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authorRajithaY <rajithax.yerrumsetty@intel.com>2017-04-25 03:31:15 -0700
committerRajitha Yerrumchetty <rajithax.yerrumsetty@intel.com>2017-05-22 06:48:08 +0000
commitbb756eebdac6fd24e8919e2c43f7d2c8c4091f59 (patch)
treeca11e03542edf2d8f631efeca5e1626d211107e3 /qemu/audio/alsaaudio.c
parenta14b48d18a9ed03ec191cf16b162206998a895ce (diff)
Adding qemu as a submodule of KVMFORNFV
This Patch includes the changes to add qemu as a submodule to kvmfornfv repo and make use of the updated latest qemu for the execution of all testcase Change-Id: I1280af507a857675c7f81d30c95255635667bdd7 Signed-off-by:RajithaY<rajithax.yerrumsetty@intel.com>
Diffstat (limited to 'qemu/audio/alsaaudio.c')
-rw-r--r--qemu/audio/alsaaudio.c1228
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)
-};