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

Abstract

The invention discloses a streaming media data transcoding method, which is suitable for being completed in computing equipment and comprises the following steps of: receiving source streaming media data, demultiplexing a source streaming media data stream, and decomposing the source streaming media data stream into source audio data and source video data, the source audio data being in a first audio coding format; performing audio data processing on the source audio data to convert the source audio data into target audio data, wherein the target audio data is in a second audio coding format;performing video data processing on the source video data to obtain target video data; and multiplexing the target audio data and the target video data to obtain target streaming media data. The invention further discloses a corresponding streaming media data transcoding device, computing equipment and a readable storage medium.

Description

technical field [0001] The invention relates to the field of audio and video processing, in particular to a streaming media data transcoding method, computing equipment and a 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 the source media data. For example, with the development of Internet technology, live video broadcasting, video conferencing, etc. 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 and convert non-webRTC data into WebRTC Data has become a technical problem that must be solved. 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 encod...

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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