| /* |
| * QEMU ALSA audio driver |
| * |
| * Copyright (c) 2008-2010 The Android Open Source Project |
| * 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 "sysemu/char.h" |
| #include "audio.h" |
| |
| #if QEMU_GNUC_PREREQ(4, 3) |
| #pragma GCC diagnostic ignored "-Waddress" |
| #endif |
| |
| #define AUDIO_CAP "alsa" |
| #include "audio_int.h" |
| #include <dlfcn.h> |
| #include <pthread.h> |
| #include "android/qemu-debug.h" |
| |
| #define DEBUG 1 |
| |
| #if DEBUG |
| # include <stdio.h> |
| # define D(...) VERBOSE_PRINT(audio,__VA_ARGS__) |
| # define D_ACTIVE VERBOSE_CHECK(audio) |
| # define O(...) VERBOSE_PRINT(audioout,__VA_ARGS__) |
| # define I(...) VERBOSE_PRINT(audioin,__VA_ARGS__) |
| #else |
| # define D(...) ((void)0) |
| # define D_ACTIVE 0 |
| # define O(...) ((void)0) |
| # define I(...) ((void)0) |
| #endif |
| |
| static void* alsa_lib; |
| |
| struct pollhlp { |
| snd_pcm_t *handle; |
| struct pollfd *pfds; |
| 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; |
| |
| 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_overridden; |
| int period_size_in_overridden; |
| |
| int buffer_size_out_overridden; |
| int period_size_out_overridden; |
| int verbose; |
| } conf = { |
| .buffer_size_out = 4096, |
| .period_size_out = 1024, |
| .pcm_name_out = "default", |
| .pcm_name_in = "default", |
| }; |
| |
| 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)) { |
| if (conf.verbose) { |
| dolog ("revents = %d\n", 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) { |
| err = qemu_set_fd_handler (pfds[i].fd, alsa_poll_handler, |
| NULL, hlp); |
| } |
| if (pfds[i].events & POLLOUT) { |
| if (conf.verbose) { |
| dolog ("POLLOUT %d %d\n", i, pfds[i].fd); |
| } |
| err = qemu_set_fd_handler (pfds[i].fd, NULL, |
| alsa_poll_handler, hlp); |
| } |
| if (conf.verbose) { |
| dolog ("Set handler events=%#x index=%d fd=%d err=%d\n", |
| pfds[i].events, i, pfds[i].fd, err); |
| } |
| |
| if (err) { |
| dolog ("Failed to set handler events=%#x index=%d fd=%d err=%d\n", |
| pfds[i].events, i, pfds[i].fd, err); |
| |
| while (i--) { |
| qemu_set_fd_handler (pfds[i].fd, NULL, NULL, NULL); |
| } |
| g_free (pfds); |
| return -1; |
| } |
| } |
| 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) |
| { |
| 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; |
| |
| case AUD_FMT_S32: |
| return SND_PCM_FORMAT_S32_LE; |
| |
| case AUD_FMT_U32: |
| 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) |
| { |
| 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 && conf.verbose) { |
| 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 (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 (conf.verbose && |
| (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: |
| if (conf.verbose) { |
| dolog ("Failed to write %d frames (wrote zero)\n", len); |
| } |
| return; |
| |
| case -EPIPE: |
| if (alsa_recover (alsa->handle)) { |
| alsa_logerr (written, "Failed to write %d frames\n", |
| len); |
| return; |
| } |
| if (conf.verbose) { |
| dolog ("Recovering from playback xrun\n"); |
| } |
| 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; |
| } |
| if (conf.verbose) { |
| dolog ("Resuming suspended output stream\n"); |
| } |
| 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); |
| |
| if (alsa->pcm_buf) { |
| g_free (alsa->pcm_buf); |
| alsa->pcm_buf = NULL; |
| } |
| } |
| |
| static int alsa_init_out (HWVoiceOut *hw, struct audsettings *as) |
| { |
| ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; |
| struct alsa_params_req req; |
| struct alsa_params_obt obt; |
| snd_pcm_t *handle; |
| struct audsettings obt_as; |
| int result = -1; |
| |
| /* shut alsa debug spew */ |
| if (!D_ACTIVE) |
| stdio_disable(); |
| |
| 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; |
| 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)) { |
| goto Exit; |
| } |
| |
| 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); |
| goto Exit; |
| } |
| |
| alsa->handle = handle; |
| result = 0; /* success */ |
| |
| Exit: |
| if (!D_ACTIVE) |
| stdio_enable(); |
| |
| return result; |
| } |
| |
| 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: |
| { |
| 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", 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, struct audsettings *as) |
| { |
| ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; |
| struct alsa_params_req req; |
| struct alsa_params_obt obt; |
| snd_pcm_t *handle; |
| struct audsettings obt_as; |
| int result = -1; |
| |
| /* shut alsa debug spew */ |
| if (!D_ACTIVE) |
| stdio_disable(); |
| |
| 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; |
| 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)) { |
| goto Exit; |
| } |
| |
| 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); |
| goto Exit; |
| } |
| |
| alsa->handle = handle; |
| result = 0; /* success */ |
| |
| Exit: |
| if (!D_ACTIVE) |
| stdio_enable(); |
| |
| return result; |
| } |
| |
| static void alsa_fini_in (HWVoiceIn *hw) |
| { |
| ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; |
| |
| alsa_anal_close (&alsa->handle, &alsa->pollhlp); |
| |
| if (alsa->pcm_buf) { |
| 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; |
| } |
| if (conf.verbose) { |
| dolog ("Resuming suspended input stream\n"); |
| } |
| break; |
| default: |
| if (conf.verbose) { |
| dolog ("No frames available and ALSA state is %d\n", 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: |
| 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: |
| { |
| 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", 0); |
| } |
| |
| 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", 1); |
| } |
| |
| return -1; |
| } |
| |
| extern int alsa_dynlink_init(void* lib); |
| |
| static void *alsa_audio_init (void) |
| { |
| void* result = NULL; |
| |
| alsa_lib = dlopen( "libasound.so", RTLD_NOW ); |
| if (alsa_lib == NULL) |
| alsa_lib = dlopen( "libasound.so.2", RTLD_NOW ); |
| |
| if (alsa_lib == NULL) { |
| ldebug("could not find libasound on this system\n"); |
| goto Exit; |
| } |
| |
| if (alsa_dynlink_init(alsa_lib) < 0) |
| goto Fail; |
| |
| result = &conf; |
| goto Exit; |
| |
| Fail: |
| ldebug("%s: failed to open library\n", __FUNCTION__); |
| dlclose(alsa_lib); |
| |
| Exit: |
| return result; |
| } |
| |
| static void alsa_audio_fini (void *opaque) |
| { |
| if (alsa_lib != NULL) { |
| dlclose(alsa_lib); |
| alsa_lib = NULL; |
| } |
| (void) opaque; |
| } |
| |
| static struct audio_option alsa_options[] = { |
| { |
| .name = "DAC_SIZE_IN_USEC", |
| .tag = AUD_OPT_BOOL, |
| .valp = &conf.size_in_usec_out, |
| .descr = "DAC period/buffer size in microseconds (otherwise in frames)" |
| }, |
| { |
| .name = "DAC_PERIOD_SIZE", |
| .tag = AUD_OPT_INT, |
| .valp = &conf.period_size_out, |
| .descr = "DAC period size (0 to go with system default)", |
| .overriddenp = &conf.period_size_out_overridden |
| }, |
| { |
| .name = "DAC_BUFFER_SIZE", |
| .tag = AUD_OPT_INT, |
| .valp = &conf.buffer_size_out, |
| .descr = "DAC buffer size (0 to go with system default)", |
| .overriddenp = &conf.buffer_size_out_overridden |
| }, |
| { |
| .name = "ADC_SIZE_IN_USEC", |
| .tag = AUD_OPT_BOOL, |
| .valp = &conf.size_in_usec_in, |
| .descr = |
| "ADC period/buffer size in microseconds (otherwise in frames)" |
| }, |
| { |
| .name = "ADC_PERIOD_SIZE", |
| .tag = AUD_OPT_INT, |
| .valp = &conf.period_size_in, |
| .descr = "ADC period size (0 to go with system default)", |
| .overriddenp = &conf.period_size_in_overridden |
| }, |
| { |
| .name = "ADC_BUFFER_SIZE", |
| .tag = AUD_OPT_INT, |
| .valp = &conf.buffer_size_in, |
| .descr = "ADC buffer size (0 to go with system default)", |
| .overriddenp = &conf.buffer_size_in_overridden |
| }, |
| { |
| .name = "THRESHOLD", |
| .tag = AUD_OPT_INT, |
| .valp = &conf.threshold, |
| .descr = "(undocumented)" |
| }, |
| { |
| .name = "DAC_DEV", |
| .tag = AUD_OPT_STR, |
| .valp = &conf.pcm_name_out, |
| .descr = "DAC device name (for instance dmix)" |
| }, |
| { |
| .name = "ADC_DEV", |
| .tag = AUD_OPT_STR, |
| .valp = &conf.pcm_name_in, |
| .descr = "ADC device name" |
| }, |
| { |
| .name = "VERBOSE", |
| .tag = AUD_OPT_BOOL, |
| .valp = &conf.verbose, |
| .descr = "Behave in a more verbose way" |
| }, |
| { /* 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) |
| }; |
