Linux ALSA：防止扬声器欠载的方法
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ALSA: Ways to prevent underrun for speaker
提问by Lunar Mushrooms
I am playing a single channel audio in non-interleaved mode. I am getting underrun when I am writing audio data into speaker : ALSA lib pcm.c:7339:(snd_pcm_recover) underrun occurred
我正在以非交错模式播放单声道音频。当我将音频数据写入扬声器时，我的速度不足：ALSA lib pcm.c:7339:(snd_pcm_recover) underrun occurred
Here is how I write:
这是我的写作方式：
printf("%d",snd_pcm_avail (spkhandle));
ret = snd_pcm_writen(spkhandle, pSpeakerBuf , framesIn18Millisec);
if(ret < 0)
{
snd_pcm_recover(spkhandle, ret, 0);
}
What are the different ways/parameter configurations to prevent ALSA under run ?
防止 ALSA 运行的不同方式/参数配置有哪些？
(I am using Linux 3.0, ARM )
（我使用的是 Linux 3.0，ARM）
Edit: Here is a buffer measurement using snd_pcm_avail() API
编辑：这是使用 snd_pcm_avail() API 的缓冲区测量
snd_pcm_avail = 2304 << snd_pcm_writen call 1 success
snd_pcm_avail = 2160 << snd_pcm_writen call 2 success
snd_pcm_avail = 2016 << snd_pcm_writen call 3 success
snd_pcm_writen error -32 Broken pipe << snd_pcm_writen call 4 failure
ALSA lib pcm.c:7339:(snd_pcm_recover) underrun occurred << And displays this message
Here is the output that Marko requested for:
这是 Marko 要求的输出：
snd_output_t* out;
....
// Do alsa parameters init ....
....
snd_output_stdio_attach(&out, stderr, 0);
snd_pcm_dump_sw_setup(spkhandle, out);
tstamp_mode : NONE
period_step : 1
avail_min : 144
period_event : 0
start_threshold : 288
stop_threshold : 2304
silence_threshold: 0
silence_size : 0
boundary : 1207959552
采纳答案by marko
I assume this code runs in a tight loop and is intended to block on snd_pcm_writen(). The sample-rate isn't given; I assume 48kHz since the numbers all divide nicely.
我假设这段代码在一个紧密的循环中运行，并且旨在阻止snd_pcm_writen(). 没有给出采样率；我假设 48kHz，因为数字都很好地划分。
What I think is going here is as follows:
我认为这里的情况如下：
snd_pcm_write()doesn't guarantee to write all frames provided (the return value is only ever checked for error conditions). Judging from the logging ofsnd_pcm_avail()it's in fact consumingavail_minor144frames on each. This is 3ms of audio.- Assuming that audio is not running at this point, after two writes, the number of frames in the buffer is equal to
start_threshold- at288samples; audio output starts - calls to
printf()block, and I seem to remember thatsnd_pcm_avail()has to synchronise with the audio output hardware and might also block. Since you are now 6ms ahead of the playback, it's entirely possible that the buffer is running dry during the time of the third call ofsnd_pcm_writen()
snd_pcm_write()不保证写入提供的所有帧（返回值只检查错误情况）。从snd_pcm_avail()它的日志记录来看，它实际上是消耗avail_min或144每个帧。这是 3 毫秒的音频。- 假设音频不会在这一点上运行时，两个写之后，在缓冲器中的帧的数目等于
start_threshold-在288样本; 音频输出开始 - 调用
printf()阻塞，我似乎记得它snd_pcm_avail()必须与音频输出硬件同步，也可能阻塞。由于您现在比回放提前 6 毫秒，因此在第三次调用snd_pcm_writen()
In summary, you shouldn't be calling printf()at this point, and you probably need to compensate for fact that snd_pcm_writen()isn't consuming all of the frames in pSpeakerBuf
总之，此时您不应该调用printf()，并且您可能需要补偿snd_pcm_writen()没有消耗所有帧的事实pSpeakerBuf
回答by Rahul
It is the buffer underrun,You can try increasing the buffer size by explicitly mentioning it in your ~/.asoundrc file ?
这是缓冲区不足，您可以尝试通过在 ~/.asoundrc 文件中明确提及来增加缓冲区大小？
