/** @file * * VBox PulseAudio backend */ /* * Copyright (C) 2006-2007 Sun Microsystems, Inc. * * This file is part of VirtualBox Open Source Edition (OSE), as * available from http://www.virtualbox.org. This file is free software; * you can redistribute it and/or modify it under the terms of the GNU * General Public License (GPL) as published by the Free Software * Foundation, in version 2 as it comes in the "COPYING" file of the * VirtualBox OSE distribution. VirtualBox OSE is distributed in the * hope that it will be useful, but WITHOUT ANY WARRANTY of any kind. * * Please contact Sun Microsystems, Inc., 4150 Network Circle, Santa * Clara, CA 95054 USA or visit http://www.sun.com if you need * additional information or have any questions. */ /******************************************************************************* * Header Files * *******************************************************************************/ #define LOG_GROUP LOG_GROUP_DEV_AUDIO #include #include #include #include "pulse_stubs.h" #include "../../vl_vbox.h" #include "audio.h" #define AUDIO_CAP "pulse" #include "audio_int.h" #include /* * We use a g_pMainLoop in a separate thread g_pContext. We have to call functions for * manipulating objects either from callback functions or we have to protect * these functions by pa_threaded_mainloop_lock() / pa_threaded_mainloop_unlock(). */ static struct pa_threaded_mainloop *g_pMainLoop; static struct pa_context *g_pContext; static void pulse_audio_fini (void *); typedef struct PulseVoice { HWVoiceOut hw; void *pPCMBuf; pa_stream *pStream; int fOpSuccess; } PulseVoice; static struct { int buffer_msecs_out; int buffer_msecs_in; } conf = { INIT_FIELD (.buffer_msecs_out = ) 100, INIT_FIELD (.buffer_msecs_in = ) 100, }; struct pulse_params_req { int freq; pa_sample_format_t pa_format; int nchannels; }; struct pulse_params_obt { int freq; pa_sample_format_t pa_format; int nchannels; unsigned long buffer_size; }; static pa_sample_format_t aud_to_pulsefmt (audfmt_e fmt) { switch (fmt) { case AUD_FMT_U8: return PA_SAMPLE_U8; case AUD_FMT_S16: return PA_SAMPLE_S16LE; #ifdef PA_SAMPLE_S32LE case AUD_FMT_S32: return PA_SAMPLE_S32LE; #endif default: dolog ("Bad audio format %d\n", fmt); return PA_SAMPLE_U8; } } static int pulse_to_audfmt (pa_sample_format_t pulsefmt, audfmt_e *fmt, int *endianess) { switch (pulsefmt) { case PA_SAMPLE_U8: *endianess = 0; *fmt = AUD_FMT_U8; break; case PA_SAMPLE_S16LE: *fmt = AUD_FMT_S16; *endianess = 0; break; case PA_SAMPLE_S16BE: *fmt = AUD_FMT_S16; *endianess = 1; break; #ifdef PA_SAMPLE_S32LE case PA_SAMPLE_S32LE: *fmt = AUD_FMT_S32; *endianess = 0; break; #endif #ifdef PA_SAMPLE_S32BE case PA_SAMPLE_S32BE: *fmt = AUD_FMT_S32; *endianess = 1; break; #endif default: return -1; } return 0; } static void context_state_callback(pa_context *c, void *userdata) { switch (pa_context_get_state(c)) { case PA_CONTEXT_READY: case PA_CONTEXT_TERMINATED: case PA_CONTEXT_FAILED: pa_threaded_mainloop_signal(g_pMainLoop, 0); break; default: break; } } static void stream_state_callback(pa_stream *s, void *userdata) { switch (pa_stream_get_state(s)) { case PA_STREAM_READY: case PA_STREAM_FAILED: case PA_STREAM_TERMINATED: pa_threaded_mainloop_signal(g_pMainLoop, 0); break; default: break; } } static void stream_latency_update_callback(pa_stream *s, void *userdata) { pa_threaded_mainloop_signal(g_pMainLoop, 0); } static int pulse_open (int fIn, struct pulse_params_req *req, struct pulse_params_obt *obt, pa_stream **ppStream) { pa_sample_spec sspec; pa_channel_map cmap; pa_stream *pStream = NULL; pa_buffer_attr bufAttr; const pa_buffer_attr *pBufAttr; const pa_sample_spec *pSampSpec; char achPCMName[64]; pa_stream_flags_t flags; int ms = fIn ? conf.buffer_msecs_in : conf.buffer_msecs_out; const char *stream_name = audio_get_stream_name(); RTStrPrintf(achPCMName, sizeof(achPCMName), "%.32s%s%s%s", stream_name ? stream_name : "", stream_name ? " (" : "", fIn ? "pcm_in" : "pcm_out", stream_name ? ")" : ""); sspec.rate = req->freq; sspec.channels = req->nchannels; sspec.format = req->pa_format; LogRel(("Pulse: open %s rate=%dHz channels=%d format=%s\n", fIn ? "PCM_IN" : "PCM_OUT", req->freq, req->nchannels, pa_sample_format_to_string(req->pa_format))); if (!pa_sample_spec_valid(&sspec)) { LogRel(("Pulse: Unsupported sample specification\n")); goto fail; } pa_channel_map_init_auto(&cmap, sspec.channels, PA_CHANNEL_MAP_ALSA); #if 0 pa_cvolume_reset(&volume, sspec.channels); #endif pa_threaded_mainloop_lock(g_pMainLoop); if (!(pStream = pa_stream_new(g_pContext, achPCMName, &sspec, &cmap))) { LogRel(("Pulse: Cannot create stream %s\n", achPCMName)); goto unlock_and_fail; } pSampSpec = pa_stream_get_sample_spec(pStream); obt->pa_format = pSampSpec->format; obt->nchannels = pSampSpec->channels; obt->freq = pSampSpec->rate; pa_stream_set_state_callback(pStream, stream_state_callback, NULL); pa_stream_set_latency_update_callback(pStream, stream_latency_update_callback, NULL); memset(&bufAttr, 0, sizeof(bufAttr)); bufAttr.tlength = (pa_bytes_per_second(pSampSpec) * ms) / 1000; bufAttr.maxlength = (bufAttr.tlength*3) / 2; bufAttr.minreq = pa_bytes_per_second(pSampSpec) / 100; /* 10ms */ bufAttr.prebuf = bufAttr.tlength - bufAttr.minreq; bufAttr.fragsize = pa_bytes_per_second(pSampSpec) / 100; /* 10ms */ flags = PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE; if (fIn) { if (pa_stream_connect_record(pStream, /*dev=*/NULL, &bufAttr, flags) < 0) { LogRel(("Pulse: Cannot connect record stream : %s\n", pa_strerror(pa_context_errno(g_pContext)))); goto disconnect_unlock_and_fail; } } else { if (pa_stream_connect_playback(pStream, /*dev=*/NULL, &bufAttr, flags, NULL, NULL) < 0) { LogRel(("Pulse: Cannot connect playback stream: %s\n", pa_strerror(pa_context_errno(g_pContext)))); goto disconnect_unlock_and_fail; } } /* Wait until the stream is ready */ for (;;) { pa_stream_state_t sstate; pa_threaded_mainloop_wait(g_pMainLoop); sstate = pa_stream_get_state(pStream); if (sstate == PA_STREAM_READY) break; else if (sstate == PA_STREAM_FAILED || sstate == PA_STREAM_TERMINATED) { LogRel(("Pulse: Failed to initialize stream (state %d)\n", sstate)); goto disconnect_unlock_and_fail; } } pBufAttr = pa_stream_get_buffer_attr(pStream); obt->buffer_size = pBufAttr->maxlength; pa_threaded_mainloop_unlock(g_pMainLoop); LogRel(("Pulse: buffer settings: max=%d tlength=%d prebuf=%d minreq=%d\n", pBufAttr->maxlength, pBufAttr->tlength, pBufAttr->prebuf, pBufAttr->minreq)); *ppStream = pStream; return 0; disconnect_unlock_and_fail: pa_stream_disconnect(pStream); unlock_and_fail: pa_threaded_mainloop_unlock(g_pMainLoop); fail: if (pStream) pa_stream_unref(pStream); *ppStream = NULL; return -1; } static int pulse_init_out (HWVoiceOut *hw, audsettings_t *as) { PulseVoice *pulse = (PulseVoice *) hw; struct pulse_params_req req; struct pulse_params_obt obt; audfmt_e effective_fmt; int endianness; audsettings_t obt_as; req.pa_format = aud_to_pulsefmt (as->fmt); req.freq = as->freq; req.nchannels = as->nchannels; if (pulse_open (/*fIn=*/0, &req, &obt, &pulse->pStream)) return -1; if (pulse_to_audfmt (obt.pa_format, &effective_fmt, &endianness)) { LogRel(("Pulse: Cannot find audio format %d\n", obt.pa_format)); return -1; } obt_as.freq = obt.freq; obt_as.nchannels = obt.nchannels; obt_as.fmt = effective_fmt; obt_as.endianness = endianness; audio_pcm_init_info (&hw->info, &obt_as); hw->samples = obt.buffer_size >> hw->info.shift; pulse->pPCMBuf = RTMemAllocZ(obt.buffer_size); if (!pulse->pPCMBuf) { LogRel(("Pulse: Could not allocate DAC buffer of %d bytes\n", obt.buffer_size)); return -1; } return 0; } static void pulse_fini_out (HWVoiceOut *hw) { PulseVoice *pulse = (PulseVoice *)hw; if (pulse->pStream) { pa_stream_disconnect(pulse->pStream); pa_stream_unref(pulse->pStream); pulse->pStream = NULL; } if (pulse->pPCMBuf) { RTMemFree (pulse->pPCMBuf); pulse->pPCMBuf = NULL; } } static int pulse_run_out (HWVoiceOut *hw) { PulseVoice *pulse = (PulseVoice *) hw; int csLive, csDecr, csSamples, csToWrite, csAvail; size_t cbAvail, cbToWrite; uint8_t *pu8Dst; st_sample_t *psSrc; csLive = audio_pcm_hw_get_live_out (hw); if (!csLive) return 0; pa_threaded_mainloop_lock(g_pMainLoop); cbAvail = pa_stream_writable_size (pulse->pStream); if (cbAvail == -1) { LogRel(("Pulse: Failed to determine the writable size: %s\n", pa_strerror(pa_context_errno(g_pContext)))); return 0; } csAvail = cbAvail >> hw->info.shift; /* bytes => samples */ csDecr = audio_MIN (csLive, csAvail); csSamples = csDecr; while (csSamples) { /* split request at the end of our samples buffer */ csToWrite = audio_MIN (csSamples, hw->samples - hw->rpos); cbToWrite = csToWrite << hw->info.shift; psSrc = hw->mix_buf + hw->rpos; pu8Dst = advance (pulse->pPCMBuf, hw->rpos << hw->info.shift); hw->clip (pu8Dst, psSrc, csToWrite); if (pa_stream_write (pulse->pStream, pu8Dst, cbToWrite, /*cleanup_callback=*/NULL, 0, PA_SEEK_RELATIVE) < 0) { LogRel(("Pulse: Failed to write %d samples: %s\n", csToWrite, pa_strerror(pa_context_errno(g_pContext)))); break; } hw->rpos = (hw->rpos + csToWrite) % hw->samples; csSamples -= csToWrite; } pa_threaded_mainloop_unlock(g_pMainLoop); return csDecr; } static int pulse_write (SWVoiceOut *sw, void *buf, int len) { return audio_pcm_sw_write (sw, buf, len); } static void stream_success_callback(pa_stream *pStream, int success, void *userdata) { PulseVoice *pulse = (PulseVoice *) userdata; pulse->fOpSuccess = success; pa_threaded_mainloop_signal(g_pMainLoop, 0); } typedef enum { Unpause = 0, Pause = 1, Flush = 2, Trigger = 3 } pulse_cmd_t; static int pulse_ctrl (HWVoiceOut *hw, pulse_cmd_t cmd) { PulseVoice *pulse = (PulseVoice *) hw; pa_operation *op = NULL; if (!pulse->pStream) return 0; pa_threaded_mainloop_lock(g_pMainLoop); switch (cmd) { case Pause: op = pa_stream_cork(pulse->pStream, 1, stream_success_callback, pulse); break; case Unpause: op = pa_stream_cork(pulse->pStream, 0, stream_success_callback, pulse); break; case Flush: op = pa_stream_flush(pulse->pStream, stream_success_callback, pulse); break; case Trigger: op = pa_stream_trigger(pulse->pStream, stream_success_callback, pulse); break; default: goto fail; } if (!op) LogRel(("Pulse: Failed ctrl cmd=%d to stream: %s\n", cmd, pa_strerror(pa_context_errno(g_pContext)))); else pa_operation_unref(op); fail: pa_threaded_mainloop_unlock(g_pMainLoop); return 0; } static int pulse_ctl_out (HWVoiceOut *hw, int cmd, ...) { switch (cmd) { case VOICE_ENABLE: pulse_ctrl(hw, Unpause); pulse_ctrl(hw, Trigger); break; case VOICE_DISABLE: pulse_ctrl(hw, Flush); break; default: return -1; } return 0; } static int pulse_init_in (HWVoiceIn *hw, audsettings_t *as) { PulseVoice *pulse = (PulseVoice *) hw; struct pulse_params_req req; struct pulse_params_obt obt; audfmt_e effective_fmt; int endianness; audsettings_t obt_as; req.pa_format = aud_to_pulsefmt (as->fmt); req.freq = as->freq; req.nchannels = as->nchannels; if (pulse_open (/*fIn=*/1, &req, &obt, &pulse->pStream)) return -1; if (pulse_to_audfmt (obt.pa_format, &effective_fmt, &endianness)) { LogRel(("Pulse: Cannot find audio format %d\n", obt.pa_format)); return -1; } obt_as.freq = obt.freq; obt_as.nchannels = obt.nchannels; obt_as.fmt = effective_fmt; obt_as.endianness = endianness; audio_pcm_init_info (&hw->info, &obt_as); /* pcm_in: reserve twice as the maximum buffer length because of peek()/drop(). */ hw->samples = 2 * (obt.buffer_size >> hw->info.shift); /* no buffer for input */ pulse->pPCMBuf = NULL; return 0; } static void pulse_fini_in (HWVoiceIn *hw) { PulseVoice *pulse = (PulseVoice *)hw; if (pulse->pStream) { pa_stream_disconnect(pulse->pStream); pa_stream_unref(pulse->pStream); pulse->pStream = NULL; } if (pulse->pPCMBuf) { RTMemFree (pulse->pPCMBuf); pulse->pPCMBuf = NULL; } } static int pulse_run_in (HWVoiceIn *hw) { PulseVoice *pulse = (PulseVoice *) hw; int csDead, csDecr = 0, csSamples, csRead, csAvail; size_t cbAvail; const void *pu8Src; st_sample_t *psDst; csDead = hw->samples - audio_pcm_hw_get_live_in (hw); if (!csDead) return 0; /* no buffer available */ pa_threaded_mainloop_lock(g_pMainLoop); if (pa_stream_peek(pulse->pStream, &pu8Src, &cbAvail) < 0) { LogRel(("Pulse: Peek failed: %s\n", pa_strerror(pa_context_errno(g_pContext)))); goto exit; } if (!pu8Src) goto exit; csAvail = cbAvail >> hw->info.shift; csDecr = audio_MIN (csDead, csAvail); csSamples = csDecr; while (csSamples) { /* split request at the end of our samples buffer */ psDst = hw->conv_buf + hw->wpos; csRead = audio_MIN (csSamples, hw->samples - hw->wpos); hw->conv (psDst, pu8Src, csRead, &nominal_volume); hw->wpos = (hw->wpos + csRead) % hw->samples; csSamples -= csRead; pu8Src = (const void*)((uint8_t*)pu8Src + (csRead << hw->info.shift)); } pa_stream_drop(pulse->pStream); exit: pa_threaded_mainloop_unlock(g_pMainLoop); return csDecr; } static int pulse_read (SWVoiceIn *sw, void *buf, int size) { return audio_pcm_sw_read (sw, buf, size); } static int pulse_ctl_in (HWVoiceIn *hw, int cmd, ...) { return 0; } static void *pulse_audio_init (void) { int rc; rc = audioLoadPulseLib(); if (RT_FAILURE(rc)) { LogRel(("Pulse: Failed to load the PulseAudio shared library! Error %Rrc\n", rc)); return NULL; } if (!(g_pMainLoop = pa_threaded_mainloop_new())) { LogRel(("Pulse: Failed to allocate main loop: %s\n", pa_strerror(pa_context_errno(g_pContext)))); goto fail; } if (!(g_pContext = pa_context_new(pa_threaded_mainloop_get_api(g_pMainLoop), "VBox"))) { LogRel(("Pulse: Failed to allocate context: %s\n", pa_strerror(pa_context_errno(g_pContext)))); goto fail; } if (pa_threaded_mainloop_start(g_pMainLoop) < 0) { LogRel(("Pulse: Failed to start threaded mainloop: %s\n", pa_strerror(pa_context_errno(g_pContext)))); goto fail; } pa_context_set_state_callback(g_pContext, context_state_callback, NULL); pa_threaded_mainloop_lock(g_pMainLoop); if (pa_context_connect(g_pContext, /*server=*/NULL, 0, NULL) < 0) { LogRel(("Pulse: Failed to connect to server: %s\n", pa_strerror(pa_context_errno(g_pContext)))); goto unlock_and_fail; } /* Wait until the g_pContext is ready */ for (;;) { pa_context_state_t cstate; pa_threaded_mainloop_wait(g_pMainLoop); cstate = pa_context_get_state(g_pContext); if (cstate == PA_CONTEXT_READY) break; else if (cstate == PA_CONTEXT_TERMINATED || cstate == PA_CONTEXT_FAILED) { LogRel(("Pulse: Failed to initialize context (state %d)\n", cstate)); goto unlock_and_fail; } } pa_threaded_mainloop_unlock(g_pMainLoop); return &conf; unlock_and_fail: if (g_pMainLoop) pa_threaded_mainloop_unlock(g_pMainLoop); fail: if (g_pMainLoop) pa_threaded_mainloop_stop(g_pMainLoop); if (g_pContext) { pa_context_disconnect(g_pContext); pa_context_unref(g_pContext); g_pContext = NULL; } if (g_pMainLoop) { pa_threaded_mainloop_free(g_pMainLoop); g_pMainLoop = NULL; } return NULL; } static void pulse_audio_fini (void *opaque) { if (g_pMainLoop) pa_threaded_mainloop_stop(g_pMainLoop); if (g_pContext) { pa_context_disconnect(g_pContext); pa_context_unref(g_pContext); g_pContext = NULL; } if (g_pMainLoop) { pa_threaded_mainloop_free(g_pMainLoop); g_pMainLoop = NULL; } (void) opaque; } static struct audio_option pulse_options[] = { {"DAC_MS", AUD_OPT_INT, &conf.buffer_msecs_out, "DAC period size in milliseconds", NULL, 0}, {"ADC_MS", AUD_OPT_INT, &conf.buffer_msecs_in, "ADC period size in milliseconds", NULL, 0}, {NULL, 0, NULL, NULL, NULL, 0} }; static struct audio_pcm_ops pulse_pcm_ops = { pulse_init_out, pulse_fini_out, pulse_run_out, pulse_write, pulse_ctl_out, pulse_init_in, pulse_fini_in, pulse_run_in, pulse_read, pulse_ctl_in }; struct audio_driver pulse_audio_driver = { INIT_FIELD (name = ) "pulse", INIT_FIELD (descr = ) "PulseAudio http://www.pulseaudio.org", INIT_FIELD (options = ) pulse_options, INIT_FIELD (init = ) pulse_audio_init, INIT_FIELD (fini = ) pulse_audio_fini, INIT_FIELD (pcm_ops = ) &pulse_pcm_ops, INIT_FIELD (can_be_default = ) 1, INIT_FIELD (max_voices_out = ) INT_MAX, INIT_FIELD (max_voices_in = ) INT_MAX, INIT_FIELD (voice_size_out = ) sizeof (PulseVoice), INIT_FIELD (voice_size_in = ) sizeof (PulseVoice) };