The State – Sort of – of HTML5 Audio Scott Schiller discusses the high level of hype around HTML5 and CSS3. The two specs render ”many years of feature hacks redundant by replacing them with native features,” he writes in an insightful blog. Blogging, he says: CSS3’s border-radius, box-shadow, text-shadow and gradients, and HTML5’s <canvas>, Read the rest...

Icons packs are among the most desirable freebies around. There are several out there, going from a wide array of topics from user interfaces to personal finance. But sometimes you can find some rather unusual but clever additions to the icons universe. This Vector Audio DJ Pack is a nice example, brought to you exclusively …

Freebie: 264 Vector Audio DJ Pack Icons Read More »

In recent years, all-neural end-to-end approaches have obtained state-of-the-art results on several challenging automatic speech recognition (ASR) tasks. However, most existing works focus on building ASR models where train and test data are drawn from the same domain. This results in poor generalization characteristics on mismatched-domains: e.g., end-to-end models trained on short segments perform poorly when evaluated on longer utterances. In this work, we analyze the generalization properties of streaming and non-streaming recurrent neural network transducer (RNN-T) based end-to-end models in order to identify model components that negatively affect generalization performance. We propose two solutions: combining multiple regularization techniques during training, and using dynamic overlapping inference. On a long-form YouTube test set, when the non-streaming RNN-T model is trained with shorter segments of data, the proposed combination improves word error rate (WER) from 22.3% to 14.8%; when the streaming RNN-T model trained on short Search queries, the proposed techniques improve WER on the YouTube set from 67.0% to 25.3%. Finally, when trained on Librispeech, we find that dynamic overlapping inference improves WER on YouTube from 99.8% to 33.0%.

Methods, systems, and computer program products are provided for email administration for rendering email on a digital audio player. Embodiments include retrieving an email message; extracting text from the email message; creating a media file; and storing the extracted text of the email message as metadata associated with the media file. Embodiments may also include storing the media file on a digital audio player and displaying the metadata describing the media file, the metadata containing the extracted text of the email message.

Low bit rate audio coding such as BWE algorithm often encounters conflict goal of achieving high time resolution and high frequency resolution at the same time. In order to achieve best possible quality, input signal can be first classified into fast signal and slow signal. This invention focuses on classifying signal into fast signal and slow signal, based on at least one of the following parameters or a combination of the following parameters: spectral sharpness, temporal sharpness, pitch correlation (pitch gain), and/or spectral envelope variation. This classification information can help to choose different BWE algorithms, different coding algorithms, and different postprocessing algorithms respectively for fast signal and slow signal.

An apparatus for processing an audio signal and method thereof are disclosed. The present invention includes receiving a downmix signal and side information; extracting control restriction information from the side information; receiving control information for controlling gain or panning at least one object signal; generating at least one of first multi-channel information and first downmix processing information based on the control information and object information, without using the control restriction information; and, generating an output signal by applying the at least one of the first multichannel information and the first downmix processing information to the downmix signal, wherein the control restriction information relates to a parameter indicating limiting degree of the control information.

A method comprising: sampling received audio at a first rate to produce a first audio signal; transforming the first audio signal into a sparse domain to produce a sparse audio signal; re-sampling of the sparse audio signal to produce a re-sampled sparse audio signal; and providing the re-sampled sparse audio signal, wherein bandwidth required for accurate audio reproduction is removed but bandwidth required for spatial audio encoding is retained AND/OR a method comprising: receiving a first sparse audio signal for a first channel; receiving a second sparse audio signal for a second channel; and processing the first sparse audio signal and the second sparse audio signal to produce one or more inter-channel spatial audio parameters.

According to one embodiment, an audio controlling apparatus includes a first receiver configured to receive audio signal, a second receiver configured to receive environmental sound, a temporary gain calculator configured to calculate temporary gain based on environmental sound received by second receiver, a sound type determination module configured to determine sound type of main component of audio signal received by first receiver, and a gain controller configured to stabilize temporary gain that is calculated by temporary gain calculator and set gain, when it is determined that sound type of main component of audio signal received by first receiver is music.

A method (700, 800) and apparatus (100, 200) processes audio frames to transition between different codecs. The method can include producing (720), using a first coding method, a first frame of coded output audio samples by coding a first audio frame in a sequence of frames. The method can include forming (730) an overlap-add portion of the first frame using the first coding method. The method can include generating (740) a combination first frame of coded audio samples based on combining the first frame of coded output audio samples with the overlap-add portion of the first frame. The method can include initializing (760) a state of a second coding method based on the combination first frame of coded audio samples. The method can include constructing (770) an output signal based on the initialized state of the second coding method.

An encoder for providing an audio stream on the basis of a transform-domain representation of an input audio signal includes a quantization error calculator configured to determine a multi-band quantization error over a plurality of frequency bands of the input audio signal for which separate band gain information is available. The encoder also includes an audio stream provider for providing the audio stream such that the audio stream includes information describing an audio content of the frequency bands and information describing the multi-band quantization error. A decoder for providing a decoded representation of an audio signal on the basis of an encoded audio stream representing spectral components of frequency bands of the audio signal includes a noise filler for introducing noise into spectral components of a plurality of frequency bands to which separate frequency band gain information is associated on the basis of a common multi-band noise intensity value.

An audio encoder for encoding an audio signal has a first coding branch, the first coding branch comprising a first converter for converting a signal from a time domain into a frequency domain. Furthermore, the audio encoder has a second coding branch comprising a second time/frequency converter. Additionally, a signal analyzer for analyzing the audio signal is provided. The signal analyzer, on the hand, determines whether an audio portion is effective in the encoder output signal as a first encoded signal from the first encoding branch or as a second encoded signal from a second encoding branch. On the other hand, the signal analyzer determines a time/frequency resolution to be applied by the converters when generating the encoded signals. An output interface includes, in addition to the first encoded signal and the second encoded signal, a resolution information identifying the resolution used by the first time/frequency converter and used by the second time/frequency converter.

An audio signal decoder has a time warp contour calculator, a time warp contour data rescaler and a warp decoder. The time warp contour calculator is configured to generate time warp contour data repeatedly restarting from a predetermined time warp contour start value, based on time warp contour evolution information describing a temporal evolution of the time warp contour. The time warp contour data rescaler is configured to rescale at least a portion of the time warp contour data such that a discontinuity at a restart is avoided, reduced or eliminated in a rescaled version of the time warp contour. The warp decoder is configured to provide the decoded audio signal representation, based on an encoded audio signal representation and using the rescaled version of the time warp contour.

An apparatus for encoding an audio signal having a stream of audio samples has: a windower for applying a prediction coding analysis window to the stream of audio samples to obtain windowed data for a prediction analysis and for applying a transform coding analysis window to the stream of audio samples to obtain windowed data for a transform analysis, wherein the transform coding analysis window is associated with audio samples within a current frame of audio samples and with audio samples of a predefined portion of a future frame of audio samples being a transform-coding look-ahead portion, wherein the prediction coding analysis window is associated with at least the portion of the audio samples of the current frame and with audio samples of a predefined portion of the future frame being a prediction coding look-ahead portion, wherein the transform coding look-ahead portion and the prediction coding look-ahead portion are identically to each other or are different from each other by less than 20%; and an encoding processor for generating prediction coded data or for generating transform coded data.

A time warp contour calculator for use in an audio signal decoder receives an encoded warp ratio information, derives a sequence of warp ratio values from the encoded warp ratio information, and obtains warp contour node values starting from a time warp contour start value. Ratios between the time warp contour node values and the time warp contour starting value are determined by the warp ratio values. The time warp contour calculator computes a time warp contour node value of a given time warp contour node, on the basis of a product-formation having a ratio between the time warp contour node values of the intermediate time warp contour node and the time warp contour starting value and a ratio between the time warp contour node values of the given time warp contour node and of the intermediate time warp contour node as factors.

An error concealment method and apparatus for an audio signal and a decoding method and apparatus for an audio signal using the error concealment method and apparatus. The error concealment method includes selecting one of an error concealment in a frequency domain and an error concealment in a time domain as an error concealment scheme for a current frame based on a predetermined criteria when an error occurs in the current frame, selecting one of a repetition scheme and an interpolation scheme in the frequency domain as the error concealment scheme for the current frame based on a predetermined criteria when the error concealment in the frequency domain is selected, and concealing the error of the current frame using the selected scheme.

A multi-participant conference system and method is described. The multi-participant system includes a PSTN client, at least one remote client and a first participant client. The PSTN client communicates audio data and the remote clients communicate audio-video data. The first participant client includes a voice over IP (VoIP) encoder, a VoIP decoder, a first audio mixer, and a second audio mixer. The VoIP encoder compresses audio data transported to the PSTN client. The VoIP decoder then decodes audio data from the PSTN client. The first audio mixer mixes the decoded audio data from the PSTN client with the audio-video data from the first participant into a first mixed audio-video data stream transmitted to the remote client. The second audio mixer mixes the audio-video data stream from the first participant with the audio-video data stream from each remote client into a second mixed audio transmitted to the PSTN client.

Audio signals are processed for use in a communication event. A data store may be queried to obtain an indication of an echo direction, which relates to a direction from which audio signals output from the audio output are likely to be received at a microphone array (plurality of microphones) of a device. Beamformer coefficients of an adaptive beamformer of the device are determined in dependence upon the received indication of the echo direction. Audio signals are received at the microphone array. The adaptive beamformer applies the determined beamformer coefficients to the received audio signals, thereby generating a beamformer output for use in the communication event. The beamformer coefficients are determined such that echo suppression is applied to audio signals received at the microphone array from the indicated echo direction.

Methods, devices, and computer program products for providing instant messaging in conjunction with an audiovisual, video, or audio program are provided. The methods include providing an audiovisual, video, or audio program to a user. Viewer/listener input is received requesting activation of a program-based instant messaging function. A viewer/listener identifier corresponding to the viewer/listener is associated with a program identifier that uniquely identifies the audiovisual, video, or audio program being provided to the user to thereby generate a program viewer/listener record. The program viewer/listener record is transmitted to an electronic database. A list of other users who are viewing or listening to the program in addition to the viewer/listener is acquired from the electronic database. The list of other users is transmitted to the viewer/listener.

A lightweight radio/CD player for vehicular application is virtually “fastenerless” and includes a case and frontal interface formed of polymer based material that is molded to provide details to accept audio devices such as playback mechanisms (if desired) and radio receivers, as well as the circuit boards required for electrical control and display. The case and frontal interface are of composite structure, including an insert molded electrically conductive wire mesh screen that has been pre-formed to contour with the molding operation. The wire mesh provides EMC, RFI, BCI and ESD shielding and grounding of the circuit boards via exposed wire mesh pads and adjacent ground clips. The PCB architecture is bifurcated into a first board carrying common circuit components in a surface mount configuration suitable for high volume production, and a second board carrying application specific circuit components in a wave soldered stick mount configuration. The major components and subassemblies are self-fixturing during the final assembly process, eliminating the need for dedicated tools, fixtures and assembly equipment. The major components and subassemblies self-interconnect by integral guide and connection features effecting “slide lock” and “snap lock” self-interconnection. The radio architecture includes improved push buttons employing 4-bar living hinge linkage and front loaded decorative trim buttons.

A lightweight radio/CD player for vehicular application includes a case and frontal interface formed of polymer based material molded to provide details to accept audio devices and radio receivers, as well as the circuit boards required for electrical control and display. The case and frontal interface are of composite structure, including an insert molded electrically conductive wire mesh screen that has been pre-formed to contour with the molding operation. The wire mesh provides shielding and grounding of the circuit boards via exposed wire mesh pads and adjacent ground clips.

A mobile wireless communications device may include a portable housing, a circuit board carried by the portable housing, a wireless communications circuit carried by the circuit board, and an audio circuit carried by the circuit board. The mobile wireless communications device may further include an antenna assembly including an antenna carrier frame coupled to the circuit board and defining a cavity therein, and an antenna element carried on the antenna carrier frame and having a plurality of spaced apart signal feed points coupled to the wireless communications circuit. In addition, an audio transducer may be carried within the cavity of the antenna carrier frame and coupled to the audio circuit.

In one embodiment, a method includes obtaining a link to a video program; obtaining metadata that relates to the program and that defines, for a specified time point in the program, annotations to be invoked at the specified time point; wherein the annotations comprise: a graphic image; one or more filters, each of the filters comprising a key and one or more matching values; and optionally a reference to a video segment, an electronic document, program code statements, or a programmatic call; during playing the video, detecting that the video program is playing at the specified time point; in response to the detecting: for each particular annotation for the specified time point, retrieving a current value for the key, and causing to display the graphic image associated with that particular annotation only when the current value of the key matches one of the matching values of one of the filters.

An adaptable audio instruction system and method allows for tailoring and modification to audio sequences used for audio instruction of users. The tailoring and modification abilities of the system regard content and presentation details of the audio sequences to comply with user preferences and user progress in learning content contained in the audio sequences.

A power converter with positive and negative supply rail outputs for feeding a single ended class D amplifier, the converter comprising a transformer arrangement, a supply pump reduction arrangement connected between the secondary windings and the positive and negative supply rail outputs, and a boost drive mode switching arrangement. A controller is adapted to control the power converter in a negate drive mode and a boost drive mode, wherein the output voltage in the boost mode is increased by means of the transformer and the boost drive mode switching arrangement. The output voltages on the positive and negative rails can be generated at two different output voltage levels without changing the duty cycle or dead time of the control signals.

An apparatus is described that includes an audio power amplifier having an input and an output. An alternating-current to direct-current power converter is coupled to the audio power amplifier in a single package to supply power to the audio power amplifier.

An audio performance system for use during an event including a shipping container containing at least a first electroacoustic device. A lifting device forms a first portion of the shipping container, the lifting device configured to movably secure the first electroacoustic device between a first position for transport of the container and a second position for the event. A second portion of the shipping container is configured to be manually removable. A third portion of the shipping container is configured to positionably secure a second electroacoustic device for the event.

Rebaudioside M compositions with improved aqueous solubility and methods for preparing the same are provided herein. The rebaudioside M compositions include (i) disordered crystalline compositions comprising rebaudioside M and rebaudioside D, (ii) spray-dried compositions comprising rebaudioside M, rebaudioside D and steviol glycoside mixtures and/or rebaudioside B and/or NSF-02, (iii) spray-dried compositions comprising rebaudioside M, rebaudioside D and at least one surfactant, polymer, saponin, carbohydrate, polyol, preservative or a combination thereof. Sweetened compositions, such a beverages, containing the rebaudioside M compositions with improved water solubility are also provided herein.

The system and corresponding process includes a system for producing a mechanical image of original audio source media and a system for encoding the mechanical image information into a digital file. A processing system recovers the mechanical image information from the digital file at a receiving end. Audio processing is used to produce the original audio source material without the standard losses associated with digital encoding of audio material

A content server generates sponsored audio including procedurally generated background music. The content server obtains reference music features describing musicological characteristics of reference songs as well as sponsored audio information received from a third-party. The content server determines music generation parameters based on the sponsored audio information and based on a music model mapping the reference music features to music generation parameters. The music model may incorporate machine learning techniques to improve the mapping. The content server generates background music by using the determined music generation parameters as input to a procedural music algorithm, and generates sponsored audio by combining the generated background music concurrently with an audio voiceover obtained from the sponsored audio information. The sponsored audio is provided to a client device, which presents the sponsored audio to a user.

A delay time counter in a DSP cyclically counts a sampling clock from zero to a delay time sampling count and issues a delay time interrupt to a CPU each time the sampling clock count reaches the delay time sampling count. The CPU measures a time difference between each time the DSP issues the delay time interrupt and each time sequence clock interrupts occur a number of times corresponding to the delay time. Then, in order to reduce this time difference, the CPU increases or decreases a maximum count that is set to the sequence clock counter. Therefore, in the next delay process, the shift between the time by which the automatic performance is advanced by the CPU (which is equal to the delay time) and the timing of the delay process executed by the DSP (which is also equal in length to the delay time) will be corrected.

The 2012 Progressive Opportunities Conference showcased the best work of nonprofits, companies, unions, activists, and government leaders, looking to shed light on some of our most pressing economic issues. The Conference sought to create an opportunity to network and exchange ideas, arming attendees with new tools to make positive economic change.…

Our friends at The Pro Audio Files have compiled a big list of the best audio articles, videos and podcast from 2014. I’ve been seriously…

Today I was reading a post on a facebook group from a guy, let’s call him Jack, concerned with a master that was clean in…

Sam (Corridor Digital) has been experimenting with 3D printing scans of his ears for binaural sound capture. He's using a pair of Sennheiser wireless lav mics inside the fake ear's canals. For a quick proof of concept experiment, this worked extremely well.

3 of my most used apps on my iPhone for audio engineering/production.

Earlier this year Apple came under some criticism after a report by the Guardian on how Apple hired contractors to listen to a small percentage of anonymized ‌Siri‌ recordings to evaluate the responses with the purpose of improving accuracy and reliability.

In response, Apple promised that it would introduce options in a forthcoming update that would allow users to opt out of sharing their audio recordings.

Those options arrived with the releases of iOS 13.2, iPadOS 13.2, and macOS 10.15.1.

In this podcast, Tyler walks us through how to access and change these new options.

This episode, we talk to Milk Dee. Along with his brother Giz, they formed the Audio Two, an influential duo whose biggest hit, “Top Billin’,” has been sampled or interpolated by pretty much every rapper alive. We chat with Milk about the Audio Two, his successful production career (he was behind Eamon’s monster hit “Fuck It”), and the work of his sister, the legendary MC Lyte.

See http://theciphershow.com/episode/40/ for full show notes and comments.

Mammoth Audio has made available a free lite version of its Density cinematic toolkit for Native Instruments Kontakt. “It’s a tough moment in our history, but one positive has been producers getting back into music making with the spare time. We want to gift the creative community with something which would inspire them to stay […]

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Spitfire Audio has announced a new LABS series instrument titled Tundra Atmos, featuring atmospheric textures. While on various shoots in the Northern hemisphere for Albion Tundra, we captured the natural sounds of near-silent environments — lava flows in the volcanic hinterlands of Iceland, a lapping Scottish loch, and a dense Estonian forest — treated and […]

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Blue Cat Audio has released a free update for its multiple tracks audio analyzers Blue Cat’s FreqAnalyst Multi, Oscilloscope Multi, and StereoScope Multi. These special audio analysis plugins can communicate between instances: It is thus possible to visualize the spectrum, stereo image and waveform of multiple audio tracks on a single graph simultaneously. These tools […]

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ModeAudio has announced its 2020 Spring Sale, offering a 30% discount on all samples and synth presets for a limited time. Our first sale of 2020 is finally here, discounting every last one of our royalty-free sound packs by a tasty 30%! From treasure troves of vintage drum samples & intricate synth presets to loop […]

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Audiofier has announced the release of VenKatt, a multi-layer ambient machine for ambient/electronic productions and film scores. The Kontakt instrument offers 4 sound layers with unique features: Attack, Sustain, Reverse and Rhythmic. The Attack layer includes only transient sounds. The Sustain layer includes only sustaining sounds without transients. When combining two different instruments’ characters, brand […]

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XLN Audio has partnered with leading music production platform Splice to make one of its most popular production tools available at a more musician-friendly price point for the first time. XLN’s RC-20 Retro Color is now available to producers for the introductory rate of $4.99 USD/month (for 20 months) through Splice Rent-to-Own, a unique, musician-friendly […]

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Lunatic Audio has announced the release of Retro Pong, a free plugin that brings the video game from the 80s to your DAW. Lunatic Audio debuts with a ground breaking revolutionary plugin that is taking the industry by storm! As the name suggests: yes it is a straight up 80’s pong video game wrapped in […]

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Numerical Audio has announced availability of the MM-1 Mute Master app for iOS, a unique AUv3 effect plugin designed to create automated or generative arrangements by automatically mixing multiple tracks inside a DAW. MM-1 allows you to create modular style arrangements where individual tracks are brought in and out based on a set number of […]

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Magix has announced the release of Samplitude Pro X5, an update to the complete audio production bundle for professionals. The update includes ARA2 to integrate Melodyne more seamlessly than before, MIDI Polyphonic Expression support, a new section for simplifying monitoring, tempo track functions, automation shapes, Convology XT Complete vintage reverb units, and coreFX VolumeFormer. Samplitude […]

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Acustica Audio has announced a sale on its Purple 3 equalizer collection, offering a 65% discount for a limited time. The Purple suite was just updated to Core15, bringing the latest in VVK Technology. It now also includes a brand ‘new’ filter for a total of 6 processors. If you are tired of looking around […]

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Spitfire Audio has announced availability of Cinematic Percussion, a new Originals series instrument for the Spitfire Audio App. Epic swells and hits, performed by London’s best percussion players and recorded at AIR Studios — home of blockbuster film scores. Cinematic Percussion features London’s best percussion players. Recorded at London’s AIR Studios, home of blockbuster film […]

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Tone2 has announced the release of a new soundset for the Icarus2 synthesizer workstation instrument titled Sparkle. Suitable for Drum & Bass, Trap, Dubstep, Electronic, Dance, Cinematic, Synthwave, HipHop and many other genres, the soundset includes bass, wobbles, arps, keys, leads, pads, atmos, chords, sequeneces, synth sounds and more. Sparkle features 166 patches. Created by […]

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