diff options
Diffstat (limited to 'audio/alsaaudio.c')
| -rw-r--r-- | audio/alsaaudio.c | 974 |
1 files changed, 974 insertions, 0 deletions
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c new file mode 100644 index 0000000..71e5235 --- /dev/null +++ b/audio/alsaaudio.c @@ -0,0 +1,974 @@ +/* + * QEMU ALSA audio driver + * + * Copyright (c) 2005 Vassili Karpov (malc) + * + * Permission is hereby granted, free of charge, to any person obtaining a copy + * of this software and associated documentation files (the "Software"), to deal + * in the Software without restriction, including without limitation the rights + * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the Software is + * furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN + * THE SOFTWARE. + */ +#include <alsa/asoundlib.h> +#include "vl.h" + +#define AUDIO_CAP "alsa" +#include "audio_int.h" + +typedef struct ALSAVoiceOut { + HWVoiceOut hw; + void *pcm_buf; + snd_pcm_t *handle; +} ALSAVoiceOut; + +typedef struct ALSAVoiceIn { + HWVoiceIn hw; + snd_pcm_t *handle; + void *pcm_buf; +} ALSAVoiceIn; + +static struct { + int size_in_usec_in; + int size_in_usec_out; + const char *pcm_name_in; + const char *pcm_name_out; + unsigned int buffer_size_in; + unsigned int period_size_in; + unsigned int buffer_size_out; + unsigned int period_size_out; + unsigned int threshold; + + int buffer_size_in_overriden; + int period_size_in_overriden; + + int buffer_size_out_overriden; + int period_size_out_overriden; + int verbose; +} conf = { +#ifdef HIGH_LATENCY + .size_in_usec_in = 1, + .size_in_usec_out = 1, +#endif + .pcm_name_out = "default", + .pcm_name_in = "default", +#ifdef HIGH_LATENCY + .buffer_size_in = 400000, + .period_size_in = 400000 / 4, + .buffer_size_out = 400000, + .period_size_out = 400000 / 4, +#else +#define DEFAULT_BUFFER_SIZE 1024 +#define DEFAULT_PERIOD_SIZE 256 + .buffer_size_in = DEFAULT_BUFFER_SIZE * 4, + .period_size_in = DEFAULT_PERIOD_SIZE * 4, + .buffer_size_out = DEFAULT_BUFFER_SIZE, + .period_size_out = DEFAULT_PERIOD_SIZE, + .buffer_size_in_overriden = 0, + .buffer_size_out_overriden = 0, + .period_size_in_overriden = 0, + .period_size_out_overriden = 0, +#endif + .threshold = 0, + .verbose = 0 +}; + +struct alsa_params_req { + int freq; + audfmt_e fmt; + int nchannels; + unsigned int buffer_size; + unsigned int period_size; +}; + +struct alsa_params_obt { + int freq; + audfmt_e fmt; + int nchannels; + snd_pcm_uframes_t samples; +}; + +static void GCC_FMT_ATTR (2, 3) alsa_logerr (int err, const char *fmt, ...) +{ + va_list ap; + + va_start (ap, fmt); + AUD_vlog (AUDIO_CAP, fmt, ap); + va_end (ap); + + AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err)); +} + +static void GCC_FMT_ATTR (3, 4) alsa_logerr2 ( + int err, + const char *typ, + const char *fmt, + ... + ) +{ + va_list ap; + + AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ); + + va_start (ap, fmt); + AUD_vlog (AUDIO_CAP, fmt, ap); + va_end (ap); + + AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err)); +} + +static void alsa_anal_close (snd_pcm_t **handlep) +{ + int err = snd_pcm_close (*handlep); + if (err) { + alsa_logerr (err, "Failed to close PCM handle %p\n", *handlep); + } + *handlep = NULL; +} + +static int alsa_write (SWVoiceOut *sw, void *buf, int len) +{ + return audio_pcm_sw_write (sw, buf, len); +} + +static int aud_to_alsafmt (audfmt_e fmt) +{ + switch (fmt) { + case AUD_FMT_S8: + return SND_PCM_FORMAT_S8; + + case AUD_FMT_U8: + return SND_PCM_FORMAT_U8; + + case AUD_FMT_S16: + return SND_PCM_FORMAT_S16_LE; + + case AUD_FMT_U16: + return SND_PCM_FORMAT_U16_LE; + + default: + dolog ("Internal logic error: Bad audio format %d\n", fmt); +#ifdef DEBUG_AUDIO + abort (); +#endif + return SND_PCM_FORMAT_U8; + } +} + +static int alsa_to_audfmt (int alsafmt, audfmt_e *fmt, int *endianness) +{ + switch (alsafmt) { + case SND_PCM_FORMAT_S8: + *endianness = 0; + *fmt = AUD_FMT_S8; + break; + + case SND_PCM_FORMAT_U8: + *endianness = 0; + *fmt = AUD_FMT_U8; + break; + + case SND_PCM_FORMAT_S16_LE: + *endianness = 0; + *fmt = AUD_FMT_S16; + break; + + case SND_PCM_FORMAT_U16_LE: + *endianness = 0; + *fmt = AUD_FMT_U16; + break; + + case SND_PCM_FORMAT_S16_BE: + *endianness = 1; + *fmt = AUD_FMT_S16; + break; + + case SND_PCM_FORMAT_U16_BE: + *endianness = 1; + *fmt = AUD_FMT_U16; + break; + + default: + dolog ("Unrecognized audio format %d\n", alsafmt); + return -1; + } + + return 0; +} + +#if defined DEBUG_MISMATCHES || defined DEBUG +static void alsa_dump_info (struct alsa_params_req *req, + struct alsa_params_obt *obt) +{ + dolog ("parameter | requested value | obtained value\n"); + dolog ("format | %10d | %10d\n", req->fmt, obt->fmt); + dolog ("channels | %10d | %10d\n", + req->nchannels, obt->nchannels); + dolog ("frequency | %10d | %10d\n", req->freq, obt->freq); + dolog ("============================================\n"); + dolog ("requested: buffer size %d period size %d\n", + req->buffer_size, req->period_size); + dolog ("obtained: samples %ld\n", obt->samples); +} +#endif + +static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold) +{ + int err; + snd_pcm_sw_params_t *sw_params; + + snd_pcm_sw_params_alloca (&sw_params); + + err = snd_pcm_sw_params_current (handle, sw_params); + if (err < 0) { + dolog ("Could not fully initialize DAC\n"); + alsa_logerr (err, "Failed to get current software parameters\n"); + return; + } + + err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold); + if (err < 0) { + dolog ("Could not fully initialize DAC\n"); + alsa_logerr (err, "Failed to set software threshold to %ld\n", + threshold); + return; + } + + err = snd_pcm_sw_params (handle, sw_params); + if (err < 0) { + dolog ("Could not fully initialize DAC\n"); + alsa_logerr (err, "Failed to set software parameters\n"); + return; + } +} + +static int alsa_open (int in, struct alsa_params_req *req, + struct alsa_params_obt *obt, snd_pcm_t **handlep) +{ + snd_pcm_t *handle; + snd_pcm_hw_params_t *hw_params; + int err, freq, nchannels; + const char *pcm_name = in ? conf.pcm_name_in : conf.pcm_name_out; + unsigned int period_size, buffer_size; + snd_pcm_uframes_t obt_buffer_size; + const char *typ = in ? "ADC" : "DAC"; + + freq = req->freq; + period_size = req->period_size; + buffer_size = req->buffer_size; + nchannels = req->nchannels; + + snd_pcm_hw_params_alloca (&hw_params); + + err = snd_pcm_open ( + &handle, + pcm_name, + in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK, + SND_PCM_NONBLOCK + ); + if (err < 0) { + alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name); + return -1; + } + + err = snd_pcm_hw_params_any (handle, hw_params); + if (err < 0) { + alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n"); + goto err; + } + + err = snd_pcm_hw_params_set_access ( + handle, + hw_params, + SND_PCM_ACCESS_RW_INTERLEAVED + ); + if (err < 0) { + alsa_logerr2 (err, typ, "Failed to set access type\n"); + goto err; + } + + err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt); + if (err < 0) { + alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt); + goto err; + } + + err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0); + if (err < 0) { + alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq); + goto err; + } + + err = snd_pcm_hw_params_set_channels_near ( + handle, + hw_params, + &nchannels + ); + if (err < 0) { + alsa_logerr2 (err, typ, "Failed to set number of channels %d\n", + req->nchannels); + goto err; + } + + if (nchannels != 1 && nchannels != 2) { + alsa_logerr2 (err, typ, + "Can not handle obtained number of channels %d\n", + nchannels); + goto err; + } + + if (!((in && conf.size_in_usec_in) || (!in && conf.size_in_usec_out))) { + if (!buffer_size) { + buffer_size = DEFAULT_BUFFER_SIZE; + period_size= DEFAULT_PERIOD_SIZE; + } + } + + if (buffer_size) { + if ((in && conf.size_in_usec_in) || (!in && conf.size_in_usec_out)) { + if (period_size) { + err = snd_pcm_hw_params_set_period_time_near ( + handle, + hw_params, + &period_size, + 0 + ); + if (err < 0) { + alsa_logerr2 (err, typ, + "Failed to set period time %d\n", + req->period_size); + goto err; + } + } + + err = snd_pcm_hw_params_set_buffer_time_near ( + handle, + hw_params, + &buffer_size, + 0 + ); + + if (err < 0) { + alsa_logerr2 (err, typ, + "Failed to set buffer time %d\n", + req->buffer_size); + goto err; + } + } + else { + int dir; + snd_pcm_uframes_t minval; + + if (period_size) { + minval = period_size; + dir = 0; + + err = snd_pcm_hw_params_get_period_size_min ( + hw_params, + &minval, + &dir + ); + if (err < 0) { + alsa_logerr ( + err, + "Could not get minmal period size for %s\n", + typ + ); + } + else { + if (period_size < minval) { + if ((in && conf.period_size_in_overriden) + || (!in && conf.period_size_out_overriden)) { + dolog ("%s period size(%d) is less " + "than minmal period size(%ld)\n", + typ, + period_size, + minval); + } + period_size = minval; + } + } + + err = snd_pcm_hw_params_set_period_size ( + handle, + hw_params, + period_size, + 0 + ); + if (err < 0) { + alsa_logerr2 (err, typ, "Failed to set period size %d\n", + req->period_size); + goto err; + } + } + + minval = buffer_size; + err = snd_pcm_hw_params_get_buffer_size_min ( + hw_params, + &minval + ); + if (err < 0) { + alsa_logerr (err, "Could not get minmal buffer size for %s\n", + typ); + } + else { + if (buffer_size < minval) { + if ((in && conf.buffer_size_in_overriden) + || (!in && conf.buffer_size_out_overriden)) { + dolog ( + "%s buffer size(%d) is less " + "than minimal buffer size(%ld)\n", + typ, + buffer_size, + minval + ); + } + buffer_size = minval; + } + } + + err = snd_pcm_hw_params_set_buffer_size ( + handle, + hw_params, + buffer_size + ); + if (err < 0) { + alsa_logerr2 (err, typ, "Failed to set buffer size %d\n", + req->buffer_size); + goto err; + } + } + } + else { + dolog ("warning: Buffer size is not set\n"); + } + + err = snd_pcm_hw_params (handle, hw_params); + if (err < 0) { + alsa_logerr2 (err, typ, "Failed to apply audio parameters\n"); + goto err; + } + + err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size); + if (err < 0) { + alsa_logerr2 (err, typ, "Failed to get buffer size\n"); + goto err; + } + + err = snd_pcm_prepare (handle); + if (err < 0) { + alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle); + goto err; + } + + if (!in && conf.threshold) { + snd_pcm_uframes_t threshold; + int bytes_per_sec; + + bytes_per_sec = freq + << (nchannels == 2) + << (req->fmt == AUD_FMT_S16 || req->fmt == AUD_FMT_U16); + + threshold = (conf.threshold * bytes_per_sec) / 1000; + alsa_set_threshold (handle, threshold); + } + + obt->fmt = req->fmt; + obt->nchannels = nchannels; + obt->freq = freq; + obt->samples = obt_buffer_size; + *handlep = handle; + +#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; + } + } + + 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; + obt_as.endianness = endianness; + + audio_pcm_init_info (&hw->info, &obt_as); + 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; + obt_as.endianness = endianness; + + audio_pcm_init_info (&hw->info, &obt_as); + 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) +}; |