📄 alsaaudio.c
字号:
#if defined DEBUG_MISMATCHES || defined DEBUG if (obt->fmt != req->fmt || obt->nchannels != req->nchannels || obt->freq != req->freq) { dolog ("Audio paramters mismatch for %s\n", typ); alsa_dump_info (req, obt); }#endif#ifdef DEBUG alsa_dump_info (req, obt);#endif return 0; err: alsa_anal_close (&handle); return -1;}static int alsa_recover (snd_pcm_t *handle){ int err = snd_pcm_prepare (handle); if (err < 0) { alsa_logerr (err, "Failed to prepare handle %p\n", handle); return -1; } return 0;}static snd_pcm_sframes_t alsa_get_avail (snd_pcm_t *handle){ snd_pcm_sframes_t avail; avail = snd_pcm_avail_update (handle); if (avail < 0) { if (avail == -EPIPE) { if (!alsa_recover (handle)) { avail = snd_pcm_avail_update (handle); } } if (avail < 0) { alsa_logerr (avail, "Could not obtain number of available frames\n"); return -1; } } return avail;}static int alsa_run_out (HWVoiceOut *hw){ ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; int rpos, live, decr; int samples; uint8_t *dst; st_sample_t *src; snd_pcm_sframes_t avail; live = audio_pcm_hw_get_live_out (hw); if (!live) { return 0; } avail = alsa_get_avail (alsa->handle); if (avail < 0) { dolog ("Could not get number of available playback frames\n"); return 0; } decr = audio_MIN (live, avail); samples = decr; rpos = hw->rpos; while (samples) { int left_till_end_samples = hw->samples - rpos; int len = audio_MIN (samples, left_till_end_samples); snd_pcm_sframes_t written; src = hw->mix_buf + rpos; dst = advance (alsa->pcm_buf, rpos << hw->info.shift); hw->clip (dst, src, len); while (len) { written = snd_pcm_writei (alsa->handle, dst, len); if (written <= 0) { switch (written) { case 0: if (conf.verbose) { dolog ("Failed to write %d frames (wrote zero)\n", len); } goto exit; case -EPIPE: if (alsa_recover (alsa->handle)) { alsa_logerr (written, "Failed to write %d frames\n", len); goto exit; } if (conf.verbose) { dolog ("Recovering from playback xrun\n"); } continue; case -EAGAIN: goto exit; default: alsa_logerr (written, "Failed to write %d frames to %p\n", len, dst); goto exit; } } mixeng_clear (src, written); rpos = (rpos + written) % hw->samples; samples -= written; len -= written; dst = advance (dst, written << hw->info.shift); src += written; } } exit: hw->rpos = rpos; return decr;}static void alsa_fini_out (HWVoiceOut *hw){ ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; ldebug ("alsa_fini\n"); alsa_anal_close (&alsa->handle); if (alsa->pcm_buf) { qemu_free (alsa->pcm_buf); alsa->pcm_buf = NULL; }}static int alsa_init_out (HWVoiceOut *hw, audsettings_t *as){ ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; struct alsa_params_req req; struct alsa_params_obt obt; audfmt_e effective_fmt; int endianness; int err; snd_pcm_t *handle; audsettings_t obt_as; req.fmt = aud_to_alsafmt (as->fmt); req.freq = as->freq; req.nchannels = as->nchannels; req.period_size = conf.period_size_out; req.buffer_size = conf.buffer_size_out; if (alsa_open (0, &req, &obt, &handle)) { return -1; } err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness); if (err) { alsa_anal_close (&handle); return -1; } obt_as.freq = obt.freq; obt_as.nchannels = obt.nchannels; obt_as.fmt = effective_fmt; audio_pcm_init_info ( &hw->info, &obt_as, audio_need_to_swap_endian (endianness) ); hw->samples = obt.samples; alsa->pcm_buf = audio_calloc (AUDIO_FUNC, obt.samples, 1 << hw->info.shift); if (!alsa->pcm_buf) { dolog ("Could not allocate DAC buffer (%d samples, each %d bytes)\n", hw->samples, 1 << hw->info.shift); alsa_anal_close (&handle); return -1; } alsa->handle = handle; return 0;}static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int pause){ int err; if (pause) { err = snd_pcm_drop (handle); if (err < 0) { alsa_logerr (err, "Could not stop %s\n", typ); return -1; } } else { err = snd_pcm_prepare (handle); if (err < 0) { alsa_logerr (err, "Could not prepare handle for %s\n", typ); return -1; } } return 0;}static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...){ ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; switch (cmd) { case VOICE_ENABLE: ldebug ("enabling voice\n"); return alsa_voice_ctl (alsa->handle, "playback", 0); case VOICE_DISABLE: ldebug ("disabling voice\n"); return alsa_voice_ctl (alsa->handle, "playback", 1); } return -1;}static int alsa_init_in (HWVoiceIn *hw, audsettings_t *as){ ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; struct alsa_params_req req; struct alsa_params_obt obt; int endianness; int err; audfmt_e effective_fmt; snd_pcm_t *handle; audsettings_t obt_as; req.fmt = aud_to_alsafmt (as->fmt); req.freq = as->freq; req.nchannels = as->nchannels; req.period_size = conf.period_size_in; req.buffer_size = conf.buffer_size_in; if (alsa_open (1, &req, &obt, &handle)) { return -1; } err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness); if (err) { alsa_anal_close (&handle); return -1; } obt_as.freq = obt.freq; obt_as.nchannels = obt.nchannels; obt_as.fmt = effective_fmt; audio_pcm_init_info ( &hw->info, &obt_as, audio_need_to_swap_endian (endianness) ); hw->samples = obt.samples; alsa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift); if (!alsa->pcm_buf) { dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n", hw->samples, 1 << hw->info.shift); alsa_anal_close (&handle); return -1; } alsa->handle = handle; return 0;}static void alsa_fini_in (HWVoiceIn *hw){ ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; alsa_anal_close (&alsa->handle); if (alsa->pcm_buf) { qemu_free (alsa->pcm_buf); alsa->pcm_buf = NULL; }}static int alsa_run_in (HWVoiceIn *hw){ ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; int hwshift = hw->info.shift; int i; int live = audio_pcm_hw_get_live_in (hw); int dead = hw->samples - live; int decr; struct { int add; int len; } bufs[2] = { { hw->wpos, 0 }, { 0, 0 } }; snd_pcm_sframes_t avail; snd_pcm_uframes_t read_samples = 0; if (!dead) { return 0; } avail = alsa_get_avail (alsa->handle); if (avail < 0) { dolog ("Could not get number of captured frames\n"); return 0; } if (!avail && (snd_pcm_state (alsa->handle) == SND_PCM_STATE_PREPARED)) { avail = hw->samples; } decr = audio_MIN (dead, avail); if (!decr) { return 0; } if (hw->wpos + decr > hw->samples) { bufs[0].len = (hw->samples - hw->wpos); bufs[1].len = (decr - (hw->samples - hw->wpos)); } else { bufs[0].len = decr; } for (i = 0; i < 2; ++i) { void *src; st_sample_t *dst; snd_pcm_sframes_t nread; snd_pcm_uframes_t len; len = bufs[i].len; src = advance (alsa->pcm_buf, bufs[i].add << hwshift); dst = hw->conv_buf + bufs[i].add; while (len) { nread = snd_pcm_readi (alsa->handle, src, len); if (nread <= 0) { switch (nread) { case 0: if (conf.verbose) { dolog ("Failed to read %ld frames (read zero)\n", len); } goto exit; case -EPIPE: if (alsa_recover (alsa->handle)) { alsa_logerr (nread, "Failed to read %ld frames\n", len); goto exit; } if (conf.verbose) { dolog ("Recovering from capture xrun\n"); } continue; case -EAGAIN: goto exit; default: alsa_logerr ( nread, "Failed to read %ld frames from %p\n", len, src ); goto exit; } } hw->conv (dst, src, nread, &nominal_volume); src = advance (src, nread << hwshift); dst += nread; read_samples += nread; len -= nread; } } exit: hw->wpos = (hw->wpos + read_samples) % hw->samples; return read_samples;}static int alsa_read (SWVoiceIn *sw, void *buf, int size){ return audio_pcm_sw_read (sw, buf, size);}static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...){ ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; switch (cmd) { case VOICE_ENABLE: ldebug ("enabling voice\n"); return alsa_voice_ctl (alsa->handle, "capture", 0); case VOICE_DISABLE: ldebug ("disabling voice\n"); return alsa_voice_ctl (alsa->handle, "capture", 1); } return -1;}static void *alsa_audio_init (void){ return &conf;}static void alsa_audio_fini (void *opaque){ (void) opaque;}static struct audio_option alsa_options[] = { {"DAC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_out, "DAC period/buffer size in microseconds (otherwise in frames)", NULL, 0}, {"DAC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_out, "DAC period size", &conf.period_size_out_overriden, 0}, {"DAC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_out, "DAC buffer size", &conf.buffer_size_out_overriden, 0}, {"ADC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_in, "ADC period/buffer size in microseconds (otherwise in frames)", NULL, 0}, {"ADC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_in, "ADC period size", &conf.period_size_in_overriden, 0}, {"ADC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_in, "ADC buffer size", &conf.buffer_size_in_overriden, 0}, {"THRESHOLD", AUD_OPT_INT, &conf.threshold, "(undocumented)", NULL, 0}, {"DAC_DEV", AUD_OPT_STR, &conf.pcm_name_out, "DAC device name (for instance dmix)", NULL, 0}, {"ADC_DEV", AUD_OPT_STR, &conf.pcm_name_in, "ADC device name", NULL, 0}, {"VERBOSE", AUD_OPT_BOOL, &conf.verbose, "Behave in a more verbose way", NULL, 0}, {NULL, 0, NULL, NULL, NULL, 0}};static struct audio_pcm_ops alsa_pcm_ops = { alsa_init_out, alsa_fini_out, alsa_run_out, alsa_write, alsa_ctl_out, alsa_init_in, alsa_fini_in, alsa_run_in, alsa_read, alsa_ctl_in};struct audio_driver alsa_audio_driver = { INIT_FIELD (name = ) "alsa", INIT_FIELD (descr = ) "ALSA http://www.alsa-project.org", INIT_FIELD (options = ) alsa_options, INIT_FIELD (init = ) alsa_audio_init, INIT_FIELD (fini = ) alsa_audio_fini, INIT_FIELD (pcm_ops = ) &alsa_pcm_ops, INIT_FIELD (can_be_default = ) 1, INIT_FIELD (max_voices_out = ) INT_MAX, INIT_FIELD (max_voices_in = ) INT_MAX, INIT_FIELD (voice_size_out = ) sizeof (ALSAVoiceOut), INIT_FIELD (voice_size_in = ) sizeof (ALSAVoiceIn)};
⌨️ 快捷键说明
复制代码
Ctrl + C
搜索代码
Ctrl + F
全屏模式
F11
切换主题
Ctrl + Shift + D
显示快捷键
?
增大字号
Ctrl + =
减小字号
Ctrl + -