I meet a problem about creating AudioTrack failed with larger audio
output latency.
It fails in AudioTrack::set(), blocked by codes below:
if (frameCount < minFrameCount) {
LOGE("Invalid buffer size: minFrameCount %d, frameCount %d",
minFrameCount, frameCount);
return BAD_VALUE;
}
I think this code is for guaranteeing there are enough buffer and
sample in AudioTrack to wait for audio output consuming the output
data.
In MediaPlayerService::AudioOutput::open(), AudioTrack is created by
codes below:
new AudioTrack(mStreamType, sampleRate, format, channelCount,
frameCount*bufferCount);
framecount is determined by frameCount = (sampleRate*afFrameCount)/
afSampleRate; In most situation, sampleRate is usually 44100,
bufferCount is hard-code in PV(4) and SONIVOX_PLAYER/VORBIS_PLAYER
(DEFAULT_AUDIOSINK_BUFFERCOUNT 4).
So if there is a larger output latency, PV/SONIVOX_PLAYER/
VORBIS_PLAYER may not create an AudioTrack with enough bufferCount.
These may happen when output is a larger latency network device, such
as BlueTooth, etc.
Is there a way to prevent from creating AudioTrack fail under this
situation?
Thanks.
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