CN112435676A

CN112435676A - Audio processing system based on live broadcast

Info

Publication number: CN112435676A
Application number: CN202011247977.5A
Authority: CN
Inventors: 黄志健
Original assignee: Individual
Current assignee: Individual
Priority date: 2020-11-10
Filing date: 2020-11-10
Publication date: 2021-03-02

Abstract

The invention discloses a live broadcast-based audio processing system, which relates to the technical field of audio processing and comprises an analog audio acquisition module, a DSP audio processing module, an audio acquisition module and an audio decoding module, wherein the analog audio acquisition module is connected with the DSP audio processing module, and the audio acquisition module and the audio decoding module are respectively connected with the DSP audio processing module. The invention realizes the extraction of I2S digital audio, the processing and switching by the DSP audio processing module, the conversion of the I2S digital audio into USB digital audio stream for recording, the decoding and outputting of the I2S digital audio, the signal loss is small, the high fidelity performance effect is realized, the good tone quality is obtained, the problem that the converter can only convert the single-channel audio or the tone loss is large at present is solved, and the problem that the microphone is connected in the live broadcasting process is solved.

Description

Audio processing system based on live broadcast

Technical Field

The invention relates to the technical field of audio processing, in particular to a live-broadcast-based audio processing system.

Background

The 3.5 audio interface of the mobile phone only includes analog stereo output and analog mono microphone input, and the microphone input is converted by the internal a/D (analog/digital) acquisition of the mobile phone, while the input and output of the real digital audio signal are all data communication through Type-c of the mobile phone or USB included in the Lightning interface.

At present, short videos of mobile phones and live broadcasts of mobile phones are a trend of the current network. In the prior art, audio signals accessed by mobile phone video recording, sound recording or network live broadcast mainly access analog audio to a mobile phone, so that the requirement of high fidelity cannot be met.

Two main approaches are used in the art today: firstly, directly insert cell-phone 3.5 headset interface with outside analog audio signal, utilize the microphone signal channel end input of this 3.5 interface, because of the purpose of this microphone passageway development is not for the input music and designs, so utilize this passageway input music can only insert monophonic (single channel) audio signal, the frequency width both ends of sound can be cut the sound and decrease great, also have certain restriction to the input level and will appear the crackle not well controlling. Secondly, the mobile phone is mainly added with a Type-c or Lightning interface Type audio converter, but only analog audio signals can be input, the inside of the audio converter on the market in the prior art is basically manufactured by an ADC chip for mobile terminal use, and the manufacturing method and effect are only equivalent to the 3.5 headset interface which is canceled from the mainstream mobile phone and is restored outside the mobile phone. In addition, a Lightning male plug is embedded in the inner core of the current mainstream Lightning interface converter, and the practical effect is an external 3.5 audio interface. Because the mobile terminal ADC (analog to digital) coding chip is used in the method, and the performance limit of the chip field causes large audio loss, lossless digital audio signals cannot be transmitted to the mobile phone, and even the most stereo cannot transmit only single-channel audio to the interior of the mobile phone. Therefore, a need exists for a live based audio processing system.

An effective solution to the problems in the related art has not been proposed yet.

Disclosure of Invention

The present invention provides a live broadcast-based audio processing system to solve the above technical problems in the related art.

The technical scheme of the invention is realized as follows:

a live broadcast-based audio processing system comprises an analog audio acquisition module, a DSP audio processing module, an audio acquisition module and an audio decoding module, wherein the analog audio acquisition module is connected with the DSP audio processing module, the audio acquisition module and the audio decoding module are respectively connected with the DSP audio processing module,

the analog audio acquisition module is used for acquiring a stereo analog signal and transmitting the stereo analog signal to the DSP audio processing module;

the DSP audio processing module is used for carrying out signal detection and identification on the acquired signals and switching and outputting the acquired signals;

the audio acquisition module; the device is used for converting I2S digital audio into USB digital audio stream for recording;

the audio decoding module; for decoding and outputting the I2S digital audio.

Furthermore, the analog audio acquisition module is connected with an analog audio input preceding stage, and the analog audio input preceding stage is used for inputting and transmitting analog audio signals to the analog audio acquisition module.

Further, the device also comprises an optical fiber coaxial input/output, wherein the optical fiber coaxial input/output is connected with the DSP audio processing module and is used for accessing an external optical fiber coaxial digital audio signal.

Further, the audio acquisition module is connected to the mobile phone terminal, and is configured to output the I2S serial data of the DSP audio processing module to the I2S recording input channel of the audio acquisition module, and convert the I2S digital audio into USB digital audio for being transmitted to the mobile phone terminal for recording.

Further, the audio decoding module is connected to the earphone end, and is configured to output the I2S serial data of the DSP audio processing module to the I2S input channel of the audio decoding module for decoding, and transmit the decoded stereo analog audio signal to the earphone end.

Further, the audio decoding module is connected with a MAX97220 linear driver.

The invention has the beneficial effects that:

the audio processing system based on live broadcasting is characterized in that a stereo analog signal is collected and transmitted to the DSP audio processing module through integrating the analog audio collecting module, the DSP audio processing module, the audio acquiring module and the audio decoding module, the collected signal is detected, identified and switched to output, I2S digital audio is extracted, the I2S digital audio is processed and switched through the DSP audio processing module, the I2S digital audio is converted into USB digital audio stream for recording, and the I2S digital audio is decoded and output, so that the signal loss is small, the high-fidelity performance effect is achieved, good tone quality is obtained, the problem that the existing converter can only convert single-channel audio or has large sound loss is solved, and the problem that a microphone is connected in the live broadcasting process is solved.

Drawings

In order to more clearly illustrate the embodiments of the present invention or the technical solutions in the prior art, the drawings needed in the embodiments will be briefly described below, and it is obvious that the drawings in the following description are only some embodiments of the present invention, and it is obvious for those skilled in the art to obtain other drawings without creative efforts.

FIG. 1 is a functional block diagram of a live based audio processing system according to an embodiment of the present invention;

FIG. 2 is a schematic circuit diagram of an analog audio capture module of a live broadcast-based audio processing system according to an embodiment of the present invention;

fig. 3 is a schematic diagram of a fiber-optic coax input/output circuit of a live-based audio processing system according to an embodiment of the present invention;

FIG. 4 is a block circuit diagram of a DSP audio processing module of a live based audio processing system according to an embodiment of the invention;

FIG. 5 is a schematic diagram of an audio acquisition module of a live based audio processing system according to an embodiment of the present invention;

FIG. 6 is a circuit diagram of an audio decoding module of a live broadcast-based audio processing system according to an embodiment of the present invention;

fig. 7 is a schematic diagram of a power circuit of a live based audio processing system according to an embodiment of the present invention.

In the figure:

1. an analog audio acquisition module; 2. a DSP audio processing module; 3. an audio acquisition module; 4. an audio decoding module; 5. analog audio is input to the preceding stage.

Detailed Description

The technical solutions in the embodiments of the present invention will be clearly and completely described below with reference to the drawings in the embodiments of the present invention, and it is obvious that the described embodiments are only a part of the embodiments of the present invention, and not all of the embodiments. All other embodiments that can be derived by one of ordinary skill in the art from the embodiments given herein are intended to be within the scope of the present invention.

According to an embodiment of the present invention, a live based audio processing system is provided.

As shown in fig. 1-7, the live broadcast-based audio processing system according to the embodiment of the present invention includes an analog audio acquisition module 1, a DSP audio processing module 2, an audio acquisition module 3, and an audio decoding module 4, where the analog audio acquisition module 1 is connected to the DSP audio processing module 2, the audio acquisition module 3 and the audio decoding module 4 are respectively connected to the DSP audio processing module 2,

the analog audio acquisition module 1 is used for acquiring stereo analog signals and transmitting the stereo analog signals to the DSP audio processing module 2;

the DSP audio processing module 2 is used for carrying out signal detection and identification on the acquired signals and switching and outputting the acquired signals;

the audio acquisition module 3; the device is used for converting I2S digital audio into USB digital audio stream for recording;

the audio decoding module 4; for decoding and outputting the I2S digital audio.

The analog audio acquisition module 1 is connected with an analog audio input front stage 5, and the analog audio input front stage 5 is used for inputting and transmitting analog audio signals to the analog audio acquisition module 1.

The digital audio signal processing device also comprises an optical fiber coaxial input/output, wherein the optical fiber coaxial input/output is connected with the DSP audio processing module 2 and is used for accessing an external optical fiber coaxial digital audio signal.

The audio acquisition module 3 is connected to the mobile phone terminal, and is configured to output the I2S serial data of the DSP audio processing module 2 to the I2S recording input channel of the audio acquisition module 3, and convert the I2S digital audio into USB digital audio for being transmitted to the mobile phone terminal for recording by the audio acquisition module 3.

The audio decoding module 4 is connected to the earphone end, and configured to output the I2S serial data of the DSP audio processing module 2 to the I2S input channel of the audio decoding module 4 for decoding, and transmit the decoded stereo analog audio signal to the earphone end.

Wherein, the audio decoding module 4 is further connected with a MAX97220 linear driver.

By means of the above-mentioned technical scheme, through integrated analog audio acquisition module, DSP audio processing module and audio acquisition module and audio decoding module, it gathers stereo analog signal and transmits extremely DSP audio processing module carries out signal detection discernment and switching output to the signal of gathering, the realization is drawed I2S digital audio, and handle and switch through DSP audio processing module, and export the defeated record of carrying out of USB digital audio stream to I2S digital audio conversion, decode the output to I2S digital audio, signal loss is little, have the performance effect of high-fidelity and obtain fine tone quality, solve present converter can only convert the monophonic audio frequency or the great problem of sound loss, and solve live broadcasting in-process company wheat card.

In addition, specifically, the analog audio acquisition module 1 may adopt an AK5720VT chip, the DSP audio processing module 2 may adopt an ADAU1452 chip, the audio acquisition module 3 may adopt an SA9023A chip, and the audio decoding module 4 may adopt a PCM5102A chip, the SA9023A chip may be a high-performance 24-bit USB full-speed compatible digital audio processor, which has a stereo play and recording pair I2S input and output, and may have a bit rate and a sampling rate configured as 16/24dit and 32/44.1/48/88.2/96KHZ, and supports external eepm to load external EEPROM specific data to set an I2S master-slave mode through an EEPROM data set sampling rate, and the like. The ADAU1452 is a fully programmable audio DSP, which can improve the sound processing performance with a sampling rate as high as 192kHz, and uses a proprietary programming tool SigmaStaudio to develop a custom signal flow to support an external self-booting EEPROM memory. The system has stereo S/PDIF input and output, four serial input ports, four serial output ports and eight stereo ASRCs. The input, output, routing, master slave, asynchronous, etc. of TDM, PCM, I2S, left and right aligned format audio can be flexibly configured. In addition, the AK5720VT is a low-voltage 24-bit A/D converter for digital audio system, which can collect and convert single-ended stereo analog signal without external filter, is compatible with I2S, left-and right-alignment formats and can be flexibly set and allocated to master-slave mode with sampling rate of 8-96 KHZ. The PCM5102A is a D/A audio decoder which can receive I2S, left-justified 16-32 bit audio data, the integrated PLL system clock phase-locked loop sampling rate is up to 384KHZ, and the drive force of the included linear driver supports 1K omega load.

In addition, the MAX97220 is a differential input linear driver/headphone amplifier, which can be set as a single-ended input, the integrated charge pump establishes a negative power supply, the output power is 32 Ω 125MW under 5V power supply, excellent THD + N is provided in the whole audio frequency band, and the 18-bit SNR performance can reach 112 dB.

Specifically, when the mobile phone is used, the Type-c or Lightning interface of the mobile phone terminal is connected with the audio acquisition module 3 to perform the mutual communication of the transmission exchange of the USB audio stream data. When the mobile phone is in a playing/recording or live broadcasting state, the audio acquisition module 3 can obtain current sampling rate information of the mobile phone, and can flexibly change along with the change of the sampling rate of the mobile phone after loading specific data of an external EEPROM. And simultaneously, the I2S input (recording) channel master mode of the audio acquisition module 3 outputs a system clock MCLK synchronous with the current sampling rate of the mobile phone, and the system clock MCLK is transmitted to the XTAL-IN clock input end of the DSP audio processing module 2. Therefore, the clock and the sampling rate of the whole system are determined and synchronized by the working state of the mobile phone, so that the whole system can stably work under the uniform clock frequency.

In addition, the four I2S serial data input channels and the four I2S serial data output channels of the DSP audio processing module 2 have very flexible programmability and various parameter settings. Thus the I2S digital audio system may be configured as follows:

first, the I2S output channel I is set as the I2S input (recording) channel connected to the audio acquisition module 3 in slave mode, and provides digital audio data for mobile phone recording through SDATA _ OUT0, BCLK _ OUT0, LRCK _ OUT0 outputs.

And the second I2S output channel is set as a master mode, is connected with the I2S input channel of the audio decoding module 4, and provides digital audio data for monitoring through SDATA _ OUT1, BCLK _ OUT0 and LRCK _ OUT 0. The purpose of using BCLK _ OUT0 and LRCK _ OUT0 is to monitor the sound for anomalies at the sampling rate of the recorded sound. The system clock MCLK of the audio decoding block 4 is provided by the CLKOUT clock output of the DSP audio processing block 2.

And thirdly, the I2S input channel I is set as a master mode and is set as an ASRC input mode, is connected with the I2S output channel of the analog audio acquisition module 1, and receives external analog stereo-to-digital audio signals through SDATA _ IN0, BCLK _ IN0 and LRCK _ IN 0. The system clock MCLK of the analog audio acquisition module 1 is provided by the CLKOUT clock output terminal of the DSP audio processing module 2.

And fourthly, the I2S input channel II is set to be IN a slave mode and is set to be IN an ASRC input mode, is connected with the I2S output (playing) channel of the audio acquisition module 3, and receives digital audio data played by the mobile phone through SDATA _ IN1, BCLK _ IN1 and LRCK _ IN 1.

Fifthly, the default configuration of the DSP audio processing module 2 integrates the ASRC external fiber/coaxial digital audio SPDIF input and supports the maximum sampling rate of 192K. ASRC belongs to asynchronous resampling, which is currently the best SPDIF-compatible technique to suppress Jitter performance.

Sixthly, the DSP audio processing module 2 is configured by default to integrate optical fiber/coaxial digital audio SPDIF output, and the output audio information is set to be consistent with the second output channel of I2S.

And seventhly, the remaining two serial output channels and LRCK _ OUT2, LRCK _ OUT3, LRCK _ IN2 and LRCK _ IN3 of the two serial input channels are combined with MP6 and MP7 to configure functions of monitoring volume plus-minus, mode indicator light and mode selector switch.

In addition, when the mobile phone records, records or broadcasts directly, the audio of the mobile phone is in a recording input state. The acquisition signal enters the DSP audio processing module 2, signal detection and identification are carried out in the DSP audio processing module 2, and the acquisition signal is automatically switched to corresponding input, the recording signal is distributed into two paths of output by the DSP audio processing module 2, one path of output is sent to an I2S recording input channel of the audio acquisition module 3 through an I2S serial data output channel of the DSP audio processing module 2, and the I2S digital audio is converted into a USB digital audio stream by the audio acquisition module 3 and is transmitted to the mobile phone end for recording. The other path is sent to the I2S input channel of the audio decoding module 4 for decoding through the I2S serial data output channel of the DSP audio processing module 2, and the decoded stereo analog audio signal is linearly amplified by the MAX97220 and then output to push the monitoring earphone. Because the recording and the monitoring of the mobile phone are the same audio signal and only have different channels, the sound monitored by the earphone is the sound recorded by the mobile phone.

Specifically, when the mobile phone live broadcasts PK with microphone, the mobile phone audio is in a recording input state and in a playing output state. At this time, the audio acquisition module 3 receives USB digital audio stream data transmitted by the mobile phone while transmitting the USB audio stream signal to the mobile phone for recording, and converts the USB digital audio stream data into I2S digital audio to be transmitted to the I2S input channel two of the DSP audio processing module 2, after the signal and the recording signal are mixed in the DSP audio processing module 2, the mixed signal is transmitted to the I2S input channel two of the DSP audio processing module 2 through the I2S serial data output channel two of the DSP audio processing module 2 for decoding, and the decoded stereo analog audio signal is linearly amplified by MAX97220 and then output to push the monitoring earphone. Therefore, the effect of connecting the microphone can be achieved by hearing the sound played by the mobile phone and the sound recorded by the mobile phone, and the sound played and output by the mobile phone cannot enter a mobile phone recording input channel, so that the reciprocating circulation of the sound cannot be caused.

In summary, according to the above technical solution of the present invention, by integrating the analog audio acquisition module, the DSP audio processing module, the audio acquisition module, and the audio decoding module, the analog stereo signal is acquired and transmitted to the DSP audio processing module, the acquired signal is detected, identified, and switched to output, so as to extract I2S digital audio, and processed and switched by the DSP audio processing module, convert the I2S digital audio into USB digital audio stream for recording, and decode and output the I2S digital audio, which has small signal loss, and has high fidelity performance effect to obtain good tone quality, and solves the problem that the existing converter can only convert monaural audio or has large sound loss, and the problem of connecting a microphone during live broadcasting.

The above description is only for the purpose of illustrating the preferred embodiments of the present invention and is not to be construed as limiting the invention, and any modifications, equivalents, improvements and the like that fall within the spirit and principle of the present invention are intended to be included therein.

Claims

1. An audio processing system based on live broadcast is characterized by comprising an analog audio acquisition module (1), a DSP audio processing module (2), an audio acquisition module (3) and an audio decoding module (4), wherein the analog audio acquisition module (1) is connected with the DSP audio processing module (2), the audio acquisition module (3) and the audio decoding module (4) are respectively connected with the DSP audio processing module (2),

the analog audio acquisition module (1) is used for acquiring stereo analog signals and transmitting the stereo analog signals to the DSP audio processing module (2);

the DSP audio processing module (2) is used for carrying out signal detection and identification on the collected signals and switching and outputting the signals;

the audio acquisition module (3); the device is used for converting I2S digital audio into USB digital audio stream for recording;

the audio decoding module (4); for decoding and outputting the I2S digital audio.

2. The live based audio processing system according to claim 1, wherein an analog audio input pre-stage (5) is connected to the analog audio capture module (1), wherein the analog audio input pre-stage (5) is configured to be input as an analog audio signal and transmitted to the analog audio capture module (1).

3. The live based audio processing system according to claim 1, further comprising a fiber-optic coaxial input/output connected to the DSP audio processing module (2) for accessing external fiber-optic coaxial digital audio signals.

4. The live broadcast-based audio processing system of claim 1, wherein the audio acquisition module (3) is connected to the mobile phone terminal, and is configured to output the I2S serial data of the DSP audio processing module (2) to the I2S recording input channel of the audio acquisition module (3), and convert the I2S digital audio from the audio acquisition module (3) to the USB digital audio transmitting mobile phone terminal for recording.

5. The live based audio processing system according to claim 4, wherein the audio decoding module (4) is connected to the headphone end for outputting the I2S serial data of the DSP audio processing module (2) to the I2S input channel of the audio decoding module (4) for decoding, and the decoded stereo analog audio signal is transmitted to the headphone end.

6. The live based audio processing system according to claim 5, wherein said audio decoding module (4) is further connected with a MAX97220 linear driver.