diff options
Diffstat (limited to 'qemu/audio/alsaaudio.c')
| -rw-r--r-- | qemu/audio/alsaaudio.c | 1228 |
1 files changed, 0 insertions, 1228 deletions
diff --git a/qemu/audio/alsaaudio.c b/qemu/audio/alsaaudio.c deleted file mode 100644 index 3652a7b5f..000000000 --- a/qemu/audio/alsaaudio.c +++ /dev/null @@ -1,1228 +0,0 @@ -/* - * 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 "qemu/osdep.h" -#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) -}; |