| /* public domain */ |
| #include "qemu-common.h" |
| #include "audio.h" |
| |
| #include <pulse/simple.h> |
| #include <pulse/error.h> |
| #include <dlfcn.h> |
| |
| #define AUDIO_CAP "pulseaudio" |
| #include "audio_int.h" |
| #include "audio_pt_int.h" |
| |
| #define DEBUG 1 |
| |
| #if DEBUG |
| # include "android/qemu-debug.h" |
| # 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 |
| |
| #define DYNLINK_FUNCTIONS \ |
| DYNLINK_FUNC(pa_simple*,pa_simple_new,(const char* server,const char* name, pa_stream_direction_t dir, const char* dev, const char* stream_name, const pa_sample_spec* ss, const pa_channel_map* map, const pa_buffer_attr *attr, int *error)) \ |
| DYNLINK_FUNC(void,pa_simple_free,(pa_simple* s))\ |
| DYNLINK_FUNC(int,pa_simple_write,(pa_simple* s, const void* data, size_t bytes, int* error))\ |
| DYNLINK_FUNC(int,pa_simple_read,(pa_simple* s,void* data, size_t bytes, int* error))\ |
| DYNLINK_FUNC(const char*,pa_strerror,(int error))\ |
| |
| #define DYNLINK_FUNCTIONS_INIT \ |
| pa_dynlink_init |
| |
| static void* pa_lib; |
| |
| #include "android/dynlink.h" |
| |
| typedef struct { |
| HWVoiceOut hw; |
| int done; |
| int live; |
| int decr; |
| int rpos; |
| pa_simple *s; |
| void *pcm_buf; |
| struct audio_pt pt; |
| } PAVoiceOut; |
| |
| typedef struct { |
| HWVoiceIn hw; |
| int done; |
| int dead; |
| int incr; |
| int wpos; |
| pa_simple *s; |
| void *pcm_buf; |
| struct audio_pt pt; |
| } PAVoiceIn; |
| |
| static struct { |
| int samples; |
| int divisor; |
| char *server; |
| char *sink; |
| char *source; |
| } conf = { |
| .samples = 1024, |
| .divisor = 2, |
| }; |
| |
| static void GCC_FMT_ATTR (2, 3) qpa_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", FF(pa_strerror) (err)); |
| } |
| |
| static void *qpa_thread_out (void *arg) |
| { |
| PAVoiceOut *pa = arg; |
| HWVoiceOut *hw = &pa->hw; |
| int threshold; |
| |
| threshold = conf.divisor ? hw->samples / conf.divisor : 0; |
| |
| if (audio_pt_lock (&pa->pt, AUDIO_FUNC)) { |
| return NULL; |
| } |
| |
| for (;;) { |
| int decr, to_mix, rpos; |
| |
| for (;;) { |
| if (pa->done) { |
| goto exit; |
| } |
| |
| if (pa->live > threshold) { |
| break; |
| } |
| |
| if (audio_pt_wait (&pa->pt, AUDIO_FUNC)) { |
| goto exit; |
| } |
| } |
| |
| decr = to_mix = pa->live; |
| rpos = hw->rpos; |
| |
| if (audio_pt_unlock (&pa->pt, AUDIO_FUNC)) { |
| return NULL; |
| } |
| |
| while (to_mix) { |
| int error; |
| int chunk = audio_MIN (to_mix, hw->samples - rpos); |
| struct st_sample *src = hw->mix_buf + rpos; |
| |
| hw->clip (pa->pcm_buf, src, chunk); |
| |
| if (FF(pa_simple_write) (pa->s, pa->pcm_buf, |
| chunk << hw->info.shift, &error) < 0) { |
| qpa_logerr (error, "pa_simple_write failed\n"); |
| return NULL; |
| } |
| |
| rpos = (rpos + chunk) % hw->samples; |
| to_mix -= chunk; |
| } |
| |
| if (audio_pt_lock (&pa->pt, AUDIO_FUNC)) { |
| return NULL; |
| } |
| |
| pa->rpos = rpos; |
| pa->live -= decr; |
| pa->decr += decr; |
| } |
| |
| exit: |
| audio_pt_unlock (&pa->pt, AUDIO_FUNC); |
| return NULL; |
| } |
| |
| static int qpa_run_out (HWVoiceOut *hw, int live) |
| { |
| int decr; |
| PAVoiceOut *pa = (PAVoiceOut *) hw; |
| |
| if (audio_pt_lock (&pa->pt, AUDIO_FUNC)) { |
| return 0; |
| } |
| |
| decr = audio_MIN (live, pa->decr); |
| pa->decr -= decr; |
| pa->live = live - decr; |
| hw->rpos = pa->rpos; |
| if (pa->live > 0) { |
| audio_pt_unlock_and_signal (&pa->pt, AUDIO_FUNC); |
| } |
| else { |
| audio_pt_unlock (&pa->pt, AUDIO_FUNC); |
| } |
| return decr; |
| } |
| |
| static int qpa_write (SWVoiceOut *sw, void *buf, int len) |
| { |
| return audio_pcm_sw_write (sw, buf, len); |
| } |
| |
| /* capture */ |
| static void *qpa_thread_in (void *arg) |
| { |
| PAVoiceIn *pa = arg; |
| HWVoiceIn *hw = &pa->hw; |
| int threshold; |
| |
| threshold = conf.divisor ? hw->samples / conf.divisor : 0; |
| |
| if (audio_pt_lock (&pa->pt, AUDIO_FUNC)) { |
| return NULL; |
| } |
| |
| for (;;) { |
| int incr, to_grab, wpos; |
| |
| for (;;) { |
| if (pa->done) { |
| goto exit; |
| } |
| |
| if (pa->dead > threshold) { |
| break; |
| } |
| |
| if (audio_pt_wait (&pa->pt, AUDIO_FUNC)) { |
| goto exit; |
| } |
| } |
| |
| incr = to_grab = pa->dead; |
| wpos = hw->wpos; |
| |
| if (audio_pt_unlock (&pa->pt, AUDIO_FUNC)) { |
| return NULL; |
| } |
| |
| while (to_grab) { |
| int error; |
| int chunk = audio_MIN (to_grab, hw->samples - wpos); |
| void *buf = advance (pa->pcm_buf, wpos); |
| |
| if (FF(pa_simple_read) (pa->s, buf, |
| chunk << hw->info.shift, &error) < 0) { |
| qpa_logerr (error, "pa_simple_read failed\n"); |
| return NULL; |
| } |
| |
| hw->conv (hw->conv_buf + wpos, buf, chunk, &nominal_volume); |
| wpos = (wpos + chunk) % hw->samples; |
| to_grab -= chunk; |
| } |
| |
| if (audio_pt_lock (&pa->pt, AUDIO_FUNC)) { |
| return NULL; |
| } |
| |
| pa->wpos = wpos; |
| pa->dead -= incr; |
| pa->incr += incr; |
| } |
| |
| exit: |
| audio_pt_unlock (&pa->pt, AUDIO_FUNC); |
| return NULL; |
| } |
| |
| static int qpa_run_in (HWVoiceIn *hw) |
| { |
| int live, incr, dead; |
| PAVoiceIn *pa = (PAVoiceIn *) hw; |
| |
| if (audio_pt_lock (&pa->pt, AUDIO_FUNC)) { |
| return 0; |
| } |
| |
| live = audio_pcm_hw_get_live_in (hw); |
| dead = hw->samples - live; |
| incr = audio_MIN (dead, pa->incr); |
| pa->incr -= incr; |
| pa->dead = dead - incr; |
| hw->wpos = pa->wpos; |
| if (pa->dead > 0) { |
| audio_pt_unlock_and_signal (&pa->pt, AUDIO_FUNC); |
| } |
| else { |
| audio_pt_unlock (&pa->pt, AUDIO_FUNC); |
| } |
| return incr; |
| } |
| |
| static int qpa_read (SWVoiceIn *sw, void *buf, int len) |
| { |
| return audio_pcm_sw_read (sw, buf, len); |
| } |
| |
| static pa_sample_format_t audfmt_to_pa (audfmt_e afmt, int endianness) |
| { |
| int format; |
| |
| switch (afmt) { |
| case AUD_FMT_S8: |
| case AUD_FMT_U8: |
| format = PA_SAMPLE_U8; |
| break; |
| case AUD_FMT_S16: |
| case AUD_FMT_U16: |
| format = endianness ? PA_SAMPLE_S16BE : PA_SAMPLE_S16LE; |
| break; |
| case AUD_FMT_S32: |
| case AUD_FMT_U32: |
| format = endianness ? PA_SAMPLE_S32BE : PA_SAMPLE_S32LE; |
| break; |
| default: |
| dolog ("Internal logic error: Bad audio format %d\n", afmt); |
| format = PA_SAMPLE_U8; |
| break; |
| } |
| return format; |
| } |
| |
| static audfmt_e pa_to_audfmt (pa_sample_format_t fmt, int *endianness) |
| { |
| switch (fmt) { |
| case PA_SAMPLE_U8: |
| return AUD_FMT_U8; |
| case PA_SAMPLE_S16BE: |
| *endianness = 1; |
| return AUD_FMT_S16; |
| case PA_SAMPLE_S16LE: |
| *endianness = 0; |
| return AUD_FMT_S16; |
| case PA_SAMPLE_S32BE: |
| *endianness = 1; |
| return AUD_FMT_S32; |
| case PA_SAMPLE_S32LE: |
| *endianness = 0; |
| return AUD_FMT_S32; |
| default: |
| dolog ("Internal logic error: Bad pa_sample_format %d\n", fmt); |
| return AUD_FMT_U8; |
| } |
| } |
| |
| static int qpa_init_out (HWVoiceOut *hw, struct audsettings *as) |
| { |
| int error; |
| static pa_sample_spec ss; |
| struct audsettings obt_as = *as; |
| PAVoiceOut *pa = (PAVoiceOut *) hw; |
| |
| ss.format = audfmt_to_pa (as->fmt, as->endianness); |
| ss.channels = as->nchannels; |
| ss.rate = as->freq; |
| |
| obt_as.fmt = pa_to_audfmt (ss.format, &obt_as.endianness); |
| |
| pa->s = FF(pa_simple_new) ( |
| conf.server, |
| "qemu", |
| PA_STREAM_PLAYBACK, |
| conf.sink, |
| "pcm.playback", |
| &ss, |
| NULL, /* channel map */ |
| NULL, /* buffering attributes */ |
| &error |
| ); |
| if (!pa->s) { |
| qpa_logerr (error, "pa_simple_new for playback failed\n"); |
| goto fail1; |
| } |
| |
| audio_pcm_init_info (&hw->info, &obt_as); |
| hw->samples = conf.samples; |
| pa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift); |
| if (!pa->pcm_buf) { |
| dolog ("Could not allocate buffer (%d bytes)\n", |
| hw->samples << hw->info.shift); |
| goto fail2; |
| } |
| |
| if (audio_pt_init (&pa->pt, qpa_thread_out, hw, AUDIO_CAP, AUDIO_FUNC)) { |
| goto fail3; |
| } |
| |
| return 0; |
| |
| fail3: |
| g_free (pa->pcm_buf); |
| pa->pcm_buf = NULL; |
| fail2: |
| FF(pa_simple_free) (pa->s); |
| pa->s = NULL; |
| fail1: |
| return -1; |
| } |
| |
| static int qpa_init_in (HWVoiceIn *hw, struct audsettings *as) |
| { |
| int error; |
| static pa_sample_spec ss; |
| struct audsettings obt_as = *as; |
| PAVoiceIn *pa = (PAVoiceIn *) hw; |
| |
| ss.format = audfmt_to_pa (as->fmt, as->endianness); |
| ss.channels = as->nchannels; |
| ss.rate = as->freq; |
| |
| obt_as.fmt = pa_to_audfmt (ss.format, &obt_as.endianness); |
| |
| pa->s = FF(pa_simple_new) ( |
| conf.server, |
| "qemu", |
| PA_STREAM_RECORD, |
| conf.source, |
| "pcm.capture", |
| &ss, |
| NULL, /* channel map */ |
| NULL, /* buffering attributes */ |
| &error |
| ); |
| if (!pa->s) { |
| qpa_logerr (error, "pa_simple_new for capture failed\n"); |
| goto fail1; |
| } |
| |
| audio_pcm_init_info (&hw->info, &obt_as); |
| hw->samples = conf.samples; |
| pa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift); |
| if (!pa->pcm_buf) { |
| dolog ("Could not allocate buffer (%d bytes)\n", |
| hw->samples << hw->info.shift); |
| goto fail2; |
| } |
| |
| if (audio_pt_init (&pa->pt, qpa_thread_in, hw, AUDIO_CAP, AUDIO_FUNC)) { |
| goto fail3; |
| } |
| |
| return 0; |
| |
| fail3: |
| g_free (pa->pcm_buf); |
| pa->pcm_buf = NULL; |
| fail2: |
| FF(pa_simple_free) (pa->s); |
| pa->s = NULL; |
| fail1: |
| return -1; |
| } |
| |
| static void qpa_fini_out (HWVoiceOut *hw) |
| { |
| void *ret; |
| PAVoiceOut *pa = (PAVoiceOut *) hw; |
| |
| audio_pt_lock (&pa->pt, AUDIO_FUNC); |
| pa->done = 1; |
| audio_pt_unlock_and_signal (&pa->pt, AUDIO_FUNC); |
| audio_pt_join (&pa->pt, &ret, AUDIO_FUNC); |
| |
| if (pa->s) { |
| FF(pa_simple_free) (pa->s); |
| pa->s = NULL; |
| } |
| |
| audio_pt_fini (&pa->pt, AUDIO_FUNC); |
| g_free (pa->pcm_buf); |
| pa->pcm_buf = NULL; |
| } |
| |
| static void qpa_fini_in (HWVoiceIn *hw) |
| { |
| void *ret; |
| PAVoiceIn *pa = (PAVoiceIn *) hw; |
| |
| audio_pt_lock (&pa->pt, AUDIO_FUNC); |
| pa->done = 1; |
| audio_pt_unlock_and_signal (&pa->pt, AUDIO_FUNC); |
| audio_pt_join (&pa->pt, &ret, AUDIO_FUNC); |
| |
| if (pa->s) { |
| FF(pa_simple_free) (pa->s); |
| pa->s = NULL; |
| } |
| |
| audio_pt_fini (&pa->pt, AUDIO_FUNC); |
| g_free (pa->pcm_buf); |
| pa->pcm_buf = NULL; |
| } |
| |
| static int qpa_ctl_out (HWVoiceOut *hw, int cmd, ...) |
| { |
| (void) hw; |
| (void) cmd; |
| return 0; |
| } |
| |
| static int qpa_ctl_in (HWVoiceIn *hw, int cmd, ...) |
| { |
| (void) hw; |
| (void) cmd; |
| return 0; |
| } |
| |
| /* common */ |
| static void *qpa_audio_init (void) |
| { |
| void* result = NULL; |
| |
| D("%s: entering", __FUNCTION__); |
| pa_lib = dlopen( "libpulse-simple.so", RTLD_NOW ); |
| if (pa_lib == NULL) |
| pa_lib = dlopen( "libpulse-simple.so.0", RTLD_NOW ); |
| |
| if (pa_lib == NULL) { |
| D("could not find libpulse on this system\n"); |
| goto Exit; |
| } |
| |
| if (pa_dynlink_init(pa_lib) < 0) |
| goto Fail; |
| |
| { |
| pa_sample_spec ss; |
| int error; |
| pa_simple* simple; |
| |
| ss.format = PA_SAMPLE_U8; |
| ss.rate = 44100; |
| ss.channels = 1; |
| |
| /* try to open it for playback */ |
| simple = FF(pa_simple_new) ( |
| conf.server, |
| "qemu", |
| PA_STREAM_PLAYBACK, |
| conf.sink, |
| "pcm.playback", |
| &ss, |
| NULL, /* channel map */ |
| NULL, /* buffering attributes */ |
| &error |
| ); |
| |
| if (simple == NULL) { |
| D("%s: error opening open pulse audio library: %s", |
| __FUNCTION__, FF(pa_strerror)(error)); |
| goto Fail; |
| } |
| FF(pa_simple_free)(simple); |
| } |
| |
| result = &conf; |
| goto Exit; |
| |
| Fail: |
| D("%s: failed to open library\n", __FUNCTION__); |
| dlclose(pa_lib); |
| |
| Exit: |
| D("%s: exiting", __FUNCTION__); |
| return result; |
| } |
| |
| static void qpa_audio_fini (void *opaque) |
| { |
| if (pa_lib != NULL) { |
| dlclose(pa_lib); |
| pa_lib = NULL; |
| } |
| (void) opaque; |
| (void) opaque; |
| } |
| |
| struct audio_option qpa_options[] = { |
| { |
| .name = "SAMPLES", |
| .tag = AUD_OPT_INT, |
| .valp = &conf.samples, |
| .descr = "buffer size in samples" |
| }, |
| { |
| .name = "DIVISOR", |
| .tag = AUD_OPT_INT, |
| .valp = &conf.divisor, |
| .descr = "threshold divisor" |
| }, |
| { |
| .name = "SERVER", |
| .tag = AUD_OPT_STR, |
| .valp = &conf.server, |
| .descr = "server address" |
| }, |
| { |
| .name = "SINK", |
| .tag = AUD_OPT_STR, |
| .valp = &conf.sink, |
| .descr = "sink device name" |
| }, |
| { |
| .name = "SOURCE", |
| .tag = AUD_OPT_STR, |
| .valp = &conf.source, |
| .descr = "source device name" |
| }, |
| { /* End of list */ } |
| }; |
| |
| static struct audio_pcm_ops qpa_pcm_ops = { |
| .init_out = qpa_init_out, |
| .fini_out = qpa_fini_out, |
| .run_out = qpa_run_out, |
| .write = qpa_write, |
| .ctl_out = qpa_ctl_out, |
| |
| .init_in = qpa_init_in, |
| .fini_in = qpa_fini_in, |
| .run_in = qpa_run_in, |
| .read = qpa_read, |
| .ctl_in = qpa_ctl_in |
| }; |
| |
| struct audio_driver pa_audio_driver = { |
| .name = "pa", |
| .descr = "http://www.pulseaudio.org/", |
| .options = qpa_options, |
| .init = qpa_audio_init, |
| .fini = qpa_audio_fini, |
| .pcm_ops = &qpa_pcm_ops, |
| .can_be_default = 1, |
| .max_voices_out = INT_MAX, |
| .max_voices_in = INT_MAX, |
| .voice_size_out = sizeof (PAVoiceOut), |
| .voice_size_in = sizeof (PAVoiceIn) |
| }; |
