Media processing method and device

• US RE48,323 E1
• Filed: 11/14/2018
• Issued: 11/24/2020
• Est. Priority Date: 08/04/2008
• Status: Active Grant

• Current Assignee: Apple Inc.
• Original Assignee: Apple Inc.
• Inventors: Conroy, David G., Schell, Steve, Corlett, Barry, Warren, Niel D., Lindahl, Aram
• Primary Examiner(s): Lee, Christopher E.
• Application Number:
  US16/190,506
• Time in Patent Office: 741 Days
• Field of Search: 710 13, 710 29, 710 53, 710316, 455403, 4555501, 4555562, 348 1408, 3484253, 381 17, 381107, 381119, 709213, 709226, 370450, 370466, 360 63, 386353, 718107, 712 35, 280294
• CPC Class Codes: N/A

Abstract

A media processing system and device with improved power usage characteristics, improved audio functionality and improved media security is provided. Embodiments of the media processing system include an audio processing subsystem that operates independently of the host processor for long periods of time, allowing the host processor to enter a low power state while the audio data is being processed. Other aspects of the media processing system provide for enhanced audio effects such as mixing stored audio samples into real-time telephone audio. Still other aspects of the media processing system provide for improved media security due to the isolation of decrypted audio data from the host processor.

Claims

1. A digital signal processor, comprising:

a core processor configured to process audio data;

at least one input buffer coupled to the core processor, the at least one input buffer configured to receive audio data from at least one audio input device;

at least one output buffer coupled to the core processor, the at least one output buffer configured to send processed audio data to at least one audio output device; and

a programmable memory configured to interface with the core processor to control the routing of audio data segments from the at least one input buffer to the at least one output buffer, wherein controlling the routing of the audio data segments comprises determining an order in which the audio data segments are processed by the core processor and routed to the output buffer.

2. The digital signal processor of claim 1, wherein the programmable memory is configured to interface with the core processor to assign a priority level to each of the audio data segments.

3. The digital signal processor of claim 1, wherein the programmable memory is configured to interface with the core processor to control the routing of audio data from the at least one audio input device to the at least one input buffer.

4. The digital signal processor of claim 1, wherein the programmable memory is configured to interface with the core processor to control the routing of processed audio data from the at least one output buffer to the at least one audio output device.

5. The digital signal processor of claim 1, wherein the core processor is configured to mix two or more audio samples;

sample rate convert audio data;

equalize audio data;

compress or decompress audio data;

encrypt or decrypt audio data;

or some combination thereof.

6. The digital signal processor of claim 1, wherein the at least one audio input device comprises one or more microphones, a headset, or a combination thereof.

7. The digital signal processor of claim 1, wherein the at least one audio output device comprises a speaker, a headphone, a headset, or a combination thereof.

8. The digital signal processor of claim 1, wherein the at least one input buffer is configured to receive the audio data from the at least one audio input device comprising first receiving the audio data from a CODEC, wherein the CODEC is configured to convert the audio data into digital audio data.

9. The digital signal processor of claim 8, wherein the CODEC comprises dual communication channels, and wherein the CODEC is configured to convert the audio data into digital audio data and convert the processed audio data into analog audio data substantially simultaneously.

10. The digital signal processor of claim 1, wherein the at least one output buffer is configured to send the processed audio data to the at least one audio output device comprising first sending the processed audio data to a CODEC, wherein the CODEC is configured to convert the processed audio data into analog audio data.

11. A method of processing audio data via a digital signal processor (DSP), comprising:

receiving audio data into an input buffer;

partitioning the audio data into audio data segments via a programmable memory configured to interface with a core processor of the DSP, wherein the audio data segments correspond to a plurality of audio processing task segments;

assigning a respective priority level to each of the plurality of audio processing task segments;

processing each audio processing task segment of the plurality of audio processing task segments via the core processor according to the respective priority level of each of the plurality of audio processing task segments; and

storing the audio data segments into an output buffer.

12. The method of claim 11, wherein receiving the audio data comprises receiving telephone audio data, music audio data, or a combination thereof.

13. The method of claim 11, wherein processing each audio processing task segment of the plurality of audio processing task segments comprises performing audio data compression, audio data decompression, audio data equalization, audio data mixing, or a combination thereof.

14. The method of claim 11, wherein assigning the respective priority level to each of the plurality of audio processing task segments comprises assigning a priority level according to a respective sampling rate of each of the plurality of audio processing task segments.

15. The method of claim 14, wherein assigning the priority level according to the respective sampling rate comprises determining a first sampling rate when the audio data segments comprise telephone audio data and determining a second sampling rate when the audio data segments comprise music audio data, wherein the first sampling rate is less than the second sampling rate.

16. The method of claim 11, wherein assigning the respective priority level to each of the plurality of audio processing task segments comprises assigning a priority level to telephone audio data according to a respective processing time delay associated with the telephone audio data.

21. The method of claim 11, wherein partitioning the audio data into audio data segments comprises dividing each of the plurality of audio processing task segments into a series of smaller task segments to increase a speed at which each of the plurality of audio processing task segments is processed.

17. An electronic device, comprising,

a digital signal processor (DSP), comprising:

an input buffer configured to receive audio data;

a programmable memory configured to interface with a core processor of the DSP to;

partition the audio data into audio data segments, wherein the audio data segments correspond to a plurality of audio processing task segments; and

assign a respective priority level to each of the plurality of audio processing task segments;

wherein the core processor is configured to process each audio processing task segment of the plurality of audio processing task segments according to the respective priority level of each of the plurality of audio processing task segments; and

;

an output buffer configured to store the audio data segments.

18. The electronic device of claim 17, wherein the DSP is configured to switch between the plurality of audio processing task segments based at least in part on whether the audio data segments comprise one of telephone audio data or music audio data.

19. The electronic device of claim 17, wherein programmable memory is configured to interface with the core processor to assign a higher priority level to the telephone audio data than the priority level assigned to the music audio data.

20. The electronic device of claim 17, wherein the core processor is configured to process each audio processing task segment of the plurality of audio processing task segments by processing a first segment of telephone audio data samples, a second segment of music audio data samples, or a combination thereof.