由于ffmpeg最新版本(从2.1开始貌似)使用avcodec_decode_audio4函数来解码音频,但解码得到的数据类型为float 4bit,而播放器播放的格式一般为S16(signed 16bit),就需要对解码得到的数据进行转换,然而,ffmpeg已经帮我们做好了,只需调用API就可以了,这个函数就是:swr_convert
以下为从ffmpeg官网的复制:
原型:int swr_convert(struct SwrContext * s, uint8_t ** out, int out_count, const uint8_t ** in, int in_count )
Convert audio.
in and in_count can be set to 0 to flush the last few samples out at the end.
If more input is provided than output space then the input will be buffered. You can avoid this buffering by providing more output space than input.
Convertion will run directly without copying whenever possible.
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Parameters
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s allocated Swr context, with parameters set out output buffers, only the first one need be set in case of packed audio out_count amount of space available for output in samples per channel in input buffers, only the first one need to be set in case of packed audio in_count number of input samples available in one channel
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Returns
- number of samples output per channel, negative value on error