CN106330907A

CN106330907A - Mobile-end audio high/low-data-rate adaptive play method

Info

Publication number: CN106330907A
Application number: CN201610715202.3A
Authority: CN
Inventors: 岳传真; 石钰; 胡蓓蓓; 李杨; 虞逸斐
Original assignee: Archimedes (shanghai) Media Co Ltd
Current assignee: Archimedes (shanghai) Media Co Ltd
Priority date: 2016-08-24
Filing date: 2016-08-24
Publication date: 2017-01-11

Abstract

The invention provides a method for a mobile end to play different data rates of audios by monitoring network signals intelligently in real time and according to signal intensity. The method is characterized in that the method comprises the following steps: 1) a slice server slices audios in different data rates; 2) the mobile end plays a common data rate audio first; 3) the network (WIFI and a mobile network) signals at the mobile end are monitored in real time; 4) the different data rates of audios are switched according to the monitored network result; 5) the audio is played continuously; and 6) the steps 3)-5) are cycled until audio play is finished.

Description

Mobile terminal audio frequency high-low code flow adaptive identifying method

Technical field

This programme is applicable to mobile terminal, according to the power of current network signal (including WIFI, mobile network), automated intelligent Identification and play the audio frequency of corresponding code stream.

Background technology

Current phone manufacturer is numerous, and the handset capability of production is different, and network signal (WIFI or mobile network) is different Power (the signal intensity that RSSI:Received Signal Strength Indicator-receives in period or different locations Instruction) different, the most off and on, even there is Caton phenomenon, so brings bad experience to user in user's listening to audio, The most sometimes normal work or life are also affected.

The Streaming Media QoS control method of a kind of network self-adapting of technology being currently known (patent No.: CN102547389B), transmitted and the height of player relief area by the message of the QoS of XMPP according to server and client side Low water level judges, increases burden in the case of original network thus can be given more weak, and performance and experience are not fine.Such as one The transmission method (patent No.: CN102098547A) of built-in network self adaptation/real-time high-definition video, is to transmit mould according to network The assessment return value of block, then the video gathered is done the coding of high code stream or the coding of low code stream, so can bring about broadcasting Time delay and the load of increase server.

Therefore, in this case, the user of mobile terminal is used, if the distributor of audio frequency uses identical code stream Audio content, the user of network difference is because network problem is experienced bad, and the good user of network signal can not be preferable because of network And experience preferably (high definition code stream) audio content.

Summary of the invention

The detailed content of this programme relates to server and mobile terminal

1, service end part

Service end is responsible for the distribution of audio code stream, according to circumstances can distribute multiple code stream during distribution: high definition code stream (128k), common code stream (56k) and low code stream (16k), use multimedia Open Framework FFMPEG, cut out corresponding according to demand Audio code stream is cut into slices, and cuts stream order as follows:

Acodec audio coding decoding storehouse codec parameters output file, ffmpeg i audio-source file address (* .m3u8)

The file format cut out approximately as:

#EXTM3U

#EXT-X-VERSION:3

#EXT-X-ALLOW-CACHE:YES

#EXT-X-STREAM-INF:PROGRAM-ID=1, BANDWIDTH=16000//low code stream

http://www.example.com/low.m3u8

#EXT-X-STREAM-INF:PROGRAM-ID=1, BANDWIDTH=56000//common code stream

http://www.example.com/middle.m3u8

#EXT-X-STREAM-INF:PROGRAM-ID=1, BANDWIDTH=128000//high definition code stream

http://www.example.com/high.m3u8

#EXT-X-ENDLIST

The voice data of all kinds of specification code streams that service end is handled well stores on server, then uses unified interface Address is externally announced, in order to mobile terminal accesses.

2, mobile terminal part

The equipment of mobile terminal, owing to location is unfixed, the network state potentially unstable caused, such as hands Machine uses WIFI in Office Area, holds mobile phone when getting around, and network signal will be dynamic, so just may feel that broadcasting During network audio, arise that pause once in a while or Caton phenomenon, if in the case of using mobile network 4G equally, may go to Another place and cause 4G signal the most weak and transfer weak 3G to, also result in audio frequency play card or stop.

We design and can define type as follows according to the size of the dB value of the RSSI of network signal:

It is weak signal during dBm≤-80db, plays low code stream audio frequency；

It is strong signal during dBm >=-25db, plays high definition code stream audio frequency；

DBm value is marginal is the most common signal, code stream audio frequency in broadcasting.

When mobile terminal player commences play out audio frequency, acquiescence plays the audio frequency of common code stream, starts periodically in playing process The detection of network signal, according to the result of detection signal, plays the audio frequency of corresponding code stream automatically, if network signal is more weak, Just it is switched to low code stream audio stream according to reproduction time stamp play, when signal is fine, is switched to high definition code also according to reproduction time The audio stream broadcast point of stream.The code stream that packet loss is more serious can also carry out application sometimes and retransmit packet loss, player is according to current The time that audio frequency is play carries out the seamless switching of high-low code flow, has the most preferably ensured the experience of user.

For broadcast audio, particularly programme televised live, if network signal is very poor, cause user's broadcast listening off and on Can not the broadcast content of the announcer of uppick normally, there may come a time when, owing to losing some contents and misinterpretation, to employ net Network intelligent adaptive audio code stream, has just ensured to miss and has listened to content, experiences more preferably.

Accompanying drawing explanation

Fig. 1: code stream self adaptation is cut stream, is play audio frequency schematic diagram；

Fig. 2: mobile terminal dynamic monitoring network signal also switches audio code stream schematic diagram.

Detailed description of the invention

The operating procedure of this programme is as follows:

1, service end downloads FFMPEG source code, and configuration compiling parameter also compiles FFMPEG, to include supporting sound in configuration Frequently AAC form and M3U8 form, in compilation process, may report an error, and needs to download some plug-in unit supports；

2, section server use slice tool FFMPEG, cut as required the audio stream source address of stream, code stream information and The duration of section, configuration cuts stream parameter accordingly, cuts out the audio file of the different code streams such as high, normal, basic respectively, stores to specifying Server；

3, storage server is responsible for all kinds of audio files that storage section server cuts out, and accesses for mobile terminal and connects；

4, mobile terminal detects the power of network signal RSSI in real time, then judges stablizing of this value according to the result of detection Property, such as there may be the instability that network signal is at times strong and at other times weak in certain short time period, in this case, cannot be right The switching of high-low sound frequency stream is so frequent, and should use the audio frequency of low code stream, waits when network signal becomes strong signal stabilization It is switched to high definition code stream.

5, switching audio code stream, first records the timestamp of the most currently playing audio frequency, then opens and need switching Audio code stream, according to the timestamp just now recorded, directly plays at timestamp, so ensures seamless switching, so gives and use The experience that family is brought is more preferable.

Claims

1. a mobile terminal audio frequency high-low code flow adaptive identifying method, it is characterised in that comprise the steps:

(1) section server cuts out the audio frequency of different code streams；

(2) common code stream audio frequency is play in mobile terminal first；

(3) mobile terminal network signal is monitored in real time；

(4) web results according to monitoring switches the audio frequency of different code streams；

(5) continue to play audio frequency；

(6) circulation step (3), (4) and (5), terminate until audio frequency is play.

2. the method for claim 1, further, the section code stream that cuts out of server include high definition code stream (128k), Common code stream (56k) and low code stream (16k).

The most the method for claim 1, wherein server of cutting into slices cuts out the voice data of different code streams and is stored in storage clothes On business device, unified interface IP address is then used externally to announce, in order to mobile terminal accesses.

4. the method as described in claim 1 or 2 or 3, further, when mobile terminal player commences play out audio frequency, acquiescence is broadcast Put the audio frequency of common code stream, playing process starts periodic detection network signal, according to the result of detection signal, automatically plays phase Answering the audio frequency of code stream, if network signal is more weak, is just switched to low code stream audio stream according to reproduction time stamp and plays, signal is very It is switched to the audio stream broadcast point of high definition code stream also according to reproduction time time good.

5. method as claimed in claim 4, further, the dB value of the RSSI of periodic detection network signal, when dB value is less than Or during equal to first threshold, be switched to low code stream audio stream and play, when dB value is more than or equal to Second Threshold, it is switched to high definition Code stream audio stream is play, and when dB value falls between, plays according to common code stream.

6. method as claimed in claim 4, further, the dB value of the RSSI of periodic detection network signal, according to detection Result judges the stability of this value, if unstable, then uses the audio frequency of low code stream.

7. the method as described in one of claim 4-6, further, the code stream that packet loss is more serious is carried out by mobile terminal player Application retransmits packet loss, and player carries out the seamless switching of high-low code flow according to the time that present video is play.

8. the method for claim 1, further, when switching audio code stream, first records the most currently playing The timestamp of audio frequency, then opens and needs the audio code stream of switching, according to the timestamp just now recorded, directly broadcast at timestamp Put, it is ensured that seamless switching.

9. the method for claim 1, further, described mobile terminal network is WI F I or mobile network.