Streaming media data transcoding method, apparatus, computing device and readable storage medium

Abstract

The invention discloses a streaming media data transcoding method, which is suitable for completion in a computing device, comprising: receiving source streaming media data, demultiplexing the source streaming media data stream, decomposing source audio data and source video data, The source audio data is in a first audio coding format; the source audio data is processed by audio data to convert it into target audio data, and the target audio data is in a second audio coding format; video data processing is performed on the source video data Obtain target video data; multiplex the target audio data and target video data to obtain target streaming media data. The invention also discloses a corresponding streaming media data transcoding device, computing device and readable storage medium.

Description

technical field [0001] The present invention relates to the field of audio and video processing, and in particular, to a streaming media data transcoding method, device, computing device and readable storage medium. Background technique [0002] In order to adapt to business needs or network conditions, it is often necessary to change the encoding method of source media data. Communication, WebRTC) technology can provide users with real-time voice calls and video calls supported by web pages, but before WebRTC is widely used in video applications, how to apply WebRTC to existing audio and video systems to convert non-webRTC data into WebRTC Data has become a technical problem that must be solved. Usually in these two systems, there are differences between the audio coding format and the video coding format. In WebRTC, the commonly used audio coding format is the OPUS coding format, while the AAC audio coding format is mostly used in non-WebRTC. For the conversion of audio c...

AI Technical Summary

Problems solved by technology

Usually, there are differences between the audio encoding format and the video encoding format in these two systems. In WebRTC, the commonly used audio encoding format is the OPUS encoding format, rather than the AAC audio encoding format used in WebRTC. For the conversion of the audio encoding format, currently FFMPEG is more commonly used, but this audio transcoding technology does not consider the impact of the frame length of the encoding format on transcoding when transcoding, although it provides a wealth of conversion functions between individual frames, audio streams in practical applications The data conversion needs to do additional processing on the basis of FFPEMG

[0003] In addition, for video data, although the video encoding in both systems adopts the H264 encoding format, the processing of video streams is also different. In WebRTC, continuous I frames, SPS frames, and PPS frames are used as the judgment of the starting frame Standard, not in webRTC, when the SPS frame and PPS frame remain unchanged, the independent I frame can be used as the sign of the start frame, and in some cases, some data frames need to be discarded in order to adapt to bad network conditions. The start frame cannot be discarded. It is necessary to be able to identify the start frame exactly before discarding it. Otherwise, the start frame may be discarded, resulting in damage to the entire streaming media data.

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