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1/ I would like to know if I am computing (with rough precision) the bandwidth for MP3/AAC audio stream correctly. Parameters of the stream are:

Sample rate: 44100,  
Bitrate: 128bit
Channels: 2,
Stream type: icecast (no metadata)
Codecs used: MP3 or AAC

The raw audio would be: 44100*128*2 = 11289600 bits/sec = 11025 kbit/sec = ~10.8 MBit/sec. I will count with 10:1 compression ratio (which I think can be more or less correct - but please correct me, if I am wrong) then approx. 1 Megabit/sec should be enough.

2/ Is the computation correct? Because this means that for GPRS (~80 kbps), EDGE (~230kbps), UMTS (~384kbps) it does not make sense to even try those streams. If my computation is correct, then the only technology would be then the HDSPA, which starts on ~1.8 Mbps.

3/ Can someone give me good parameters for the streaming for the mobile audio streaming?

Thanks a lot BR STeN

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您已经弄清楚了第一部分，音频实际上是 16 位的，并且这里指定的比特率是针对压缩器的。

第二部分是选择适合移动流媒体的比特率。在我的测试中，我发现你并不想超过 96kbit，但 64kbit 通常是可以接受的。

要在 64kbit 上获得不错的音频质量，您可以将流编码为单声道。如果您更喜欢立体声，您将在此比特率下获得许多压缩伪像。那时只推荐使用 AAC，它在低比特率下效果很好。

至少在我所在的地区，带宽无法可靠地做任何更高的事情，这是在 EvDO 上。

于 2011-05-24T13:14:47.673 回答