WO2017124904A1 - Procédé et dispositif de lecture audio - Google Patents

Procédé et dispositif de lecture audio Download PDF

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Publication number
WO2017124904A1
WO2017124904A1 PCT/CN2016/113252 CN2016113252W WO2017124904A1 WO 2017124904 A1 WO2017124904 A1 WO 2017124904A1 CN 2016113252 W CN2016113252 W CN 2016113252W WO 2017124904 A1 WO2017124904 A1 WO 2017124904A1
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WO
WIPO (PCT)
Prior art keywords
audio data
speed
buffer
decoding
data packet
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PCT/CN2016/113252
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English (en)
Chinese (zh)
Inventor
张龙华
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广州视睿电子科技有限公司
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Publication of WO2017124904A1 publication Critical patent/WO2017124904A1/fr

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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04N—PICTORIAL COMMUNICATION, e.g. TELEVISION
    • H04N21/00—Selective content distribution, e.g. interactive television or video on demand [VOD]
    • H04N21/40—Client devices specifically adapted for the reception of or interaction with content, e.g. set-top-box [STB]; Operations thereof
    • H04N21/43—Processing of content or additional data, e.g. demultiplexing additional data from a digital video stream; Elementary client operations, e.g. monitoring of home network or synchronising decoder's clock; Client middleware
    • H04N21/439—Processing of audio elementary streams
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04N—PICTORIAL COMMUNICATION, e.g. TELEVISION
    • H04N21/00—Selective content distribution, e.g. interactive television or video on demand [VOD]
    • H04N21/40—Client devices specifically adapted for the reception of or interaction with content, e.g. set-top-box [STB]; Operations thereof
    • H04N21/43—Processing of content or additional data, e.g. demultiplexing additional data from a digital video stream; Elementary client operations, e.g. monitoring of home network or synchronising decoder's clock; Client middleware
    • H04N21/439—Processing of audio elementary streams
    • H04N21/4392—Processing of audio elementary streams involving audio buffer management
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04N—PICTORIAL COMMUNICATION, e.g. TELEVISION
    • H04N21/00—Selective content distribution, e.g. interactive television or video on demand [VOD]
    • H04N21/40—Client devices specifically adapted for the reception of or interaction with content, e.g. set-top-box [STB]; Operations thereof
    • H04N21/43—Processing of content or additional data, e.g. demultiplexing additional data from a digital video stream; Elementary client operations, e.g. monitoring of home network or synchronising decoder's clock; Client middleware
    • H04N21/439—Processing of audio elementary streams
    • H04N21/4398—Processing of audio elementary streams involving reformatting operations of audio signals
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04N—PICTORIAL COMMUNICATION, e.g. TELEVISION
    • H04N21/00—Selective content distribution, e.g. interactive television or video on demand [VOD]
    • H04N21/40—Client devices specifically adapted for the reception of or interaction with content, e.g. set-top-box [STB]; Operations thereof
    • H04N21/43—Processing of content or additional data, e.g. demultiplexing additional data from a digital video stream; Elementary client operations, e.g. monitoring of home network or synchronising decoder's clock; Client middleware
    • H04N21/442—Monitoring of processes or resources, e.g. detecting the failure of a recording device, monitoring the downstream bandwidth, the number of times a movie has been viewed, the storage space available from the internal hard disk
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04N—PICTORIAL COMMUNICATION, e.g. TELEVISION
    • H04N21/00—Selective content distribution, e.g. interactive television or video on demand [VOD]
    • H04N21/40—Client devices specifically adapted for the reception of or interaction with content, e.g. set-top-box [STB]; Operations thereof
    • H04N21/43—Processing of content or additional data, e.g. demultiplexing additional data from a digital video stream; Elementary client operations, e.g. monitoring of home network or synchronising decoder's clock; Client middleware
    • H04N21/442—Monitoring of processes or resources, e.g. detecting the failure of a recording device, monitoring the downstream bandwidth, the number of times a movie has been viewed, the storage space available from the internal hard disk
    • H04N21/44204—Monitoring of content usage, e.g. the number of times a movie has been viewed, copied or the amount which has been watched

Definitions

  • the present invention relates to the field of network technologies, and in particular, to an audio playing method and apparatus.
  • the ideal audio interactive communication is that the interval at which the transmitting end sends the data packet is equal to the interval at which the receiving end receives the data packet, and the receiving end directly decodes and plays the data packet after receiving the data packet.
  • UDP User Datagram Protocol
  • UDP transmission may be out of order and packet loss, and the time when each data packet arrives at the receiving end may also be There will be different delays, so that the interval at which the receiving end receives the data packet is not fixed. If the network becomes better, the interval at which the receiving end receives the data packet becomes smaller.
  • the receiving end can normally decode and play the data packet; if the network is degraded, the interval at which the receiving end receives the data packet becomes larger, and the decoding and playback of the data is inevitable. Waiting occurs, making the playback inconsistent, resulting in a problem that the receiving end receives data quickly and slowly, and the data distortion is serious.
  • the embodiment of the invention provides an audio playing method and device, which can improve the smoothness of audio playing.
  • An embodiment of the present invention provides an audio playing method, including:
  • the audio data packet is read from the first buffer at the decoding speed for decoding playback.
  • the detecting the number of the audio data packets buffered in the first buffer, and adjusting the decoding speed in real time according to the number of the audio data packets specifically includes:
  • the first speed is expressed as decoding the next audio data packet after each audio data packet is played; the second speed is expressed as decoding the next audio data after each audio data packet is played and the preset duration is hibernated.
  • the third speed is represented as decoding the next audio data packet of the discarded audio data packet after each audio data packet is played and the next audio data packet is discarded.
  • the audio playing method further includes:
  • the decoding operation is suspended, and the first buffer is expanded into a second buffer to buffer the received audio data packet;
  • the second threshold is the number of audio data packets that can be buffered in the first buffer
  • the first threshold is half of the number of audio data packets that can be buffered in the first buffer
  • the third threshold is the number of audio data packets that can be buffered in the second buffer.
  • an audio playback device including:
  • a cache module configured to receive an audio data packet, and cache the audio data packet into a first buffer
  • a detecting module configured to detect a number of audio data packets buffered in the first buffer, and adjust a decoding speed in real time according to the number of the audio data packets
  • a playing module configured to read the audio data packet from the first buffer according to the decoding speed for decoding and playing.
  • the detecting module specifically includes:
  • a detecting unit configured to detect the number of audio data packets buffered in the first buffer
  • a first adjusting unit configured to adjust the decoding speed to a first speed when the number of the audio data packets is greater than a preset first threshold and less than a preset second threshold;
  • a second adjusting unit configured to adjust the decoding speed to a second speed when the number of the audio data packets is less than a preset first threshold; the second speed is smaller than the first speed;
  • the third adjusting unit is configured to adjust the playing speed to a third speed when the number of the audio data packets is greater than a preset second threshold; the third speed is greater than the first speed.
  • the first speed is expressed as decoding the next audio data packet after each audio data packet is played; the second speed is expressed as decoding the next audio data after each audio data packet is played and the preset duration is hibernated.
  • the third speed is expressed as decoding the discarded audio data after each audio data packet is played and the next audio data packet is discarded. The next audio packet of the package.
  • the audio playback device further includes:
  • a loop detection module configured to start timing when the decoding speed is adjusted to the second speed, and cyclically detect whether the decoding speed is still the second speed;
  • a buffer expansion module configured to: if the decoding speed is still the second speed, if the time period of the timer reaches a preset duration threshold, pause the decoding operation, and expand the first buffer to the first buffer Two buffers to buffer the received audio packets; and,
  • a replay module configured to re-read from the second buffer according to the first speed when detecting that the number of audio data packets buffered in the second buffer reaches a preset third threshold The audio data packet is decoded and played.
  • the second threshold is the number of audio data packets that can be buffered in the first buffer
  • the first threshold is half of the number of audio data packets that can be buffered in the first buffer
  • the third threshold is the number of audio data packets that can be buffered in the second buffer.
  • the audio playing method and device provided by the embodiments of the present invention can buffer the received audio data packet, and adjust the speed of decoding and playing in real time according to the number of buffered audio data packets to adapt to different network conditions and ensure audio.
  • the audio data packet in the playback buffer area is accelerated, and when the network condition is poor, the audio data packet in the playback buffer area is decelerated and decoded, so that the speed of normal decoding playback can be restored as soon as possible, and the audio playback is improved.
  • the fluency when the network situation is very poor, expand the buffer capacity, in order to cache more audio data packets before decoding and playback, improve user experience.
  • FIG. 1 is a schematic flow chart of an embodiment of an audio playing method provided by the present invention.
  • step S2 is a schematic flow chart of an embodiment of step S2 in the audio playing method provided by the present invention
  • FIG. 3 is a schematic structural diagram of an embodiment of an audio playback device provided by the present invention.
  • FIG. 4 is a schematic structural diagram of an embodiment of a detection module in an audio playback device provided by the present invention.
  • a schematic flowchart of an embodiment of an audio playing method provided by the present invention includes:
  • a buffer is initialized, and the received audio data packets are cached in the buffer for queuing.
  • the buffer area is full, that is, after the audio data packets buffered in the buffer area reach their capacity, the audio data packets are read from the buffer area for decoding and playing according to the order of the cache.
  • the number of audio data packets buffered in the buffer area is detected in real time, and the decoding speed is adjusted in real time according to the number of audio data packets, so that the audio data packets are decoded and played according to the decoding speed.
  • the decoding speed is adjusted in real time according to different network conditions to ensure the smoothness of audio playback and improve user experience.
  • the detecting the number of audio data packets buffered in the first buffer, and adjusting the decoding speed in real time according to the number of the audio data packets specifically includes:
  • the speed of receiving the audio data packet is fast and slow, so that the network condition is determined according to the number of audio data packets buffered in the first buffer. If the number of the audio data packets is greater than the preset first threshold and less than the preset second threshold, the network condition is normal, and the decoding is performed according to the first speed, that is, the normal speed of decoding; if the audio data packet If the number is less than the preset first threshold, the network condition is poor, and the audio data packet needs to be decelerated and decoded, so that the decoding speed is adjusted to the second speed; if the number of audio data packets is greater than the preset number The second threshold indicates that the network condition is good, and the audio data packet needs to be acceleratedly decoded and played, thereby adjusting the decoding speed to the third speed.
  • the first speed is expressed as decoding the next audio data packet after each audio data packet is played; the second speed is expressed as decoding the next audio data after each audio data packet is played and the preset duration is hibernated.
  • the third speed is expressed as decoding the discarded audio data after each audio data packet is played and the next audio data packet is discarded. The next audio packet of the package.
  • the playback time of each audio packet is 10ms.
  • the first speed is to read and decode the audio data packet in the first buffer every 10 ms.
  • the second speed is 10 ms after each 10 ms audio data packet is played, and then the audio data packet in the first buffer is read and played, so that the speech rate heard by the user is slowed down, thereby achieving the purpose of deceleration.
  • the third speed is that after a 10ms audio data packet is played, one 10ms audio data packet in the first buffer is discarded, and the next 10ms audio data packet of the discarded audio data packet in the first buffer is read, so that the next 10ms audio data packet is discarded.
  • the speed of speech that the user hears is accelerated, thereby achieving the purpose of acceleration.
  • the audio playing method further includes:
  • the decoding operation is suspended, and the first buffer is expanded into a second buffer to buffer the received audio data packet;
  • the timing starts. If the duration of the timing reaches the preset duration threshold, such as 5s, the decoding speed cannot be restored to the normal speed, that is, the audio data packet is still in the deceleration decoding state. , indicating that the network situation is very poor, will first suspend the decoding operation, and expand the capacity of the first buffer, generally double the capacity of the buffer, thereby expanding the first buffer to the second buffer, and Wait until the number of audio data buffers buffered in the second buffer reaches the preset third threshold, and then restart the decoding operation. If the decoding speed cannot be restored to normal speed after the decoding operation is performed, continue. Expand the capacity of the second buffer. If the decoding speed returns to the normal speed before the timer duration reaches the preset duration threshold, the network condition is improved, and the audio data packet can be decoded according to the normal speed.
  • the preset duration threshold such as 5s
  • the second threshold is the number of audio data packets that can be buffered in the first buffer
  • the first threshold is half of the number of audio data packets that can be buffered in the first buffer
  • the third threshold is the number of audio data packets that can be buffered in the second buffer.
  • each audio data packet has a play duration of 10 ms, and the first buffer can buffer 10 audio data packets, then the second threshold is set to 10, and the first threshold is set to 5. If the number of audio data packets buffered in the first buffer is maintained at 5 to 10, the network condition is normal, and the audio data packet is normally decoded and played; if the audio data packet is buffered in the first buffer If the number is less than 5, the network condition is poor, there is a certain delay, and the audio data packet needs to be decelerated and decoded. If the number of audio data packets buffered in the first buffer exceeds 10, the network is described. The situation is better, and the audio data packet needs to be acceleratedly decoded and played.
  • the first buffer is expanded.
  • the second buffer is buffered so that the second buffer can buffer 20 audio data packets, that is, the third threshold is set to 20, and the decoding operation is suspended, waiting for the reception of the audio data packet.
  • the audio data packets in the second buffer are re-read at normal speed for decoding and playing.
  • the audio playing method provided by the embodiment of the invention can buffer the received audio data packet, and adjust the speed of decoding and playing in real time according to the number of buffered audio data packets, so as to adapt to different network conditions and ensure audio playback. Fluency while improving user experience. Moreover, when the network condition is good, the audio data packet in the playback buffer area is accelerated, and when the network condition is poor, the audio data packet in the playback buffer area is decelerated and decoded, so that the speed of normal decoding playback can be restored as soon as possible, and the audio playback is improved. The fluency; when the network situation is very poor, expand the buffer capacity, in order to cache more audio data packets before decoding and playback, improve user experience.
  • the present invention also provides an audio playback device capable of implementing all the processes of the audio playback method in the above embodiments.
  • FIG. 3 is a schematic structural diagram of an embodiment of an audio playback device provided by the present invention, including:
  • a cache module 1 configured to receive an audio data packet and buffer the audio data packet into a first buffer
  • the detecting module 2 is configured to detect the number of audio data packets buffered in the first buffer, and adjust the decoding speed in real time according to the number of the audio data packets;
  • the playing module 3 is configured to read the audio data packet from the first buffer according to the decoding speed for decoding and playing.
  • the detecting module 2 specifically includes:
  • the detecting unit 21 is configured to detect the number of audio data packets buffered in the first buffer.
  • the first adjusting unit 22 is configured to adjust the decoding speed to a first speed when the number of the audio data packets is greater than a preset first threshold and less than a preset second threshold;
  • the second adjusting unit 23 is configured to adjust the decoding speed to a second speed when the number of the audio data packets is less than a preset first threshold; the second speed is smaller than the first speed;
  • the third adjusting unit 24 is configured to adjust the playing speed to a third speed when the number of the audio data packets is greater than a preset second threshold; the third speed is greater than the first speed.
  • the first speed is expressed as decoding the next audio data packet after each audio data packet is played; the second speed is expressed as decoding the next audio data after each audio data packet is played and the preset duration is hibernated.
  • the third speed is represented as decoding the next audio data packet of the discarded audio data packet after each audio data packet is played and the next audio data packet is discarded.
  • the audio playback device further includes:
  • a loop detection module configured to start timing when the decoding speed is adjusted to the second speed, and cycle detection Whether the decoding speed is still the second speed
  • a buffer expansion module configured to: if the decoding speed is still the second speed, if the time period of the timer reaches a preset duration threshold, pause the decoding operation, and expand the first buffer to the first buffer Two buffers to buffer the received audio packets; and,
  • a replay module configured to re-read from the second buffer according to the first speed when detecting that the number of audio data packets buffered in the second buffer reaches a preset third threshold The audio data packet is decoded and played.
  • the second threshold is the number of audio data packets that can be buffered in the first buffer
  • the first threshold is half of the number of audio data packets that can be buffered in the first buffer
  • the third threshold is the number of audio data packets that can be buffered in the second buffer.
  • the audio playing device can buffer the received audio data packet, and adjust the speed of the decoding and playing in real time according to the number of the buffered audio data packets, so as to adapt to different network conditions and ensure audio playback. Fluency while improving user experience. Moreover, when the network condition is good, the audio data packet in the playback buffer area is accelerated, and when the network condition is poor, the audio data packet in the playback buffer area is decelerated and decoded, so that the speed of normal decoding playback can be restored as soon as possible, and the audio playback is improved. The fluency; when the network situation is very poor, expand the buffer capacity, in order to cache more audio data packets before decoding and playback, improve user experience.

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  • Engineering & Computer Science (AREA)
  • Multimedia (AREA)
  • Signal Processing (AREA)
  • Databases & Information Systems (AREA)
  • Computer Networks & Wireless Communication (AREA)
  • Data Exchanges In Wide-Area Networks (AREA)

Abstract

La présente invention concerne un procédé de lecture audio consistant : à recevoir un paquet de données audio, et à mettre en mémoire tampon le paquet de données audio dans une première zone de mémoire tampon ; à déterminer le nombre de paquets de données audio mis en mémoire tampon dans la première zone de mémoire tampon, et à régler une vitesse de décodage en temps réel selon le nombre de paquets de données audio ; à lire le paquet de données audio de la première zone de mémoire tampon à la vitesse de décodage en vue du décodage et de la lecture. De manière correspondante, la présente invention concerne également un dispositif de lecture audio. Les modes de réalisation de la présente invention peuvent améliorer la fluidité de lecture audio.
PCT/CN2016/113252 2016-01-22 2016-12-29 Procédé et dispositif de lecture audio WO2017124904A1 (fr)

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
CN201610046305.5 2016-01-22
CN201610046305.5A CN105704554A (zh) 2016-01-22 2016-01-22 一种音频播放方法及装置

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CN112398912A (zh) * 2020-10-26 2021-02-23 北京佳讯飞鸿电气股份有限公司 一种语音信号加速方法、装置、计算机设备及存储介质
CN112398912B (zh) * 2020-10-26 2024-02-27 北京佳讯飞鸿电气股份有限公司 一种语音信号加速方法、装置、计算机设备及存储介质
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CN114339406B (zh) * 2022-01-27 2023-01-24 重庆紫光华山智安科技有限公司 送解码速度调整方法、系统、设备及介质

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