinary skill in the art prior to the effective filing date of the claimed invention to modify Jean-Marc with Lee.
Regarding claim 12, Jean-Marc teaches the apparatus of claim 1, Jean-Marc does not specifically disclose the apparatus further comprising wherein the instructions, when executed with the at least one processor, cause the apparatus to obtain a bitstream, wherein the bitstream comprises information of the amount of average distance gain attenuation, wherein the instructions, when executed with the at least one processor, cause the apparatus to determine the average attenuation parameter value based on the information of the amount of average distance gain attenuation however,
Since it is known in the art as evidenced by Lee for an apparatus to further comprise wherein the instructions, when executed with the at least one processor, cause the apparatus to obtain a bitstream, wherein the bitstream comprises information of the amount of average distance gain attenuation (“A method and apparatus for rendering an object-based audio signal” in ¶[ABSTRACT] and “The metadata 103 may include a gain of the audio signal 102, a distance between a sound source and the audio signal 102, a location of a listener, a location of the sound source” in ¶[0041]), wherein the instructions, when executed with the at least one processor, cause the apparatus to determine the average attenuation parameter value based on the information of the amount of average distance gain attenuation (“the rendering apparatus 101 may modify the direct sound control information so as to reduce a gain of the direct sound. For example, when it is determined that the obstacle is present in the direct sound transmission path, the rendering apparatus 101 may modify the direct sound control information using a low pass filter” in ¶[0057]),
An ordinary skilled in the art would be motivated to modify the invention of Jean-Marc with the teachings of Lee for the benefit of saving processing resources of the device, therefore it would have been obvious to a person of ordinary skill in the art prior to the effective filing date of the claimed invention to modify Jean-Marc with Lee.
Claim(s) 18 is/are rejected under 35 U.S.C. 103 as being unpatentable over Eronen et al (WO 2021186102 A1) hereinafter Eronen in view of Laaksonen et al. (US 20210076153 A1) hereinafter Laaksonen.
Regarding claim 18, Eronen teaches the apparatus of claim 13, Eronen further teaches the apparatus further comprising wherein the instructions, when executed with the at least one processor, cause the apparatus to: determine parameters for a diffuse-to-direct ratio control filter for the reverberator within the spatial rendering (“The reverberation time can be indicated for a set of frequency bands; for example, octave bands. In addition, other parameters such as diffuse-to-direct ratio can be provided and used for finding the match” in paragraph 1, page 41);
Eronen does not specifically disclose the apparatus further comprising adjust the parameters for the diffuse-to-direct ratio control filter based on the average attenuation parameter value , wherein the bitstream comprises the adjusted parameters for the diffuse-to-direct ratio control filter however,
Since it is known in the art as evidenced by Laaksonen for an apparatus to further comprise adjust the parameters for the diffuse-to-direct ratio control filter based on the average attenuation parameter value, wherein the bitstream comprises the adjusted parameters for the diffuse-to-direct ratio control filter (“binaural rendering, positioning in width and/or height dimension is obtained by selecting suitable head related transfer function (HRTF) filters (one for left ear, one for right ear) for each of the spectrally distinct sound objects depending on its position. A pair of HRTF filters model the path from a point in space to the listener's ears. The HRFT coefficient pairs are stored for all the possible directions of arrival for a sound. Similarly, distance dimension of a spectrally distinct sound object 12 is controlled by modelling distance attenuation with gain control and optionally direct to reverberant (indirect) ratio” in ¶[0096]),
An ordinary skilled in the art would be motivated to modify the invention of Eronen with the teachings of Laaksonen for the benefit of improving the accuracy of the apparatus, therefore it would have been obvious to a person of ordinary skilled in the art prior to the effective filing date of the claimed invention to modify Eronen with Laaksonen.
Claim(s) 4 is/are rejected under 35 U.S.C. 103 as being unpatentable over Jean-Marc et al “Analysis and synthesis of room reverberation based on statistical time-frequency model” by Jean-Marc Jot et al. AES September 1007 103rd Convention, Hereinafter Jean-Marc in view of Eronen et al. (WO 2021186102 A1) hereinafter Eronen.
Regarding claim 4, Jean-Marc as modified teaches the apparatus of claim 3, Jean-Marc does not specifically disclose the apparatus further comprising wherein the instructions, when executed with the at least one processor, cause the apparatus to apply one of: a closed form expression to calculate the average distance of two points in a geometric shape associated with the at least one audio environment geometry; or a sampling procedure to simulate possible source and listener positions in the at least one audio environment geometry however,
Since it is known in the art as evidenced by Eronen for an apparatus to further comprise wherein the instructions, when executed with the at least one processor, cause the apparatus to apply one of: a closed form expression to calculate the average distance of two points in a geometric shape associated with the at least one audio environment geometry; or a sampling procedure to simulate possible source and listener positions in the at least one audio environment geometry (“These embodiments can be summarized as: receiving a spatial room impulse response (RIR) containing at least one clean individual reflection; performing spatial decomposition to determine the direction of arrival (DOA) for time samples in the spatial RIR; using the determined DOA and a sound pressure level of the spatial RIR to determine the position of at least one clean individual reflection” in last paragraph , page 24 to first paragraph page 25),
An ordinary skilled in the art would be motivated to modify the invention of Jean-Marc with the teachings of Eronen for the benefit of saving processing resources of the apparatus, therefore it would have been obvious to a person of ordinary skill in the art prior to the effective filing date of the claimed invention to modify Jean-Marc with Eronen.
Conclusion
Any inquiry concerning this communication or earlier communications from the examiner should be directed to AMMAR T HAMID whose telephone number is (571)272-1953. The examiner can normally be reached M-F 9-5, Eastern time.
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AMMAR T. HAMID
Primary Examiner
Art Unit 2695
/AMMAR T HAMID/ Primary Examiner, Art Unit 2695
Prosecution Insights