PCM3501E/2K Allicdata Electronics
Allicdata Part #:

PCM3501E/2K-ND

Manufacturer Part#:

PCM3501E/2K

Price: $ 0.00
Product Category:

Integrated Circuits (ICs)

Manufacturer: Texas Instruments
Short Description: IC VOICE/MODEM CODEC 24-SSOP
More Detail: Voice-Band Interface 16 b Serial 24-SSOP
DataSheet: PCM3501E/2K datasheetPCM3501E/2K Datasheet/PDF
Quantity: 1000
Stock 1000Can Ship Immediately
Specifications
S/N Ratio, ADCs / DACs (db) Typ: 88 / 94
Base Part Number: PCM3501
Supplier Device Package: 24-SSOP
Package / Case: 24-SSOP (0.209", 5.30mm Width)
Mounting Type: Surface Mount
Operating Temperature: -25°C ~ 85°C
Voltage - Supply, Digital: 2.7 V ~ 3.6 V
Voltage - Supply, Analog: 2.7 V ~ 3.6 V
Dynamic Range, ADCs / DACs (db) Typ: 88 / 92
Series: SoundPlus™
Sigma Delta: Yes
Number of ADCs / DACs: 1 / 1
Resolution (Bits): 16 b
Data Interface: Serial
Type: Voice-Band
Part Status: Active
Packaging: Tape & Reel (TR) 
Description

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PCM3501E/2K is an industry-leading mono input codec and mono output codec, which provides the most commonly used compression and decompression functions. The product is the first portable solution on the market featuring a compact, low-power design and industry-leading audio accuracy.

The PCM3501E/2K device is suitable for a variety of audio applications, including audio playback, recording and product evaluation, and is an ideal choice for developers who require industry-leading audio accuracy.

Application Field

The PCM3501E/2K device can be used in a variety of applications, such as computer sound cards and peripheral audio-related products, portable personal audio devices, automotive audio systems and consumer electronics products. It can also be used in telephone systems and medical equipment.

Working Principle

The PCM3501E/2K device uses two types of compression algorithms, pulse code modulation (PCM) and adaptive differential pulse code modulation (ADPCM). The PCM codec converts digital signals into analog signals, while the ADPCM codec performs data compression.

The PCM codec processes audio signals by converting them into digital signals in four stages: analog-to-digital conversion, quantization, encoding and data compression. Once the signal is converted into digital format, it is compressed using a constant bit rate (CBR) algorithm. The CBR algorithm stores audio data in a compressed format but does not perform any actual data compression.

The ADPCM codec then performs data compression by storing audio signals in a compressed format. This compression format can reduce the amount of data required for the transmission of audio signals by up to 80 percent. ADPCM also offers a low latency time and ensures high audio quality.

The PCM3501E/2K device supports a variety of sampling frequencies, including 8, 11.025, 12, 16, 22.05, 24, 32 and 48 kHz. It also supports a wide range of audio formats, including mono, stereo and multi-channel, as well as a variety of bit depths and data rates.

In addition, the PCM3501E/2K device features a variety of features, such as dynamic range control, de-emphasize and gain control, and support for automatic gain control (AGC) and sample rate conversion (SRC). It also supports stereo enhancement, which improves the stereo separation of a signal.

The PCM3501E/2K device is designed to meet the needs of a wide range of applications. It offers excellent audio accuracy and is an ideal solution for developers who require industry-leading audio accuracy.

The specific data is subject to PDF, and the above content is for reference

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