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打印报错如下，请问使用alsa需要注意些什么？这个错误是什么导致？
<audiodev.cpp><sound_alsa_create:694>the para as follow:
<audiodev.cpp><sound_alsa_create:695>m_snd_attrs.soundStd:1
<audiodev.cpp><sound_alsa_create:696>m_snd_attrs.sampleRate:16000
<audiodev.cpp><sound_alsa_create:697>m_snd_handle->channels:2
<audiodev.cpp><sound_alsa_create:698>m_snd_handle->periodSize:512
<audiodev.cpp><sound_alsa_create:702>snd_pcm_open enter 0
<audiodev.cpp><sound_alsa_create:706>snd_pcm_open enter 1
<audiodev.cpp><sound_alsa_create:713>snd_pcm_open over
[ 586.588446] rockchip-i2s ff890000.i2s: Fail to set mclk -22
[ 586.599372] asoc-simple-card rt5651-sound: ASoC: machine hw_params failed: -22
<audiodev.cpp><_init_hwparams:496>Failed to set ALSA parameters on hw:0,0 (Invalid argument)
<audioserver.cpp><audio_init:223>sound_alsa_create error!
<audioserver.cpp><audioserver_start:282>audio_init error!
alsa初始化部分如下：
int Csound_alsa_soc::_init_hwparams(snd_pcm_t *rc, snd_pcm_hw_params_t *hwParams)
{
int my_res = 0;
snd_pcm_uframes_t bufSize = 0;
snd_pcm_uframes_t periodSize = 0;
unsigned int buffer_time = 0;
unsigned int sampleRate = 0;
ADEBUG(1, "Enter\n");
/* Fill the params with the defaults */
my_res = snd_pcm_hw_params_any(rc, hwParams);
if (my_res < 0) {
ADEBUG(0, "Failed to init ALSA hardware param structure on %s (%s)\n",
AUDIO_DEVICE, snd_strerror(my_res));
return SCERR_FAIL;
}
/* Set the access type to interleaved */
my_res = snd_pcm_hw_params_set_access(rc, hwParams,
DEF_PCM_ACCESS_MODE);//SND_PCM_ACCESS_RW_INTERLEAVED DEF_PCM_ACCESS_MODE
if (my_res < 0) {
ADEBUG(0, "Failed to set ALSA access type on %s (%s)\n",
AUDIO_DEVICE, snd_strerror(my_res));
return SCERR_FAIL;
}
/* Set the format to 16 bit little endian */
my_res = snd_pcm_hw_params_set_format(rc, hwParams, DEF_PCM_FORMAT);
if (my_res < 0) {
ADEBUG(0, "Failed to set sample format on %s (%s)\n",
AUDIO_DEVICE, snd_strerror(my_res));
return SCERR_FAIL;
}
/* Set the sample rate */
sampleRate = m_snd_attrs.sampleRate;
my_res = snd_pcm_hw_params_set_rate_near(rc, hwParams,
&sampleRate, 0);
if (my_res < 0) {
ADEBUG(0, "Failed to set sample rate to %d on %s (%s)\n",
m_snd_attrs.sampleRate, AUDIO_DEVICE, snd_strerror(my_res));
return SCERR_FAIL;
}
/* Set the number of channels to 2 */
my_res = snd_pcm_hw_params_set_channels(rc, hwParams, m_snd_attrs.channels);
if (my_res < 0) {
ADEBUG(0, "Failed to set channel count to %d on %s (%s)\n",
m_snd_attrs.channels, AUDIO_DEVICE, snd_strerror(my_res));
return SCERR_FAIL;
}
/* Set period size equal to input buffer size */
m_snd_handle->periodSize = 256;
periodSize = m_snd_handle->periodSize;
my_res = snd_pcm_hw_params_set_period_size_near (rc, hwParams,
&periodSize, 0);
if (my_res < 0) {
ADEBUG(0, "Failed to set the period size to %ld on %s (%s)\n",
periodSize, AUDIO_DEVICE, snd_strerror(my_res));
return SCERR_FAIL;
}
if (periodSize != m_snd_handle->periodSize) {
ADEBUG(0, "Failed to set period size (%ld) , got (%ld) on %s (%s)\n",
m_snd_handle->periodSize, periodSize, AUDIO_DEVICE,
snd_strerror(my_res));
m_snd_handle->periodSize = periodSize;
}
/* Set the maximum buffer time */
my_res = snd_pcm_hw_params_get_buffer_time_max(hwParams, &buffer_time, 0);
ADEBUG(3, "maximum buffer time<%d>\n", buffer_time);
if (my_res < 0) {
ADEBUG(0, "Failed to get maximum buffer time on %s (%s)\n",
AUDIO_DEVICE, snd_strerror(my_res));
return SCERR_FAIL;
}
if (buffer_time > DEF_BUF_TIME) {
buffer_time = DEF_BUF_TIME;
}
my_res = snd_pcm_hw_params_set_buffer_time_near(rc, hwParams,
&buffer_time, 0);
if (my_res < 0) {
ADEBUG(0, "Failed to set maximum buffer time on %s (%s)\n",
AUDIO_DEVICE, snd_strerror(my_res));
return SCERR_FAIL;
}
/*! check hardware can support pause operation or not */
if (snd_pcm_hw_params_can_pause(hwParams) == 1)
{
ADEBUG(3, "hardware do support pause operation\n");
}else{
ADEBUG(3, "hardware not support pause operation\n");
}
。。。
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