diff options
Diffstat (limited to 'qemu/audio/alsaaudio.c')
| -rw-r--r-- | qemu/audio/alsaaudio.c | 1227 |
1 files changed, 1227 insertions, 0 deletions
diff --git a/qemu/audio/alsaaudio.c b/qemu/audio/alsaaudio.c new file mode 100644 index 000000000..6315b2d74 --- /dev/null +++ b/qemu/audio/alsaaudio.c @@ -0,0 +1,1227 @@ +/* + * 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 "qemu-common.h" +#include "qemu/main-loop.h" +#include "audio.h" +#include "trace.h" + +#if QEMU_GNUC_PREREQ(4, 3) +#pragma GCC diagnostic ignored "-Waddress" +#endif + +#define AUDIO_CAP "alsa" +#include "audio_int.h" + +typedef struct ALSAConf { + 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_overridden; + int period_size_in_overridden; + + int buffer_size_out_overridden; + int period_size_out_overridden; +} ALSAConf; + +struct pollhlp { + snd_pcm_t *handle; + struct pollfd *pfds; + ALSAConf *conf; + int count; + int mask; +}; + +typedef struct ALSAVoiceOut { + HWVoiceOut hw; + int wpos; + int pending; + void *pcm_buf; + snd_pcm_t *handle; + struct pollhlp pollhlp; +} ALSAVoiceOut; + +typedef struct ALSAVoiceIn { + HWVoiceIn hw; + snd_pcm_t *handle; + void *pcm_buf; + struct pollhlp pollhlp; +} ALSAVoiceIn; + +struct alsa_params_req { + int freq; + snd_pcm_format_t fmt; + int nchannels; + int size_in_usec; + int override_mask; + unsigned int buffer_size; + unsigned int period_size; +}; + +struct alsa_params_obt { + int freq; + audfmt_e fmt; + int endianness; + 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_fini_poll (struct pollhlp *hlp) +{ + int i; + struct pollfd *pfds = hlp->pfds; + + if (pfds) { + for (i = 0; i < hlp->count; ++i) { + qemu_set_fd_handler (pfds[i].fd, NULL, NULL, NULL); + } + g_free (pfds); + } + hlp->pfds = NULL; + hlp->count = 0; + hlp->handle = NULL; +} + +static void alsa_anal_close1 (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 void alsa_anal_close (snd_pcm_t **handlep, struct pollhlp *hlp) +{ + alsa_fini_poll (hlp); + alsa_anal_close1 (handlep); +} + +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 int alsa_resume (snd_pcm_t *handle) +{ + int err = snd_pcm_resume (handle); + if (err < 0) { + alsa_logerr (err, "Failed to resume handle %p\n", handle); + return -1; + } + return 0; +} + +static void alsa_poll_handler (void *opaque) +{ + int err, count; + snd_pcm_state_t state; + struct pollhlp *hlp = opaque; + unsigned short revents; + + count = poll (hlp->pfds, hlp->count, 0); + if (count < 0) { + dolog ("alsa_poll_handler: poll %s\n", strerror (errno)); + return; + } + + if (!count) { + return; + } + + /* XXX: ALSA example uses initial count, not the one returned by + poll, correct? */ + err = snd_pcm_poll_descriptors_revents (hlp->handle, hlp->pfds, + hlp->count, &revents); + if (err < 0) { + alsa_logerr (err, "snd_pcm_poll_descriptors_revents"); + return; + } + + if (!(revents & hlp->mask)) { + trace_alsa_revents(revents); + return; + } + + state = snd_pcm_state (hlp->handle); + switch (state) { + case SND_PCM_STATE_SETUP: + alsa_recover (hlp->handle); + break; + + case SND_PCM_STATE_XRUN: + alsa_recover (hlp->handle); + break; + + case SND_PCM_STATE_SUSPENDED: + alsa_resume (hlp->handle); + break; + + case SND_PCM_STATE_PREPARED: + audio_run ("alsa run (prepared)"); + break; + + case SND_PCM_STATE_RUNNING: + audio_run ("alsa run (running)"); + break; + + default: + dolog ("Unexpected state %d\n", state); + } +} + +static int alsa_poll_helper (snd_pcm_t *handle, struct pollhlp *hlp, int mask) +{ + int i, count, err; + struct pollfd *pfds; + + count = snd_pcm_poll_descriptors_count (handle); + if (count <= 0) { + dolog ("Could not initialize poll mode\n" + "Invalid number of poll descriptors %d\n", count); + return -1; + } + + pfds = audio_calloc ("alsa_poll_helper", count, sizeof (*pfds)); + if (!pfds) { + dolog ("Could not initialize poll mode\n"); + return -1; + } + + err = snd_pcm_poll_descriptors (handle, pfds, count); + if (err < 0) { + alsa_logerr (err, "Could not initialize poll mode\n" + "Could not obtain poll descriptors\n"); + g_free (pfds); + return -1; + } + + for (i = 0; i < count; ++i) { + if (pfds[i].events & POLLIN) { + qemu_set_fd_handler (pfds[i].fd, alsa_poll_handler, NULL, hlp); + } + if (pfds[i].events & POLLOUT) { + trace_alsa_pollout(i, pfds[i].fd); + qemu_set_fd_handler (pfds[i].fd, NULL, alsa_poll_handler, hlp); + } + trace_alsa_set_handler(pfds[i].events, i, pfds[i].fd, err); + + } + hlp->pfds = pfds; + hlp->count = count; + hlp->handle = handle; + hlp->mask = mask; + return 0; +} + +static int alsa_poll_out (HWVoiceOut *hw) +{ + ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; + + return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLOUT); +} + +static int alsa_poll_in (HWVoiceIn *hw) +{ + ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; + + return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLIN); +} + +static int alsa_write (SWVoiceOut *sw, void *buf, int len) +{ + return audio_pcm_sw_write (sw, buf, len); +} + +static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt, int endianness) +{ + switch (fmt) { + case AUD_FMT_S8: + return SND_PCM_FORMAT_S8; + + case AUD_FMT_U8: + return SND_PCM_FORMAT_U8; + + case AUD_FMT_S16: + if (endianness) { + return SND_PCM_FORMAT_S16_BE; + } + else { + return SND_PCM_FORMAT_S16_LE; + } + + case AUD_FMT_U16: + if (endianness) { + return SND_PCM_FORMAT_U16_BE; + } + else { + return SND_PCM_FORMAT_U16_LE; + } + + case AUD_FMT_S32: + if (endianness) { + return SND_PCM_FORMAT_S32_BE; + } + else { + return SND_PCM_FORMAT_S32_LE; + } + + case AUD_FMT_U32: + if (endianness) { + return SND_PCM_FORMAT_U32_BE; + } + else { + return SND_PCM_FORMAT_U32_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 (snd_pcm_format_t 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; + + case SND_PCM_FORMAT_S32_LE: + *endianness = 0; + *fmt = AUD_FMT_S32; + break; + + case SND_PCM_FORMAT_U32_LE: + *endianness = 0; + *fmt = AUD_FMT_U32; + break; + + case SND_PCM_FORMAT_S32_BE: + *endianness = 1; + *fmt = AUD_FMT_S32; + break; + + case SND_PCM_FORMAT_U32_BE: + *endianness = 1; + *fmt = AUD_FMT_U32; + break; + + default: + dolog ("Unrecognized audio format %d\n", alsafmt); + return -1; + } + + return 0; +} + +static void alsa_dump_info (struct alsa_params_req *req, + struct alsa_params_obt *obt, + snd_pcm_format_t obtfmt) +{ + dolog ("parameter | requested value | obtained value\n"); + dolog ("format | %10d | %10d\n", req->fmt, obtfmt); + 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); +} + +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, + ALSAConf *conf) +{ + snd_pcm_t *handle; + snd_pcm_hw_params_t *hw_params; + int err; + int size_in_usec; + unsigned int freq, nchannels; + const char *pcm_name = in ? conf->pcm_name_in : conf->pcm_name_out; + snd_pcm_uframes_t obt_buffer_size; + const char *typ = in ? "ADC" : "DAC"; + snd_pcm_format_t obtfmt; + + freq = req->freq; + nchannels = req->nchannels; + size_in_usec = req->size_in_usec; + + 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); + } + + 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 (req->buffer_size) { + unsigned long obt; + + if (size_in_usec) { + int dir = 0; + unsigned int btime = req->buffer_size; + + err = snd_pcm_hw_params_set_buffer_time_near ( + handle, + hw_params, + &btime, + &dir + ); + obt = btime; + } + else { + snd_pcm_uframes_t bsize = req->buffer_size; + + err = snd_pcm_hw_params_set_buffer_size_near ( + handle, + hw_params, + &bsize + ); + obt = bsize; + } + if (err < 0) { + alsa_logerr2 (err, typ, "Failed to set buffer %s to %d\n", + size_in_usec ? "time" : "size", req->buffer_size); + goto err; + } + + if ((req->override_mask & 2) && (obt - req->buffer_size)) + dolog ("Requested buffer %s %u was rejected, using %lu\n", + size_in_usec ? "time" : "size", req->buffer_size, obt); + } + + if (req->period_size) { + unsigned long obt; + + if (size_in_usec) { + int dir = 0; + unsigned int ptime = req->period_size; + + err = snd_pcm_hw_params_set_period_time_near ( + handle, + hw_params, + &ptime, + &dir + ); + obt = ptime; + } + else { + int dir = 0; + snd_pcm_uframes_t psize = req->period_size; + + err = snd_pcm_hw_params_set_period_size_near ( + handle, + hw_params, + &psize, + &dir + ); + obt = psize; + } + + if (err < 0) { + alsa_logerr2 (err, typ, "Failed to set period %s to %d\n", + size_in_usec ? "time" : "size", req->period_size); + goto err; + } + + if (((req->override_mask & 1) && (obt - req->period_size))) + dolog ("Requested period %s %u was rejected, using %lu\n", + size_in_usec ? "time" : "size", req->period_size, obt); + } + + 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_hw_params_get_format (hw_params, &obtfmt); + if (err < 0) { + alsa_logerr2 (err, typ, "Failed to get format\n"); + goto err; + } + + if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) { + dolog ("Invalid format was returned %d\n", obtfmt); + 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); + + switch (obt->fmt) { + case AUD_FMT_S8: + case AUD_FMT_U8: + break; + + case AUD_FMT_S16: + case AUD_FMT_U16: + bytes_per_sec <<= 1; + break; + + case AUD_FMT_S32: + case AUD_FMT_U32: + bytes_per_sec <<= 2; + break; + } + + threshold = (conf->threshold * bytes_per_sec) / 1000; + alsa_set_threshold (handle, threshold); + } + + obt->nchannels = nchannels; + obt->freq = freq; + obt->samples = obt_buffer_size; + + *handlep = handle; + + if (obtfmt != req->fmt || + obt->nchannels != req->nchannels || + obt->freq != req->freq) { + dolog ("Audio parameters for %s\n", typ); + alsa_dump_info (req, obt, obtfmt); + } + +#ifdef DEBUG + alsa_dump_info (req, obt, obtfmt); +#endif + return 0; + + err: + alsa_anal_close1 (&handle); + return -1; +} + +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 void alsa_write_pending (ALSAVoiceOut *alsa) +{ + HWVoiceOut *hw = &alsa->hw; + + while (alsa->pending) { + int left_till_end_samples = hw->samples - alsa->wpos; + int len = audio_MIN (alsa->pending, left_till_end_samples); + char *src = advance (alsa->pcm_buf, alsa->wpos << hw->info.shift); + + while (len) { + snd_pcm_sframes_t written; + + written = snd_pcm_writei (alsa->handle, src, len); + + if (written <= 0) { + switch (written) { + case 0: + trace_alsa_wrote_zero(len); + return; + + case -EPIPE: + if (alsa_recover (alsa->handle)) { + alsa_logerr (written, "Failed to write %d frames\n", + len); + return; + } + trace_alsa_xrun_out(); + continue; + + case -ESTRPIPE: + /* stream is suspended and waiting for an + application recovery */ + if (alsa_resume (alsa->handle)) { + alsa_logerr (written, "Failed to write %d frames\n", + len); + return; + } + trace_alsa_resume_out(); + continue; + + case -EAGAIN: + return; + + default: + alsa_logerr (written, "Failed to write %d frames from %p\n", + len, src); + return; + } + } + + alsa->wpos = (alsa->wpos + written) % hw->samples; + alsa->pending -= written; + len -= written; + } + } +} + +static int alsa_run_out (HWVoiceOut *hw, int live) +{ + ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; + int decr; + snd_pcm_sframes_t avail; + + 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); + decr = audio_pcm_hw_clip_out (hw, alsa->pcm_buf, decr, alsa->pending); + alsa->pending += decr; + alsa_write_pending (alsa); + return decr; +} + +static void alsa_fini_out (HWVoiceOut *hw) +{ + ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; + + ldebug ("alsa_fini\n"); + alsa_anal_close (&alsa->handle, &alsa->pollhlp); + + g_free(alsa->pcm_buf); + alsa->pcm_buf = NULL; +} + +static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as, + void *drv_opaque) +{ + ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; + struct alsa_params_req req; + struct alsa_params_obt obt; + snd_pcm_t *handle; + struct audsettings obt_as; + ALSAConf *conf = drv_opaque; + + req.fmt = aud_to_alsafmt (as->fmt, as->endianness); + req.freq = as->freq; + req.nchannels = as->nchannels; + req.period_size = conf->period_size_out; + req.buffer_size = conf->buffer_size_out; + req.size_in_usec = conf->size_in_usec_out; + req.override_mask = + (conf->period_size_out_overridden ? 1 : 0) | + (conf->buffer_size_out_overridden ? 2 : 0); + + if (alsa_open (0, &req, &obt, &handle, conf)) { + return -1; + } + + obt_as.freq = obt.freq; + obt_as.nchannels = obt.nchannels; + obt_as.fmt = obt.fmt; + obt_as.endianness = obt.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_close1 (&handle); + return -1; + } + + alsa->handle = handle; + alsa->pollhlp.conf = conf; + return 0; +} + +#define VOICE_CTL_PAUSE 0 +#define VOICE_CTL_PREPARE 1 +#define VOICE_CTL_START 2 + +static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int ctl) +{ + int err; + + if (ctl == VOICE_CTL_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; + } + if (ctl == VOICE_CTL_START) { + err = snd_pcm_start(handle); + if (err < 0) { + alsa_logerr (err, "Could not start 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: + { + va_list ap; + int poll_mode; + + va_start (ap, cmd); + poll_mode = va_arg (ap, int); + va_end (ap); + + ldebug ("enabling voice\n"); + if (poll_mode && alsa_poll_out (hw)) { + poll_mode = 0; + } + hw->poll_mode = poll_mode; + return alsa_voice_ctl (alsa->handle, "playback", VOICE_CTL_PREPARE); + } + + case VOICE_DISABLE: + ldebug ("disabling voice\n"); + if (hw->poll_mode) { + hw->poll_mode = 0; + alsa_fini_poll (&alsa->pollhlp); + } + return alsa_voice_ctl (alsa->handle, "playback", VOICE_CTL_PAUSE); + } + + return -1; +} + +static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque) +{ + ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; + struct alsa_params_req req; + struct alsa_params_obt obt; + snd_pcm_t *handle; + struct audsettings obt_as; + ALSAConf *conf = drv_opaque; + + req.fmt = aud_to_alsafmt (as->fmt, as->endianness); + req.freq = as->freq; + req.nchannels = as->nchannels; + req.period_size = conf->period_size_in; + req.buffer_size = conf->buffer_size_in; + req.size_in_usec = conf->size_in_usec_in; + req.override_mask = + (conf->period_size_in_overridden ? 1 : 0) | + (conf->buffer_size_in_overridden ? 2 : 0); + + if (alsa_open (1, &req, &obt, &handle, conf)) { + return -1; + } + + obt_as.freq = obt.freq; + obt_as.nchannels = obt.nchannels; + obt_as.fmt = obt.fmt; + obt_as.endianness = obt.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_close1 (&handle); + return -1; + } + + alsa->handle = handle; + alsa->pollhlp.conf = conf; + return 0; +} + +static void alsa_fini_in (HWVoiceIn *hw) +{ + ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; + + alsa_anal_close (&alsa->handle, &alsa->pollhlp); + + g_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] = { + { .add = hw->wpos, .len = 0 }, + { .add = 0, .len = 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_t state; + + state = snd_pcm_state (alsa->handle); + switch (state) { + case SND_PCM_STATE_PREPARED: + avail = hw->samples; + break; + case SND_PCM_STATE_SUSPENDED: + /* stream is suspended and waiting for an application recovery */ + if (alsa_resume (alsa->handle)) { + dolog ("Failed to resume suspended input stream\n"); + return 0; + } + trace_alsa_resume_in(); + break; + default: + trace_alsa_no_frames(state); + return 0; + } + } + + 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; + struct st_sample *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: + trace_alsa_read_zero(len); + goto exit; + + case -EPIPE: + if (alsa_recover (alsa->handle)) { + alsa_logerr (nread, "Failed to read %ld frames\n", len); + goto exit; + } + trace_alsa_xrun_in(); + 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); + + 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: + { + va_list ap; + int poll_mode; + + va_start (ap, cmd); + poll_mode = va_arg (ap, int); + va_end (ap); + + ldebug ("enabling voice\n"); + if (poll_mode && alsa_poll_in (hw)) { + poll_mode = 0; + } + hw->poll_mode = poll_mode; + + return alsa_voice_ctl (alsa->handle, "capture", VOICE_CTL_START); + } + + case VOICE_DISABLE: + ldebug ("disabling voice\n"); + if (hw->poll_mode) { + hw->poll_mode = 0; + alsa_fini_poll (&alsa->pollhlp); + } + return alsa_voice_ctl (alsa->handle, "capture", VOICE_CTL_PAUSE); + } + + return -1; +} + +static ALSAConf glob_conf = { + .buffer_size_out = 4096, + .period_size_out = 1024, + .pcm_name_out = "default", + .pcm_name_in = "default", +}; + +static void *alsa_audio_init (void) +{ + ALSAConf *conf = g_malloc(sizeof(ALSAConf)); + *conf = glob_conf; + return conf; +} + +static void alsa_audio_fini (void *opaque) +{ + g_free(opaque); +} + +static struct audio_option alsa_options[] = { + { + .name = "DAC_SIZE_IN_USEC", + .tag = AUD_OPT_BOOL, + .valp = &glob_conf.size_in_usec_out, + .descr = "DAC period/buffer size in microseconds (otherwise in frames)" + }, + { + .name = "DAC_PERIOD_SIZE", + .tag = AUD_OPT_INT, + .valp = &glob_conf.period_size_out, + .descr = "DAC period size (0 to go with system default)", + .overriddenp = &glob_conf.period_size_out_overridden + }, + { + .name = "DAC_BUFFER_SIZE", + .tag = AUD_OPT_INT, + .valp = &glob_conf.buffer_size_out, + .descr = "DAC buffer size (0 to go with system default)", + .overriddenp = &glob_conf.buffer_size_out_overridden + }, + { + .name = "ADC_SIZE_IN_USEC", + .tag = AUD_OPT_BOOL, + .valp = &glob_conf.size_in_usec_in, + .descr = + "ADC period/buffer size in microseconds (otherwise in frames)" + }, + { + .name = "ADC_PERIOD_SIZE", + .tag = AUD_OPT_INT, + .valp = &glob_conf.period_size_in, + .descr = "ADC period size (0 to go with system default)", + .overriddenp = &glob_conf.period_size_in_overridden + }, + { + .name = "ADC_BUFFER_SIZE", + .tag = AUD_OPT_INT, + .valp = &glob_conf.buffer_size_in, + .descr = "ADC buffer size (0 to go with system default)", + .overriddenp = &glob_conf.buffer_size_in_overridden + }, + { + .name = "THRESHOLD", + .tag = AUD_OPT_INT, + .valp = &glob_conf.threshold, + .descr = "(undocumented)" + }, + { + .name = "DAC_DEV", + .tag = AUD_OPT_STR, + .valp = &glob_conf.pcm_name_out, + .descr = "DAC device name (for instance dmix)" + }, + { + .name = "ADC_DEV", + .tag = AUD_OPT_STR, + .valp = &glob_conf.pcm_name_in, + .descr = "ADC device name" + }, + { /* End of list */ } +}; + +static struct audio_pcm_ops alsa_pcm_ops = { + .init_out = alsa_init_out, + .fini_out = alsa_fini_out, + .run_out = alsa_run_out, + .write = alsa_write, + .ctl_out = alsa_ctl_out, + + .init_in = alsa_init_in, + .fini_in = alsa_fini_in, + .run_in = alsa_run_in, + .read = alsa_read, + .ctl_in = alsa_ctl_in, +}; + +struct audio_driver alsa_audio_driver = { + .name = "alsa", + .descr = "ALSA http://www.alsa-project.org", + .options = alsa_options, + .init = alsa_audio_init, + .fini = alsa_audio_fini, + .pcm_ops = &alsa_pcm_ops, + .can_be_default = 1, + .max_voices_out = INT_MAX, + .max_voices_in = INT_MAX, + .voice_size_out = sizeof (ALSAVoiceOut), + .voice_size_in = sizeof (ALSAVoiceIn) +}; |