merged 15a_aqemu.patch audio patch (malc)

git-svn-id: svn://svn.savannah.nongnu.org/qemu/trunk@1584 c046a42c-6fe2-441c-8c8c-71466251a162

Changelog

@@ -1,3 +1,10 @@
+version 0.7.3:
+
+  - Mac OS X cocoa improvements (Mike Kronenberg)
+  - DirectSound driver (malc)
+  - new audio options: '-soundhw' and 'audio-help' (malc)
+  - ES1370 PCI audio device (malc)
+   
 version 0.7.2:
   
   - x86_64 fixes (Win2000 and Linux 2.6 boot in 32 bit)

Makefile.target

@@ -262,7 +262,7 @@ endif
 VL_OBJS=vl.o osdep.o block.o readline.o monitor.o pci.o console.o
 VL_OBJS+=block-cow.o block-qcow.o aes.o block-vmdk.o block-cloop.o block-dmg.o block-bochs.o block-vpc.o block-vvfat.o
 
-SOUND_HW = sb16.o
+SOUND_HW = sb16.o es1370.o
 AUDIODRV = audio.o noaudio.o wavaudio.o
 ifdef CONFIG_SDL
 AUDIODRV += sdlaudio.o
@@ -270,29 +270,38 @@ endif
 ifdef CONFIG_OSS
 AUDIODRV += ossaudio.o
 endif
-
+ifdef CONFIG_COREAUDIO
-pc.o: DEFINES := -DUSE_SB16 $(DEFINES)
+AUDIODRV += coreaudio.o
-
+endif
-ifdef CONFIG_ADLIB
+ifdef CONFIG_ALSA
-SOUND_HW += fmopl.o adlib.o
+AUDIODRV += alsaaudio.o
+LIBS += -lasound
+endif
+ifdef CONFIG_DSOUND
+AUDIODRV += dsoundaudio.o
+LIBS += -lole32 -ldxguid
 endif
-
 ifdef CONFIG_FMOD
 AUDIODRV += fmodaudio.o
 audio.o fmodaudio.o: DEFINES := -I$(CONFIG_FMOD_INC) $(DEFINES)
 LIBS += $(CONFIG_FMOD_LIB)
 endif
+ifdef CONFIG_ADLIB
+SOUND_HW += fmopl.o adlib.o
+endif
 
 ifeq ($(TARGET_BASE_ARCH), i386)
 # Hardware support
 VL_OBJS+= ide.o ne2000.o pckbd.o vga.o $(SOUND_HW) dma.o $(AUDIODRV)
 VL_OBJS+= fdc.o mc146818rtc.o serial.o i8259.o i8254.o pc.o
 VL_OBJS+= cirrus_vga.o mixeng.o apic.o parallel.o
+DEFINES += -DHAS_AUDIO
 endif
 ifeq ($(TARGET_BASE_ARCH), ppc)
 VL_OBJS+= ppc.o ide.o ne2000.o pckbd.o vga.o $(SOUND_HW) dma.o $(AUDIODRV)
 VL_OBJS+= mc146818rtc.o serial.o i8259.o i8254.o fdc.o m48t59.o
 VL_OBJS+= ppc_prep.o ppc_chrp.o cuda.o adb.o openpic.o heathrow_pic.o mixeng.o
+DEFINES += -DHAS_AUDIO
 endif
 ifeq ($(TARGET_ARCH), mips)
 VL_OBJS+= mips_r4k.o dma.o vga.o serial.o ne2000.o i8254.o i8259.o
@@ -317,7 +326,10 @@ VL_OBJS+=sdl.o
 endif
 ifdef CONFIG_COCOA
 VL_OBJS+=cocoa.o
-COCOA_LIBS=-F/System/Library/Frameworks -framework Cocoa
+COCOA_LIBS=-F/System/Library/Frameworks -framework Cocoa -framework IOKit
+ifdef CONFIG_COREAUDIO
+COCOA_LIBS+=-framework CoreAudio
+endif
 endif
 ifdef CONFIG_SLIRP
 DEFINES+=-I$(SRC_PATH)/slirp
@@ -349,6 +361,10 @@ ifeq ($(ARCH),ia64)
 VL_LDFLAGS+=-Wl,-G0 -Wl,-T,$(SRC_PATH)/ia64.ld
 endif
 
+ifdef CONFIG_WIN32
+SDL_LIBS := $(filter-out -mwindows, $(SDL_LIBS)) -mconsole
+endif
+
 $(QEMU_SYSTEM): $(VL_OBJS) libqemu.a
 	$(CC) $(VL_LDFLAGS) -o $@ $^ $(LIBS) $(SDL_LIBS) $(COCOA_LIBS) $(VL_LIBS)
 
@@ -364,6 +380,9 @@ sdlaudio.o: sdlaudio.c
 depend: $(SRCS)
 	$(CC) -MM $(CFLAGS) $(DEFINES) $^ 1>.depend
 
+vldepend: $(VL_OBJS:.o=.c)
+	$(CC) -MM $(CFLAGS) $(DEFINES) $^ 1>.depend
+
 # libqemu 
 
 libqemu.a: $(LIBOBJS)
@@ -415,8 +434,6 @@ op.o: op.c op_template.c op_mem.c
 op_helper.o: op_helper_mem.c
 endif
 
-mixeng.o: mixeng.c mixeng.h mixeng_template.h
-
 %.o: %.c
 	$(CC) $(CFLAGS) $(DEFINES) -c -o $@ $<
 
@@ -434,3 +451,9 @@ endif
 ifneq ($(wildcard .depend),)
 include .depend
 endif
+
+ifeq (0, 1)
+audio.o sdlaudio.o dsoundaudio.o ossaudio.o wavaudio.o noaudio.o \
+fmodaudio.o alsaaudio.o mixeng.o: \
+CFLAGS := $(CFLAGS) -Wall -Werror -W -Wsign-compare
+endif

audio/alsaaudio.c

@@ -0,0 +1,926 @@
+/*
+ * 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 <alsa/asoundlib.h>
+#include "vl.h"
+
+#define AUDIO_CAP "alsa"
+#include "audio_int.h"
+
+typedef struct ALSAVoiceOut {
+    HWVoiceOut hw;
+    void *pcm_buf;
+    snd_pcm_t *handle;
+    int can_pause;
+    int was_enabled;
+} ALSAVoiceOut;
+
+typedef struct ALSAVoiceIn {
+    HWVoiceIn hw;
+    snd_pcm_t *handle;
+    void *pcm_buf;
+    int can_pause;
+} 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_overriden;
+    int period_size_in_overriden;
+
+    int buffer_size_out_overriden;
+    int period_size_out_overriden;
+} conf = {
+#ifdef HIGH_LATENCY
+    .size_in_usec_in = 1,
+    .size_in_usec_out = 1,
+#endif
+    .pcm_name_out = "hw:0,0",
+    .pcm_name_in = "hw:0,0",
+#ifdef HIGH_LATENCY
+    .buffer_size_in = 400000,
+    .period_size_in = 400000 / 4,
+    .buffer_size_out = 400000,
+    .period_size_out = 400000 / 4,
+#else
+#define DEFAULT_BUFFER_SIZE 1024
+#define DEFAULT_PERIOD_SIZE 256
+    .buffer_size_in = DEFAULT_BUFFER_SIZE,
+    .period_size_in = DEFAULT_PERIOD_SIZE,
+    .buffer_size_out = DEFAULT_BUFFER_SIZE,
+    .period_size_out = DEFAULT_PERIOD_SIZE,
+    .buffer_size_in_overriden = 0,
+    .buffer_size_out_overriden = 0,
+    .period_size_in_overriden = 0,
+    .period_size_out_overriden = 0,
+#endif
+    .threshold = 0
+};
+
+struct alsa_params_req {
+    int freq;
+    audfmt_e fmt;
+    int nchannels;
+    unsigned int buffer_size;
+    unsigned int period_size;
+};
+
+struct alsa_params_obt {
+    int freq;
+    audfmt_e fmt;
+    int nchannels;
+    int can_pause;
+    snd_pcm_uframes_t buffer_size;
+};
+
+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, "Can 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_anal_close (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 int alsa_write (SWVoiceOut *sw, void *buf, int len)
+{
+    return audio_pcm_sw_write (sw, buf, len);
+}
+
+static int 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;
+
+    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 (int 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;
+
+    default:
+        dolog ("Unrecognized audio format %d\n", alsafmt);
+        return -1;
+    }
+
+    return 0;
+}
+
+#ifdef DEBUG_MISMATCHES
+static void alsa_dump_info (struct alsa_params_req *req,
+                            struct alsa_params_obt *obt)
+{
+    dolog ("parameter | requested value | obtained value\n");
+    dolog ("format    |      %10d |     %10d\n", req->fmt, obt->fmt);
+    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: buffer size %ld\n", obt->buffer_size);
+}
+#endif
+
+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 ("Can 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 ("Can 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 ("Can 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, freq, nchannels;
+    const char *pcm_name = in ? conf.pcm_name_in : conf.pcm_name_out;
+    unsigned int period_size, buffer_size;
+    snd_pcm_uframes_t obt_buffer_size;
+    const char *typ = in ? "ADC" : "DAC";
+
+    freq = req->freq;
+    period_size = req->period_size;
+    buffer_size = req->buffer_size;
+    nchannels = req->nchannels;
+
+    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);
+        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 (!((in && conf.size_in_usec_in) || (!in && conf.size_in_usec_out))) {
+        if (!buffer_size) {
+            buffer_size = DEFAULT_BUFFER_SIZE;
+            period_size= DEFAULT_PERIOD_SIZE;
+        }
+    }
+
+    if (buffer_size) {
+        if ((in && conf.size_in_usec_in) || (!in && conf.size_in_usec_out)) {
+            if (period_size) {
+                err = snd_pcm_hw_params_set_period_time_near (
+                    handle,
+                    hw_params,
+                    &period_size,
+                    0);
+                if (err < 0) {
+                    alsa_logerr2 (err, typ,
+                                  "Failed to set period time %d\n",
+                                  req->period_size);
+                    goto err;
+                }
+            }
+
+            err = snd_pcm_hw_params_set_buffer_time_near (
+                handle,
+                hw_params,
+                &buffer_size,
+                0);
+
+            if (err < 0) {
+                alsa_logerr2 (err, typ,
+                              "Failed to set buffer time %d\n",
+                              req->buffer_size);
+                goto err;
+            }
+        }
+        else {
+            int dir;
+            snd_pcm_uframes_t minval;
+
+            if (period_size) {
+                minval = period_size;
+                dir = 0;
+
+                err = snd_pcm_hw_params_get_period_size_min (
+                    hw_params,
+                    &minval,
+                    &dir
+                    );
+                if (err < 0) {
+                    alsa_logerr (
+                        err,
+                        "Can not get minmal period size for %s\n",
+                        typ
+                        );
+                }
+                else {
+                    if (period_size < minval) {
+                        if ((in && conf.period_size_in_overriden)
+                            || (!in && conf.period_size_out_overriden)) {
+                            dolog ("%s period size(%d) is less "
+                                   "than minmal period size(%ld)\n",
+                                   typ,
+                                   period_size,
+                                   minval);
+                        }
+                        period_size = minval;
+                    }
+                }
+
+                err = snd_pcm_hw_params_set_period_size (
+                    handle,
+                    hw_params,
+                    period_size,
+                    0
+                    );
+                if (err < 0) {
+                    alsa_logerr2 (err, typ, "Failed to set period size %d\n",
+                                  req->period_size);
+                    goto err;
+                }
+            }
+
+            minval = buffer_size;
+            err = snd_pcm_hw_params_get_buffer_size_min (
+                hw_params,
+                &minval
+                );
+            if (err < 0) {
+                alsa_logerr (err, "Can not get minmal buffer size for %s\n",
+                             typ);
+            }
+            else {
+                if (buffer_size < minval) {
+                    if ((in && conf.buffer_size_in_overriden)
+                        || (!in && conf.buffer_size_out_overriden)) {
+                        dolog (
+                            "%s buffer size(%d) is less "
+                            "than minimal buffer size(%ld)\n",
+                            typ,
+                            buffer_size,
+                            minval
+                            );
+                    }
+                    buffer_size = minval;
+                }
+            }
+
+            err = snd_pcm_hw_params_set_buffer_size (
+                handle,
+                hw_params,
+                buffer_size
+                );
+            if (err < 0) {
+                alsa_logerr2 (err, typ, "Failed to set buffer size %d\n",
+                              req->buffer_size);
+                goto err;
+            }
+        }
+    }
+    else {
+        dolog ("warning: buffer size is not set\n");
+    }
+
+    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_prepare (handle);
+    if (err < 0) {
+        alsa_logerr2 (err, typ, "Can not prepare handle %p\n", handle);
+        goto err;
+    }
+
+    obt->can_pause = snd_pcm_hw_params_can_pause (hw_params);
+    if (obt->can_pause < 0) {
+        alsa_logerr (err, "Can not get pause capability for %s\n", typ);
+        obt->can_pause = 0;
+    }
+
+    if (!in && conf.threshold) {
+        snd_pcm_uframes_t threshold;
+        int bytes_per_sec;
+
+        bytes_per_sec = freq
+            << (nchannels == 2)
+            << (req->fmt == AUD_FMT_S16 || req->fmt == AUD_FMT_U16);
+
+        threshold = (conf.threshold * bytes_per_sec) / 1000;
+        alsa_set_threshold (handle, threshold);
+    }
+
+    obt->fmt = req->fmt;
+    obt->nchannels = nchannels;
+    obt->freq = freq;
+    obt->buffer_size = snd_pcm_frames_to_bytes (handle, obt_buffer_size);
+    *handlep = handle;
+
+    if (obt->fmt != req->fmt ||
+        obt->nchannels != req->nchannels ||
+        obt->freq != req->freq) {
+#ifdef DEBUG_MISMATCHES
+        dolog ("Audio paramters mismatch for %s\n", typ);
+        alsa_dump_info (req, obt);
+#endif
+    }
+
+#ifdef DEBUG
+    alsa_dump_info (req, obt);
+#endif
+    return 0;
+
+ err:
+    alsa_anal_close (&handle);
+    return -1;
+}
+
+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_run_out (HWVoiceOut *hw)
+{
+    ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
+    int rpos, live, decr;
+    int samples;
+    uint8_t *dst;
+    st_sample_t *src;
+    snd_pcm_sframes_t avail;
+
+    live = audio_pcm_hw_get_live_out (hw);
+    if (!live) {
+        return 0;
+    }
+
+    avail = snd_pcm_avail_update (alsa->handle);
+    if (avail < 0) {
+        if (avail == -EPIPE) {
+            if (!alsa_recover (alsa->handle)) {
+                avail = snd_pcm_avail_update (alsa->handle);
+                if (avail >= 0) {
+                    goto ok;
+                }
+            }
+        }
+
+        alsa_logerr (avail, "Can not get amount free space\n");
+        return 0;
+    }
+
+ ok:
+    decr = audio_MIN (live, avail);
+    samples = decr;
+    rpos = hw->rpos;
+    while (samples) {
+        int left_till_end_samples = hw->samples - rpos;
+        int convert_samples = audio_MIN (samples, left_till_end_samples);
+        snd_pcm_sframes_t written;
+
+        src = hw->mix_buf + rpos;
+        dst = advance (alsa->pcm_buf, rpos << hw->info.shift);
+
+        hw->clip (dst, src, convert_samples);
+
+    again:
+        written = snd_pcm_writei (alsa->handle, dst, convert_samples);
+
+        if (written < 0) {
+            switch (written) {
+            case -EPIPE:
+                if (!alsa_recover (alsa->handle)) {
+                    goto again;
+                }
+                dolog (
+                    "Failed to write %d frames to %p, handle %p not prepared\n",
+                    convert_samples,
+                    dst,
+                    alsa->handle
+                    );
+                goto exit;
+
+            case -EAGAIN:
+                goto again;
+
+            default:
+                alsa_logerr (written, "Failed to write %d frames to %p\n",
+                             convert_samples, dst);
+                goto exit;
+            }
+        }
+
+        mixeng_clear (src, written);
+        rpos = (rpos + written) % hw->samples;
+        samples -= written;
+    }
+
+ exit:
+    hw->rpos = rpos;
+    return decr;
+}
+
+static void alsa_fini_out (HWVoiceOut *hw)
+{
+    ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
+
+    ldebug ("alsa_fini\n");
+    alsa_anal_close (&alsa->handle);
+
+    if (alsa->pcm_buf) {
+        qemu_free (alsa->pcm_buf);
+        alsa->pcm_buf = NULL;
+    }
+}
+
+static int alsa_init_out (HWVoiceOut *hw, int freq, int nchannels, audfmt_e fmt)
+{
+    ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
+    struct alsa_params_req req;
+    struct alsa_params_obt obt;
+    audfmt_e effective_fmt;
+    int endianness;
+    int err;
+    snd_pcm_t *handle;
+
+    req.fmt = aud_to_alsafmt (fmt);
+    req.freq = freq;
+    req.nchannels = nchannels;
+    req.period_size = conf.period_size_out;
+    req.buffer_size = conf.buffer_size_out;
+
+    if (alsa_open (0, &req, &obt, &handle)) {
+        return -1;
+    }
+
+    err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness);
+    if (err) {
+        alsa_anal_close (&handle);
+        return -1;
+    }
+
+    audio_pcm_init_info (
+        &hw->info,
+        obt.freq,
+        obt.nchannels,
+        effective_fmt,
+        audio_need_to_swap_endian (endianness)
+        );
+    alsa->can_pause = obt.can_pause;
+    hw->bufsize = obt.buffer_size;
+
+    alsa->pcm_buf = qemu_mallocz (hw->bufsize);
+    if (!alsa->pcm_buf) {
+        alsa_anal_close (&handle);
+        return -1;
+    }
+
+    alsa->handle = handle;
+    alsa->was_enabled = 0;
+    return 0;
+}
+
+static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...)
+{
+    int err;
+    ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
+
+    switch (cmd) {
+    case VOICE_ENABLE:
+        ldebug ("enabling voice\n");
+        audio_pcm_info_clear_buf (&hw->info, alsa->pcm_buf, hw->samples);
+        if (alsa->can_pause) {
+            /* Why this was_enabled madness is needed at all?? */
+            if (alsa->was_enabled) {
+                err = snd_pcm_pause (alsa->handle, 0);
+                if (err < 0) {
+                    alsa_logerr (err, "Failed to resume playing\n");
+                    /* not fatal really */
+                }
+            }
+            else {
+                alsa->was_enabled = 1;
+            }
+        }
+        break;
+
+    case VOICE_DISABLE:
+        ldebug ("disabling voice\n");
+        if (alsa->can_pause) {
+            err = snd_pcm_pause (alsa->handle, 1);
+            if (err < 0) {
+                alsa_logerr (err, "Failed to stop playing\n");
+                /* not fatal really */
+            }
+        }
+        break;
+    }
+    return 0;
+}
+
+static int alsa_init_in (HWVoiceIn *hw,
+                        int freq, int nchannels, audfmt_e fmt)
+{
+    ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
+    struct alsa_params_req req;
+    struct alsa_params_obt obt;
+    int endianness;
+    int err;
+    audfmt_e effective_fmt;
+    snd_pcm_t *handle;
+
+    req.fmt = aud_to_alsafmt (fmt);
+    req.freq = freq;
+    req.nchannels = nchannels;
+    req.period_size = conf.period_size_in;
+    req.buffer_size = conf.buffer_size_in;
+
+    if (alsa_open (1, &req, &obt, &handle)) {
+        return -1;
+    }
+
+    err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness);
+    if (err) {
+        alsa_anal_close (&handle);
+        return -1;
+    }
+
+    audio_pcm_init_info (
+        &hw->info,
+        obt.freq,
+        obt.nchannels,
+        effective_fmt,
+        audio_need_to_swap_endian (endianness)
+        );
+    alsa->can_pause = obt.can_pause;
+    hw->bufsize = obt.buffer_size;
+    alsa->pcm_buf = qemu_mallocz (hw->bufsize);
+    if (!alsa->pcm_buf) {
+        alsa_anal_close (&handle);
+        return -1;
+    }
+
+    alsa->handle = handle;
+    return 0;
+}
+
+static void alsa_fini_in (HWVoiceIn *hw)
+{
+    ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
+
+    alsa_anal_close (&alsa->handle);
+
+    if (alsa->pcm_buf) {
+        qemu_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;
+    struct {
+        int add;
+        int len;
+    } bufs[2] = {
+        { hw->wpos, 0 },
+        { 0, 0 }
+    };
+
+    snd_pcm_uframes_t read_samples = 0;
+
+    if (!dead) {
+        return 0;
+    }
+
+    if (hw->wpos + dead > hw->samples) {
+        bufs[0].len = (hw->samples - hw->wpos);
+        bufs[1].len = (dead - (hw->samples - hw->wpos));
+    }
+    else {
+        bufs[0].len = dead;
+    }
+
+
+    for (i = 0; i < 2; ++i) {
+        void *src;
+        st_sample_t *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 -EPIPE:
+                    if (!alsa_recover (alsa->handle)) {
+                        continue;
+                    }
+                    dolog (
+                        "Failed to read %ld frames from %p, "
+                        "handle %p not prepared\n",
+                        len,
+                        src,
+                        alsa->handle
+                        );
+                    goto exit;
+
+                case -EAGAIN:
+                    continue;
+
+                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, ...)
+{
+    (void) hw;
+    (void) cmd;
+    return 0;
+}
+
+static void *alsa_audio_init (void)
+{
+    return &conf;
+}
+
+static void alsa_audio_fini (void *opaque)
+{
+    (void) opaque;
+}
+
+static struct audio_option alsa_options[] = {
+    {"DAC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_out,
+     "DAC period/buffer size in microseconds (otherwise in frames)", NULL, 0},
+    {"DAC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_out,
+     "DAC period size", &conf.period_size_out_overriden, 0},
+    {"DAC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_out,
+     "DAC buffer size", &conf.buffer_size_out_overriden, 0},
+
+    {"ADC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_in,
+     "ADC period/buffer size in microseconds (otherwise in frames)", NULL, 0},
+    {"ADC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_in,
+     "ADC period size", &conf.period_size_in_overriden, 0},
+    {"ADC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_in,
+     "ADC buffer size", &conf.buffer_size_in_overriden, 0},
+
+    {"THRESHOLD", AUD_OPT_INT, &conf.threshold,
+     "(undocumented)", NULL, 0},
+
+    {"DAC_DEV", AUD_OPT_STR, &conf.pcm_name_out,
+     "DAC device name (for instance dmix)", NULL, 0},
+
+    {"ADC_DEV", AUD_OPT_STR, &conf.pcm_name_in,
+     "ADC device name", NULL, 0},
+    {NULL, 0, NULL, NULL, NULL, 0}
+};
+
+static struct audio_pcm_ops alsa_pcm_ops = {
+    alsa_init_out,
+    alsa_fini_out,
+    alsa_run_out,
+    alsa_write,
+    alsa_ctl_out,
+
+    alsa_init_in,
+    alsa_fini_in,
+    alsa_run_in,
+    alsa_read,
+    alsa_ctl_in
+};
+
+struct audio_driver alsa_audio_driver = {
+    INIT_FIELD (name           = ) "alsa",
+    INIT_FIELD (descr          = ) "ALSA http://www.alsa-project.org",
+    INIT_FIELD (options        = ) alsa_options,
+    INIT_FIELD (init           = ) alsa_audio_init,
+    INIT_FIELD (fini           = ) alsa_audio_fini,
+    INIT_FIELD (pcm_ops        = ) &alsa_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 (ALSAVoiceOut),
+    INIT_FIELD (voice_size_in  = ) sizeof (ALSAVoiceIn)
+};

audio/audio.h

@@ -1,7 +1,7 @@
 /*
  * QEMU Audio subsystem header
  *
- * Copyright (c) 2003-2004 Vassili Karpov (malc)
+ * Copyright (c) 2003-2005 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
@@ -24,7 +24,7 @@
 #ifndef QEMU_AUDIO_H
 #define QEMU_AUDIO_H
 
-#include "mixeng.h"
+typedef void (*audio_callback_fn_t) (void *opaque, int avail);
 
 typedef enum {
   AUD_FMT_U8,
@@ -33,22 +33,60 @@ typedef enum {
   AUD_FMT_S16
 } audfmt_e;
 
-typedef struct SWVoice SWVoice;
+typedef struct SWVoiceOut SWVoiceOut;
+typedef struct SWVoiceIn SWVoiceIn;
 
-SWVoice * AUD_open (SWVoice *sw, const char *name, int freq,
+typedef struct QEMUAudioTimeStamp {
-                    int nchannels, audfmt_e fmt);
+    uint64_t old_ts;
-void   AUD_init (void);
+} QEMUAudioTimeStamp;
+
+void AUD_vlog (const char *cap, const char *fmt, va_list ap);
 void AUD_log (const char *cap, const char *fmt, ...)
-    __attribute__ ((__format__ (__printf__, 2, 3)));;
+#ifdef __GNUC__
-void   AUD_close (SWVoice *sw);
+    __attribute__ ((__format__ (__printf__, 2, 3)))
-int    AUD_write (SWVoice *sw, void *pcm_buf, int size);
+#endif
-void   AUD_adjust (SWVoice *sw, int leftover);
+    ;
-void   AUD_reset (SWVoice *sw);
+
-int    AUD_get_free (SWVoice *sw);
+void AUD_init (void);
-int    AUD_get_buffer_size (SWVoice *sw);
+void AUD_help (void);
-void   AUD_run (void);
+
-void   AUD_enable (SWVoice *sw, int on);
+SWVoiceOut  *AUD_open_out (
-int    AUD_calc_elapsed (SWVoice *sw);
+    SWVoiceOut *sw,
+    const char *name,
+    void *callback_opaque,
+    audio_callback_fn_t callback_fn,
+    int freq,
+    int nchannels,
+    audfmt_e fmt
+    );
+void         AUD_close_out (SWVoiceOut *sw);
+int          AUD_write (SWVoiceOut *sw, void *pcm_buf, int size);
+int          AUD_get_buffer_size_out (SWVoiceOut *sw);
+void         AUD_set_active_out (SWVoiceOut *sw, int on);
+int          AUD_is_active_out (SWVoiceOut *sw);
+void         AUD_init_time_stamp_out (SWVoiceOut *sw,
+                                      QEMUAudioTimeStamp *ts);
+uint64_t     AUD_time_stamp_get_elapsed_usec_out (SWVoiceOut *sw,
+                                                  QEMUAudioTimeStamp *ts);
+
+SWVoiceIn   *AUD_open_in (
+    SWVoiceIn *sw,
+    const char *name,
+    void *callback_opaque,
+    audio_callback_fn_t callback_fn,
+    int freq,
+    int nchannels,
+    audfmt_e fmt
+    );
+void         AUD_close_in (SWVoiceIn *sw);
+int          AUD_read (SWVoiceIn *sw, void *pcm_buf, int size);
+void         AUD_adjust_in (SWVoiceIn *sw, int leftover);
+void         AUD_set_active_in (SWVoiceIn *sw, int on);
+int          AUD_is_active_in (SWVoiceIn *sw);
+void         AUD_init_time_stamp_in (SWVoiceIn *sw,
+                                     QEMUAudioTimeStamp *ts);
+uint64_t     AUD_time_stamp_get_elapsed_usec_in (SWVoiceIn *sw,
+                                                 QEMUAudioTimeStamp *ts);
 
 static inline void *advance (void *p, int incr)
 {
@@ -59,7 +97,21 @@ static inline void *advance (void *p, int incr)
 uint32_t popcount (uint32_t u);
 inline uint32_t lsbindex (uint32_t u);
 
+#ifdef __GNUC__
+#define audio_MIN(a, b) ( __extension__ ({      \
+    __typeof (a) ta = a;                        \
+    __typeof (b) tb = b;                        \
+    ((ta)>(tb)?(tb):(ta));                      \
+}))
+
+#define audio_MAX(a, b) ( __extension__ ({      \
+    __typeof (a) ta = a;                        \
+    __typeof (b) tb = b;                        \
+    ((ta)<(tb)?(tb):(ta));                      \
+}))
+#else
 #define audio_MIN(a, b) ((a)>(b)?(b):(a))
 #define audio_MAX(a, b) ((a)<(b)?(b):(a))
+#endif
 
 #endif  /* audio.h */

audio/audio_int.h

@@ -1,7 +1,7 @@
 /*
  * QEMU Audio subsystem header
  *
- * Copyright (c) 2003-2004 Vassili Karpov (malc)
+ * Copyright (c) 2003-2005 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
@@ -24,140 +24,266 @@
 #ifndef QEMU_AUDIO_INT_H
 #define QEMU_AUDIO_INT_H
 
-#include "vl.h"
+#include "sys-queue.h"
 
-struct pcm_ops;
+#ifdef CONFIG_COREAUDIO
+#define FLOAT_MIXENG
+/* #define RECIPROCAL */
+#endif
+#include "mixeng.h"
 
-typedef struct HWVoice {
+int audio_bug (const char *funcname, int cond);
+
+struct audio_pcm_ops;
+
+typedef enum {
+    AUD_OPT_INT,
+    AUD_OPT_FMT,
+    AUD_OPT_STR,
+    AUD_OPT_BOOL
+} audio_option_tag_e;
+
+struct audio_option {
+    const char *name;
+    audio_option_tag_e tag;
+    void *valp;
+    const char *descr;
+    int *overridenp;
+    int overriden;
+};
+
+struct audio_callback {
+    void *opaque;
+    audio_callback_fn_t fn;
+};
+
+struct audio_pcm_info {
+    int bits;
+    int sign;
+    int freq;
+    int nchannels;
+    int align;
+    int shift;
+    int bytes_per_second;
+    int swap_endian;
+};
+
+typedef struct HWVoiceOut {
     int active;
     int enabled;
     int pending_disable;
     int valid;
-    int freq;
+    struct audio_pcm_info info;
 
     f_sample *clip;
-    audfmt_e fmt;
-    int nchannels;
-
-    int align;
-    int shift;
 
     int rpos;
     int bufsize;
+    uint64_t ts_helper;
 
-    int bytes_per_second;
     st_sample_t *mix_buf;
 
     int samples;
-    int64_t old_ticks;
+    LIST_HEAD (sw_out_listhead, SWVoiceOut) sw_head;
-    int nb_voices;
+    struct audio_pcm_ops *pcm_ops;
-    struct SWVoice **pvoice;
+    LIST_ENTRY (HWVoiceOut) entries;
-    struct pcm_ops *pcm_ops;
+} HWVoiceOut;
-} HWVoice;
 
-extern struct pcm_ops no_pcm_ops;
+typedef struct HWVoiceIn {
-extern struct audio_output_driver no_output_driver;
+    int enabled;
-
+    int active;
-extern struct pcm_ops oss_pcm_ops;
+    struct audio_pcm_info info;
-extern struct audio_output_driver oss_output_driver;
-
-extern struct pcm_ops sdl_pcm_ops;
-extern struct audio_output_driver sdl_output_driver;
-
-extern struct pcm_ops wav_pcm_ops;
-extern struct audio_output_driver wav_output_driver;
-
-extern struct pcm_ops fmod_pcm_ops;
-extern struct audio_output_driver fmod_output_driver;
-
-struct audio_output_driver {
-    const char *name;
-    void *(*init) (void);
-    void (*fini) (void *);
-    struct pcm_ops *pcm_ops;
-    int can_be_default;
-    int max_voices;
-    int voice_size;
-};
-
-typedef struct AudioState {
-    int fixed_format;
-    int fixed_freq;
-    int fixed_channels;
-    int fixed_fmt;
-    int nb_hw_voices;
-    int64_t ticks_threshold;
-    int freq_threshold;
-    void *opaque;
-    struct audio_output_driver *drv;
-} AudioState;
-extern AudioState audio_state;
-
-struct SWVoice {
-    int freq;
-    audfmt_e fmt;
-    int nchannels;
-
-    int shift;
-    int align;
 
     t_sample *conv;
 
-    int left;
+    int wpos;
-    int pos;
+    int bufsize;
-    int bytes_per_second;
+    int total_samples_captured;
+    uint64_t ts_helper;
+
+    st_sample_t *conv_buf;
+
+    int samples;
+    LIST_HEAD (sw_in_listhead, SWVoiceIn) sw_head;
+    struct audio_pcm_ops *pcm_ops;
+    LIST_ENTRY (HWVoiceIn) entries;
+} HWVoiceIn;
+
+extern struct audio_driver no_audio_driver;
+extern struct audio_driver oss_audio_driver;
+extern struct audio_driver sdl_audio_driver;
+extern struct audio_driver wav_audio_driver;
+extern struct audio_driver fmod_audio_driver;
+extern struct audio_driver alsa_audio_driver;
+extern struct audio_driver coreaudio_audio_driver;
+extern struct audio_driver dsound_audio_driver;
+extern volume_t nominal_volume;
+
+struct audio_driver {
+    const char *name;
+    const char *descr;
+    struct audio_option *options;
+    void *(*init) (void);
+    void (*fini) (void *);
+    struct audio_pcm_ops *pcm_ops;
+    int can_be_default;
+    int max_voices_out;
+    int max_voices_in;
+    int voice_size_out;
+    int voice_size_in;
+};
+
+typedef struct AudioState {
+    int fixed_settings_out;
+    int fixed_freq_out;
+    int fixed_channels_out;
+    int fixed_fmt_out;
+    int nb_hw_voices_out;
+    int greedy_out;
+
+    int fixed_settings_in;
+    int fixed_freq_in;
+    int fixed_channels_in;
+    int fixed_fmt_in;
+    int nb_hw_voices_in;
+    int greedy_in;
+
+    void *opaque;
+    struct audio_driver *drv;
+
+    QEMUTimer *ts;
+    union {
+        int usec;
+        int64_t ticks;
+    } period;
+
+    int plive;
+} AudioState;
+extern AudioState audio_state;
+
+struct SWVoiceOut {
+    struct audio_pcm_info info;
+    t_sample *conv;
     int64_t ratio;
     st_sample_t *buf;
     void *rate;
-
+    int total_hw_samples_mixed;
-    int wpos;
-    int live;
     int active;
-    int64_t old_ticks;
+    int empty;
-    HWVoice *hw;
+    HWVoiceOut *hw;
     char *name;
+    volume_t vol;
+    struct audio_callback callback;
+    LIST_ENTRY (SWVoiceOut) entries;
 };
 
-struct pcm_ops {
+struct SWVoiceIn {
-    int  (*init)  (HWVoice *hw, int freq, int nchannels, audfmt_e fmt);
+    int active;
-    void (*fini)  (HWVoice *hw);
+    struct audio_pcm_info info;
-    void (*run)   (HWVoice *hw);
+    int64_t ratio;
-    int  (*write) (SWVoice *sw, void *buf, int size);
+    void *rate;
-    int  (*ctl)   (HWVoice *hw, int cmd, ...);
+    int total_hw_samples_acquired;
+    st_sample_t *conv_buf;
+    f_sample *clip;
+    HWVoiceIn *hw;
+    char *name;
+    volume_t vol;
+    struct audio_callback callback;
+    LIST_ENTRY (SWVoiceIn) entries;
 };
 
-void      pcm_sw_free_resources (SWVoice *sw);
+struct audio_pcm_ops {
-int       pcm_sw_alloc_resources (SWVoice *sw);
+    int  (*init_out)(HWVoiceOut *hw, int freq, int nchannels, audfmt_e fmt);
-void      pcm_sw_fini (SWVoice *sw);
+    void (*fini_out)(HWVoiceOut *hw);
-int       pcm_sw_init (SWVoice *sw, HWVoice *hw, int freq,
+    int  (*run_out) (HWVoiceOut *hw);
-                       int nchannels, audfmt_e fmt);
+    int  (*write)   (SWVoiceOut *sw, void *buf, int size);
+    int  (*ctl_out) (HWVoiceOut *hw, int cmd, ...);
 
-void      pcm_hw_clear (HWVoice *hw, void *buf, int len);
+    int  (*init_in) (HWVoiceIn *hw, int freq, int nchannels, audfmt_e fmt);
-HWVoice * pcm_hw_find_any (HWVoice *hw);
+    void (*fini_in) (HWVoiceIn *hw);
-HWVoice * pcm_hw_find_any_active (HWVoice *hw);
+    int  (*run_in)  (HWVoiceIn *hw);
-HWVoice * pcm_hw_find_any_passive (HWVoice *hw);
+    int  (*read)    (SWVoiceIn *sw, void *buf, int size);
-HWVoice * pcm_hw_find_specific (HWVoice *hw, int freq,
+    int  (*ctl_in)  (HWVoiceIn *hw, int cmd, ...);
-                                int nchannels, audfmt_e fmt);
+};
-HWVoice * pcm_hw_add (int freq, int nchannels, audfmt_e fmt);
-int       pcm_hw_add_sw (HWVoice *hw, SWVoice *sw);
-int       pcm_hw_del_sw (HWVoice *hw, SWVoice *sw);
-SWVoice * pcm_create_voice_pair (int freq, int nchannels, audfmt_e fmt);
 
-void      pcm_hw_free_resources (HWVoice *hw);
+void audio_pcm_init_info (struct audio_pcm_info *info, int freq,
-int       pcm_hw_alloc_resources (HWVoice *hw);
+                          int nchannels, audfmt_e fmt, int swap_endian);
-void      pcm_hw_fini (HWVoice *hw);
+void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len);
-void      pcm_hw_gc (HWVoice *hw);
-int       pcm_hw_get_live (HWVoice *hw);
-int       pcm_hw_get_live2 (HWVoice *hw, int *nb_active);
-void      pcm_hw_dec_live (HWVoice *hw, int decr);
-int       pcm_hw_write (SWVoice *sw, void *buf, int len);
 
-int         audio_get_conf_int (const char *key, int defval);
+int  audio_pcm_sw_write (SWVoiceOut *sw, void *buf, int len);
-const char *audio_get_conf_str (const char *key, const char *defval);
+int  audio_pcm_hw_get_live_in (HWVoiceIn *hw);
 
-struct audio_output_driver;
+int  audio_pcm_sw_read (SWVoiceIn *sw, void *buf, int len);
+int  audio_pcm_hw_get_live_out (HWVoiceOut *hw);
+int  audio_pcm_hw_get_live_out2 (HWVoiceOut *hw, int *nb_live);
 
 #define VOICE_ENABLE 1
 #define VOICE_DISABLE 2
 
+static inline int audio_ring_dist (int dst, int src, int len)
+{
+    return (dst >= src) ? (dst - src) : (len - src + dst);
+}
+
+static inline int audio_need_to_swap_endian (int endianness)
+{
+#ifdef WORDS_BIGENDIAN
+    return endianness != 1;
+#else
+    return endianness != 0;
+#endif
+}
+
+#if defined __GNUC__
+#define GCC_ATTR __attribute__ ((__unused__, __format__ (__printf__, 1, 2)))
+#define INIT_FIELD(f) . f
+#define GCC_FMT_ATTR(n, m) __attribute__ ((__format__ (printf, n, m)))
+#else
+#define GCC_ATTR /**/
+#define INIT_FIELD(f) /**/
+#define GCC_FMT_ATTR(n, m)
+#endif
+
+static void GCC_ATTR dolog (const char *fmt, ...)
+{
+    va_list ap;
+
+    va_start (ap, fmt);
+    AUD_vlog (AUDIO_CAP, fmt, ap);
+    va_end (ap);
+}
+
+#ifdef DEBUG
+static void GCC_ATTR ldebug (const char *fmt, ...)
+{
+    va_list ap;
+
+    va_start (ap, fmt);
+    AUD_vlog (AUDIO_CAP, fmt, ap);
+    va_end (ap);
+}
+#else
+#if defined NDEBUG && defined __GNUC__
+#define ldebug(...)
+#elif defined NDEBUG && defined _MSC_VER
+#define ldebug __noop
+#else
+static void GCC_ATTR ldebug (const char *fmt, ...)
+{
+    (void) fmt;
+}
+#endif
+#endif
+
+#undef GCC_ATTR
+
+#define AUDIO_STRINGIFY_(n) #n
+#define AUDIO_STRINGIFY(n) AUDIO_STRINGIFY_(n)
+
+#if defined _MSC_VER || defined __GNUC__
+#define AUDIO_FUNC __FUNCTION__
+#else
+#define AUDIO_FUNC __FILE__ ":" AUDIO_STRINGIFY (__LINE__)
+#endif
+
 #endif /* audio_int.h */

audio/audio_template.h

@@ -0,0 +1,401 @@
+/*
+ * QEMU Audio subsystem header
+ *
+ * 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.
+ */
+
+#ifdef DAC
+#define TYPE out
+#define HW glue (HWVoice, Out)
+#define SW glue (SWVoice, Out)
+#else
+#define TYPE in
+#define HW glue (HWVoice, In)
+#define SW glue (SWVoice, In)
+#endif
+
+static void glue (audio_pcm_sw_fini_, TYPE) (SW *sw)
+{
+    glue (audio_pcm_sw_free_resources_, TYPE) (sw);
+    if (sw->name) {
+        qemu_free (sw->name);
+        sw->name = NULL;
+    }
+}
+
+static void glue (audio_pcm_hw_add_sw_, TYPE) (HW *hw, SW *sw)
+{
+    LIST_INSERT_HEAD (&hw->sw_head, sw, entries);
+}
+
+static void glue (audio_pcm_hw_del_sw_, TYPE) (SW *sw)
+{
+    LIST_REMOVE (sw, entries);
+}
+
+static void glue (audio_pcm_hw_fini_, TYPE) (HW *hw)
+{
+    if (hw->active) {
+        glue (audio_pcm_hw_free_resources_ ,TYPE) (hw);
+        glue (hw->pcm_ops->fini_, TYPE) (hw);
+        memset (hw, 0, glue (audio_state.drv->voice_size_, TYPE));
+    }
+}
+
+static void glue (audio_pcm_hw_gc_, TYPE) (HW *hw)
+{
+    if (!hw->sw_head.lh_first) {
+        glue (audio_pcm_hw_fini_, TYPE) (hw);
+    }
+}
+
+static HW *glue (audio_pcm_hw_find_any_, TYPE) (HW *hw)
+{
+    return hw ? hw->entries.le_next : glue (hw_head_, TYPE).lh_first;
+}
+
+static HW *glue (audio_pcm_hw_find_any_active_, TYPE) (HW *hw)
+{
+    while ((hw = glue (audio_pcm_hw_find_any_, TYPE) (hw))) {
+        if (hw->active) {
+            return hw;
+        }
+    }
+    return NULL;
+}
+
+static HW *glue (audio_pcm_hw_find_any_active_enabled_, TYPE) (HW *hw)
+{
+    while ((hw = glue (audio_pcm_hw_find_any_, TYPE) (hw))) {
+        if (hw->active && hw->enabled) {
+            return hw;
+        }
+    }
+    return NULL;
+}
+
+static HW *glue (audio_pcm_hw_find_any_passive_, TYPE) (HW *hw)
+{
+    while ((hw = glue (audio_pcm_hw_find_any_, TYPE) (hw))) {
+        if (!hw->active) {
+            return hw;
+        }
+    }
+    return NULL;
+}
+
+static HW *glue (audio_pcm_hw_find_specific_, TYPE) (
+    HW *hw,
+    int freq,
+    int nchannels,
+    audfmt_e fmt
+    )
+{
+    while ((hw = glue (audio_pcm_hw_find_any_active_, TYPE) (hw))) {
+        if (audio_pcm_info_eq (&hw->info, freq, nchannels, fmt)) {
+            return hw;
+        }
+    }
+    return NULL;
+}
+
+static HW *glue (audio_pcm_hw_add_new_, TYPE) (
+    int freq,
+    int nchannels,
+    audfmt_e fmt
+    )
+{
+    HW *hw;
+
+    hw = glue (audio_pcm_hw_find_any_passive_, TYPE) (NULL);
+    if (hw) {
+        hw->pcm_ops = audio_state.drv->pcm_ops;
+        if (!hw->pcm_ops) {
+            return NULL;
+        }
+
+        if (glue (audio_pcm_hw_init_, TYPE) (hw, freq, nchannels, fmt)) {
+            glue (audio_pcm_hw_gc_, TYPE) (hw);
+            return NULL;
+        }
+        else {
+            return hw;
+        }
+    }
+
+    return NULL;
+}
+
+static HW *glue (audio_pcm_hw_add_, TYPE) (
+    int freq,
+    int nchannels,
+    audfmt_e fmt
+    )
+{
+    HW *hw;
+
+    if (glue (audio_state.greedy_, TYPE)) {
+        hw = glue (audio_pcm_hw_add_new_, TYPE) (freq, nchannels, fmt);
+        if (hw) {
+            return hw;
+        }
+    }
+
+    hw = glue (audio_pcm_hw_find_specific_, TYPE) (NULL, freq, nchannels, fmt);
+    if (hw) {
+        return hw;
+    }
+
+    hw = glue (audio_pcm_hw_add_new_, TYPE) (freq, nchannels, fmt);
+    if (hw) {
+        return hw;
+    }
+
+    return glue (audio_pcm_hw_find_any_active_, TYPE) (NULL);
+}
+
+static SW *glue (audio_pcm_create_voice_pair_, TYPE) (
+    const char *name,
+    int freq,
+    int nchannels,
+    audfmt_e fmt
+    )
+{
+    SW *sw;
+    HW *hw;
+    int hw_freq = freq;
+    int hw_nchannels = nchannels;
+    int hw_fmt = fmt;
+
+    if (glue (audio_state.fixed_settings_, TYPE)) {
+        hw_freq = glue (audio_state.fixed_freq_, TYPE);
+        hw_nchannels = glue (audio_state.fixed_channels_, TYPE);
+        hw_fmt = glue (audio_state.fixed_fmt_, TYPE);
+    }
+
+    sw = qemu_mallocz (sizeof (*sw));
+    if (!sw) {
+        goto err1;
+    }
+
+    hw = glue (audio_pcm_hw_add_, TYPE) (hw_freq, hw_nchannels, hw_fmt);
+    if (!hw) {
+        goto err2;
+    }
+
+    glue (audio_pcm_hw_add_sw_, TYPE) (hw, sw);
+
+    if (glue (audio_pcm_sw_init_, TYPE) (sw, hw, name, freq, nchannels, fmt)) {
+        goto err3;
+    }
+
+    return sw;
+
+err3:
+    glue (audio_pcm_hw_del_sw_, TYPE) (sw);
+    glue (audio_pcm_hw_gc_, TYPE) (hw);
+err2:
+    qemu_free (sw);
+err1:
+    return NULL;
+}
+
+void glue (AUD_close_, TYPE) (SW *sw)
+{
+    if (sw) {
+        glue (audio_pcm_sw_fini_, TYPE) (sw);
+        glue (audio_pcm_hw_del_sw_, TYPE) (sw);
+        glue (audio_pcm_hw_gc_, TYPE) (sw->hw);
+        qemu_free (sw);
+    }
+}
+
+SW *glue (AUD_open_, TYPE) (
+    SW *sw,
+    const char *name,
+    void *callback_opaque ,
+    audio_callback_fn_t callback_fn,
+    int freq,
+    int nchannels,
+    audfmt_e fmt
+    )
+{
+#ifdef DAC
+    int live = 0;
+    SW *old_sw = NULL;
+#endif
+
+    if (!callback_fn) {
+        dolog ("No callback specifed for voice `%s'\n", name);
+        goto fail;
+    }
+
+    if (nchannels != 1 && nchannels != 2) {
+        dolog ("Bogus channel count %d for voice `%s'\n", nchannels, name);
+        goto fail;
+    }
+
+    if (!audio_state.drv) {
+        dolog ("No audio driver defined\n");
+        goto fail;
+    }
+
+    if (sw && audio_pcm_info_eq (&sw->info, freq, nchannels, fmt)) {
+        return sw;
+    }
+
+#ifdef DAC
+    if (audio_state.plive && sw && (!sw->active && !sw->empty)) {
+        live = sw->total_hw_samples_mixed;
+
+#ifdef DEBUG_PLIVE
+        dolog ("Replacing voice %s with %d live samples\n", sw->name, live);
+        dolog ("Old %s freq %d, bits %d, channels %d\n",
+               sw->name, sw->info.freq, sw->info.bits, sw->info.nchannels);
+        dolog ("New %s freq %d, bits %d, channels %d\n",
+               name, freq, (fmt == AUD_FMT_S16 || fmt == AUD_FMT_U16) ? 16 : 8,
+               nchannels);
+#endif
+
+        if (live) {
+            old_sw = sw;
+            old_sw->callback.fn = NULL;
+            sw = NULL;
+        }
+    }
+#endif
+
+    if (!glue (audio_state.fixed_settings_, TYPE) && sw) {
+        glue (AUD_close_, TYPE) (sw);
+        sw = NULL;
+    }
+
+    if (sw) {
+        HW *hw = sw->hw;
+
+        if (!hw) {
+            dolog ("Internal logic error voice %s has no hardware store\n",
+                   name);
+            goto fail;
+        }
+
+        if (glue (audio_pcm_sw_init_, TYPE) (
+                sw,
+                hw,
+                name,
+                freq,
+                nchannels,
+                fmt
+                )) {
+            goto fail;
+        }
+    }
+    else {
+        sw = glue (audio_pcm_create_voice_pair_, TYPE) (
+            name,
+            freq,
+            nchannels,
+            fmt);
+        if (!sw) {
+            dolog ("Failed to create voice %s\n", name);
+            goto fail;
+        }
+    }
+
+    if (sw) {
+        sw->vol = nominal_volume;
+        sw->callback.fn = callback_fn;
+        sw->callback.opaque = callback_opaque;
+
+#ifdef DAC
+        if (live) {
+            int mixed =
+                (live << old_sw->info.shift)
+                * old_sw->info.bytes_per_second
+                / sw->info.bytes_per_second;
+
+#ifdef DEBUG_PLIVE
+            dolog ("Silence will be mixed %d\n", mixed);
+#endif
+            sw->total_hw_samples_mixed += mixed;
+        }
+#endif
+
+#ifdef DEBUG_AUDIO
+        dolog ("%s\n", name);
+        audio_pcm_print_info ("hw", &sw->hw->info);
+        audio_pcm_print_info ("sw", &sw->info);
+#endif
+    }
+
+    return sw;
+
+ fail:
+    glue (AUD_close_, TYPE) (sw);
+    return NULL;
+}
+
+int glue (AUD_is_active_, TYPE) (SW *sw)
+{
+    return sw ? sw->active : 0;
+}
+
+void glue (AUD_init_time_stamp_, TYPE) (SW *sw, QEMUAudioTimeStamp *ts)
+{
+    if (!sw) {
+        return;
+    }
+
+    ts->old_ts = sw->hw->ts_helper;
+}
+
+uint64_t glue (AUD_time_stamp_get_elapsed_usec_, TYPE) (
+    SW *sw,
+    QEMUAudioTimeStamp *ts
+    )
+{
+    uint64_t delta, cur_ts, old_ts;
+
+    if (!sw) {
+        return 0;
+    }
+
+    cur_ts = sw->hw->ts_helper;
+    old_ts = ts->old_ts;
+    /* dolog ("cur %lld old %lld\n", cur_ts, old_ts); */
+
+    if (cur_ts >= old_ts) {
+        delta = cur_ts - old_ts;
+    }
+    else {
+        delta = UINT64_MAX - old_ts + cur_ts;
+    }
+
+    if (!delta) {
+        return 0;
+    }
+
+    return (delta * sw->hw->info.freq) / 1000000;
+}
+
+#undef TYPE
+#undef HW
+#undef SW

audio/coreaudio.c

@@ -0,0 +1,513 @@
+/*
+ * QEMU OS X CoreAudio audio driver
+ *
+ * Copyright (c) 2005 Mike Kronenberg
+ *
+ * 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 <CoreAudio/CoreAudio.h>
+#include <string.h>             /* strerror */
+#include <pthread.h>            /* pthread_X */
+
+#include "vl.h"
+
+#define AUDIO_CAP "coreaudio"
+#include "audio_int.h"
+
+#define DEVICE_BUFFER_FRAMES  (512)
+
+struct {
+    int buffer_frames;
+} conf = {
+    .buffer_frames = 512
+};
+
+typedef struct coreaudioVoiceOut {
+    HWVoiceOut hw;
+    pthread_mutex_t mutex;
+    AudioDeviceID outputDeviceID;
+    UInt32 audioDevicePropertyBufferSize;
+    AudioStreamBasicDescription outputStreamBasicDescription;
+    int isPlaying;
+    int live;
+    int decr;
+    int rpos;
+} coreaudioVoiceOut;
+
+static void coreaudio_logstatus (OSStatus status)
+{
+    char *str = "BUG";
+
+    switch(status) {
+    case kAudioHardwareNoError:
+        str = "kAudioHardwareNoError";
+        break;
+
+    case kAudioHardwareNotRunningError:
+        str = "kAudioHardwareNotRunningError";
+        break;
+
+    case kAudioHardwareUnspecifiedError:
+        str = "kAudioHardwareUnspecifiedError";
+        break;
+
+    case kAudioHardwareUnknownPropertyError:
+        str = "kAudioHardwareUnknownPropertyError";
+        break;
+
+    case kAudioHardwareBadPropertySizeError:
+        str = "kAudioHardwareBadPropertySizeError";
+        break;
+
+    case kAudioHardwareIllegalOperationError:
+        str = "kAudioHardwareIllegalOperationError";
+        break;
+
+    case kAudioHardwareBadDeviceError:
+        str = "kAudioHardwareBadDeviceError";
+        break;
+
+    case kAudioHardwareBadStreamError:
+        str = "kAudioHardwareBadStreamError";
+        break;
+
+    case kAudioHardwareUnsupportedOperationError:
+        str = "kAudioHardwareUnsupportedOperationError";
+        break;
+
+    case kAudioDeviceUnsupportedFormatError:
+        str = "kAudioDeviceUnsupportedFormatError";
+        break;
+
+    case kAudioDevicePermissionsError:
+        str = "kAudioDevicePermissionsError";
+        break;
+
+    default:
+        AUD_log (AUDIO_CAP, "Reason: status code %ld\n", status);
+        return;
+    }
+
+    AUD_log (AUDIO_CAP, "Reason: %s\n", str);
+}
+
+static void GCC_FMT_ATTR (2, 3) coreaudio_logerr (
+    OSStatus status,
+    const char *fmt,
+    ...
+    )
+{
+    va_list ap;
+
+    va_start (ap, fmt);
+    AUD_log (AUDIO_CAP, fmt, ap);
+    va_end (ap);
+
+    coreaudio_logstatus (status);
+}
+
+static void GCC_FMT_ATTR (3, 4) coreaudio_logerr2 (
+    OSStatus status,
+    const char *typ,
+    const char *fmt,
+    ...
+    )
+{
+    va_list ap;
+
+    AUD_log (AUDIO_CAP, "Can not initialize %s\n", typ);
+
+    va_start (ap, fmt);
+    AUD_vlog (AUDIO_CAP, fmt, ap);
+    va_end (ap);
+
+    coreaudio_logstatus (status);
+}
+
+static int coreaudio_lock (coreaudioVoiceOut *core, const char *fn_name)
+{
+    int err;
+
+    err = pthread_mutex_lock (&core->mutex);
+    if (err) {
+        dolog ("Can not lock voice for %s\nReason: %s\n",
+               fn_name, strerror (err));
+        return -1;
+    }
+    return 0;
+}
+
+static int coreaudio_unlock (coreaudioVoiceOut *core, const char *fn_name)
+{
+    int err;
+
+    err = pthread_mutex_unlock (&core->mutex);
+    if (err) {
+        dolog ("Can not unlock voice for %s\nReason: %s\n",
+               fn_name, strerror (err));
+        return -1;
+    }
+    return 0;
+}
+
+static int coreaudio_run_out (HWVoiceOut *hw)
+{
+    int live, decr;
+    coreaudioVoiceOut *core = (coreaudioVoiceOut *) hw;
+
+    if (coreaudio_lock (core, "coreaudio_run_out")) {
+        return 0;
+    }
+
+    live = audio_pcm_hw_get_live_out (hw);
+
+    if (core->decr > live) {
+        ldebug ("core->decr %d live %d core->live %d\n",
+                core->decr,
+                live,
+                core->live);
+    }
+
+    decr = audio_MIN (core->decr, live);
+    core->decr -= decr;
+
+    core->live = live - decr;
+    hw->rpos = core->rpos;
+
+    coreaudio_unlock (core, "coreaudio_run_out");
+    return decr;
+}
+
+/* callback to feed audiooutput buffer */
+static OSStatus audioDeviceIOProc(
+    AudioDeviceID inDevice,
+    const AudioTimeStamp* inNow,
+    const AudioBufferList* inInputData,
+    const AudioTimeStamp* inInputTime,
+    AudioBufferList* outOutputData,
+    const AudioTimeStamp* inOutputTime,
+    void* hwptr)
+{
+    unsigned int frame, frameCount;
+    float *out = outOutputData->mBuffers[0].mData;
+    HWVoiceOut *hw = hwptr;
+    coreaudioVoiceOut *core = (coreaudioVoiceOut *) hwptr;
+    int rpos, live;
+    st_sample_t *src;
+#ifndef FLOAT_MIXENG
+#ifdef RECIPROCAL
+    const float scale = 1.f / UINT_MAX;
+#else
+    const float scale = UINT_MAX;
+#endif
+#endif
+
+    if (coreaudio_lock (core, "audioDeviceIOProc")) {
+        inInputTime = 0;
+        return 0;
+    }
+
+    frameCount = conf.buffer_frames;
+    live = core->live;
+
+    /* if there are not enough samples, set signal and return */
+    if (live < frameCount) {
+        inInputTime = 0;
+        coreaudio_unlock (core, "audioDeviceIOProc(empty)");
+        return 0;
+    }
+
+    rpos = core->rpos;
+    src = hw->mix_buf + rpos;
+
+    /* fill buffer */
+    for (frame = 0; frame < frameCount; frame++) {
+#ifdef FLOAT_MIXENG
+        *out++ = src[frame].l; /* left channel */
+        *out++ = src[frame].r; /* right channel */
+#else
+#ifdef RECIPROCAL
+        *out++ = src[frame].l * scale; /* left channel */
+        *out++ = src[frame].r * scale; /* right channel */
+#else
+        *out++ = src[frame].l / scale; /* left channel */
+        *out++ = src[frame].r / scale; /* right channel */
+#endif
+#endif
+    }
+
+    /* cleanup */
+    mixeng_clear (src, frameCount);
+    rpos = (rpos + frameCount) % hw->samples;
+    core->decr = frameCount;
+    core->rpos = rpos;
+
+    coreaudio_unlock (core, "audioDeviceIOProc");
+    return 0;
+}
+
+static int coreaudio_write (SWVoiceOut *sw, void *buf, int len)
+{
+    return audio_pcm_sw_write (sw, buf, len);
+}
+
+static int coreaudio_init_out (HWVoiceOut *hw, int freq,
+                               int nchannels, audfmt_e fmt)
+{
+    OSStatus status;
+    coreaudioVoiceOut *core = (coreaudioVoiceOut *) hw;
+    UInt32 propertySize;
+    int err;
+    int bits = 8;
+    int endianess = 0;
+    const char *typ = "DAC";
+
+    /* create mutex */
+    err = pthread_mutex_init(&core->mutex, NULL);
+    if (err) {
+        dolog("Can not create mutex\nReason: %s\n", strerror (err));
+        return -1;
+    }
+
+    if (fmt == AUD_FMT_S16 || fmt == AUD_FMT_U16) {
+        bits = 16;
+        endianess = 1;
+    }
+
+    audio_pcm_init_info (
+        &hw->info,
+        freq,
+        nchannels,
+        fmt,
+        /* Following is irrelevant actually since we do not use
+           mixengs clipping routines */
+        audio_need_to_swap_endian (endianess)
+        );
+    hw->bufsize = 4 * conf.buffer_frames * nchannels * bits;
+
+    /* open default output device */
+    propertySize = sizeof(core->outputDeviceID);
+    status = AudioHardwareGetProperty(
+        kAudioHardwarePropertyDefaultOutputDevice,
+        &propertySize,
+        &core->outputDeviceID);
+    if (status != kAudioHardwareNoError) {
+        coreaudio_logerr2 (status, typ,
+                           "Can not get default output Device\n");
+        return -1;
+    }
+    if (core->outputDeviceID == kAudioDeviceUnknown) {
+        dolog ("Can not initialize %s - Unknown Audiodevice\n", typ);
+        return -1;
+    }
+
+    /* set Buffersize to conf.buffer_frames frames */
+    propertySize = sizeof(core->audioDevicePropertyBufferSize);
+    core->audioDevicePropertyBufferSize =
+        conf.buffer_frames * sizeof(float) * 2;
+    status = AudioDeviceSetProperty(
+        core->outputDeviceID,
+        NULL,
+        0,
+        false,
+        kAudioDevicePropertyBufferSize,
+        propertySize,
+        &core->audioDevicePropertyBufferSize);
+    if (status != kAudioHardwareNoError) {
+        coreaudio_logerr2 (status, typ,
+                           "Can not set device buffer size %d\n",
+                           kAudioDevicePropertyBufferSize);
+        return -1;
+    }
+
+    /* get Buffersize */
+    propertySize = sizeof(core->audioDevicePropertyBufferSize);
+    status = AudioDeviceGetProperty(
+        core->outputDeviceID,
+        0,
+        false,
+        kAudioDevicePropertyBufferSize,
+        &propertySize,
+        &core->audioDevicePropertyBufferSize);
+    if (status != kAudioHardwareNoError) {
+        coreaudio_logerr2 (status, typ, "Can not get device buffer size\n");
+        return -1;
+    }
+
+    /* get StreamFormat */
+    propertySize = sizeof(core->outputStreamBasicDescription);
+    status = AudioDeviceGetProperty(
+        core->outputDeviceID,
+        0,
+        false,
+        kAudioDevicePropertyStreamFormat,
+        &propertySize,
+        &core->outputStreamBasicDescription);
+    if (status != kAudioHardwareNoError) {
+        coreaudio_logerr2 (status, typ,
+                           "Can not get Device Stream properties\n");
+        core->outputDeviceID = kAudioDeviceUnknown;
+        return -1;
+    }
+
+    /* set Samplerate */
+    core->outputStreamBasicDescription.mSampleRate = (Float64)freq;
+    propertySize = sizeof(core->outputStreamBasicDescription);
+    status = AudioDeviceSetProperty(
+        core->outputDeviceID,
+        0,
+        0,
+        0,
+        kAudioDevicePropertyStreamFormat,
+        propertySize,
+        &core->outputStreamBasicDescription);
+    if (status != kAudioHardwareNoError) {
+        coreaudio_logerr2 (status, typ, "Can not set samplerate %d\n", freq);
+        core->outputDeviceID = kAudioDeviceUnknown;
+        return -1;
+    }
+
+    /* set Callback */
+    status = AudioDeviceAddIOProc(core->outputDeviceID, audioDeviceIOProc, hw);
+    if (status != kAudioHardwareNoError) {
+        coreaudio_logerr2 (status, typ, "Can not set IOProc\n");
+        core->outputDeviceID = kAudioDeviceUnknown;
+        return -1;
+    }
+
+    /* start Playback */
+    if (!core->isPlaying) {
+        status = AudioDeviceStart(core->outputDeviceID, audioDeviceIOProc);
+        if (status != kAudioHardwareNoError) {
+            coreaudio_logerr2 (status, typ, "Can not start playback\n");
+            AudioDeviceRemoveIOProc(core->outputDeviceID, audioDeviceIOProc);
+            core->outputDeviceID = kAudioDeviceUnknown;
+            return -1;
+        }
+        core->isPlaying = 1;
+    }
+
+    return 0;
+}
+
+static void coreaudio_fini_out (HWVoiceOut *hw)
+{
+    OSStatus status;
+    int err;
+    coreaudioVoiceOut *core = (coreaudioVoiceOut *) hw;
+
+    /* stop playback */
+    if (core->isPlaying) {
+        status = AudioDeviceStop(core->outputDeviceID, audioDeviceIOProc);
+        if (status != kAudioHardwareNoError) {
+            coreaudio_logerr (status, "Can not stop playback\n");
+        }
+        core->isPlaying = 0;
+    }
+
+    /* remove callback */
+    status = AudioDeviceRemoveIOProc(core->outputDeviceID, audioDeviceIOProc);
+    if (status != kAudioHardwareNoError) {
+        coreaudio_logerr (status, "Can not remove IOProc\n");
+    }
+    core->outputDeviceID = kAudioDeviceUnknown;
+
+    /* destroy mutex */
+    err = pthread_mutex_destroy(&core->mutex);
+    if (err) {
+        dolog("Can not destroy mutex\nReason: %s\n", strerror (err));
+    }
+}
+
+static int coreaudio_ctl_out (HWVoiceOut *hw, int cmd, ...)
+{
+    OSStatus status;
+    coreaudioVoiceOut *core = (coreaudioVoiceOut *) hw;
+
+    switch (cmd) {
+    case VOICE_ENABLE:
+        /* start playback */
+        if (!core->isPlaying) {
+            status = AudioDeviceStart(core->outputDeviceID, audioDeviceIOProc);
+            if (status != kAudioHardwareNoError) {
+                coreaudio_logerr (status, "Can not unpause playback\n");
+            }
+            core->isPlaying = 1;
+        }
+        break;
+
+    case VOICE_DISABLE:
+        /* stop playback */
+        if (core->isPlaying) {
+            status = AudioDeviceStop(core->outputDeviceID, audioDeviceIOProc);
+            if (status != kAudioHardwareNoError) {
+                coreaudio_logerr (status, "Can not pause playback\n");
+            }
+            core->isPlaying = 0;
+        }
+        break;
+    }
+    return 0;
+}
+
+static void *coreaudio_audio_init (void)
+{
+    return &coreaudio_audio_init;
+}
+
+static void coreaudio_audio_fini (void *opaque)
+{
+    (void) opaque;
+}
+
+static struct audio_option coreaudio_options[] = {
+    {"BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_frames,
+     "Size of the buffer in frames", NULL, 0},
+    {NULL, 0, NULL, NULL, NULL, 0}
+};
+
+static struct audio_pcm_ops coreaudio_pcm_ops = {
+    coreaudio_init_out,
+    coreaudio_fini_out,
+    coreaudio_run_out,
+    coreaudio_write,
+    coreaudio_ctl_out,
+
+    NULL,
+    NULL,
+    NULL,
+    NULL,
+    NULL
+};
+
+struct audio_driver coreaudio_audio_driver = {
+    INIT_FIELD (name           = ) "coreaudio",
+    INIT_FIELD (descr          = )
+    "CoreAudio http://developer.apple.com/audio/coreaudio.html",
+    INIT_FIELD (options        = ) coreaudio_options,
+    INIT_FIELD (init           = ) coreaudio_audio_init,
+    INIT_FIELD (fini           = ) coreaudio_audio_fini,
+    INIT_FIELD (pcm_ops        = ) &coreaudio_pcm_ops,
+    INIT_FIELD (can_be_default = ) 1,
+    INIT_FIELD (max_voices_out = ) 1,
+    INIT_FIELD (max_voices_in  = ) 0,
+    INIT_FIELD (voice_size_out = ) sizeof (coreaudioVoiceOut),
+    INIT_FIELD (voice_size_in  = ) 0
+};

audio/dsound_template.h

@@ -0,0 +1,298 @@
+/*
+ * QEMU DirectSound audio driver header
+ *
+ * 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.
+ */
+#ifdef DSBTYPE_IN
+#define NAME "capture buffer"
+#define TYPE in
+#define IFACE IDirectSoundCaptureBuffer
+#define BUFPTR LPDIRECTSOUNDCAPTUREBUFFER
+#define FIELD dsound_capture_buffer
+#else
+#define NAME "playback buffer"
+#define TYPE out
+#define IFACE IDirectSoundBuffer
+#define BUFPTR LPDIRECTSOUNDBUFFER
+#define FIELD dsound_buffer
+#endif
+
+static int glue (dsound_unlock_, TYPE) (
+    BUFPTR buf,
+    LPVOID p1,
+    LPVOID p2,
+    DWORD blen1,
+    DWORD blen2
+    )
+{
+    HRESULT hr;
+
+    hr = glue (IFACE, _Unlock) (buf, p1, blen1, p2, blen2);
+    if (FAILED (hr)) {
+        dsound_logerr (hr, "Can not unlock " NAME "\n");
+        return -1;
+    }
+
+    return 0;
+}
+
+static int glue (dsound_lock_, TYPE) (
+    BUFPTR buf,
+    struct audio_pcm_info *info,
+    DWORD pos,
+    DWORD len,
+    LPVOID *p1p,
+    LPVOID *p2p,
+    DWORD *blen1p,
+    DWORD *blen2p,
+    int entire
+    )
+{
+    HRESULT hr;
+    int i;
+    LPVOID p1 = NULL, p2 = NULL;
+    DWORD blen1 = 0, blen2 = 0;
+
+    for (i = 0; i < conf.lock_retries; ++i) {
+        hr = glue (IFACE, _Lock) (
+            buf,
+            pos,
+            len,
+            &p1,
+            &blen1,
+            &p2,
+            &blen2,
+            (entire
+#ifdef DSBTYPE_IN
+             ? DSCBLOCK_ENTIREBUFFER
+#else
+             ? DSBLOCK_ENTIREBUFFER
+#endif
+             : 0)
+            );
+
+        if (FAILED (hr)) {
+#ifndef DSBTYPE_IN
+            if (hr == DSERR_BUFFERLOST) {
+                if (glue (dsound_restore_, TYPE) (buf)) {
+                    dsound_logerr (hr, "Can not lock " NAME "\n");
+                    goto fail;
+                }
+                continue;
+            }
+#endif
+            dsound_logerr (hr, "Can not lock " NAME "\n");
+            goto fail;
+        }
+
+        break;
+    }
+
+    if (i == conf.lock_retries) {
+        dolog ("%d attempts to lock " NAME " failed\n", i);
+        goto fail;
+    }
+
+    if ((p1 && (blen1 & info->align)) || (p2 && (blen2 & info->align))) {
+        dolog ("DirectSound returned misaligned buffer %ld %ld\n",
+               blen1, blen2);
+        glue (dsound_unlock_, TYPE) (buf, p1, p2, blen1, blen2);
+        goto fail;
+    }
+
+    if (!p1 && blen1) {
+        dolog ("warning: !p1 && blen1=%ld\n", blen1);
+        blen1 = 0;
+    }
+
+    if (!p2 && blen2) {
+        dolog ("warning: !p2 && blen2=%ld\n", blen2);
+        blen2 = 0;
+    }
+
+    *p1p = p1;
+    *p2p = p2;
+    *blen1p = blen1;
+    *blen2p = blen2;
+    return 0;
+
+ fail:
+    *p1p = NULL - 1;
+    *p2p = NULL - 1;
+    *blen1p = -1;
+    *blen2p = -1;
+    return -1;
+}
+
+#ifdef DSBTYPE_IN
+static void dsound_fini_in (HWVoiceIn *hw)
+#else
+static void dsound_fini_out (HWVoiceOut *hw)
+#endif
+{
+    HRESULT hr;
+#ifdef DSBTYPE_IN
+    DSoundVoiceIn *ds = (DSoundVoiceIn *) hw;
+#else
+    DSoundVoiceOut *ds = (DSoundVoiceOut *) hw;
+#endif
+
+    if (ds->FIELD) {
+        hr = glue (IFACE, _Stop) (ds->FIELD);
+        if (FAILED (hr)) {
+            dsound_logerr (hr, "Can not stop " NAME "\n");
+        }
+
+        hr = glue (IFACE, _Release) (ds->FIELD);
+        if (FAILED (hr)) {
+            dsound_logerr (hr, "Can not release " NAME "\n");
+        }
+        ds->FIELD = NULL;
+    }
+}
+
+#ifdef DSBTYPE_IN
+static int dsound_init_in (
+    HWVoiceIn *hw,
+    int freq,
+    int nchannels,
+    audfmt_e fmt
+    )
+#else
+static int dsound_init_out (
+    HWVoiceOut *hw,
+    int freq,
+    int nchannels,
+    audfmt_e fmt
+    )
+#endif
+{
+    int err;
+    HRESULT hr;
+    dsound *s = &glob_dsound;
+    WAVEFORMATEX wfx;
+    struct full_fmt full_fmt;
+#ifdef DSBTYPE_IN
+    const char *typ = "ADC";
+    DSoundVoiceIn *ds = (DSoundVoiceIn *) hw;
+    DSCBUFFERDESC bd;
+    DSCBCAPS bc;
+#else
+    const char *typ = "DAC";
+    DSoundVoiceOut *ds = (DSoundVoiceOut *) hw;
+    DSBUFFERDESC bd;
+    DSBCAPS bc;
+#endif
+
+    full_fmt.freq = freq;
+    full_fmt.nchannels = nchannels;
+    full_fmt.fmt = fmt;
+    err = waveformat_from_full_fmt (&wfx, &full_fmt);
+    if (err) {
+        return -1;
+    }
+
+    memset (&bd, 0, sizeof (bd));
+    bd.dwSize = sizeof (bd);
+    bd.lpwfxFormat = &wfx;
+#ifdef DSBTYPE_IN
+    bd.dwBufferBytes = conf.bufsize_in;
+    hr = IDirectSoundCapture_CreateCaptureBuffer (
+        s->dsound_capture,
+        &bd,
+        &ds->dsound_capture_buffer,
+        NULL
+        );
+#else
+    bd.dwFlags = DSBCAPS_STICKYFOCUS | DSBCAPS_GETCURRENTPOSITION2;
+    bd.dwBufferBytes = conf.bufsize_out;
+    hr = IDirectSound_CreateSoundBuffer (
+        s->dsound,
+        &bd,
+        &ds->dsound_buffer,
+        NULL
+        );
+#endif
+
+    if (FAILED (hr)) {
+        dsound_logerr2 (hr, typ, "Can not create " NAME "\n");
+        return -1;
+    }
+
+    hr = glue (IFACE, _GetFormat) (
+        ds->FIELD,
+        &wfx,
+        sizeof (wfx),
+        NULL
+        );
+    if (FAILED (hr)) {
+        dsound_logerr2 (hr, typ, "Can not get " NAME " format\n");
+        goto fail0;
+    }
+
+#ifdef DEBUG_DSOUND
+    dolog (NAME "\n");
+    print_wave_format (&wfx);
+#endif
+
+    memset (&bc, 0, sizeof (bc));
+    bc.dwSize = sizeof (bc);
+
+    hr = glue (IFACE, _GetCaps) (ds->FIELD, &bc);
+    if (FAILED (hr)) {
+        dsound_logerr2 (hr, typ, "Can not get " NAME " format\n");
+        goto fail0;
+    }
+
+    err = waveformat_to_full_fmt (&wfx, &full_fmt);
+    if (err) {
+        goto fail0;
+    }
+
+    ds->first_time = 1;
+    hw->bufsize = bc.dwBufferBytes;
+    audio_pcm_init_info (
+        &hw->info,
+        full_fmt.freq,
+        full_fmt.nchannels,
+        full_fmt.fmt,
+        audio_need_to_swap_endian (0)
+        );
+
+#ifdef DEBUG_DSOUND
+    dolog ("caps %ld, desc %ld\n",
+           bc.dwBufferBytes, bd.dwBufferBytes);
+
+    dolog ("bufsize %d, freq %d, chan %d, fmt %d\n",
+           hw->bufsize, full_fmt.freq, full_fmt.nchannels, full_fmt.fmt);
+#endif
+    return 0;
+
+ fail0:
+    glue (dsound_fini_, TYPE) (hw);
+    return -1;
+}
+
+#undef NAME
+#undef TYPE
+#undef IFACE
+#undef BUFPTR
+#undef FIELD

audio/fmodaudio.c

@@ -1,7 +1,7 @@
 /*
- * QEMU FMOD audio output driver
+ * QEMU FMOD audio driver
  *
- * Copyright (c) 2004 Vassili Karpov (malc)
+ * Copyright (c) 2004-2005 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
@@ -25,53 +25,77 @@
 #include <fmod_errors.h>
 #include "vl.h"
 
-#include "audio/audio_int.h"
+#define AUDIO_CAP "fmod"
+#include "audio_int.h"
 
-typedef struct FMODVoice {
+typedef struct FMODVoiceOut {
-    HWVoice hw;
+    HWVoiceOut hw;
     unsigned int old_pos;
     FSOUND_SAMPLE *fmod_sample;
     int channel;
-} FMODVoice;
+} FMODVoiceOut;
 
-#define dolog(...) AUD_log ("fmod", __VA_ARGS__)
+typedef struct FMODVoiceIn {
-#ifdef DEBUG
+    HWVoiceIn hw;
-#define ldebug(...) dolog (__VA_ARGS__)
+    FSOUND_SAMPLE *fmod_sample;
-#else
+} FMODVoiceIn;
-#define ldebug(...)
-#endif
-
-#define QC_FMOD_DRV "QEMU_FMOD_DRV"
-#define QC_FMOD_FREQ "QEMU_FMOD_FREQ"
-#define QC_FMOD_SAMPLES "QEMU_FMOD_SAMPLES"
-#define QC_FMOD_CHANNELS "QEMU_FMOD_CHANNELS"
-#define QC_FMOD_BUFSIZE "QEMU_FMOD_BUFSIZE"
-#define QC_FMOD_THRESHOLD "QEMU_FMOD_THRESHOLD"
 
 static struct {
+    const char *drvname;
     int nb_samples;
     int freq;
     int nb_channels;
     int bufsize;
     int threshold;
+    int broken_adc;
 } conf = {
-    2048,
+    NULL,
+    2048 * 2,
     44100,
-    1,
+    2,
     0,
-    128
+    0,
+    0
 };
 
-#define errstr() FMOD_ErrorString (FSOUND_GetError ())
+static void GCC_FMT_ATTR (1, 2) fmod_logerr (const char *fmt, ...)
-
-static int fmod_hw_write (SWVoice *sw, void *buf, int len)
 {
-    return pcm_hw_write (sw, buf, len);
+    va_list ap;
+
+    va_start (ap, fmt);
+    AUD_vlog (AUDIO_CAP, fmt, ap);
+    va_end (ap);
+
+    AUD_log (AUDIO_CAP, "Reason: %s\n",
+             FMOD_ErrorString (FSOUND_GetError ()));
 }
 
-static void fmod_clear_sample (FMODVoice *fmd)
+static void GCC_FMT_ATTR (2, 3) fmod_logerr2 (
+    const char *typ,
+    const char *fmt,
+    ...
+    )
 {
-    HWVoice *hw = &fmd->hw;
+    va_list ap;
+
+    AUD_log (AUDIO_CAP, "Can 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",
+             FMOD_ErrorString (FSOUND_GetError ()));
+}
+
+static int fmod_write (SWVoiceOut *sw, void *buf, int len)
+{
+    return audio_pcm_sw_write (sw, buf, len);
+}
+
+static void fmod_clear_sample (FMODVoiceOut *fmd)
+{
+    HWVoiceOut *hw = &fmd->hw;
     int status;
     void *p1 = 0, *p2 = 0;
     unsigned int len1 = 0, len2 = 0;
@@ -79,7 +103,7 @@ static void fmod_clear_sample (FMODVoice *fmd)
     status = FSOUND_Sample_Lock (
         fmd->fmod_sample,
         0,
-        hw->samples << hw->shift,
+        hw->samples << hw->info.shift,
         &p1,
         &p2,
         &len1,
@@ -87,78 +111,88 @@ static void fmod_clear_sample (FMODVoice *fmd)
         );
 
     if (!status) {
-        dolog ("Failed to lock sample\nReason: %s\n", errstr ());
+        fmod_logerr ("Failed to lock sample\n");
         return;
     }
 
-    if ((len1 & hw->align) || (len2 & hw->align)) {
+    if ((len1 & hw->info.align) || (len2 & hw->info.align)) {
-        dolog ("Locking sample returned unaligned length %d, %d\n",
+        dolog ("Lock returned misaligned length %d, %d, alignment %d\n",
-               len1, len2);
+               len1, len2, hw->info.align + 1);
         goto fail;
     }
 
-    if (len1 + len2 != hw->samples << hw->shift) {
+    if ((len1 + len2) - (hw->samples << hw->info.shift)) {
-        dolog ("Locking sample returned incomplete length %d, %d\n",
+        dolog ("Lock returned incomplete length %d, %d\n",
-               len1 + len2, hw->samples << hw->shift);
+               len1 + len2, hw->samples << hw->info.shift);
         goto fail;
     }
-    pcm_hw_clear (hw, p1, hw->samples);
+
+    audio_pcm_info_clear_buf (&hw->info, p1, hw->samples);
 
  fail:
     status = FSOUND_Sample_Unlock (fmd->fmod_sample, p1, p2, len1, len2);
     if (!status) {
-        dolog ("Failed to unlock sample\nReason: %s\n", errstr ());
+        fmod_logerr ("Failed to unlock sample\n");
     }
 }
 
-static int fmod_write_sample (HWVoice *hw, uint8_t *dst, st_sample_t *src,
+static void fmod_write_sample (HWVoiceOut *hw, uint8_t *dst, int dst_len)
-                              int src_size, int src_pos, int dst_len)
 {
-    int src_len1 = dst_len, src_len2 = 0, pos = src_pos + dst_len;
+    int src_len1 = dst_len;
-    st_sample_t *src1 = src + src_pos, *src2 = 0;
+    int src_len2 = 0;
+    int pos = hw->rpos + dst_len;
+    st_sample_t *src1 = hw->mix_buf + hw->rpos;
+    st_sample_t *src2 = NULL;
 
-    if (src_pos + dst_len > src_size) {
+    if (pos > hw->samples) {
-        src_len1 = src_size - src_pos;
+        src_len1 = hw->samples - hw->rpos;
-        src2 = src;
+        src2 = hw->mix_buf;
         src_len2 = dst_len - src_len1;
         pos = src_len2;
     }
 
     if (src_len1) {
         hw->clip (dst, src1, src_len1);
-        memset (src1, 0, src_len1 * sizeof (st_sample_t));
+        mixeng_clear (src1, src_len1);
-        advance (dst, src_len1);
     }
 
     if (src_len2) {
+        dst = advance (dst, src_len1 << hw->info.shift);
         hw->clip (dst, src2, src_len2);
-        memset (src2, 0, src_len2 * sizeof (st_sample_t));
+        mixeng_clear (src2, src_len2);
     }
-    return pos;
+
+    hw->rpos = pos % hw->samples;
 }
 
-static int fmod_unlock_sample (FMODVoice *fmd, void *p1, void *p2,
+static int fmod_unlock_sample (FSOUND_SAMPLE *sample, void *p1, void *p2,
                                unsigned int blen1, unsigned int blen2)
 {
-    int status = FSOUND_Sample_Unlock (fmd->fmod_sample, p1, p2, blen1, blen2);
+    int status = FSOUND_Sample_Unlock (sample, p1, p2, blen1, blen2);
     if (!status) {
-        dolog ("Failed to unlock sample\nReason: %s\n", errstr ());
+        fmod_logerr ("Failed to unlock sample\n");
         return -1;
     }
     return 0;
 }
 
-static int fmod_lock_sample (FMODVoice *fmd, int pos, int len,
+static int fmod_lock_sample (
-                             void **p1, void **p2,
+    FSOUND_SAMPLE *sample,
-                             unsigned int *blen1, unsigned int *blen2)
+    struct audio_pcm_info *info,
+    int pos,
+    int len,
+    void **p1,
+    void **p2,
+    unsigned int *blen1,
+    unsigned int *blen2
+    )
 {
-    HWVoice *hw = &fmd->hw;
     int status;
 
     status = FSOUND_Sample_Lock (
-        fmd->fmod_sample,
+        sample,
-        pos << hw->shift,
+        pos << info->shift,
-        len << hw->shift,
+        len << info->shift,
         p1,
         p2,
         blen1,
@@ -166,89 +200,117 @@ static int fmod_lock_sample (FMODVoice *fmd, int pos, int len,
         );
 
     if (!status) {
-        dolog ("Failed to lock sample\nReason: %s\n", errstr ());
+        fmod_logerr ("Failed to lock sample\n");
         return -1;
     }
 
-    if ((*blen1 & hw->align) || (*blen2 & hw->align)) {
+    if ((*blen1 & info->align) || (*blen2 & info->align)) {
-        dolog ("Locking sample returned unaligned length %d, %d\n",
+        dolog ("Lock returned misaligned length %d, %d, alignment %d\n",
-               *blen1, *blen2);
+               *blen1, *blen2, info->align + 1);
-        fmod_unlock_sample (fmd, *p1, *p2, *blen1, *blen2);
+
+        fmod_unlock_sample (sample, *p1, *p2, *blen1, *blen2);
+
+        *p1 = NULL - 1;
+        *p2 = NULL - 1;
+        *blen1 = ~0U;
+        *blen2 = ~0U;
         return -1;
     }
+
+    if (!*p1 && *blen1) {
+        dolog ("warning: !p1 && blen1=%d\n", *blen1);
+        *blen1 = 0;
+    }
+
+    if (!p2 && *blen2) {
+        dolog ("warning: !p2 && blen2=%d\n", *blen2);
+        *blen2 = 0;
+    }
+
     return 0;
 }
 
-static void fmod_hw_run (HWVoice *hw)
+static int fmod_run_out (HWVoiceOut *hw)
 {
-    FMODVoice *fmd = (FMODVoice *) hw;
+    FMODVoiceOut *fmd = (FMODVoiceOut *) hw;
-    int rpos, live, decr;
+    int live, decr;
     void *p1 = 0, *p2 = 0;
     unsigned int blen1 = 0, blen2 = 0;
     unsigned int len1 = 0, len2 = 0;
-    int nb_active;
+    int nb_live;
 
-    live = pcm_hw_get_live2 (hw, &nb_active);
+    live = audio_pcm_hw_get_live_out2 (hw, &nb_live);
-    if (live <= 0) {
+    if (!live) {
-        return;
+        return 0;
     }
 
     if (!hw->pending_disable
-        && nb_active
+        && nb_live
-        && conf.threshold
+        && (conf.threshold && live <= conf.threshold)) {
-        && live <= conf.threshold) {
+        ldebug ("live=%d nb_live=%d\n", live, nb_live);
-        ldebug ("live=%d nb_active=%d\n", live, nb_active);
+        return 0;
-        return;
     }
 
     decr = live;
 
-#if 1
     if (fmd->channel >= 0) {
-        int pos2 = (fmd->old_pos + decr) % hw->samples;
+        int len = decr;
-        int pos = FSOUND_GetCurrentPosition (fmd->channel);
+        int old_pos = fmd->old_pos;
+        int ppos = FSOUND_GetCurrentPosition (fmd->channel);
 
-        if (fmd->old_pos < pos && pos2 >= pos) {
+        if (ppos == old_pos || !ppos) {
-            decr = pos - fmd->old_pos - (pos2 == pos) - 1;
+            return 0;
-        }
-        else if (fmd->old_pos > pos && pos2 >= pos && pos2 < fmd->old_pos) {
-            decr = (hw->samples - fmd->old_pos) + pos - (pos2 == pos) - 1;
-        }
-/*         ldebug ("pos=%d pos2=%d old=%d live=%d decr=%d\n", */
-/*                 pos, pos2, fmd->old_pos, live, decr); */
-    }
-#endif
-
-    if (decr <= 0) {
-        return;
         }
 
-    if (fmod_lock_sample (fmd, fmd->old_pos, decr, &p1, &p2, &blen1, &blen2)) {
+        if ((old_pos < ppos) && ((old_pos + len) > ppos)) {
-        return;
+            len = ppos - old_pos;
+        }
+        else {
+            if ((old_pos > ppos) && ((old_pos + len) > (ppos + hw->samples))) {
+                len = hw->samples - old_pos + ppos;
+            }
+        }
+        decr = len;
+
+        if (audio_bug (AUDIO_FUNC, decr < 0)) {
+            dolog ("decr=%d live=%d ppos=%d old_pos=%d len=%d\n",
+                   decr, live, ppos, old_pos, len);
+            return 0;
+        }
     }
 
-    len1 = blen1 >> hw->shift;
+
-    len2 = blen2 >> hw->shift;
+    if (!decr) {
+        return 0;
+    }
+
+    if (fmod_lock_sample (fmd->fmod_sample, &fmd->hw.info,
+                          fmd->old_pos, decr,
+                          &p1, &p2,
+                          &blen1, &blen2)) {
+        return 0;
+    }
+
+    len1 = blen1 >> hw->info.shift;
+    len2 = blen2 >> hw->info.shift;
     ldebug ("%p %p %d %d %d %d\n", p1, p2, len1, len2, blen1, blen2);
     decr = len1 + len2;
-    rpos = hw->rpos;
 
-    if (len1) {
+    if (p1 && len1) {
-        rpos = fmod_write_sample (hw, p1, hw->mix_buf, hw->samples, rpos, len1);
+        fmod_write_sample (hw, p1, len1);
     }
 
-    if (len2) {
+    if (p2 && len2) {
-        rpos = fmod_write_sample (hw, p2, hw->mix_buf, hw->samples, rpos, len2);
+        fmod_write_sample (hw, p2, len2);
     }
 
-    fmod_unlock_sample (fmd, p1, p2, blen1, blen2);
+    fmod_unlock_sample (fmd->fmod_sample, p1, p2, blen1, blen2);
 
-    pcm_hw_dec_live (hw, decr);
-    hw->rpos = rpos % hw->samples;
     fmd->old_pos = (fmd->old_pos + decr) % hw->samples;
+    return decr;
 }
 
-static int AUD_to_fmodfmt (audfmt_e fmt, int stereo)
+static int aud_to_fmodfmt (audfmt_e fmt, int stereo)
 {
     int mode = FSOUND_LOOP_NORMAL;
 
@@ -270,16 +332,19 @@ static int AUD_to_fmodfmt (audfmt_e fmt, int stereo)
         break;
 
     default:
-        dolog ("Internal logic error: Bad audio format %d\nAborting\n", fmt);
+        dolog ("Internal logic error: Bad audio format %d\n", fmt);
-        exit (EXIT_FAILURE);
+#ifdef DEBUG_FMOD
+        abort ();
+#endif
+        mode |= FSOUND_8BITS;
     }
     mode |= stereo ? FSOUND_STEREO : FSOUND_MONO;
     return mode;
 }
 
-static void fmod_hw_fini (HWVoice *hw)
+static void fmod_fini_out (HWVoiceOut *hw)
 {
-    FMODVoice *fmd = (FMODVoice *) hw;
+    FMODVoiceOut *fmd = (FMODVoiceOut *) hw;
 
     if (fmd->fmod_sample) {
         FSOUND_Sample_Free (fmd->fmod_sample);
@@ -291,12 +356,12 @@ static void fmod_hw_fini (HWVoice *hw)
     }
 }
 
-static int fmod_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt)
+static int fmod_init_out (HWVoiceOut *hw, int freq, int nchannels, audfmt_e fmt)
 {
     int bits16, mode, channel;
-    FMODVoice *fmd = (FMODVoice *) hw;
+    FMODVoiceOut *fmd = (FMODVoiceOut *) hw;
 
-    mode = AUD_to_fmodfmt (fmt, nchannels == 2 ? 1 : 0);
+    mode = aud_to_fmodfmt (fmt, nchannels == 2 ? 1 : 0);
     fmd->fmod_sample = FSOUND_Sample_Alloc (
         FSOUND_FREE,            /* index */
         conf.nb_samples,        /* length */
@@ -308,52 +373,145 @@ static int fmod_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt)
         );
 
     if (!fmd->fmod_sample) {
-        dolog ("Failed to allocate FMOD sample\nReason: %s\n", errstr ());
+        fmod_logerr2 ("DAC", "Failed to allocate FMOD sample\n");
         return -1;
     }
 
     channel = FSOUND_PlaySoundEx (FSOUND_FREE, fmd->fmod_sample, 0, 1);
     if (channel < 0) {
-        dolog ("Failed to start playing sound\nReason: %s\n", errstr ());
+        fmod_logerr2 ("DAC", "Failed to start playing sound\n");
         FSOUND_Sample_Free (fmd->fmod_sample);
         return -1;
     }
     fmd->channel = channel;
 
-    hw->freq = freq;
+    /* FMOD always operates on little endian frames? */
-    hw->fmt = fmt;
+    audio_pcm_init_info (&hw->info, freq, nchannels, fmt,
-    hw->nchannels = nchannels;
+                         audio_need_to_swap_endian (0));
-    bits16 = fmt == AUD_FMT_U16 || fmt == AUD_FMT_S16;
+    bits16 = (mode & FSOUND_16BITS) != 0;
     hw->bufsize = conf.nb_samples << (nchannels == 2) << bits16;
     return 0;
 }
 
-static int fmod_hw_ctl (HWVoice *hw, int cmd, ...)
+static int fmod_ctl_out (HWVoiceOut *hw, int cmd, ...)
 {
     int status;
-    FMODVoice *fmd = (FMODVoice *) hw;
+    FMODVoiceOut *fmd = (FMODVoiceOut *) hw;
 
     switch (cmd) {
     case VOICE_ENABLE:
         fmod_clear_sample (fmd);
         status = FSOUND_SetPaused (fmd->channel, 0);
         if (!status) {
-            dolog ("Failed to resume channel %d\nReason: %s\n",
+            fmod_logerr ("Failed to resume channel %d\n", fmd->channel);
-                   fmd->channel, errstr ());
         }
         break;
 
     case VOICE_DISABLE:
         status = FSOUND_SetPaused (fmd->channel, 1);
         if (!status) {
-            dolog ("Failed to pause channel %d\nReason: %s\n",
+            fmod_logerr ("Failed to pause channel %d\n", fmd->channel);
-                   fmd->channel, errstr ());
         }
         break;
     }
     return 0;
 }
 
+static int fmod_init_in (HWVoiceIn *hw, int freq, int nchannels, audfmt_e fmt)
+{
+    int bits16, mode;
+    FMODVoiceIn *fmd = (FMODVoiceIn *) hw;
+
+    if (conf.broken_adc) {
+        return -1;
+    }
+
+    mode = aud_to_fmodfmt (fmt, nchannels == 2 ? 1 : 0);
+    fmd->fmod_sample = FSOUND_Sample_Alloc (
+        FSOUND_FREE,            /* index */
+        conf.nb_samples,        /* length */
+        mode,                   /* mode */
+        freq,                   /* freq */
+        255,                    /* volume */
+        128,                    /* pan */
+        255                     /* priority */
+        );
+
+    if (!fmd->fmod_sample) {
+        fmod_logerr2 ("ADC", "Failed to allocate FMOD sample\n");
+        return -1;
+    }
+
+    /* FMOD always operates on little endian frames? */
+    audio_pcm_init_info (&hw->info, freq, nchannels, fmt,
+                         audio_need_to_swap_endian (0));
+    bits16 = (mode & FSOUND_16BITS) != 0;
+    hw->bufsize = conf.nb_samples << (nchannels == 2) << bits16;
+    return 0;
+}
+
+static void fmod_fini_in (HWVoiceIn *hw)
+{
+    FMODVoiceIn *fmd = (FMODVoiceIn *) hw;
+
+    if (fmd->fmod_sample) {
+        FSOUND_Record_Stop ();
+        FSOUND_Sample_Free (fmd->fmod_sample);
+        fmd->fmod_sample = 0;
+    }
+}
+
+static int fmod_run_in (HWVoiceIn *hw)
+{
+    FMODVoiceIn *fmd = (FMODVoiceIn *) hw;
+    int hwshift = hw->info.shift;
+    int live, dead, new_pos, len;
+    unsigned int blen1 = 0, blen2 = 0;
+    unsigned int len1, len2;
+    unsigned int decr;
+    void *p1, *p2;
+
+    live = audio_pcm_hw_get_live_in (hw);
+    dead = hw->samples - live;
+    if (!dead) {
+        return 0;
+    }
+
+    new_pos = FSOUND_Record_GetPosition ();
+    if (new_pos < 0) {
+        fmod_logerr ("Can not get recording position\n");
+        return 0;
+    }
+
+    len = audio_ring_dist (new_pos,  hw->wpos, hw->samples);
+    if (!len) {
+        return 0;
+    }
+    len = audio_MIN (len, dead);
+
+    if (fmod_lock_sample (fmd->fmod_sample, &fmd->hw.info,
+                          hw->wpos, len,
+                          &p1, &p2,
+                          &blen1, &blen2)) {
+        return 0;
+    }
+
+    len1 = blen1 >> hwshift;
+    len2 = blen2 >> hwshift;
+    decr = len1 + len2;
+
+    if (p1 && blen1) {
+        hw->conv (hw->conv_buf + hw->wpos, p1, len1, &nominal_volume);
+    }
+    if (p2 && len2) {
+        hw->conv (hw->conv_buf, p2, len2, &nominal_volume);
+    }
+
+    fmod_unlock_sample (fmd->fmod_sample, p1, p2, blen1, blen2);
+    hw->wpos = (hw->wpos + decr) % hw->samples;
+    return decr;
+}
+
 static struct {
     const char *name;
     int type;
@@ -378,16 +536,16 @@ static struct {
     {"ps2", FSOUND_OUTPUT_PS2},
     {"gcube", FSOUND_OUTPUT_GC},
 #endif
-    {"nort", FSOUND_OUTPUT_NOSOUND_NONREALTIME}
+    {"none-realtime", FSOUND_OUTPUT_NOSOUND_NONREALTIME}
 };
 
 static void *fmod_audio_init (void)
 {
-    int i;
+    size_t i;
     double ver;
     int status;
     int output_type = -1;
-    const char *drv = audio_get_conf_str (QC_FMOD_DRV, NULL);
+    const char *drv = conf.drvname;
 
     ver = FSOUND_GetVersion ();
     if (ver < FMOD_VERSION) {
@@ -395,6 +553,14 @@ static void *fmod_audio_init (void)
         return NULL;
     }
 
+#ifdef __linux__
+    if (ver < 3.75) {
+        dolog ("FMOD before 3.75 has bug preventing ADC from working\n"
+               "ADC will be disabled.\n");
+        conf.broken_adc = 1;
+    }
+#endif
+
     if (drv) {
         int found = 0;
         for (i = 0; i < sizeof (drvtab) / sizeof (drvtab[0]); i++) {
@@ -405,65 +571,115 @@ static void *fmod_audio_init (void)
             }
         }
         if (!found) {
-            dolog ("Unknown FMOD output driver `%s'\n", drv);
+            dolog ("Unknown FMOD driver `%s'\n", drv);
+            dolog ("Valid drivers:\n");
+            for (i = 0; i < sizeof (drvtab) / sizeof (drvtab[0]); i++) {
+                dolog ("  %s\n", drvtab[i].name);
+            }
         }
     }
 
     if (output_type != -1) {
         status = FSOUND_SetOutput (output_type);
         if (!status) {
-            dolog ("FSOUND_SetOutput(%d) failed\nReason: %s\n",
+            fmod_logerr ("FSOUND_SetOutput(%d) failed\n", output_type);
-                   output_type, errstr ());
             return NULL;
         }
     }
 
-    conf.freq = audio_get_conf_int (QC_FMOD_FREQ, conf.freq);
-    conf.nb_samples = audio_get_conf_int (QC_FMOD_SAMPLES, conf.nb_samples);
-    conf.nb_channels =
-        audio_get_conf_int (QC_FMOD_CHANNELS,
-                            (audio_state.nb_hw_voices > 1
-                             ? audio_state.nb_hw_voices
-                             : conf.nb_channels));
-    conf.bufsize = audio_get_conf_int (QC_FMOD_BUFSIZE, conf.bufsize);
-    conf.threshold = audio_get_conf_int (QC_FMOD_THRESHOLD, conf.threshold);
-
     if (conf.bufsize) {
         status = FSOUND_SetBufferSize (conf.bufsize);
         if (!status) {
-            dolog ("FSOUND_SetBufferSize (%d) failed\nReason: %s\n",
+            fmod_logerr ("FSOUND_SetBufferSize (%d) failed\n", conf.bufsize);
-                   conf.bufsize, errstr ());
         }
     }
 
     status = FSOUND_Init (conf.freq, conf.nb_channels, 0);
     if (!status) {
-        dolog ("FSOUND_Init failed\nReason: %s\n", errstr ());
+        fmod_logerr ("FSOUND_Init failed\n");
         return NULL;
     }
 
     return &conf;
 }
 
+static int fmod_read (SWVoiceIn *sw, void *buf, int size)
+{
+    return audio_pcm_sw_read (sw, buf, size);
+}
+
+static int fmod_ctl_in (HWVoiceIn *hw, int cmd, ...)
+{
+    int status;
+    FMODVoiceIn *fmd = (FMODVoiceIn *) hw;
+
+    switch (cmd) {
+    case VOICE_ENABLE:
+        status = FSOUND_Record_StartSample (fmd->fmod_sample, 1);
+        if (!status) {
+            fmod_logerr ("Failed to start recording\n");
+        }
+        break;
+
+    case VOICE_DISABLE:
+        status = FSOUND_Record_Stop ();
+        if (!status) {
+            fmod_logerr ("Failed to stop recording\n");
+        }
+        break;
+    }
+    return 0;
+}
+
 static void fmod_audio_fini (void *opaque)
 {
+    (void) opaque;
     FSOUND_Close ();
 }
 
-struct pcm_ops fmod_pcm_ops = {
+static struct audio_option fmod_options[] = {
-    fmod_hw_init,
+    {"DRV", AUD_OPT_STR, &conf.drvname,
-    fmod_hw_fini,
+     "FMOD driver", NULL, 0},
-    fmod_hw_run,
+    {"FREQ", AUD_OPT_INT, &conf.freq,
-    fmod_hw_write,
+     "Default frequency", NULL, 0},
-    fmod_hw_ctl
+    {"SAMPLES", AUD_OPT_INT, &conf.nb_samples,
+     "Buffer size in samples", NULL, 0},
+    {"CHANNELS", AUD_OPT_INT, &conf.nb_channels,
+     "Number of default channels (1 - mono, 2 - stereo)", NULL, 0},
+    {"BUFSIZE", AUD_OPT_INT, &conf.bufsize,
+     "(undocumented)", NULL, 0},
+#if 0
+    {"THRESHOLD", AUD_OPT_INT, &conf.threshold,
+     "(undocumented)"},
+#endif
+
+    {NULL, 0, NULL, NULL, NULL, 0}
 };
 
-struct audio_output_driver fmod_output_driver = {
+static struct audio_pcm_ops fmod_pcm_ops = {
-    "fmod",
+    fmod_init_out,
-    fmod_audio_init,
+    fmod_fini_out,
-    fmod_audio_fini,
+    fmod_run_out,
-    &fmod_pcm_ops,
+    fmod_write,
-    1,
+    fmod_ctl_out,
-    INT_MAX,
+
-    sizeof (FMODVoice)
+    fmod_init_in,
+    fmod_fini_in,
+    fmod_run_in,
+    fmod_read,
+    fmod_ctl_in
+};
+
+struct audio_driver fmod_audio_driver = {
+    INIT_FIELD (name           = ) "fmod",
+    INIT_FIELD (descr          = ) "FMOD 3.xx http://www.fmod.org",
+    INIT_FIELD (options        = ) fmod_options,
+    INIT_FIELD (init           = ) fmod_audio_init,
+    INIT_FIELD (fini           = ) fmod_audio_fini,
+    INIT_FIELD (pcm_ops        = ) &fmod_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 (FMODVoiceOut),
+    INIT_FIELD (voice_size_in  = ) sizeof (FMODVoiceIn)
 };

audio/mixeng.c

@@ -1,7 +1,7 @@
 /*
  * QEMU Mixing engine
  *
- * Copyright (c) 2004 Vassili Karpov (malc)
+ * Copyright (c) 2004-2005 Vassili Karpov (malc)
  * Copyright (c) 1998 Fabrice Bellard
  *
  * Permission is hereby granted, free of charge, to any person obtaining a copy
@@ -23,87 +23,174 @@
  * THE SOFTWARE.
  */
 #include "vl.h"
-//#define DEBUG_FP
-#include "audio/mixeng.h"
 
+#define AUDIO_CAP "mixeng"
+#include "audio_int.h"
+
+#define NOVOL
+
+/* 8 bit */
+#define ENDIAN_CONVERSION natural
+#define ENDIAN_CONVERT(v) (v)
+
+/* Signed 8 bit */
 #define IN_T int8_t
-#define IN_MIN CHAR_MIN
+#define IN_MIN SCHAR_MIN
-#define IN_MAX CHAR_MAX
+#define IN_MAX SCHAR_MAX
 #define SIGNED
+#define SHIFT 8
 #include "mixeng_template.h"
 #undef SIGNED
 #undef IN_MAX
 #undef IN_MIN
 #undef IN_T
+#undef SHIFT
 
+/* Unsigned 8 bit */
 #define IN_T uint8_t
 #define IN_MIN 0
 #define IN_MAX UCHAR_MAX
+#define SHIFT 8
 #include "mixeng_template.h"
 #undef IN_MAX
 #undef IN_MIN
 #undef IN_T
+#undef SHIFT
 
+#undef ENDIAN_CONVERT
+#undef ENDIAN_CONVERSION
+
+/* Signed 16 bit */
 #define IN_T int16_t
 #define IN_MIN SHRT_MIN
 #define IN_MAX SHRT_MAX
 #define SIGNED
+#define SHIFT 16
+#define ENDIAN_CONVERSION natural
+#define ENDIAN_CONVERT(v) (v)
 #include "mixeng_template.h"
+#undef ENDIAN_CONVERT
+#undef ENDIAN_CONVERSION
+#define ENDIAN_CONVERSION swap
+#define ENDIAN_CONVERT(v) bswap16 (v)
+#include "mixeng_template.h"
+#undef ENDIAN_CONVERT
+#undef ENDIAN_CONVERSION
 #undef SIGNED
 #undef IN_MAX
 #undef IN_MIN
 #undef IN_T
+#undef SHIFT
 
 #define IN_T uint16_t
 #define IN_MIN 0
 #define IN_MAX USHRT_MAX
+#define SHIFT 16
+#define ENDIAN_CONVERSION natural
+#define ENDIAN_CONVERT(v) (v)
 #include "mixeng_template.h"
+#undef ENDIAN_CONVERT
+#undef ENDIAN_CONVERSION
+#define ENDIAN_CONVERSION swap
+#define ENDIAN_CONVERT(v) bswap16 (v)
+#include "mixeng_template.h"
+#undef ENDIAN_CONVERT
+#undef ENDIAN_CONVERSION
 #undef IN_MAX
 #undef IN_MIN
 #undef IN_T
+#undef SHIFT
 
-t_sample *mixeng_conv[2][2][2] = {
+t_sample *mixeng_conv[2][2][2][2] = {
     {
         {
-            conv_uint8_t_to_mono,
+            {
-            conv_uint16_t_to_mono
+                conv_natural_uint8_t_to_mono,
+                conv_natural_uint16_t_to_mono
             },
             {
-            conv_int8_t_to_mono,
+                conv_natural_uint8_t_to_mono,
-            conv_int16_t_to_mono
+                conv_swap_uint16_t_to_mono
             }
         },
         {
             {
-            conv_uint8_t_to_stereo,
+                conv_natural_int8_t_to_mono,
-            conv_uint16_t_to_stereo
+                conv_natural_int16_t_to_mono
             },
             {
-            conv_int8_t_to_stereo,
+                conv_natural_int8_t_to_mono,
-            conv_int16_t_to_stereo
+                conv_swap_int16_t_to_mono
+            }
+        }
+    },
+    {
+        {
+            {
+                conv_natural_uint8_t_to_stereo,
+                conv_natural_uint16_t_to_stereo
+            },
+            {
+                conv_natural_uint8_t_to_stereo,
+                conv_swap_uint16_t_to_stereo
+            }
+        },
+        {
+            {
+                conv_natural_int8_t_to_stereo,
+                conv_natural_int16_t_to_stereo
+            },
+            {
+                conv_natural_int8_t_to_stereo,
+                conv_swap_int16_t_to_stereo
+            }
         }
     }
 };
 
-f_sample *mixeng_clip[2][2][2] = {
+f_sample *mixeng_clip[2][2][2][2] = {
     {
         {
-            clip_uint8_t_from_mono,
+            {
-            clip_uint16_t_from_mono
+                clip_natural_uint8_t_from_mono,
+                clip_natural_uint16_t_from_mono
             },
             {
-            clip_int8_t_from_mono,
+                clip_natural_uint8_t_from_mono,
-            clip_int16_t_from_mono
+                clip_swap_uint16_t_from_mono
             }
         },
         {
             {
-            clip_uint8_t_from_stereo,
+                clip_natural_int8_t_from_mono,
-            clip_uint16_t_from_stereo
+                clip_natural_int16_t_from_mono
             },
             {
-            clip_int8_t_from_stereo,
+                clip_natural_int8_t_from_mono,
-            clip_int16_t_from_stereo
+                clip_swap_int16_t_from_mono
+            }
+        }
+    },
+    {
+        {
+            {
+                clip_natural_uint8_t_from_stereo,
+                clip_natural_uint16_t_from_stereo
+            },
+            {
+                clip_natural_uint8_t_from_stereo,
+                clip_swap_uint16_t_from_stereo
+            }
+        },
+        {
+            {
+                clip_natural_int8_t_from_stereo,
+                clip_natural_int16_t_from_stereo
+            },
+            {
+                clip_natural_int8_t_from_stereo,
+                clip_swap_int16_t_from_stereo
+            }
         }
     }
 };
@@ -156,21 +243,13 @@ void *st_rate_start (int inrate, int outrate)
     rate_t rate = (rate_t) qemu_mallocz (sizeof (struct ratestuff));
 
     if (!rate) {
-        exit (EXIT_FAILURE);
+        return NULL;
-    }
-
-    if (inrate == outrate) {
-        // exit (EXIT_FAILURE);
-    }
-
-    if (inrate >= 65535 || outrate >= 65535) {
-        // exit (EXIT_FAILURE);
     }
 
     rate->opos = 0;
 
     /* increment */
-    rate->opos_inc = (inrate * ((int64_t) UINT_MAX)) / outrate;
+    rate->opos_inc = ((uint64_t) inrate << 32) / outrate;
 
     rate->ipos = 0;
     rate->ilast.l = 0;
@@ -178,78 +257,20 @@ void *st_rate_start (int inrate, int outrate)
     return rate;
 }
 
-/*
+#define NAME st_rate_flow_mix
- * Processed signed long samples from ibuf to obuf.
+#define OP(a, b) a += b
- * Return number of samples processed.
+#include "rate_template.h"
- */
-void st_rate_flow (void *opaque, st_sample_t *ibuf, st_sample_t *obuf,
-                   int *isamp, int *osamp)
-{
-    rate_t rate = (rate_t) opaque;
-    st_sample_t *istart, *iend;
-    st_sample_t *ostart, *oend;
-    st_sample_t ilast, icur, out;
-    int64_t t;
 
-    ilast = rate->ilast;
+#define NAME st_rate_flow
-
+#define OP(a, b) a = b
-    istart = ibuf;
+#include "rate_template.h"
-    iend = ibuf + *isamp;
-
-    ostart = obuf;
-    oend = obuf + *osamp;
-
-    if (rate->opos_inc == 1ULL << 32) {
-        int i, n = *isamp > *osamp ? *osamp : *isamp;
-        for (i = 0; i < n; i++) {
-            obuf[i].l += ibuf[i].r;
-            obuf[i].r += ibuf[i].r;
-        }
-        *isamp = n;
-        *osamp = n;
-        return;
-    }
-
-    while (obuf < oend) {
-
-        /* Safety catch to make sure we have input samples.  */
-        if (ibuf >= iend)
-            break;
-
-        /* read as many input samples so that ipos > opos */
-
-        while (rate->ipos <= (rate->opos >> 32)) {
-            ilast = *ibuf++;
-            rate->ipos++;
-            /* See if we finished the input buffer yet */
-            if (ibuf >= iend) goto the_end;
-        }
-
-        icur = *ibuf;
-
-        /* interpolate */
-        t = rate->opos & 0xffffffff;
-        out.l = (ilast.l * (INT_MAX - t) + icur.l * t) / INT_MAX;
-        out.r = (ilast.r * (INT_MAX - t) + icur.r * t) / INT_MAX;
-
-        /* output sample & increment position */
-#if 0
-        *obuf++ = out;
-#else
-        obuf->l += out.l;
-        obuf->r += out.r;
-        obuf += 1;
-#endif
-        rate->opos += rate->opos_inc;
-    }
-
-the_end:
-    *isamp = ibuf - istart;
-    *osamp = obuf - ostart;
-    rate->ilast = ilast;
-}
 
 void st_rate_stop (void *opaque)
 {
     qemu_free (opaque);
 }
+
+void mixeng_clear (st_sample_t *buf, int len)
+{
+    memset (buf, 0, len * sizeof (st_sample_t));
+}

audio/mixeng.h

@@ -1,7 +1,7 @@
 /*
  * QEMU Mixing engine header
  *
- * Copyright (c) 2004 Vassili Karpov (malc)
+ * Copyright (c) 2004-2005 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
@@ -24,16 +24,28 @@
 #ifndef QEMU_MIXENG_H
 #define QEMU_MIXENG_H
 
-typedef void (t_sample) (void *dst, const void *src, int samples);
+#ifdef FLOAT_MIXENG
-typedef void (f_sample) (void *dst, const void *src, int samples);
+typedef float real_t;
+typedef struct { int mute; real_t r; real_t l; } volume_t;
+typedef struct { real_t l; real_t r; } st_sample_t;
+#else
+typedef struct { int mute; int64_t r; int64_t l; } volume_t;
 typedef struct { int64_t l; int64_t r; } st_sample_t;
+#endif
 
-extern t_sample *mixeng_conv[2][2][2];
+typedef void (t_sample) (st_sample_t *dst, const void *src,
-extern f_sample *mixeng_clip[2][2][2];
+                         int samples, volume_t *vol);
+typedef void (f_sample) (void *dst, const st_sample_t *src, int samples);
+
+extern t_sample *mixeng_conv[2][2][2][2];
+extern f_sample *mixeng_clip[2][2][2][2];
 
 void *st_rate_start (int inrate, int outrate);
 void st_rate_flow (void *opaque, st_sample_t *ibuf, st_sample_t *obuf,
                    int *isamp, int *osamp);
+void st_rate_flow_mix (void *opaque, st_sample_t *ibuf, st_sample_t *obuf,
+                       int *isamp, int *osamp);
 void st_rate_stop (void *opaque);
+void mixeng_clear (st_sample_t *buf, int len);
 
 #endif  /* mixeng.h */

audio/mixeng_template.h

@@ -1,7 +1,7 @@
 /*
  * QEMU Mixing engine
  *
- * Copyright (c) 2004 Vassili Karpov (malc)
+ * Copyright (c) 2004-2005 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
@@ -27,85 +27,151 @@
  * dec++'ified by Dscho
  */
 
-#ifdef SIGNED
+#ifndef SIGNED
-#define HALFT IN_MAX
+#define HALF (IN_MAX >> 1)
-#define HALF IN_MAX
-#else
-#define HALFT ((IN_MAX)>>1)
-#define HALF HALFT
 #endif
 
-static int64_t inline glue(conv_,IN_T) (IN_T v)
+#ifdef NOVOL
-{
+#define VOL(a, b) a
-#ifdef SIGNED
-    return (INT_MAX*(int64_t)v)/HALF;
 #else
-    return (INT_MAX*((int64_t)v-HALFT))/HALF;
+#ifdef FLOAT_MIXENG
+#define VOL(a, b) ((a) * (b))
+#else
+#define VOL(a, b) ((a) * (b)) >> 32
+#endif
+#endif
+
+#define ET glue (ENDIAN_CONVERSION, glue (_, IN_T))
+
+#ifdef FLOAT_MIXENG
+static real_t inline glue (conv_, ET) (IN_T v)
+{
+    IN_T nv = ENDIAN_CONVERT (v);
+
+#ifdef RECIPROCAL
+#ifdef SIGNED
+    return nv * (1.f / (real_t) (IN_MAX - IN_MIN));
+#else
+    return (nv - HALF) * (1.f / (real_t) IN_MAX);
+#endif
+#else  /* !RECIPROCAL */
+#ifdef SIGNED
+    return nv / (real_t) (IN_MAX - IN_MIN);
+#else
+    return (nv - HALF) / (real_t) IN_MAX;
+#endif
 #endif
 }
 
-static IN_T inline glue(clip_,IN_T) (int64_t v)
+static IN_T inline glue (clip_, ET) (real_t v)
 {
-    if (v >= INT_MAX)
+    if (v >= 0.5) {
         return IN_MAX;
-    else if (v < -INT_MAX)
+    }
+    else if (v < -0.5) {
         return IN_MIN;
+    }
 
 #ifdef SIGNED
-    return (IN_T) (v*HALF/INT_MAX);
+    return ENDIAN_CONVERT ((IN_T) (v * (IN_MAX - IN_MIN)));
 #else
-    return (IN_T) (v+INT_MAX/2)*HALF/INT_MAX;
+    return ENDIAN_CONVERT ((IN_T) ((v * IN_MAX) + HALF));
 #endif
 }
 
-static void glue(glue(conv_,IN_T),_to_stereo) (void *dst, const void *src,
+#else  /* !FLOAT_MIXENG */
-                                               int samples)
+
+static inline int64_t glue (conv_, ET) (IN_T v)
 {
-    st_sample_t *out = (st_sample_t *) dst;
+    IN_T nv = ENDIAN_CONVERT (v);
+#ifdef SIGNED
+    return ((int64_t) nv) << (32 - SHIFT);
+#else
+    return ((int64_t) nv - HALF) << (32 - SHIFT);
+#endif
+}
+
+static inline IN_T glue (clip_, ET) (int64_t v)
+{
+    if (v >= 0x7f000000) {
+        return IN_MAX;
+    }
+    else if (v < -2147483648LL) {
+        return IN_MIN;
+    }
+
+#ifdef SIGNED
+    return ENDIAN_CONVERT ((IN_T) (v >> (32 - SHIFT)));
+#else
+    return ENDIAN_CONVERT ((IN_T) ((v >> (32 - SHIFT)) + HALF));
+#endif
+}
+#endif
+
+static void glue (glue (conv_, ET), _to_stereo)
+    (st_sample_t *dst, const void *src, int samples, volume_t *vol)
+{
+    st_sample_t *out = dst;
     IN_T *in = (IN_T *) src;
+#ifndef NOVOL
+    if (vol->mute) {
+        mixeng_clear (dst, samples);
+        return;
+    }
+#else
+    (void) vol;
+#endif
     while (samples--) {
-        out->l = glue(conv_,IN_T) (*in++);
+        out->l = VOL (glue (conv_, ET) (*in++), vol->l);
-        out->r = glue(conv_,IN_T) (*in++);
+        out->r = VOL (glue (conv_, ET) (*in++), vol->r);
         out += 1;
     }
 }
 
-static void glue(glue(conv_,IN_T),_to_mono) (void *dst, const void *src,
+static void glue (glue (conv_, ET), _to_mono)
-                                             int samples)
+    (st_sample_t *dst, const void *src, int samples, volume_t *vol)
 {
-    st_sample_t *out = (st_sample_t *) dst;
+    st_sample_t *out = dst;
     IN_T *in = (IN_T *) src;
+#ifndef NOVOL
+    if (vol->mute) {
+        mixeng_clear (dst, samples);
+        return;
+    }
+#else
+    (void) vol;
+#endif
     while (samples--) {
-        out->l = glue(conv_,IN_T) (in[0]);
+        out->l = VOL (glue (conv_, ET) (in[0]), vol->l);
         out->r = out->l;
         out += 1;
         in += 1;
     }
 }
 
-static void glue(glue(clip_,IN_T),_from_stereo) (void *dst, const void *src,
+static void glue (glue (clip_, ET), _from_stereo)
-                                                 int samples)
+    (void *dst, const st_sample_t *src, int samples)
 {
-    st_sample_t *in = (st_sample_t *) src;
+    const st_sample_t *in = src;
     IN_T *out = (IN_T *) dst;
     while (samples--) {
-        *out++ = glue(clip_,IN_T) (in->l);
+        *out++ = glue (clip_, ET) (in->l);
-        *out++ = glue(clip_,IN_T) (in->r);
+        *out++ = glue (clip_, ET) (in->r);
         in += 1;
     }
 }
 
-static void glue(glue(clip_,IN_T),_from_mono) (void *dst, const void *src,
+static void glue (glue (clip_, ET), _from_mono)
-                                               int samples)
+    (void *dst, const st_sample_t *src, int samples)
 {
-    st_sample_t *in = (st_sample_t *) src;
+    const st_sample_t *in = src;
     IN_T *out = (IN_T *) dst;
     while (samples--) {
-        *out++ = glue(clip_,IN_T) (in->l + in->r);
+        *out++ = glue (clip_, ET) (in->l + in->r);
         in += 1;
     }
 }
 
+#undef ET
 #undef HALF
-#undef HALFT
+#undef VOL
-

audio/noaudio.c

@@ -1,7 +1,7 @@
 /*
- * QEMU NULL audio output driver
+ * QEMU Timer based audio emulation
  *
- * Copyright (c) 2004 Vassili Karpov (malc)
+ * Copyright (c) 2004-2005 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
@@ -23,77 +23,110 @@
  */
 #include "vl.h"
 
-#include "audio/audio_int.h"
+#define AUDIO_CAP "noaudio"
+#include "audio_int.h"
 
-typedef struct NoVoice {
+typedef struct NoVoiceOut {
-    HWVoice hw;
+    HWVoiceOut hw;
     int64_t old_ticks;
-} NoVoice;
+} NoVoiceOut;
 
-#define dolog(...) AUD_log ("noaudio", __VA_ARGS__)
+typedef struct NoVoiceIn {
-#ifdef DEBUG
+    HWVoiceIn hw;
-#define ldebug(...) dolog (__VA_ARGS__)
+    int64_t old_ticks;
-#else
+} NoVoiceIn;
-#define ldebug(...)
-#endif
 
-static void no_hw_run (HWVoice *hw)
+static int no_run_out (HWVoiceOut *hw)
 {
-    NoVoice *no = (NoVoice *) hw;
+    NoVoiceOut *no = (NoVoiceOut *) hw;
-    int rpos, live, decr, samples;
+    int live, decr, samples;
-    st_sample_t *src;
     int64_t now = qemu_get_clock (vm_clock);
     int64_t ticks = now - no->old_ticks;
-    int64_t bytes = (ticks * hw->bytes_per_second) / ticks_per_sec;
+    int64_t bytes = (ticks * hw->info.bytes_per_second) / ticks_per_sec;
 
-    if (bytes > INT_MAX)
+    if (bytes > INT_MAX) {
-        samples = INT_MAX >> hw->shift;
+        samples = INT_MAX >> hw->info.shift;
-    else
+    }
-        samples = bytes >> hw->shift;
+    else {
+        samples = bytes >> hw->info.shift;
+    }
 
-    live = pcm_hw_get_live (hw);
+    live = audio_pcm_hw_get_live_out (&no->hw);
-    if (live <= 0)
+    if (!live) {
-        return;
+        return 0;
+    }
 
     no->old_ticks = now;
     decr = audio_MIN (live, samples);
-    samples = decr;
+    hw->rpos = (hw->rpos + decr) % hw->samples;
-    rpos = hw->rpos;
+    return decr;
-    while (samples) {
-        int left_till_end_samples = hw->samples - rpos;
-        int convert_samples = audio_MIN (samples, left_till_end_samples);
-
-        src = advance (hw->mix_buf, rpos * sizeof (st_sample_t));
-        memset (src, 0, convert_samples * sizeof (st_sample_t));
-
-        rpos = (rpos + convert_samples) % hw->samples;
-        samples -= convert_samples;
-    }
-
-    pcm_hw_dec_live (hw, decr);
-    hw->rpos = rpos;
 }
 
-static int no_hw_write (SWVoice *sw, void *buf, int len)
+static int no_write (SWVoiceOut *sw, void *buf, int len)
 {
-    return pcm_hw_write (sw, buf, len);
+    return audio_pcm_sw_write (sw, buf, len);
 }
 
-static int no_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt)
+static int no_init_out (HWVoiceOut *hw, int freq,
+                        int nchannels, audfmt_e fmt)
 {
-    hw->freq = freq;
+    audio_pcm_init_info (&hw->info, freq, nchannels, fmt, 0);
-    hw->nchannels = nchannels;
-    hw->fmt = fmt;
     hw->bufsize = 4096;
     return 0;
 }
 
-static void no_hw_fini (HWVoice *hw)
+static void no_fini_out (HWVoiceOut *hw)
 {
     (void) hw;
 }
 
-static int no_hw_ctl (HWVoice *hw, int cmd, ...)
+static int no_ctl_out (HWVoiceOut *hw, int cmd, ...)
+{
+    (void) hw;
+    (void) cmd;
+    return 0;
+}
+
+static int no_init_in (HWVoiceIn *hw, int freq,
+                       int nchannels, audfmt_e fmt)
+{
+    audio_pcm_init_info (&hw->info, freq, nchannels, fmt, 0);
+    hw->bufsize = 4096;
+    return 0;
+}
+
+static void no_fini_in (HWVoiceIn *hw)
+{
+    (void) hw;
+}
+
+static int no_run_in (HWVoiceIn *hw)
+{
+    NoVoiceIn *no = (NoVoiceIn *) hw;
+    int64_t now = qemu_get_clock (vm_clock);
+    int64_t ticks = now - no->old_ticks;
+    int64_t bytes = (ticks * hw->info.bytes_per_second) / ticks_per_sec;
+    int live = audio_pcm_hw_get_live_in (hw);
+    int dead = hw->samples - live;
+    int samples;
+
+    bytes = audio_MIN (bytes, INT_MAX);
+    samples = bytes >> hw->info.shift;
+    samples = audio_MIN (samples, dead);
+
+    return samples;
+}
+
+static int no_read (SWVoiceIn *sw, void *buf, int size)
+{
+    int samples = size >> sw->info.shift;
+    int total = sw->hw->total_samples_captured - sw->total_hw_samples_acquired;
+    int to_clear = audio_MIN (samples, total);
+    audio_pcm_info_clear_buf (&sw->info, buf, to_clear);
+    return to_clear;
+}
+
+static int no_ctl_in (HWVoiceIn *hw, int cmd, ...)
 {
     (void) hw;
     (void) cmd;
@@ -107,22 +140,33 @@ static void *no_audio_init (void)
 
 static void no_audio_fini (void *opaque)
 {
+    (void) opaque;
 }
 
-struct pcm_ops no_pcm_ops = {
+static struct audio_pcm_ops no_pcm_ops = {
-    no_hw_init,
+    no_init_out,
-    no_hw_fini,
+    no_fini_out,
-    no_hw_run,
+    no_run_out,
-    no_hw_write,
+    no_write,
-    no_hw_ctl
+    no_ctl_out,
+
+    no_init_in,
+    no_fini_in,
+    no_run_in,
+    no_read,
+    no_ctl_in
 };
 
-struct audio_output_driver no_output_driver = {
+struct audio_driver no_audio_driver = {
-    "none",
+    INIT_FIELD (name           = ) "none",
-    no_audio_init,
+    INIT_FIELD (descr          = ) "Timer based audio emulation",
-    no_audio_fini,
+    INIT_FIELD (options        = ) NULL,
-    &no_pcm_ops,
+    INIT_FIELD (init           = ) no_audio_init,
-    1,
+    INIT_FIELD (fini           = ) no_audio_fini,
-    1,
+    INIT_FIELD (pcm_ops        = ) &no_pcm_ops,
-    sizeof (NoVoice)
+    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 (NoVoiceOut),
+    INIT_FIELD (voice_size_in  = ) sizeof (NoVoiceIn)
 };

audio/ossaudio.c

@@ -1,7 +1,7 @@
 /*
- * QEMU OSS audio output driver
+ * QEMU OSS audio driver
  *
- * Copyright (c) 2003-2004 Vassili Karpov (malc)
+ * Copyright (c) 2003-2005 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
@@ -25,45 +25,42 @@
 #include <sys/types.h>
 #include <sys/ioctl.h>
 #include <sys/soundcard.h>
-#include <assert.h>
 #include "vl.h"
 
-#include "audio/audio_int.h"
+#define AUDIO_CAP "oss"
+#include "audio_int.h"
 
-typedef struct OSSVoice {
+typedef struct OSSVoiceOut {
-    HWVoice hw;
+    HWVoiceOut hw;
     void *pcm_buf;
     int fd;
     int nfrags;
     int fragsize;
     int mmapped;
     int old_optr;
-} OSSVoice;
+} OSSVoiceOut;
 
-#define dolog(...) AUD_log ("oss", __VA_ARGS__)
+typedef struct OSSVoiceIn {
-#ifdef DEBUG
+    HWVoiceIn hw;
-#define ldebug(...) dolog (__VA_ARGS__)
+    void *pcm_buf;
-#else
+    int fd;
-#define ldebug(...)
+    int nfrags;
-#endif
+    int fragsize;
-
+    int old_optr;
-#define QC_OSS_FRAGSIZE "QEMU_OSS_FRAGSIZE"
+} OSSVoiceIn;
-#define QC_OSS_NFRAGS   "QEMU_OSS_NFRAGS"
-#define QC_OSS_MMAP     "QEMU_OSS_MMAP"
-#define QC_OSS_DEV      "QEMU_OSS_DEV"
-
-#define errstr() strerror (errno)
 
 static struct {
     int try_mmap;
     int nfrags;
     int fragsize;
-    const char *dspname;
+    const char *devpath_out;
+    const char *devpath_in;
 } conf = {
     .try_mmap = 0,
     .nfrags = 4,
     .fragsize = 4096,
-    .dspname = "/dev/dsp"
+    .devpath_out = "/dev/dsp",
+    .devpath_in = "/dev/dsp"
 };
 
 struct oss_params {
@@ -74,65 +71,141 @@ struct oss_params {
     int fragsize;
 };
 
-static int oss_hw_write (SWVoice *sw, void *buf, int len)
+static void GCC_FMT_ATTR (2, 3) oss_logerr (int err, const char *fmt, ...)
 {
-    return pcm_hw_write (sw, buf, len);
+    va_list ap;
+
+    AUD_vlog (AUDIO_CAP, fmt, ap);
+
+    va_start (ap, fmt);
+    AUD_log (AUDIO_CAP, "Reason: %s\n", strerror (err));
+    va_end (ap);
 }
 
-static int AUD_to_ossfmt (audfmt_e fmt)
+static void GCC_FMT_ATTR (3, 4) oss_logerr2 (
+    int err,
+    const char *typ,
+    const char *fmt,
+    ...
+    )
+{
+    va_list ap;
+
+    AUD_log (AUDIO_CAP, "Can 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", strerror (err));
+}
+
+static void oss_anal_close (int *fdp)
+{
+    int err = close (*fdp);
+    if (err) {
+        oss_logerr (errno, "Failed to close file(fd=%d)\n", *fdp);
+    }
+    *fdp = -1;
+}
+
+static int oss_write (SWVoiceOut *sw, void *buf, int len)
+{
+    return audio_pcm_sw_write (sw, buf, len);
+}
+
+static int aud_to_ossfmt (audfmt_e fmt)
 {
     switch (fmt) {
-    case AUD_FMT_S8: return AFMT_S8;
+    case AUD_FMT_S8:
-    case AUD_FMT_U8: return AFMT_U8;
+        return AFMT_S8;
-    case AUD_FMT_S16: return AFMT_S16_LE;
+
-    case AUD_FMT_U16: return AFMT_U16_LE;
+    case AUD_FMT_U8:
+        return AFMT_U8;
+
+    case AUD_FMT_S16:
+        return AFMT_S16_LE;
+
+    case AUD_FMT_U16:
+        return AFMT_U16_LE;
+
     default:
-        dolog ("Internal logic error: Bad audio format %d\nAborting\n", fmt);
+        dolog ("Internal logic error: Bad audio format %d\n", fmt);
-        exit (EXIT_FAILURE);
+#ifdef DEBUG_AUDIO
+        abort ();
+#endif
+        return AFMT_U8;
     }
 }
 
-static int oss_to_audfmt (int fmt)
+static int oss_to_audfmt (int ossfmt, audfmt_e *fmt, int *endianness)
 {
-    switch (fmt) {
+    switch (ossfmt) {
-    case AFMT_S8: return AUD_FMT_S8;
+    case AFMT_S8:
-    case AFMT_U8: return AUD_FMT_U8;
+        *endianness =0;
-    case AFMT_S16_LE: return AUD_FMT_S16;
+        *fmt = AUD_FMT_S8;
-    case AFMT_U16_LE: return AUD_FMT_U16;
+        break;
+
+    case AFMT_U8:
+        *endianness = 0;
+        *fmt = AUD_FMT_U8;
+        break;
+
+    case AFMT_S16_LE:
+        *endianness = 0;
+        *fmt = AUD_FMT_S16;
+        break;
+
+    case AFMT_U16_LE:
+        *endianness = 0;
+        *fmt = AUD_FMT_U16;
+        break;
+
+    case AFMT_S16_BE:
+        *endianness = 1;
+        *fmt = AUD_FMT_S16;
+        break;
+
+    case AFMT_U16_BE:
+        *endianness = 1;
+        *fmt = AUD_FMT_U16;
+        break;
+
     default:
-        dolog ("Internal logic error: Unrecognized OSS audio format %d\n"
+        dolog ("Unrecognized audio format %d\n", ossfmt);
-               "Aborting\n",
+        return -1;
-               fmt);
-        exit (EXIT_FAILURE);
     }
+
+    return 0;
 }
 
-#ifdef DEBUG_PCM
+#ifdef DEBUG_MISMATCHES
-static void oss_dump_pcm_info (struct oss_params *req, struct oss_params *obt)
+static void oss_dump_info (struct oss_params *req, struct oss_params *obt)
 {
     dolog ("parameter | requested value | obtained value\n");
     dolog ("format    |      %10d |     %10d\n", req->fmt, obt->fmt);
-    dolog ("channels  |      %10d |     %10d\n", req->nchannels, obt->nchannels);
+    dolog ("channels  |      %10d |     %10d\n",
+           req->nchannels, obt->nchannels);
     dolog ("frequency |      %10d |     %10d\n", req->freq, obt->freq);
     dolog ("nfrags    |      %10d |     %10d\n", req->nfrags, obt->nfrags);
-    dolog ("fragsize  |      %10d |     %10d\n", req->fragsize, obt->fragsize);
+    dolog ("fragsize  |      %10d |     %10d\n",
+           req->fragsize, obt->fragsize);
 }
 #endif
 
-static int oss_open (struct oss_params *req, struct oss_params *obt, int *pfd)
+static int oss_open (int in, struct oss_params *req,
+                     struct oss_params *obt, int *pfd)
 {
     int fd;
     int mmmmssss;
     audio_buf_info abinfo;
     int fmt, freq, nchannels;
-    const char *dspname = conf.dspname;
+    const char *dspname = in ? conf.devpath_in : conf.devpath_out;
+    const char *typ = in ? "ADC" : "DAC";
 
-    fd = open (dspname, O_WRONLY | O_NONBLOCK);
+    fd = open (dspname, (in ? O_RDONLY : O_WRONLY) | O_NONBLOCK);
     if (-1 == fd) {
-        dolog ("Could not initialize audio hardware. Failed to open `%s':\n"
+        oss_logerr2 (errno, typ, "Failed to open `%s'\n", dspname);
-               "Reason:%s\n",
-               dspname,
-               errstr ());
         return -1;
     }
 
@@ -141,52 +214,35 @@ static int oss_open (struct oss_params *req, struct oss_params *obt, int *pfd)
     fmt = req->fmt;
 
     if (ioctl (fd, SNDCTL_DSP_SAMPLESIZE, &fmt)) {
-        dolog ("Could not initialize audio hardware\n"
+        oss_logerr2 (errno, typ, "Failed to set sample size %d\n", req->fmt);
-               "Failed to set sample size\n"
-               "Reason: %s\n",
-               errstr ());
         goto err;
     }
 
     if (ioctl (fd, SNDCTL_DSP_CHANNELS, &nchannels)) {
-        dolog ("Could not initialize audio hardware\n"
+        oss_logerr2 (errno, typ, "Failed to set number of channels %d\n",
-               "Failed to set number of channels\n"
+                     req->nchannels);
-               "Reason: %s\n",
-               errstr ());
         goto err;
     }
 
     if (ioctl (fd, SNDCTL_DSP_SPEED, &freq)) {
-        dolog ("Could not initialize audio hardware\n"
+        oss_logerr2 (errno, typ, "Failed to set frequency %d\n", req->freq);
-               "Failed to set frequency\n"
-               "Reason: %s\n",
-               errstr ());
         goto err;
     }
 
     if (ioctl (fd, SNDCTL_DSP_NONBLOCK)) {
-        dolog ("Could not initialize audio hardware\n"
+        oss_logerr2 (errno, typ, "Failed to set non-blocking mode\n");
-               "Failed to set non-blocking mode\n"
-               "Reason: %s\n",
-               errstr ());
         goto err;
     }
 
     mmmmssss = (req->nfrags << 16) | lsbindex (req->fragsize);
     if (ioctl (fd, SNDCTL_DSP_SETFRAGMENT, &mmmmssss)) {
-        dolog ("Could not initialize audio hardware\n"
+        oss_logerr2 (errno, typ, "Failed to set buffer length (%d, %d)\n",
-               "Failed to set buffer length (%d, %d)\n"
+                     req->nfrags, req->fragsize);
-               "Reason:%s\n",
-               conf.nfrags, conf.fragsize,
-               errstr ());
         goto err;
     }
 
-    if (ioctl (fd, SNDCTL_DSP_GETOSPACE, &abinfo)) {
+    if (ioctl (fd, in ? SNDCTL_DSP_GETISPACE : SNDCTL_DSP_GETOSPACE, &abinfo)) {
-        dolog ("Could not initialize audio hardware\n"
+        oss_logerr2 (errno, typ, "Failed to get buffer length\n");
-               "Failed to get buffer length\n"
-               "Reason:%s\n",
-               errstr ());
         goto err;
     }
 
@@ -202,25 +258,25 @@ static int oss_open (struct oss_params *req, struct oss_params *obt, int *pfd)
         (req->freq != obt->freq) ||
         (req->fragsize != obt->fragsize) ||
         (req->nfrags != obt->nfrags)) {
-#ifdef DEBUG_PCM
+#ifdef DEBUG_MISMATCHES
         dolog ("Audio parameters mismatch\n");
-        oss_dump_pcm_info (req, obt);
+        oss_dump_info (req, obt);
 #endif
     }
 
-#ifdef DEBUG_PCM
+#ifdef DEBUG
-    oss_dump_pcm_info (req, obt);
+    oss_dump_info (req, obt);
 #endif
     return 0;
 
-err:
+ err:
-    close (fd);
+    oss_anal_close (&fd);
     return -1;
 }
 
-static void oss_hw_run (HWVoice *hw)
+static int oss_run_out (HWVoiceOut *hw)
 {
-    OSSVoice *oss = (OSSVoice *) hw;
+    OSSVoiceOut *oss = (OSSVoiceOut *) hw;
     int err, rpos, live, decr;
     int samples;
     uint8_t *dst;
@@ -228,23 +284,25 @@ static void oss_hw_run (HWVoice *hw)
     struct audio_buf_info abinfo;
     struct count_info cntinfo;
 
-    live = pcm_hw_get_live (hw);
+    live = audio_pcm_hw_get_live_out (hw);
-    if (live <= 0)
+    if (!live) {
-        return;
+        return 0;
+    }
 
     if (oss->mmapped) {
         int bytes;
 
         err = ioctl (oss->fd, SNDCTL_DSP_GETOPTR, &cntinfo);
         if (err < 0) {
-            dolog ("SNDCTL_DSP_GETOPTR failed\nReason: %s\n", errstr ());
+            oss_logerr (errno, "SNDCTL_DSP_GETOPTR failed\n");
-            return;
+            return 0;
         }
 
         if (cntinfo.ptr == oss->old_optr) {
-            if (abs (hw->samples - live) < 64)
+            if (abs (hw->samples - live) < 64) {
-                dolog ("overrun\n");
+                dolog ("warning: overrun\n");
-            return;
+            }
+            return 0;
         }
 
         if (cntinfo.ptr > oss->old_optr) {
@@ -254,18 +312,25 @@ static void oss_hw_run (HWVoice *hw)
             bytes = hw->bufsize + cntinfo.ptr - oss->old_optr;
         }
 
-        decr = audio_MIN (bytes >> hw->shift, live);
+        decr = audio_MIN (bytes >> hw->info.shift, live);
     }
     else {
         err = ioctl (oss->fd, SNDCTL_DSP_GETOSPACE, &abinfo);
         if (err < 0) {
-            dolog ("SNDCTL_DSP_GETOSPACE failed\nReason: %s\n", errstr ());
+            oss_logerr (errno, "SNDCTL_DSP_GETOPTR failed\n");
-            return;
+            return 0;
         }
 
-        decr = audio_MIN (abinfo.bytes >> hw->shift, live);
+        if (abinfo.bytes < 0 || abinfo.bytes > hw->bufsize) {
-        if (decr <= 0)
+            ldebug ("warning: invalid available size, size=%d bufsize=%d\n",
-            return;
+                    abinfo.bytes, hw->bufsize);
+            return 0;
+        }
+
+        decr = audio_MIN (abinfo.bytes >> hw->info.shift, live);
+        if (!decr) {
+            return 0;
+        }
     }
 
     samples = decr;
@@ -274,33 +339,41 @@ static void oss_hw_run (HWVoice *hw)
         int left_till_end_samples = hw->samples - rpos;
         int convert_samples = audio_MIN (samples, left_till_end_samples);
 
-        src = advance (hw->mix_buf, rpos * sizeof (st_sample_t));
+        src = hw->mix_buf + rpos;
-        dst = advance (oss->pcm_buf, rpos << hw->shift);
+        dst = advance (oss->pcm_buf, rpos << hw->info.shift);
 
         hw->clip (dst, src, convert_samples);
         if (!oss->mmapped) {
             int written;
 
-            written = write (oss->fd, dst, convert_samples << hw->shift);
+            written = write (oss->fd, dst, convert_samples << hw->info.shift);
             /* XXX: follow errno recommendations ? */
             if (written == -1) {
-                dolog ("Failed to write audio\nReason: %s\n", errstr ());
+                oss_logerr (
+                    errno,
+                    "Failed to write %d bytes of audio data from %p\n",
+                    convert_samples << hw->info.shift,
+                    dst
+                    );
                 continue;
             }
 
-            if (written != convert_samples << hw->shift) {
+            if (written != convert_samples << hw->info.shift) {
-                int wsamples = written >> hw->shift;
+                int wsamples = written >> hw->info.shift;
-                int wbytes = wsamples << hw->shift;
+                int wbytes = wsamples << hw->info.shift;
                 if (wbytes != written) {
-                    dolog ("Unaligned write %d, %d\n", wbytes, written);
+                    dolog ("warning: misaligned write %d (requested %d), "
+                           "alignment %d\n",
+                           wbytes, written, hw->info.align + 1);
                 }
-                memset (src, 0, wbytes);
+                mixeng_clear (src, wsamples);
-                decr -= samples;
+                decr -= wsamples;
                 rpos = (rpos + wsamples) % hw->samples;
                 break;
             }
         }
-        memset (src, 0, convert_samples * sizeof (st_sample_t));
+
+        mixeng_clear (src, convert_samples);
 
         rpos = (rpos + convert_samples) % hw->samples;
         samples -= convert_samples;
@@ -309,28 +382,24 @@ static void oss_hw_run (HWVoice *hw)
         oss->old_optr = cntinfo.ptr;
     }
 
-    pcm_hw_dec_live (hw, decr);
     hw->rpos = rpos;
+    return decr;
 }
 
-static void oss_hw_fini (HWVoice *hw)
+static void oss_fini_out (HWVoiceOut *hw)
 {
     int err;
-    OSSVoice *oss = (OSSVoice *) hw;
+    OSSVoiceOut *oss = (OSSVoiceOut *) hw;
 
-    ldebug ("oss_hw_fini\n");
+    ldebug ("oss_fini\n");
-    err = close (oss->fd);
+    oss_anal_close (&oss->fd);
-    if (err) {
-        dolog ("Failed to close OSS descriptor\nReason: %s\n", errstr ());
-    }
-    oss->fd = -1;
 
     if (oss->pcm_buf) {
         if (oss->mmapped) {
             err = munmap (oss->pcm_buf, hw->bufsize);
             if (err) {
-                dolog ("Failed to unmap OSS buffer\nReason: %s\n",
+                oss_logerr (errno, "Failed to unmap buffer %p, size %d\n",
-                       errstr ());
+                            oss->pcm_buf, hw->bufsize);
             }
         }
         else {
@@ -340,25 +409,38 @@ static void oss_hw_fini (HWVoice *hw)
     }
 }
 
-static int oss_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt)
+static int oss_init_out (HWVoiceOut *hw, int freq, int nchannels, audfmt_e fmt)
 {
-    OSSVoice *oss = (OSSVoice *) hw;
+    OSSVoiceOut *oss = (OSSVoiceOut *) hw;
     struct oss_params req, obt;
+    int endianness;
+    int err;
+    int fd;
+    audfmt_e effective_fmt;
 
-    assert (!oss->fd);
+    req.fmt = aud_to_ossfmt (fmt);
-    req.fmt = AUD_to_ossfmt (fmt);
     req.freq = freq;
     req.nchannels = nchannels;
     req.fragsize = conf.fragsize;
     req.nfrags = conf.nfrags;
 
-    if (oss_open (&req, &obt, &oss->fd))
+    if (oss_open (0, &req, &obt, &fd)) {
         return -1;
+    }
 
-    hw->freq = obt.freq;
+    err = oss_to_audfmt (obt.fmt, &effective_fmt, &endianness);
-    hw->fmt = oss_to_audfmt (obt.fmt);
+    if (err) {
-    hw->nchannels = obt.nchannels;
+        oss_anal_close (&fd);
+        return -1;
+    }
 
+    audio_pcm_init_info (
+        &hw->info,
+        obt.freq,
+        obt.nchannels,
+        effective_fmt,
+        audio_need_to_swap_endian (endianness)
+        );
     oss->nfrags = obt.nfrags;
     oss->fragsize = obt.fragsize;
     hw->bufsize = obt.nfrags * obt.fragsize;
@@ -366,22 +448,23 @@ static int oss_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt)
     oss->mmapped = 0;
     if (conf.try_mmap) {
         oss->pcm_buf = mmap (0, hw->bufsize, PROT_READ | PROT_WRITE,
-                             MAP_SHARED, oss->fd, 0);
+                             MAP_SHARED, fd, 0);
         if (oss->pcm_buf == MAP_FAILED) {
-            dolog ("Failed to mmap OSS device\nReason: %s\n",
+            oss_logerr (errno, "Failed to map %d bytes of DAC\n",
-                   errstr ());
+                        hw->bufsize);
         } else {
             int err;
             int trig = 0;
-            if (ioctl (oss->fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) {
+            if (ioctl (fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) {
-                dolog ("SNDCTL_DSP_SETTRIGGER 0 failed\nReason: %s\n",
+                oss_logerr (errno, "SNDCTL_DSP_SETTRIGGER 0 failed\n");
-                       errstr ());
             }
             else {
                 trig = PCM_ENABLE_OUTPUT;
-                if (ioctl (oss->fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) {
+                if (ioctl (fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) {
-                    dolog ("SNDCTL_DSP_SETTRIGGER PCM_ENABLE_OUTPUT failed\n"
+                    oss_logerr (
-                           "Reason: %s\n", errstr ());
+                        errno,
+                        "SNDCTL_DSP_SETTRIGGER PCM_ENABLE_OUTPUT failed\n"
+                        );
                 }
                 else {
                     oss->mmapped = 1;
@@ -391,8 +474,8 @@ static int oss_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt)
             if (!oss->mmapped) {
                 err = munmap (oss->pcm_buf, hw->bufsize);
                 if (err) {
-                    dolog ("Failed to unmap OSS device\nReason: %s\n",
+                    oss_logerr (errno, "Failed to unmap buffer %p size %d\n",
-                           errstr ());
+                                oss->pcm_buf, hw->bufsize);
                 }
             }
         }
@@ -401,31 +484,34 @@ static int oss_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt)
     if (!oss->mmapped) {
         oss->pcm_buf = qemu_mallocz (hw->bufsize);
         if (!oss->pcm_buf) {
-            close (oss->fd);
+            oss_anal_close (&fd);
-            oss->fd = -1;
             return -1;
         }
     }
 
+    oss->fd = fd;
     return 0;
 }
 
-static int oss_hw_ctl (HWVoice *hw, int cmd, ...)
+static int oss_ctl_out (HWVoiceOut *hw, int cmd, ...)
 {
     int trig;
-    OSSVoice *oss = (OSSVoice *) hw;
+    OSSVoiceOut *oss = (OSSVoiceOut *) hw;
 
-    if (!oss->mmapped)
+    if (!oss->mmapped) {
         return 0;
+    }
 
     switch (cmd) {
     case VOICE_ENABLE:
         ldebug ("enabling voice\n");
-        pcm_hw_clear (hw, oss->pcm_buf, hw->samples);
+        audio_pcm_info_clear_buf (&hw->info, oss->pcm_buf, hw->samples);
         trig = PCM_ENABLE_OUTPUT;
         if (ioctl (oss->fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) {
-            dolog ("SNDCTL_DSP_SETTRIGGER PCM_ENABLE_OUTPUT failed\n"
+            oss_logerr (
-                   "Reason: %s\n", errstr ());
+                errno,
+                "SNDCTL_DSP_SETTRIGGER PCM_ENABLE_OUTPUT failed\n"
+                );
             return -1;
         }
         break;
@@ -434,8 +520,7 @@ static int oss_hw_ctl (HWVoice *hw, int cmd, ...)
         ldebug ("disabling voice\n");
         trig = 0;
         if (ioctl (oss->fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) {
-            dolog ("SNDCTL_DSP_SETTRIGGER 0 failed\nReason: %s\n",
+            oss_logerr (errno, "SNDCTL_DSP_SETTRIGGER 0 failed\n");
-                   errstr ());
             return -1;
         }
         break;
@@ -443,33 +528,194 @@ static int oss_hw_ctl (HWVoice *hw, int cmd, ...)
     return 0;
 }
 
+static int oss_init_in (HWVoiceIn *hw,
+                        int freq, int nchannels, audfmt_e fmt)
+{
+    OSSVoiceIn *oss = (OSSVoiceIn *) hw;
+    struct oss_params req, obt;
+    int endianness;
+    int err;
+    int fd;
+    audfmt_e effective_fmt;
+
+    req.fmt = aud_to_ossfmt (fmt);
+    req.freq = freq;
+    req.nchannels = nchannels;
+    req.fragsize = conf.fragsize;
+    req.nfrags = conf.nfrags;
+    if (oss_open (1, &req, &obt, &fd)) {
+        return -1;
+    }
+
+    err = oss_to_audfmt (obt.fmt, &effective_fmt, &endianness);
+    if (err) {
+        oss_anal_close (&fd);
+        return -1;
+    }
+
+    audio_pcm_init_info (
+        &hw->info,
+        obt.freq,
+        obt.nchannels,
+        effective_fmt,
+        audio_need_to_swap_endian (endianness)
+        );
+    oss->nfrags = obt.nfrags;
+    oss->fragsize = obt.fragsize;
+    hw->bufsize = obt.nfrags * obt.fragsize;
+    oss->pcm_buf = qemu_mallocz (hw->bufsize);
+    if (!oss->pcm_buf) {
+        oss_anal_close (&fd);
+        return -1;
+    }
+
+    oss->fd = fd;
+    return 0;
+}
+
+static void oss_fini_in (HWVoiceIn *hw)
+{
+    OSSVoiceIn *oss = (OSSVoiceIn *) hw;
+
+    oss_anal_close (&oss->fd);
+
+    if (oss->pcm_buf) {
+        qemu_free (oss->pcm_buf);
+        oss->pcm_buf = NULL;
+    }
+}
+
+static int oss_run_in (HWVoiceIn *hw)
+{
+    OSSVoiceIn *oss = (OSSVoiceIn *) hw;
+    int hwshift = hw->info.shift;
+    int i;
+    int live = audio_pcm_hw_get_live_in (hw);
+    int dead = hw->samples - live;
+    size_t read_samples = 0;
+    struct {
+        int add;
+        int len;
+    } bufs[2] = {
+        { hw->wpos, 0 },
+        { 0, 0 }
+    };
+
+    if (!dead) {
+        return 0;
+    }
+
+    if (hw->wpos + dead > hw->samples) {
+        bufs[0].len = (hw->samples - hw->wpos) << hwshift;
+        bufs[1].len = (dead - (hw->samples - hw->wpos)) << hwshift;
+    }
+    else {
+        bufs[0].len = dead << hwshift;
+    }
+
+
+    for (i = 0; i < 2; ++i) {
+        ssize_t nread;
+
+        if (bufs[i].len) {
+            void *p = advance (oss->pcm_buf, bufs[i].add << hwshift);
+            nread = read (oss->fd, p, bufs[i].len);
+
+            if (nread > 0) {
+                if (nread & hw->info.align) {
+                    dolog ("warning: misaligned read %d (requested %d), "
+                           "alignment %d\n", nread, bufs[i].add << hwshift,
+                           hw->info.align + 1);
+                }
+                read_samples += nread >> hwshift;
+                hw->conv (hw->conv_buf + bufs[i].add, p, nread >> hwshift,
+                          &nominal_volume);
+            }
+
+            if (bufs[i].len - nread) {
+                if (nread == -1) {
+                    switch (errno) {
+                    case EINTR:
+                    case EAGAIN:
+                        break;
+                    default:
+                        oss_logerr (
+                            errno,
+                            "Failed to read %d bytes of audio (to %p)\n",
+                            bufs[i].len, p
+                            );
+                        break;
+                    }
+                }
+                break;
+            }
+        }
+    }
+
+    hw->wpos = (hw->wpos + read_samples) % hw->samples;
+    return read_samples;
+}
+
+static int oss_read (SWVoiceIn *sw, void *buf, int size)
+{
+    return audio_pcm_sw_read (sw, buf, size);
+}
+
+static int oss_ctl_in (HWVoiceIn *hw, int cmd, ...)
+{
+    (void) hw;
+    (void) cmd;
+    return 0;
+}
+
 static void *oss_audio_init (void)
 {
-    conf.fragsize = audio_get_conf_int (QC_OSS_FRAGSIZE, conf.fragsize);
-    conf.nfrags = audio_get_conf_int (QC_OSS_NFRAGS, conf.nfrags);
-    conf.try_mmap = audio_get_conf_int (QC_OSS_MMAP, conf.try_mmap);
-    conf.dspname = audio_get_conf_str (QC_OSS_DEV, conf.dspname);
     return &conf;
 }
 
 static void oss_audio_fini (void *opaque)
 {
+    (void) opaque;
 }
 
-struct pcm_ops oss_pcm_ops = {
+static struct audio_option oss_options[] = {
-    oss_hw_init,
+    {"FRAGSIZE", AUD_OPT_INT, &conf.fragsize,
-    oss_hw_fini,
+     "Fragment size in bytes", NULL, 0},
-    oss_hw_run,
+    {"NFRAGS", AUD_OPT_INT, &conf.nfrags,
-    oss_hw_write,
+     "Number of fragments", NULL, 0},
-    oss_hw_ctl
+    {"MMAP", AUD_OPT_BOOL, &conf.try_mmap,
+     "Try using memory mapped access", NULL, 0},
+    {"DAC_DEV", AUD_OPT_STR, &conf.devpath_out,
+     "Path to DAC device", NULL, 0},
+    {"ADC_DEV", AUD_OPT_STR, &conf.devpath_in,
+     "Path to ADC device", NULL, 0},
+    {NULL, 0, NULL, NULL, NULL, 0}
 };
 
-struct audio_output_driver oss_output_driver = {
+static struct audio_pcm_ops oss_pcm_ops = {
-    "oss",
+    oss_init_out,
-    oss_audio_init,
+    oss_fini_out,
-    oss_audio_fini,
+    oss_run_out,
-    &oss_pcm_ops,
+    oss_write,
-    1,
+    oss_ctl_out,
-    INT_MAX,
+
-    sizeof (OSSVoice)
+    oss_init_in,
+    oss_fini_in,
+    oss_run_in,
+    oss_read,
+    oss_ctl_in
+};
+
+struct audio_driver oss_audio_driver = {
+    INIT_FIELD (name           = ) "oss",
+    INIT_FIELD (descr          = ) "OSS http://www.opensound.com",
+    INIT_FIELD (options        = ) oss_options,
+    INIT_FIELD (init           = ) oss_audio_init,
+    INIT_FIELD (fini           = ) oss_audio_fini,
+    INIT_FIELD (pcm_ops        = ) &oss_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 (OSSVoiceOut),
+    INIT_FIELD (voice_size_in  = ) sizeof (OSSVoiceIn)
 };

audio/rate_template.h

@@ -0,0 +1,111 @@
+/*
+ * QEMU Mixing engine
+ *
+ * Copyright (c) 2004-2005 Vassili Karpov (malc)
+ * Copyright (c) 1998 Fabrice Bellard
+ *
+ * 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.
+ */
+
+/*
+ * Processed signed long samples from ibuf to obuf.
+ * Return number of samples processed.
+ */
+void NAME (void *opaque, st_sample_t *ibuf, st_sample_t *obuf,
+           int *isamp, int *osamp)
+{
+    rate_t rate = (rate_t) opaque;
+    st_sample_t *istart, *iend;
+    st_sample_t *ostart, *oend;
+    st_sample_t ilast, icur, out;
+#ifdef FLOAT_MIXENG
+    real_t t;
+#else
+    int64_t t;
+#endif
+
+    ilast = rate->ilast;
+
+    istart = ibuf;
+    iend = ibuf + *isamp;
+
+    ostart = obuf;
+    oend = obuf + *osamp;
+
+    if (rate->opos_inc == (1ULL + UINT_MAX)) {
+        int i, n = *isamp > *osamp ? *osamp : *isamp;
+        for (i = 0; i < n; i++) {
+            OP (obuf[i].l, ibuf[i].r);
+            OP (obuf[i].r, ibuf[i].r);
+        }
+        *isamp = n;
+        *osamp = n;
+        return;
+    }
+
+    while (obuf < oend) {
+
+        /* Safety catch to make sure we have input samples.  */
+        if (ibuf >= iend) {
+            break;
+        }
+
+        /* read as many input samples so that ipos > opos */
+
+        while (rate->ipos <= (rate->opos >> 32)) {
+            ilast = *ibuf++;
+            rate->ipos++;
+            /* See if we finished the input buffer yet */
+            if (ibuf >= iend) {
+                goto the_end;
+            }
+        }
+
+        icur = *ibuf;
+
+        /* interpolate */
+#ifdef FLOAT_MIXENG
+#ifdef RECIPROCAL
+        t = (rate->opos & UINT_MAX) * (1.f / UINT_MAX);
+#else
+        t = (rate->opos & UINT_MAX) / (real_t) UINT_MAX;
+#endif
+        out.l = (ilast.l * (1.0 - t)) + icur.l * t;
+        out.r = (ilast.r * (1.0 - t)) + icur.r * t;
+#else
+        t = rate->opos & 0xffffffff;
+        out.l = (ilast.l * ((int64_t) UINT_MAX - t) + icur.l * t) >> 32;
+        out.r = (ilast.r * ((int64_t) UINT_MAX - t) + icur.r * t) >> 32;
+#endif
+
+        /* output sample & increment position */
+        OP (obuf->l, out.l);
+        OP (obuf->r, out.r);
+        obuf += 1;
+        rate->opos += rate->opos_inc;
+    }
+
+the_end:
+    *isamp = ibuf - istart;
+    *osamp = obuf - ostart;
+    rate->ilast = ilast;
+}
+
+#undef NAME
+#undef OP

audio/sdlaudio.c

@@ -1,7 +1,7 @@
 /*
- * QEMU SDL audio output driver
+ * QEMU SDL audio driver
  *
- * Copyright (c) 2004 Vassili Karpov (malc)
+ * Copyright (c) 2004-2005 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
@@ -25,22 +25,15 @@
 #include <SDL_thread.h>
 #include "vl.h"
 
-#include "audio/audio_int.h"
+#define AUDIO_CAP "sdl"
+#include "audio_int.h"
 
-typedef struct SDLVoice {
+typedef struct SDLVoiceOut {
-    HWVoice hw;
+    HWVoiceOut hw;
-} SDLVoice;
+    int live;
-
+    int rpos;
-#define dolog(...) AUD_log ("sdl", __VA_ARGS__)
+    int decr;
-#ifdef DEBUG
+} SDLVoiceOut;
-#define ldebug(...) dolog (__VA_ARGS__)
-#else
-#define ldebug(...)
-#endif
-
-#define QC_SDL_SAMPLES "QEMU_SDL_SAMPLES"
-
-#define errstr() SDL_GetError ()
 
 static struct {
     int nb_samples;
@@ -56,91 +49,129 @@ struct SDLAudioState {
 } glob_sdl;
 typedef struct SDLAudioState SDLAudioState;
 
-static void sdl_hw_run (HWVoice *hw)
+static void GCC_FMT_ATTR (1, 2) sdl_logerr (const char *fmt, ...)
 {
-    (void) hw;
+    va_list ap;
+
+    va_start (ap, fmt);
+    AUD_vlog (AUDIO_CAP, fmt, ap);
+    va_end (ap);
+
+    AUD_log (AUDIO_CAP, "Reason: %s\n", SDL_GetError ());
 }
 
-static int sdl_lock (SDLAudioState *s)
+static int sdl_lock (SDLAudioState *s, const char *forfn)
 {
     if (SDL_LockMutex (s->mutex)) {
-        dolog ("SDL_LockMutex failed\nReason: %s\n", errstr ());
+        sdl_logerr ("SDL_LockMutex for %s failed\n", forfn);
         return -1;
     }
     return 0;
 }
 
-static int sdl_unlock (SDLAudioState *s)
+static int sdl_unlock (SDLAudioState *s, const char *forfn)
 {
     if (SDL_UnlockMutex (s->mutex)) {
-        dolog ("SDL_UnlockMutex failed\nReason: %s\n", errstr ());
+        sdl_logerr ("SDL_UnlockMutex for %s failed\n", forfn);
         return -1;
     }
     return 0;
 }
 
-static int sdl_post (SDLAudioState *s)
+static int sdl_post (SDLAudioState *s, const char *forfn)
 {
     if (SDL_SemPost (s->sem)) {
-        dolog ("SDL_SemPost failed\nReason: %s\n", errstr ());
+        sdl_logerr ("SDL_SemPost for %s failed\n", forfn);
         return -1;
     }
     return 0;
 }
 
-static int sdl_wait (SDLAudioState *s)
+static int sdl_wait (SDLAudioState *s, const char *forfn)
 {
     if (SDL_SemWait (s->sem)) {
-        dolog ("SDL_SemWait failed\nReason: %s\n", errstr ());
+        sdl_logerr ("SDL_SemWait for %s failed\n", forfn);
         return -1;
     }
     return 0;
 }
 
-static int sdl_unlock_and_post (SDLAudioState *s)
+static int sdl_unlock_and_post (SDLAudioState *s, const char *forfn)
 {
-    if (sdl_unlock (s))
+    if (sdl_unlock (s, forfn)) {
         return -1;
+    }
 
-    return sdl_post (s);
+    return sdl_post (s, forfn);
 }
 
-static int sdl_hw_write (SWVoice *sw, void *buf, int len)
+static int aud_to_sdlfmt (audfmt_e fmt, int *shift)
-{
-    int ret;
-    SDLAudioState *s = &glob_sdl;
-    sdl_lock (s);
-    ret = pcm_hw_write (sw, buf, len);
-    sdl_unlock_and_post (s);
-    return ret;
-}
-
-static int AUD_to_sdlfmt (audfmt_e fmt, int *shift)
 {
+    switch (fmt) {
+    case AUD_FMT_S8:
         *shift = 0;
-    switch (fmt) {
+        return AUDIO_S8;
-    case AUD_FMT_S8: return AUDIO_S8;
+
-    case AUD_FMT_U8: return AUDIO_U8;
+    case AUD_FMT_U8:
-    case AUD_FMT_S16: *shift = 1; return AUDIO_S16LSB;
+        *shift = 0;
-    case AUD_FMT_U16: *shift = 1; return AUDIO_U16LSB;
+        return AUDIO_U8;
+
+    case AUD_FMT_S16:
+        *shift = 1;
+        return AUDIO_S16LSB;
+
+    case AUD_FMT_U16:
+        *shift = 1;
+        return AUDIO_U16LSB;
+
     default:
-        dolog ("Internal logic error: Bad audio format %d\nAborting\n", fmt);
+        dolog ("Internal logic error: Bad audio format %d\n", fmt);
-        exit (EXIT_FAILURE);
+#ifdef DEBUG_AUDIO
+        abort ();
+#endif
+        return AUDIO_U8;
     }
 }
 
-static int sdl_to_audfmt (int fmt)
+static int sdl_to_audfmt (int sdlfmt, audfmt_e *fmt, int *endianess)
 {
-    switch (fmt) {
+    switch (sdlfmt) {
-    case AUDIO_S8: return AUD_FMT_S8;
+    case AUDIO_S8:
-    case AUDIO_U8: return AUD_FMT_U8;
+        *endianess = 0;
-    case AUDIO_S16LSB: return AUD_FMT_S16;
+        *fmt = AUD_FMT_S8;
-    case AUDIO_U16LSB: return AUD_FMT_U16;
+        break;
+
+    case AUDIO_U8:
+        *endianess = 0;
+        *fmt = AUD_FMT_U8;
+        break;
+
+    case AUDIO_S16LSB:
+        *endianess = 0;
+        *fmt = AUD_FMT_S16;
+        break;
+
+    case AUDIO_U16LSB:
+        *endianess = 0;
+        *fmt = AUD_FMT_U16;
+        break;
+
+    case AUDIO_S16MSB:
+        *endianess = 1;
+        *fmt = AUD_FMT_S16;
+        break;
+
+    case AUDIO_U16MSB:
+        *endianess = 1;
+        *fmt = AUD_FMT_U16;
+        break;
+
     default:
-        dolog ("Internal logic error: Unrecognized SDL audio format %d\n"
+        dolog ("Unrecognized SDL audio format %d\n", sdlfmt);
-               "Aborting\n", fmt);
+        return -1;
-        exit (EXIT_FAILURE);
     }
+
+    return 0;
 }
 
 static int sdl_open (SDL_AudioSpec *req, SDL_AudioSpec *obt)
@@ -149,7 +180,7 @@ static int sdl_open (SDL_AudioSpec *req, SDL_AudioSpec *obt)
 
     status = SDL_OpenAudio (req, obt);
     if (status) {
-        dolog ("SDL_OpenAudio failed\nReason: %s\n", errstr ());
+        sdl_logerr ("SDL_OpenAudio failed\n");
     }
     return status;
 }
@@ -157,9 +188,9 @@ static int sdl_open (SDL_AudioSpec *req, SDL_AudioSpec *obt)
 static void sdl_close (SDLAudioState *s)
 {
     if (s->initialized) {
-        sdl_lock (s);
+        sdl_lock (s, "sdl_close");
         s->exit = 1;
-        sdl_unlock_and_post (s);
+        sdl_unlock_and_post (s, "sdl_close");
         SDL_PauseAudio (1);
         SDL_CloseAudio ();
         s->initialized = 0;
@@ -168,31 +199,40 @@ static void sdl_close (SDLAudioState *s)
 
 static void sdl_callback (void *opaque, Uint8 *buf, int len)
 {
-    SDLVoice *sdl = opaque;
+    SDLVoiceOut *sdl = opaque;
     SDLAudioState *s = &glob_sdl;
-    HWVoice *hw = &sdl->hw;
+    HWVoiceOut *hw = &sdl->hw;
-    int samples = len >> hw->shift;
+    int samples = len >> hw->info.shift;
 
     if (s->exit) {
         return;
     }
 
     while (samples) {
-        int to_mix, live, decr;
+        int to_mix, decr;
 
         /* dolog ("in callback samples=%d\n", samples); */
-        sdl_wait (s);
+        sdl_wait (s, "sdl_callback");
         if (s->exit) {
             return;
         }
 
-        sdl_lock (s);
+        if (sdl_lock (s, "sdl_callback")) {
-        live = pcm_hw_get_live (hw);
+            return;
-        if (live <= 0)
+        }
+
+        if (audio_bug (AUDIO_FUNC, sdl->live < 0 || sdl->live > hw->samples)) {
+            dolog ("sdl->live=%d hw->samples=%d\n",
+                   sdl->live, hw->samples);
+            return;
+        }
+
+        if (!sdl->live) {
             goto again;
+        }
 
         /* dolog ("in callback live=%d\n", live); */
-        to_mix = audio_MIN (samples, live);
+        to_mix = audio_MIN (samples, sdl->live);
         decr = to_mix;
         while (to_mix) {
             int chunk = audio_MIN (to_mix, hw->samples - hw->rpos);
@@ -200,44 +240,86 @@ static void sdl_callback (void *opaque, Uint8 *buf, int len)
 
             /* dolog ("in callback to_mix %d, chunk %d\n", to_mix, chunk); */
             hw->clip (buf, src, chunk);
-            memset (src, 0, chunk * sizeof (st_sample_t));
+            mixeng_clear (src, chunk);
-            hw->rpos = (hw->rpos + chunk) % hw->samples;
+            sdl->rpos = (sdl->rpos + chunk) % hw->samples;
             to_mix -= chunk;
-            buf += chunk << hw->shift;
+            buf += chunk << hw->info.shift;
         }
         samples -= decr;
-        pcm_hw_dec_live (hw, decr);
+        sdl->live -= decr;
+        sdl->decr += decr;
 
     again:
-        sdl_unlock (s);
+        if (sdl_unlock (s, "sdl_callback")) {
+            return;
+        }
     }
     /* dolog ("done len=%d\n", len); */
 }
 
-static void sdl_hw_fini (HWVoice *hw)
+static int sdl_write_out (SWVoiceOut *sw, void *buf, int len)
 {
-    ldebug ("sdl_hw_fini %d fixed=%d\n",
+    return audio_pcm_sw_write (sw, buf, len);
-             glob_sdl.initialized, audio_state.fixed_format);
+}
+
+static int sdl_run_out (HWVoiceOut *hw)
+{
+    int decr, live;
+    SDLVoiceOut *sdl = (SDLVoiceOut *) hw;
+    SDLAudioState *s = &glob_sdl;
+
+    if (sdl_lock (s, "sdl_callback")) {
+        return 0;
+    }
+
+    live = audio_pcm_hw_get_live_out (hw);
+
+    if (sdl->decr > live) {
+        ldebug ("sdl->decr %d live %d sdl->live %d\n",
+                sdl->decr,
+                live,
+                sdl->live);
+    }
+
+    decr = audio_MIN (sdl->decr, live);
+    sdl->decr -= decr;
+
+    sdl->live = live - decr;
+    hw->rpos = sdl->rpos;
+
+    if (sdl->live > 0) {
+        sdl_unlock_and_post (s, "sdl_callback");
+    }
+    else {
+        sdl_unlock (s, "sdl_callback");
+    }
+    return decr;
+}
+
+static void sdl_fini_out (HWVoiceOut *hw)
+{
+    (void) hw;
+
     sdl_close (&glob_sdl);
 }
 
-static int sdl_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt)
+static int sdl_init_out (HWVoiceOut *hw, int freq, int nchannels, audfmt_e fmt)
 {
-    SDLVoice *sdl = (SDLVoice *) hw;
+    SDLVoiceOut *sdl = (SDLVoiceOut *) hw;
     SDLAudioState *s = &glob_sdl;
     SDL_AudioSpec req, obt;
     int shift;
-
+    int endianess;
-    ldebug ("sdl_hw_init %d freq=%d fixed=%d\n",
+    int err;
-            s->initialized, freq, audio_state.fixed_format);
+    audfmt_e effective_fmt;
 
     if (nchannels != 2) {
-        dolog ("Bogus channel count %d\n", nchannels);
+        dolog ("Can not init DAC. Bogus channel count %d\n", nchannels);
         return -1;
     }
 
     req.freq = freq;
-    req.format = AUD_to_sdlfmt (fmt, &shift);
+    req.format = aud_to_sdlfmt (fmt, &shift);
     req.channels = nchannels;
     req.samples = conf.nb_samples;
     shift <<= nchannels == 2;
@@ -245,12 +327,23 @@ static int sdl_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt)
     req.callback = sdl_callback;
     req.userdata = sdl;
 
-    if (sdl_open (&req, &obt))
+    if (sdl_open (&req, &obt)) {
         return -1;
+    }
 
-    hw->freq = obt.freq;
+    err = sdl_to_audfmt (obt.format, &effective_fmt, &endianess);
-    hw->fmt = sdl_to_audfmt (obt.format);
+    if (err) {
-    hw->nchannels = obt.channels;
+        sdl_close (s);
+        return -1;
+    }
+
+    audio_pcm_init_info (
+        &hw->info,
+        obt.freq,
+        obt.channels,
+        effective_fmt,
+        audio_need_to_swap_endian (endianess)
+        );
     hw->bufsize = obt.samples << shift;
 
     s->initialized = 1;
@@ -259,7 +352,7 @@ static int sdl_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt)
     return 0;
 }
 
-static int sdl_hw_ctl (HWVoice *hw, int cmd, ...)
+static int sdl_ctl_out (HWVoiceOut *hw, int cmd, ...)
 {
     (void) hw;
 
@@ -278,24 +371,22 @@ static int sdl_hw_ctl (HWVoice *hw, int cmd, ...)
 static void *sdl_audio_init (void)
 {
     SDLAudioState *s = &glob_sdl;
-    conf.nb_samples = audio_get_conf_int (QC_SDL_SAMPLES, conf.nb_samples);
 
     if (SDL_InitSubSystem (SDL_INIT_AUDIO)) {
-        dolog ("SDL failed to initialize audio subsystem\nReason: %s\n",
+        sdl_logerr ("SDL failed to initialize audio subsystem\n");
-               errstr ());
         return NULL;
     }
 
     s->mutex = SDL_CreateMutex ();
     if (!s->mutex) {
-        dolog ("Failed to create SDL mutex\nReason: %s\n", errstr ());
+        sdl_logerr ("Failed to create SDL mutex\n");
         SDL_QuitSubSystem (SDL_INIT_AUDIO);
         return NULL;
     }
 
     s->sem = SDL_CreateSemaphore (0);
     if (!s->sem) {
-        dolog ("Failed to create SDL semaphore\nReason: %s\n", errstr ());
+        sdl_logerr ("Failed to create SDL semaphore\n");
         SDL_DestroyMutex (s->mutex);
         SDL_QuitSubSystem (SDL_INIT_AUDIO);
         return NULL;
@@ -313,20 +404,36 @@ static void sdl_audio_fini (void *opaque)
     SDL_QuitSubSystem (SDL_INIT_AUDIO);
 }
 
-struct pcm_ops sdl_pcm_ops = {
+static struct audio_option sdl_options[] = {
-    sdl_hw_init,
+    {"SAMPLES", AUD_OPT_INT, &conf.nb_samples,
-    sdl_hw_fini,
+     "Size of SDL buffer in samples", NULL, 0},
-    sdl_hw_run,
+    {NULL, 0, NULL, NULL, NULL, 0}
-    sdl_hw_write,
-    sdl_hw_ctl
 };
 
-struct audio_output_driver sdl_output_driver = {
+static struct audio_pcm_ops sdl_pcm_ops = {
-    "sdl",
+    sdl_init_out,
-    sdl_audio_init,
+    sdl_fini_out,
-    sdl_audio_fini,
+    sdl_run_out,
-    &sdl_pcm_ops,
+    sdl_write_out,
-    1,
+    sdl_ctl_out,
-    1,
+
-    sizeof (SDLVoice)
+    NULL,
+    NULL,
+    NULL,
+    NULL,
+    NULL
+};
+
+struct audio_driver sdl_audio_driver = {
+    INIT_FIELD (name           = ) "sdl",
+    INIT_FIELD (descr          = ) "SDL http://www.libsdl.org",
+    INIT_FIELD (options        = ) sdl_options,
+    INIT_FIELD (init           = ) sdl_audio_init,
+    INIT_FIELD (fini           = ) sdl_audio_fini,
+    INIT_FIELD (pcm_ops        = ) &sdl_pcm_ops,
+    INIT_FIELD (can_be_default = ) 1,
+    INIT_FIELD (max_voices_out = ) 1,
+    INIT_FIELD (max_voices_in  = ) 0,
+    INIT_FIELD (voice_size_out = ) sizeof (SDLVoiceOut),
+    INIT_FIELD (voice_size_in  = ) 0
 };

audio/sys-queue.h

@@ -0,0 +1,241 @@
+/*
+ * Copyright (c) 1991, 1993
+ *	The Regents of the University of California.  All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ * 1. Redistributions of source code must retain the above copyright
+ *    notice, this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright
+ *    notice, this list of conditions and the following disclaimer in the
+ *    documentation and/or other materials provided with the distribution.
+ * 4. Neither the name of the University nor the names of its contributors
+ *    may be used to endorse or promote products derived from this software
+ *    without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE REGENTS AND CONTRIBUTORS ``AS IS'' AND
+ * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+ * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ * ARE DISCLAIMED.  IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE
+ * FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+ * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
+ * OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+ * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+ * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+ * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
+ * SUCH DAMAGE.
+ *
+ *	@(#)queue.h	8.3 (Berkeley) 12/13/93
+ */
+
+#ifndef	_SYS_QUEUE_H
+#define	_SYS_QUEUE_H 1
+
+/*
+ * This file defines three types of data structures: lists, tail queues,
+ * and circular queues.
+ *
+ * A list is headed by a single forward pointer (or an array of forward
+ * pointers for a hash table header). The elements are doubly linked
+ * so that an arbitrary element can be removed without a need to
+ * traverse the list. New elements can be added to the list after
+ * an existing element or at the head of the list. A list may only be
+ * traversed in the forward direction.
+ *
+ * A tail queue is headed by a pair of pointers, one to the head of the
+ * list and the other to the tail of the list. The elements are doubly
+ * linked so that an arbitrary element can be removed without a need to
+ * traverse the list. New elements can be added to the list after
+ * an existing element, at the head of the list, or at the end of the
+ * list. A tail queue may only be traversed in the forward direction.
+ *
+ * A circle queue is headed by a pair of pointers, one to the head of the
+ * list and the other to the tail of the list. The elements are doubly
+ * linked so that an arbitrary element can be removed without a need to
+ * traverse the list. New elements can be added to the list before or after
+ * an existing element, at the head of the list, or at the end of the list.
+ * A circle queue may be traversed in either direction, but has a more
+ * complex end of list detection.
+ *
+ * For details on the use of these macros, see the queue(3) manual page.
+ */
+
+/*
+ * List definitions.
+ */
+#define LIST_HEAD(name, type)						\
+struct name {								\
+	struct type *lh_first;	/* first element */			\
+}
+
+#define LIST_ENTRY(type)						\
+struct {								\
+	struct type *le_next;	/* next element */			\
+	struct type **le_prev;	/* address of previous next element */	\
+}
+
+/*
+ * List functions.
+ */
+#define	LIST_INIT(head) {						\
+	(head)->lh_first = NULL;					\
+}
+
+#define LIST_INSERT_AFTER(listelm, elm, field) {			\
+	if (((elm)->field.le_next = (listelm)->field.le_next) != NULL)	\
+		(listelm)->field.le_next->field.le_prev =		\
+		    &(elm)->field.le_next;				\
+	(listelm)->field.le_next = (elm);				\
+	(elm)->field.le_prev = &(listelm)->field.le_next;		\
+}
+
+#define LIST_INSERT_HEAD(head, elm, field) {				\
+	if (((elm)->field.le_next = (head)->lh_first) != NULL)		\
+		(head)->lh_first->field.le_prev = &(elm)->field.le_next;\
+	(head)->lh_first = (elm);					\
+	(elm)->field.le_prev = &(head)->lh_first;			\
+}
+
+#define LIST_REMOVE(elm, field) {					\
+	if ((elm)->field.le_next != NULL)				\
+		(elm)->field.le_next->field.le_prev = 			\
+		    (elm)->field.le_prev;				\
+	*(elm)->field.le_prev = (elm)->field.le_next;			\
+}
+
+/*
+ * Tail queue definitions.
+ */
+#define TAILQ_HEAD(name, type)						\
+struct name {								\
+	struct type *tqh_first;	/* first element */			\
+	struct type **tqh_last;	/* addr of last next element */		\
+}
+
+#define TAILQ_ENTRY(type)						\
+struct {								\
+	struct type *tqe_next;	/* next element */			\
+	struct type **tqe_prev;	/* address of previous next element */	\
+}
+
+/*
+ * Tail queue functions.
+ */
+#define	TAILQ_INIT(head) {						\
+	(head)->tqh_first = NULL;					\
+	(head)->tqh_last = &(head)->tqh_first;				\
+}
+
+#define TAILQ_INSERT_HEAD(head, elm, field) {				\
+	if (((elm)->field.tqe_next = (head)->tqh_first) != NULL)	\
+		(elm)->field.tqe_next->field.tqe_prev =			\
+		    &(elm)->field.tqe_next;				\
+	else								\
+		(head)->tqh_last = &(elm)->field.tqe_next;		\
+	(head)->tqh_first = (elm);					\
+	(elm)->field.tqe_prev = &(head)->tqh_first;			\
+}
+
+#define TAILQ_INSERT_TAIL(head, elm, field) {				\
+	(elm)->field.tqe_next = NULL;					\
+	(elm)->field.tqe_prev = (head)->tqh_last;			\
+	*(head)->tqh_last = (elm);					\
+	(head)->tqh_last = &(elm)->field.tqe_next;			\
+}
+
+#define TAILQ_INSERT_AFTER(head, listelm, elm, field) {			\
+	if (((elm)->field.tqe_next = (listelm)->field.tqe_next) != NULL)\
+		(elm)->field.tqe_next->field.tqe_prev = 		\
+		    &(elm)->field.tqe_next;				\
+	else								\
+		(head)->tqh_last = &(elm)->field.tqe_next;		\
+	(listelm)->field.tqe_next = (elm);				\
+	(elm)->field.tqe_prev = &(listelm)->field.tqe_next;		\
+}
+
+#define TAILQ_REMOVE(head, elm, field) {				\
+	if (((elm)->field.tqe_next) != NULL)				\
+		(elm)->field.tqe_next->field.tqe_prev = 		\
+		    (elm)->field.tqe_prev;				\
+	else								\
+		(head)->tqh_last = (elm)->field.tqe_prev;		\
+	*(elm)->field.tqe_prev = (elm)->field.tqe_next;			\
+}
+
+/*
+ * Circular queue definitions.
+ */
+#define CIRCLEQ_HEAD(name, type)					\
+struct name {								\
+	struct type *cqh_first;		/* first element */		\
+	struct type *cqh_last;		/* last element */		\
+}
+
+#define CIRCLEQ_ENTRY(type)						\
+struct {								\
+	struct type *cqe_next;		/* next element */		\
+	struct type *cqe_prev;		/* previous element */		\
+}
+
+/*
+ * Circular queue functions.
+ */
+#define	CIRCLEQ_INIT(head) {						\
+	(head)->cqh_first = (void *)(head);				\
+	(head)->cqh_last = (void *)(head);				\
+}
+
+#define CIRCLEQ_INSERT_AFTER(head, listelm, elm, field) {		\
+	(elm)->field.cqe_next = (listelm)->field.cqe_next;		\
+	(elm)->field.cqe_prev = (listelm);				\
+	if ((listelm)->field.cqe_next == (void *)(head))		\
+		(head)->cqh_last = (elm);				\
+	else								\
+		(listelm)->field.cqe_next->field.cqe_prev = (elm);	\
+	(listelm)->field.cqe_next = (elm);				\
+}
+
+#define CIRCLEQ_INSERT_BEFORE(head, listelm, elm, field) {		\
+	(elm)->field.cqe_next = (listelm);				\
+	(elm)->field.cqe_prev = (listelm)->field.cqe_prev;		\
+	if ((listelm)->field.cqe_prev == (void *)(head))		\
+		(head)->cqh_first = (elm);				\
+	else								\
+		(listelm)->field.cqe_prev->field.cqe_next = (elm);	\
+	(listelm)->field.cqe_prev = (elm);				\
+}
+
+#define CIRCLEQ_INSERT_HEAD(head, elm, field) {				\
+	(elm)->field.cqe_next = (head)->cqh_first;			\
+	(elm)->field.cqe_prev = (void *)(head);				\
+	if ((head)->cqh_last == (void *)(head))				\
+		(head)->cqh_last = (elm);				\
+	else								\
+		(head)->cqh_first->field.cqe_prev = (elm);		\
+	(head)->cqh_first = (elm);					\
+}
+
+#define CIRCLEQ_INSERT_TAIL(head, elm, field) {				\
+	(elm)->field.cqe_next = (void *)(head);				\
+	(elm)->field.cqe_prev = (head)->cqh_last;			\
+	if ((head)->cqh_first == (void *)(head))			\
+		(head)->cqh_first = (elm);				\
+	else								\
+		(head)->cqh_last->field.cqe_next = (elm);		\
+	(head)->cqh_last = (elm);					\
+}
+
+#define	CIRCLEQ_REMOVE(head, elm, field) {				\
+	if ((elm)->field.cqe_next == (void *)(head))			\
+		(head)->cqh_last = (elm)->field.cqe_prev;		\
+	else								\
+		(elm)->field.cqe_next->field.cqe_prev =			\
+		    (elm)->field.cqe_prev;				\
+	if ((elm)->field.cqe_prev == (void *)(head))			\
+		(head)->cqh_first = (elm)->field.cqe_next;		\
+	else								\
+		(elm)->field.cqe_prev->field.cqe_next =			\
+		    (elm)->field.cqe_next;				\
+}
+#endif	/* sys/queue.h */

audio/wavaudio.c

@@ -1,7 +1,7 @@
 /*
- * QEMU WAV audio output driver
+ * QEMU WAV audio driver
  *
- * Copyright (c) 2004 Vassili Karpov (malc)
+ * Copyright (c) 2004-2005 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
@@ -23,22 +23,16 @@
  */
 #include "vl.h"
 
-#include "audio/audio_int.h"
+#define AUDIO_CAP "wav"
+#include "audio_int.h"
 
-typedef struct WAVVoice {
+typedef struct WAVVoiceOut {
-    HWVoice hw;
+    HWVoiceOut hw;
     QEMUFile *f;
     int64_t old_ticks;
     void *pcm_buf;
     int total_samples;
-} WAVVoice;
+} WAVVoiceOut;
-
-#define dolog(...) AUD_log ("wav", __VA_ARGS__)
-#ifdef DEBUG
-#define ldebug(...) dolog (__VA_ARGS__)
-#else
-#define ldebug(...)
-#endif
 
 static struct {
     const char *wav_path;
@@ -46,24 +40,27 @@ static struct {
     .wav_path = "qemu.wav"
 };
 
-static void wav_hw_run (HWVoice *hw)
+static int wav_run_out (HWVoiceOut *hw)
 {
-    WAVVoice *wav = (WAVVoice *) hw;
+    WAVVoiceOut *wav = (WAVVoiceOut *) hw;
     int rpos, live, decr, samples;
     uint8_t *dst;
     st_sample_t *src;
     int64_t now = qemu_get_clock (vm_clock);
     int64_t ticks = now - wav->old_ticks;
-    int64_t bytes = (ticks * hw->bytes_per_second) / ticks_per_sec;
+    int64_t bytes = (ticks * hw->info.bytes_per_second) / ticks_per_sec;
 
-    if (bytes > INT_MAX)
+    if (bytes > INT_MAX) {
-        samples = INT_MAX >> hw->shift;
+        samples = INT_MAX >> hw->info.shift;
-    else
+    }
-        samples = bytes >> hw->shift;
+    else {
+        samples = bytes >> hw->info.shift;
+    }
 
-    live = pcm_hw_get_live (hw);
+    live = audio_pcm_hw_get_live_out (hw);
-    if (live <= 0)
+    if (!live) {
-        return;
+        return 0;
+    }
 
     wav->old_ticks = now;
     decr = audio_MIN (live, samples);
@@ -73,25 +70,25 @@ static void wav_hw_run (HWVoice *hw)
         int left_till_end_samples = hw->samples - rpos;
         int convert_samples = audio_MIN (samples, left_till_end_samples);
 
-        src = advance (hw->mix_buf, rpos * sizeof (st_sample_t));
+        src = hw->mix_buf + rpos;
-        dst = advance (wav->pcm_buf, rpos << hw->shift);
+        dst = advance (wav->pcm_buf, rpos << hw->info.shift);
 
         hw->clip (dst, src, convert_samples);
-        qemu_put_buffer (wav->f, dst, convert_samples << hw->shift);
+        qemu_put_buffer (wav->f, dst, convert_samples << hw->info.shift);
-        memset (src, 0, convert_samples * sizeof (st_sample_t));
+        mixeng_clear (src, convert_samples);
 
         rpos = (rpos + convert_samples) % hw->samples;
         samples -= convert_samples;
         wav->total_samples += convert_samples;
     }
 
-    pcm_hw_dec_live (hw, decr);
     hw->rpos = rpos;
+    return decr;
 }
 
-static int wav_hw_write (SWVoice *sw, void *buf, int len)
+static int wav_write_out (SWVoiceOut *sw, void *buf, int len)
 {
-    return pcm_hw_write (sw, buf, len);
+    return audio_pcm_sw_write (sw, buf, len);
 }
 
 /* VICE code: Store number as little endian. */
@@ -104,10 +101,10 @@ static void le_store (uint8_t *buf, uint32_t val, int len)
     }
 }
 
-static int wav_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt)
+static int wav_init_out (HWVoiceOut *hw, int freq, int nchannels, audfmt_e fmt)
 {
-    WAVVoice *wav = (WAVVoice *) hw;
+    WAVVoiceOut *wav = (WAVVoiceOut *) hw;
-    int bits16 = 0, stereo = audio_state.fixed_channels == 2;
+    int bits16;
     uint8_t hdr[] = {
         0x52, 0x49, 0x46, 0x46, 0x00, 0x00, 0x00, 0x00, 0x57, 0x41, 0x56,
         0x45, 0x66, 0x6d, 0x74, 0x20, 0x10, 0x00, 0x00, 0x00, 0x01, 0x00,
@@ -115,34 +112,50 @@ static int wav_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt)
         0x00, 0x10, 0x00, 0x64, 0x61, 0x74, 0x61, 0x00, 0x00, 0x00, 0x00
     };
 
-    switch (audio_state.fixed_fmt) {
+    freq = audio_state.fixed_freq_out;
+    fmt = audio_state.fixed_fmt_out;
+    nchannels = audio_state.fixed_channels_out;
+
+    switch (fmt) {
     case AUD_FMT_S8:
     case AUD_FMT_U8:
+        bits16 = 0;
         break;
 
     case AUD_FMT_S16:
     case AUD_FMT_U16:
         bits16 = 1;
         break;
+
+    default:
+        dolog ("Internal logic error bad format %d\n", fmt);
+        return -1;
     }
 
     hdr[34] = bits16 ? 0x10 : 0x08;
-    hw->freq = 44100;
+    audio_pcm_init_info (
-    hw->nchannels = stereo ? 2 : 1;
+        &hw->info,
-    hw->fmt = bits16 ? AUD_FMT_S16 : AUD_FMT_U8;
+        freq,
+        nchannels,
+        bits16 ? AUD_FMT_S16 : AUD_FMT_U8,
+        audio_need_to_swap_endian (0)
+        );
     hw->bufsize = 4096;
     wav->pcm_buf = qemu_mallocz (hw->bufsize);
-    if (!wav->pcm_buf)
+    if (!wav->pcm_buf) {
+        dolog ("Can not initialize WAV buffer of %d bytes\n",
+               hw->bufsize);
         return -1;
+    }
 
-    le_store (hdr + 22, hw->nchannels, 2);
+    le_store (hdr + 22, hw->info.nchannels, 2);
-    le_store (hdr + 24, hw->freq, 4);
+    le_store (hdr + 24, hw->info.freq, 4);
-    le_store (hdr + 28, hw->freq << (bits16 + stereo), 4);
+    le_store (hdr + 28, hw->info.freq << (bits16 + (nchannels == 2)), 4);
-    le_store (hdr + 32, 1 << (bits16 + stereo), 2);
+    le_store (hdr + 32, 1 << (bits16 + (nchannels == 2)), 2);
 
     wav->f = fopen (conf.wav_path, "wb");
     if (!wav->f) {
-        dolog ("failed to open wave file `%s'\nReason: %s\n",
+        dolog ("Failed to open wave file `%s'\nReason: %s\n",
                conf.wav_path, strerror (errno));
         qemu_free (wav->pcm_buf);
         wav->pcm_buf = NULL;
@@ -153,17 +166,18 @@ static int wav_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt)
     return 0;
 }
 
-static void wav_hw_fini (HWVoice *hw)
+static void wav_fini_out (HWVoiceOut *hw)
 {
-    WAVVoice *wav = (WAVVoice *) hw;
+    WAVVoiceOut *wav = (WAVVoiceOut *) hw;
-    int stereo = hw->nchannels == 2;
+    int stereo = hw->info.nchannels == 2;
     uint8_t rlen[4];
     uint8_t dlen[4];
     uint32_t rifflen = (wav->total_samples << stereo) + 36;
     uint32_t datalen = wav->total_samples << stereo;
 
-    if (!wav->f || !hw->active)
+    if (!wav->f || !hw->active) {
         return;
+    }
 
     le_store (rlen, rifflen, 4);
     le_store (dlen, datalen, 4);
@@ -181,7 +195,7 @@ static void wav_hw_fini (HWVoice *hw)
     wav->pcm_buf = NULL;
 }
 
-static int wav_hw_ctl (HWVoice *hw, int cmd, ...)
+static int wav_ctl_out (HWVoiceOut *hw, int cmd, ...)
 {
     (void) hw;
     (void) cmd;
@@ -195,23 +209,41 @@ static void *wav_audio_init (void)
 
 static void wav_audio_fini (void *opaque)
 {
+    (void) opaque;
     ldebug ("wav_fini");
 }
 
-struct pcm_ops wav_pcm_ops = {
+struct audio_option wav_options[] = {
-    wav_hw_init,
+    {"PATH", AUD_OPT_STR, &conf.wav_path,
-    wav_hw_fini,
+     "Path to wave file", NULL, 0},
-    wav_hw_run,
+    {NULL, 0, NULL, NULL, NULL, 0}
-    wav_hw_write,
-    wav_hw_ctl
 };
 
-struct audio_output_driver wav_output_driver = {
+struct audio_pcm_ops wav_pcm_ops = {
-    "wav",
+    wav_init_out,
-    wav_audio_init,
+    wav_fini_out,
-    wav_audio_fini,
+    wav_run_out,
-    &wav_pcm_ops,
+    wav_write_out,
-    1,
+    wav_ctl_out,
-    1,
+
-    sizeof (WAVVoice)
+    NULL,
+    NULL,
+    NULL,
+    NULL,
+    NULL
+};
+
+struct audio_driver wav_audio_driver = {
+    INIT_FIELD (name           = ) "wav",
+    INIT_FIELD (descr          = )
+    "WAV renderer http://wikipedia.org/wiki/WAV",
+    INIT_FIELD (options        = ) wav_options,
+    INIT_FIELD (init           = ) wav_audio_init,
+    INIT_FIELD (fini           = ) wav_audio_fini,
+    INIT_FIELD (pcm_ops        = ) &wav_pcm_ops,
+    INIT_FIELD (can_be_default = ) 0,
+    INIT_FIELD (max_voices_out = ) 1,
+    INIT_FIELD (max_voices_in  = ) 0,
+    INIT_FIELD (voice_size_out = ) sizeof (WAVVoiceOut),
+    INIT_FIELD (voice_size_in  = ) 0
 };

configure

@@ -77,6 +77,9 @@ gdbstub="yes"
 slirp="yes"
 adlib="no"
 oss="no"
+dsound="no"
+coreaudio="no"
+alsa="no"
 fmod="no"
 fmod_lib=""
 fmod_inc=""
@@ -171,6 +174,12 @@ for opt do
   ;;
   --disable-sdl) sdl="no"
   ;;
+  --enable-coreaudio) coreaudio="yes"
+  ;;
+  --enable-alsa) alsa="yes"
+  ;;
+  --enable-dsound) dsound="yes"
+  ;;
   --enable-fmod) fmod="yes"
   ;;
   --fmod-lib=*) fmod_lib=${opt#--fmod-lib=}
@@ -187,7 +196,7 @@ for opt do
   ;;
   --kernel-path=*) kernel_path=${opt#--kernel-path=}
   ;;
-  --enable-cocoa) cocoa="yes" ; sdl="no"
+  --enable-cocoa) cocoa="yes" ; coreaudio="yes" ; sdl="no"
   ;;
   --disable-gfx-check) check_gfx="no"
   ;;
@@ -346,7 +355,10 @@ echo "  --make=MAKE              use specified make [$make]"
 echo "  --static                 enable static build [$static]"
 echo "  --enable-mingw32         enable Win32 cross compilation with mingw32"
 echo "  --enable-adlib           enable Adlib emulation"
-echo "  --enable-fmod            enable FMOD audio output driver"
+echo "  --enable-coreaudio       enable Coreaudio audio driver"
+echo "  --enable-alsa            enable ALSA audio driver"
+echo "  --enable-fmod            enable FMOD audio driver"
+echo "  --enabled-dsound         enable DirectSound audio driver"
 echo "  --fmod-lib               path to FMOD library"
 echo "  --fmod-inc               path to FMOD includes"
 echo ""
@@ -439,8 +451,18 @@ if test "$sdl" != "no" ; then
 fi
 echo "mingw32 support   $mingw32"
 echo "Adlib support     $adlib"
+echo "CoreAudio support $coreaudio"
+echo "ALSA support      $alsa"
+echo "DSound support    $dsound"
 echo -n "FMOD support      $fmod"
-if test $fmod = "yes"; then
+if test "$fmod" = "yes"; then
+    if test -z $fmod_lib || test -z $fmod_inc; then
+        echo
+        echo "Error: You must specify path to FMOD library and headers"
+        echo "Example: --fmod-inc=/path/include/fmod --fmod-lib=/path/lib/libfmod-3.74.so"
+        echo
+        exit 1
+    fi
     echo -n " (lib='$fmod_lib' include='$fmod_inc')"
 fi
 echo ""
@@ -568,6 +590,18 @@ if test "$oss" = "yes" ; then
   echo "CONFIG_OSS=yes" >> $config_mak
   echo "#define CONFIG_OSS 1" >> $config_h
 fi
+if test "$coreaudio" = "yes" ; then
+  echo "CONFIG_COREAUDIO=yes" >> $config_mak
+  echo "#define CONFIG_COREAUDIO 1" >> $config_h
+fi
+if test "$alsa" = "yes" ; then
+  echo "CONFIG_ALSA=yes" >> $config_mak
+  echo "#define CONFIG_ALSA 1" >> $config_h
+fi
+if test "$dsound" = "yes" ; then
+  echo "CONFIG_DSOUND=yes" >> $config_mak
+  echo "#define CONFIG_DSOUND 1" >> $config_h
+fi
 if test "$fmod" = "yes" ; then
   echo "CONFIG_FMOD=yes" >> $config_mak
   echo "CONFIG_FMOD_LIB=$fmod_lib" >> $config_mak

hw/adlib.c

@@ -1,7 +1,7 @@
 /*
- * QEMU Adlib emulation
+ * QEMU Proxy for OPL2/3 emulation by MAME team
  *
- * Copyright (c) 2004 Vassili Karpov (malc)
+ * Copyright (c) 2004-2005 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
@@ -21,8 +21,11 @@
  * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
  * THE SOFTWARE.
  */
+#include <assert.h>
 #include "vl.h"
 
+#define ADLIB_KILL_TIMERS 1
+
 #define dolog(...) AUD_log ("adlib", __VA_ARGS__)
 #ifdef DEBUG
 #define ldebug(...) dolog (__VA_ARGS__)
@@ -30,23 +33,15 @@
 #define ldebug(...)
 #endif
 
-#ifdef USE_YMF262
+#ifdef HAS_YMF262
-#define HAS_YMF262 1
 #include "ymf262.h"
-void YMF262UpdateOneQEMU(int which, INT16 *dst, int length);
+void YMF262UpdateOneQEMU (int which, INT16 *dst, int length);
 #define SHIFT 2
 #else
 #include "fmopl.h"
 #define SHIFT 1
 #endif
 
-#ifdef _WIN32
-#include <windows.h>
-#define small_delay() Sleep (1)
-#else
-#define small_delay() usleep (1)
-#endif
-
 #define IO_READ_PROTO(name) \
     uint32_t name (void *opaque, uint32_t nport)
 #define IO_WRITE_PROTO(name) \
@@ -58,35 +53,70 @@ static struct {
 } conf = {0x220, 44100};
 
 typedef struct {
+    int ticking[2];
     int enabled;
     int active;
-    int cparam;
-    int64_t ticks;
     int bufpos;
+#ifdef DEBUG
+    int64_t exp[2];
+#endif
     int16_t *mixbuf;
-    double interval;
+    uint64_t dexp[2];
-    QEMUTimer *ts, *opl_ts;
+    SWVoiceOut *voice;
-    SWVoice *voice;
+    int left, pos, samples;
-    int left, pos, samples, bytes_per_second, old_free;
+    QEMUAudioTimeStamp ats;
-    int refcount;
+#ifndef HAS_YMF262
-#ifndef USE_YMF262
     FM_OPL *opl;
 #endif
 } AdlibState;
 
 static AdlibState adlib;
 
+static void adlib_stop_opl_timer (AdlibState *s, size_t n)
+{
+#ifdef HAS_YMF262
+    YMF262TimerOver (0, n);
+#else
+    OPLTimerOver (s->opl, n);
+#endif
+    s->ticking[n] = 0;
+}
+
+static void adlib_kill_timers (AdlibState *s)
+{
+    size_t i;
+
+    for (i = 0; i < 2; ++i) {
+        if (s->ticking[i]) {
+            uint64_t delta;
+
+            delta = AUD_time_stamp_get_elapsed_usec_out (s->voice, &s->ats);
+            ldebug (
+                "delta = %f dexp = %f expired => %d\n",
+                delta / 1000000.0,
+                s->dexp[i] / 1000000.0,
+                delta >= s->dexp[i]
+                );
+            if (ADLIB_KILL_TIMERS || delta >= s->dexp[i]) {
+                adlib_stop_opl_timer (s, i);
+                AUD_init_time_stamp_out (s->voice, &s->ats);
+            }
+        }
+    }
+}
+
 static IO_WRITE_PROTO(adlib_write)
 {
     AdlibState *s = opaque;
     int a = nport & 3;
     int status;
 
-    s->ticks = qemu_get_clock (vm_clock);
     s->active = 1;
-    AUD_enable (s->voice, 1);
+    AUD_set_active_out (s->voice, 1);
 
-#ifdef USE_YMF262
+    adlib_kill_timers (s);
+
+#ifdef HAS_YMF262
     status = YMF262Write (0, a, val);
 #else
     status = OPLWrite (s->opl, a, val);
@@ -99,8 +129,9 @@ static IO_READ_PROTO(adlib_read)
     uint8_t data;
     int a = nport & 3;
 
-#ifdef USE_YMF262
+    adlib_kill_timers (s);
-    (void) s;
+
+#ifdef HAS_YMF262
     data = YMF262Read (0, a);
 #else
     data = OPLRead (s->opl, a);
@@ -108,119 +139,115 @@ static IO_READ_PROTO(adlib_read)
     return data;
 }
 
-static void OPL_timer (void *opaque)
+static void timer_handler (int c, double interval_Sec)
-{
-    AdlibState *s = opaque;
-#ifdef USE_YMF262
-    YMF262TimerOver (s->cparam >> 1, s->cparam & 1);
-#else
-    OPLTimerOver (s->opl, s->cparam);
-#endif
-    qemu_mod_timer (s->opl_ts, qemu_get_clock (vm_clock) + s->interval);
-}
-
-static void YMF262TimerHandler (int c, double interval_Sec)
 {
     AdlibState *s = &adlib;
+    unsigned n = c & 1;
+#ifdef DEBUG
+    double interval;
+#endif
+
     if (interval_Sec == 0.0) {
-        qemu_del_timer (s->opl_ts);
+        s->ticking[n] = 0;
         return;
     }
-    s->cparam = c;
+
-    s->interval = ticks_per_sec * interval_Sec;
+    s->ticking[n] = 1;
-    qemu_mod_timer (s->opl_ts, qemu_get_clock (vm_clock) + s->interval);
+#ifdef DEBUG
-    small_delay ();
+    interval = ticks_per_sec * interval_Sec;
+    exp = qemu_get_clock (vm_clock) + interval;
+    s->exp[n] = exp;
+#endif
+
+    s->dexp[n] = interval_Sec * 1000000.0;
+    AUD_init_time_stamp_out (s->voice, &s->ats);
 }
 
 static int write_audio (AdlibState *s, int samples)
 {
     int net = 0;
-    int ss = samples;
+    int pos = s->pos;
+
     while (samples) {
-        int nbytes = samples << SHIFT;
+        int nbytes, wbytes, wsampl;
-        int wbytes = AUD_write (s->voice,
+
-                                s->mixbuf + (s->pos << (SHIFT - 1)),
+        nbytes = samples << SHIFT;
-                                nbytes);
+        wbytes = AUD_write (
-        int wsampl = wbytes >> SHIFT;
+            s->voice,
+            s->mixbuf + (pos << (SHIFT - 1)),
+            nbytes
+            );
+
+        if (wbytes) {
+            wsampl = wbytes >> SHIFT;
+
             samples -= wsampl;
-        s->pos = (s->pos + wsampl) % s->samples;
+            pos = (pos + wsampl) % s->samples;
+
             net += wsampl;
-        if (!wbytes)
+        }
+        else {
             break;
         }
-    if (net > ss) {
-        dolog ("WARNING: net > ss\n");
     }
+
     return net;
 }
 
-static void timer (void *opaque)
+static void adlib_callback (void *opaque, int free)
 {
     AdlibState *s = opaque;
-    int elapsed, samples, net = 0;
+    int samples, net = 0, to_play, written;
 
-    if (s->refcount)
+    samples = free >> SHIFT;
-        dolog ("refcount=%d\n", s->refcount);
+    if (!(s->active && s->enabled) || !samples) {
-
+        return;
-    s->refcount += 1;
-    if (!(s->active && s->enabled))
-        goto reset;
-
-    AUD_run ();
-
-    while (s->left) {
-        int written = write_audio (s, s->left);
-        net += written;
-        if (!written)
-            goto reset2;
-        s->left -= written;
     }
-    s->pos = 0;
 
-    elapsed = AUD_calc_elapsed (s->voice);
+    to_play = audio_MIN (s->left, samples);
-    if (!elapsed)
+    while (to_play) {
-        goto reset2;
+        written = write_audio (s, to_play);
 
-    /* elapsed = AUD_get_free (s->voice); */
+        if (written) {
-    samples = elapsed >> SHIFT;
+            s->left -= written;
-    if (!samples)
+            samples -= written;
-        goto reset2;
+            to_play -= written;
+            s->pos = (s->pos + written) % s->samples;
+        }
+        else {
+            return;
+        }
+    }
 
     samples = audio_MIN (samples, s->samples - s->pos);
-    if (s->left)
+    if (!samples) {
-        dolog ("left=%d samples=%d elapsed=%d free=%d\n",
+        return;
-               s->left, samples, elapsed, AUD_get_free (s->voice));
+    }
 
-    if (!samples)
+#ifdef HAS_YMF262
-        goto reset2;
-
-#ifdef USE_YMF262
     YMF262UpdateOneQEMU (0, s->mixbuf + s->pos * 2, samples);
 #else
     YM3812UpdateOne (s->opl, s->mixbuf + s->pos, samples);
 #endif
 
     while (samples) {
-        int written = write_audio (s, samples);
+        written = write_audio (s, samples);
-        net += written;
-        if (!written)
-            break;
-        samples -= written;
-    }
-    if (!samples)
-        s->pos = 0;
-    s->left = samples;
 
-reset2:
+        if (written) {
-    AUD_adjust (s->voice, net << SHIFT);
+            net += written;
-reset:
+            samples -= written;
-    qemu_mod_timer (s->ts, qemu_get_clock (vm_clock) + ticks_per_sec / 1024);
+            s->pos = (s->pos + written) % s->samples;
-    s->refcount -= 1;
+        }
+        else {
+            s->left = samples;
+            return;
+        }
+    }
 }
 
 static void Adlib_fini (AdlibState *s)
 {
-#ifdef USE_YMF262
+#ifdef HAS_YMF262
     YMF262Shutdown ();
 #else
     if (s->opl) {
@@ -229,15 +256,9 @@ static void Adlib_fini (AdlibState *s)
     }
 #endif
 
-    if (s->opl_ts)
+    if (s->mixbuf) {
-        qemu_free_timer (s->opl_ts);
+        qemu_free (s->mixbuf);
-
+    }
-    if (s->ts)
-        qemu_free_timer (s->ts);
-
-#define maybe_free(p) if (p) qemu_free (p)
-    maybe_free (s->mixbuf);
-#undef maybe_free
 
     s->active = 0;
     s->enabled = 0;
@@ -247,15 +268,13 @@ void Adlib_init (void)
 {
     AdlibState *s = &adlib;
 
-    memset (s, 0, sizeof (*s));
+#ifdef HAS_YMF262
-
-#ifdef USE_YMF262
     if (YMF262Init (1, 14318180, conf.freq)) {
         dolog ("YMF262Init %d failed\n", conf.freq);
         return;
     }
     else {
-        YMF262SetTimerHandler (0, YMF262TimerHandler, 0);
+        YMF262SetTimerHandler (0, timer_handler, 0);
         s->enabled = 1;
     }
 #else
@@ -265,33 +284,26 @@ void Adlib_init (void)
         return;
     }
     else {
-        OPLSetTimerHandler (s->opl, YMF262TimerHandler, 0);
+        OPLSetTimerHandler (s->opl, timer_handler, 0);
         s->enabled = 1;
     }
 #endif
 
-    s->opl_ts = qemu_new_timer (vm_clock, OPL_timer, s);
+    s->voice = AUD_open_out (
-    if (!s->opl_ts) {
+        s->voice,
-        dolog ("Can not get timer for adlib emulation\n");
+        "adlib",
-        Adlib_fini (s);
+        s,
-        return;
+        adlib_callback,
-    }
+        conf.freq,
-
+        SHIFT,
-    s->ts = qemu_new_timer (vm_clock, timer, s);
+        AUD_FMT_S16
-    if (!s->opl_ts) {
+        );
-        dolog ("Can not get timer for adlib emulation\n");
-        Adlib_fini (s);
-        return;
-    }
-
-    s->voice = AUD_open (s->voice, "adlib", conf.freq, SHIFT, AUD_FMT_S16);
     if (!s->voice) {
         Adlib_fini (s);
         return;
     }
 
-    s->bytes_per_second = conf.freq << SHIFT;
+    s->samples = AUD_get_buffer_size_out (s->voice) >> SHIFT;
-    s->samples = AUD_get_buffer_size (s->voice) >> SHIFT;
     s->mixbuf = qemu_mallocz (s->samples << SHIFT);
 
     if (!s->mixbuf) {
@@ -300,6 +312,7 @@ void Adlib_init (void)
         Adlib_fini (s);
         return;
     }
+
     register_ioport_read (0x388, 4, 1, adlib_read, s);
     register_ioport_write (0x388, 4, 1, adlib_write, s);
 
@@ -308,6 +321,4 @@ void Adlib_init (void)
 
     register_ioport_read (conf.port + 8, 2, 1, adlib_read, s);
     register_ioport_write (conf.port + 8, 2, 1, adlib_write, s);
-
-    qemu_mod_timer (s->ts, qemu_get_clock (vm_clock) + 1);
 }

hw/pc.c

@@ -602,18 +602,18 @@ static void pc_init1(int ram_size, int vga_ram_size, int boot_device,
 
     if (audio_enabled) {
         AUD_init();
-#ifdef USE_SB16
         if (sb16_enabled)
             SB16_init ();
-#endif
 #ifdef CONFIG_ADLIB
         if (adlib_enabled)
             Adlib_init ();
 #endif
-#ifdef USE_GUS
+#ifdef CONFIG_GUS
         if (gus_enabled)
             GUS_init ();
 #endif
+        if (pci_enabled && es1370_enabled)
+            es1370_init (pci_bus);
     }
 
     floppy_controller = fdctrl_init(6, 2, 0, 0x3f0, fd_table);

hw/sb16.c

@@ -1,7 +1,7 @@
 /*
  * QEMU Soundblaster 16 emulation
  *
- * Copyright (c) 2003-2004 Vassili Karpov (malc)
+ * Copyright (c) 2003-2005 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
@@ -99,9 +99,10 @@ typedef struct SB16State {
     int dma_running;
     int bytes_per_second;
     int align;
-    SWVoice *voice;
+    int audio_free;
+    SWVoiceOut *voice;
 
-    QEMUTimer *ts, *aux_ts;
+    QEMUTimer *aux_ts;
     /* mixer state */
     int mixer_nreg;
     uint8_t mixer_regs[256];
@@ -110,6 +111,8 @@ typedef struct SB16State {
 /* XXX: suppress that and use a context */
 static struct SB16State dsp;
 
+static void SB_audio_callback (void *opaque, int free);
+
 static int magic_of_irq (int irq)
 {
     switch (irq) {
@@ -174,11 +177,11 @@ static void control (SB16State *s, int hold)
 
     if (hold) {
         DMA_hold_DREQ (dma);
-        AUD_enable (s->voice, 1);
+        AUD_set_active_out (s->voice, 1);
     }
     else {
         DMA_release_DREQ (dma);
-        AUD_enable (s->voice, 0);
+        AUD_set_active_out (s->voice, 0);
     }
 }
 
@@ -207,8 +210,9 @@ static void dma_cmd8 (SB16State *s, int mask, int dma_len)
         s->freq = (1000000 + (tmp / 2)) / tmp;
     }
 
-    if (dma_len != -1)
+    if (dma_len != -1) {
         s->block_size = dma_len << s->fmt_stereo;
+    }
     else {
         /* This is apparently the only way to make both Act1/PL
            and SecondReality/FC work
@@ -227,17 +231,28 @@ static void dma_cmd8 (SB16State *s, int mask, int dma_len)
     s->dma_auto = (mask & DMA8_AUTO) != 0;
     s->align = (1 << s->fmt_stereo) - 1;
 
-    if (s->block_size & s->align)
+    if (s->block_size & s->align) {
-        dolog ("warning: unaligned buffer\n");
+        dolog ("warning: misaligned block size %d, alignment %d\n",
+               s->block_size, s->align + 1);
+    }
 
     ldebug ("freq %d, stereo %d, sign %d, bits %d, "
             "dma %d, auto %d, fifo %d, high %d\n",
             s->freq, s->fmt_stereo, s->fmt_signed, s->fmt_bits,
             s->block_size, s->dma_auto, s->fifo, s->highspeed);
 
-    if (s->freq)
+    if (s->freq) {
-        s->voice = AUD_open (s->voice, "sb16", s->freq,
+        s->audio_free = 0;
-                             1 << s->fmt_stereo, s->fmt);
+        s->voice = AUD_open_out (
+            s->voice,
+            "sb16",
+            s,
+            SB_audio_callback,
+            s->freq,
+            1 << s->fmt_stereo,
+            s->fmt
+            );
+    }
 
     control (s, 1);
     speaker (s, 1);
@@ -309,12 +324,23 @@ static void dma_cmd (SB16State *s, uint8_t cmd, uint8_t d0, int dma_len)
     s->bytes_per_second = (s->freq << s->fmt_stereo) << (s->fmt_bits == 16);
     s->highspeed = 0;
     s->align = (1 << (s->fmt_stereo + (s->fmt_bits == 16))) - 1;
-    if (s->block_size & s->align)
+    if (s->block_size & s->align) {
-        dolog ("warning: unaligned buffer\n");
+        dolog ("warning: misaligned block size %d, alignment %d\n",
+               s->block_size, s->align + 1);
+    }
 
-    if (s->freq)
+    if (s->freq) {
-        s->voice = AUD_open (s->voice, "sb16", s->freq,
+        s->audio_free = 0;
-                             1 << s->fmt_stereo, s->fmt);
+        s->voice = AUD_open_out (
+            s->voice,
+            "sb16",
+            s,
+            SB_audio_callback,
+            s->freq,
+            1 << s->fmt_stereo,
+            s->fmt
+            );
+    }
 
     control (s, 1);
     speaker (s, 1);
@@ -323,14 +349,16 @@ static void dma_cmd (SB16State *s, uint8_t cmd, uint8_t d0, int dma_len)
 static inline void dsp_out_data (SB16State *s, uint8_t val)
 {
     ldebug ("outdata %#x\n", val);
-    if (s->out_data_len < sizeof (s->out_data))
+    if (s->out_data_len < sizeof (s->out_data)) {
         s->out_data[s->out_data_len++] = val;
+    }
 }
 
 static inline uint8_t dsp_get_data (SB16State *s)
 {
-    if (s->in_index)
+    if (s->in_index) {
         return s->in2_data[--s->in_index];
+    }
     else {
         dolog ("buffer underflow\n");
         return 0;
@@ -356,6 +384,8 @@ static void command (SB16State *s, uint8_t cmd)
         s->needed_bytes = 3;
     }
     else {
+        s->needed_bytes = 0;
+
         switch (cmd) {
         case 0x03:
             dsp_out_data (s, 0x10); /* s->csp_param); */
@@ -403,7 +433,7 @@ static void command (SB16State *s, uint8_t cmd)
             goto warn;
 
         case 0x35:
-            dolog ("MIDI command(0x35) not implemented\n");
+            dolog ("0x35 - MIDI command not implemented\n");
             break;
 
         case 0x40:
@@ -435,6 +465,38 @@ static void command (SB16State *s, uint8_t cmd)
             s->needed_bytes = 2;
             break;
 
+        case 0x74:
+            s->needed_bytes = 2; /* DMA DAC, 4-bit ADPCM */
+            dolog ("0x75 - DMA DAC, 4-bit ADPCM not implemented\n");
+            break;
+
+        case 0x75:              /* DMA DAC, 4-bit ADPCM Reference */
+            s->needed_bytes = 2;
+            dolog ("0x74 - DMA DAC, 4-bit ADPCM Reference not implemented\n");
+            break;
+
+        case 0x76:              /* DMA DAC, 2.6-bit ADPCM */
+            s->needed_bytes = 2;
+            dolog ("0x74 - DMA DAC, 2.6-bit ADPCM not implemented\n");
+            break;
+
+        case 0x77:              /* DMA DAC, 2.6-bit ADPCM Reference */
+            s->needed_bytes = 2;
+            dolog ("0x74 - DMA DAC, 2.6-bit ADPCM Reference not implemented\n");
+            break;
+
+        case 0x7d:
+            dolog ("0x7d - Autio-Initialize DMA DAC, 4-bit ADPCM Reference\n");
+            dolog ("not implemented\n");
+            break;
+
+        case 0x7f:
+            dolog (
+                "0x7d - Autio-Initialize DMA DAC, 2.6-bit ADPCM Reference\n"
+                );
+            dolog ("not implemented\n");
+            break;
+
         case 0x80:
             s->needed_bytes = 2;
             break;
@@ -476,9 +538,9 @@ static void command (SB16State *s, uint8_t cmd)
             s->dma_auto = 0;
             break;
 
-        case 0xe0:
+        case 0xe0:              /* DSP identification */
             s->needed_bytes = 1;
-            goto warn;
+            break;
 
         case 0xe1:
             dsp_out_data (s, s->ver & 0xff);
@@ -503,7 +565,7 @@ static void command (SB16State *s, uint8_t cmd)
 
         case 0xe7:
             dolog ("Attempt to probe for ESS (0xe7)?\n");
-            return;
+            break;
 
         case 0xe8:              /* read test reg */
             dsp_out_data (s, s->test_reg);
@@ -529,21 +591,29 @@ static void command (SB16State *s, uint8_t cmd)
             goto warn;
 
         default:
-            dolog ("unrecognized command %#x\n", cmd);
+            dolog ("Unrecognized command %#x\n", cmd);
-            return;
+            break;
         }
     }
 
-    s->cmd = cmd;
+    if (!s->needed_bytes) {
-    if (!s->needed_bytes)
         ldebug ("\n");
+    }
+
+ exit:
+    if (!s->needed_bytes) {
+        s->cmd = -1;
+    }
+    else {
+        s->cmd = cmd;
+    }
     return;
 
  warn:
     dolog ("warning: command %#x,%d is not truly understood yet\n",
            cmd, s->needed_bytes);
-    s->cmd = cmd;
+    goto exit;
-    return;
+
 }
 
 static uint16_t dsp_get_lohi (SB16State *s)
@@ -607,8 +677,9 @@ static void complete (SB16State *s)
                 s->csp_reg83[s->csp_reg83r % 4] = d0;
                 s->csp_reg83r += 1;
             }
-            else
+            else {
                 s->csp_regs[d1] = d0;
+            }
             break;
 
         case 0x0f:
@@ -622,8 +693,9 @@ static void complete (SB16State *s)
                 dsp_out_data (s, s->csp_reg83[s->csp_reg83w % 4]);
                 s->csp_reg83w += 1;
             }
-            else
+            else {
                 dsp_out_data (s, s->csp_regs[d0]);
+            }
             break;
 
         case 0x10:
@@ -641,8 +713,9 @@ static void complete (SB16State *s)
             break;
 
         case 0x42:              /* FT2 sets output freq with this, go figure */
+#if 0
             dolog ("cmd 0x42 might not do what it think it should\n");
-
+#endif
         case 0x41:
             s->freq = dsp_get_hilo (s);
             ldebug ("set freq %d\n", s->freq);
@@ -653,6 +726,13 @@ static void complete (SB16State *s)
             ldebug ("set dma block len %d\n", s->block_size);
             break;
 
+        case 0x74:
+        case 0x75:
+        case 0x76:
+        case 0x77:
+            /* ADPCM stuff, ignore */
+            break;
+
         case 0x80:
             {
                 int freq, samples, bytes;
@@ -662,10 +742,17 @@ static void complete (SB16State *s)
                 samples = dsp_get_lohi (s) + 1;
                 bytes = samples << s->fmt_stereo << (s->fmt_bits == 16);
                 ticks = (bytes * ticks_per_sec) / freq;
-                if (ticks < ticks_per_sec / 1024)
+                if (ticks < ticks_per_sec / 1024) {
                     pic_set_irq (s->irq, 1);
-                else
+                }
-                    qemu_mod_timer (s->aux_ts, qemu_get_clock (vm_clock) + ticks);
+                else {
+                    if (s->aux_ts) {
+                        qemu_mod_timer (
+                            s->aux_ts,
+                            qemu_get_clock (vm_clock) + ticks
+                            );
+                    }
+                }
                 ldebug ("mix silence %d %d %lld\n", samples, bytes, ticks);
             }
             break;
@@ -674,7 +761,7 @@ static void complete (SB16State *s)
             d0 = dsp_get_data (s);
             s->out_data_len = 0;
             ldebug ("E0 data = %#x\n", d0);
-            dsp_out_data(s, ~d0);
+            dsp_out_data (s, ~d0);
             break;
 
         case 0xe2:
@@ -737,6 +824,7 @@ static void reset (SB16State *s)
     s->nzero = 0;
     s->highspeed = 0;
     s->v2x6 = 0;
+    s->cmd = -1;
 
     dsp_out_data(s, 0xaa);
     speaker (s, 0);
@@ -761,9 +849,10 @@ static IO_WRITE_PROTO (dsp_write)
                     pic_set_irq (s->irq, 0);
                     control (s, 0);
                 }
-                else
+                else {
                     reset (s);
                 }
+            }
             s->v2x6 = 0;
             break;
 
@@ -845,7 +934,10 @@ static IO_READ_PROTO (dsp_read)
             s->last_read_byte = retval;
         }
         else {
-            dolog ("empty output buffer\n");
+            if (s->cmd != -1) {
+                dolog ("empty output buffer for command %#x\n",
+                       s->cmd);
+            }
             retval = s->last_read_byte;
             /* goto error; */
         }
@@ -882,13 +974,14 @@ static IO_READ_PROTO (dsp_read)
         goto error;
     }
 
-    if (!ack)
+    if (!ack) {
         ldebug ("read %#x -> %#x\n", nport, retval);
+    }
 
     return retval;
 
  error:
-    dolog ("WARNING dsp_read %#x error\n", nport);
+    dolog ("warning: dsp_read %#x error\n", nport);
     return 0xff;
 }
 
@@ -933,8 +1026,9 @@ static IO_WRITE_PROTO(mixer_write_datab)
     SB16State *s = opaque;
 
     ldebug ("mixer_write [%#x] <- %#x\n", s->mixer_nreg, val);
-    if (s->mixer_nreg > sizeof (s->mixer_regs))
+    if (s->mixer_nreg > sizeof (s->mixer_regs)) {
         return;
+    }
 
     switch (s->mixer_nreg) {
     case 0x00:
@@ -945,9 +1039,10 @@ static IO_WRITE_PROTO(mixer_write_datab)
         {
             int irq = irq_of_magic (val);
             ldebug ("setting irq to %d (val=%#x)\n", irq, val);
-            if (irq > 0)
+            if (irq > 0) {
                 s->irq = irq;
             }
+        }
         break;
 
     case 0x81:
@@ -956,8 +1051,12 @@ static IO_WRITE_PROTO(mixer_write_datab)
 
             dma = lsbindex (val & 0xf);
             hdma = lsbindex (val & 0xf0);
-            dolog ("attempt to set DMA register 8bit %d, 16bit %d (val=%#x)\n",
+            if (dma != s->dma || hdma != s->hdma) {
-                   dma, hdma, val);
+                dolog (
+                    "attempt to change DMA "
+                    "8bit %d(%d), 16bit %d(%d) (val=%#x)\n",
+                    dma, s->dma, hdma, s->hdma, val);
+            }
 #if 0
             s->dma = dma;
             s->hdma = hdma;
@@ -971,8 +1070,9 @@ static IO_WRITE_PROTO(mixer_write_datab)
         return;
 
     default:
-        if (s->mixer_nreg >= 0x80)
+        if (s->mixer_nreg >= 0x80) {
-            dolog ("attempt to write mixer[%#x] <- %#x\n", s->mixer_nreg, val);
+            ldebug ("attempt to write mixer[%#x] <- %#x\n", s->mixer_nreg, val);
+        }
         break;
     }
 
@@ -989,10 +1089,14 @@ static IO_READ_PROTO(mixer_read)
 {
     SB16State *s = opaque;
 #ifndef DEBUG_SB16_MOST
-    if (s->mixer_nreg != 0x82)
+    if (s->mixer_nreg != 0x82) {
-#endif
         ldebug ("mixer_read[%#x] -> %#x\n",
                 s->mixer_nreg, s->mixer_regs[s->mixer_nreg]);
+    }
+#else
+    ldebug ("mixer_read[%#x] -> %#x\n",
+            s->mixer_nreg, s->mixer_regs[s->mixer_nreg]);
+#endif
     return s->mixer_regs[s->mixer_nreg];
 }
 
@@ -1010,8 +1114,9 @@ static int write_audio (SB16State *s, int nchan, int dma_pos,
         int to_copy, copied;
 
         to_copy = audio_MIN (temp, left);
-        if (to_copy > sizeof(tmpbuf))
+        if (to_copy > sizeof(tmpbuf)) {
             to_copy = sizeof(tmpbuf);
+        }
 
         copied = DMA_read_memory (nchan, tmpbuf, dma_pos, to_copy);
         copied = AUD_write (s->voice, tmpbuf, copied);
@@ -1020,9 +1125,10 @@ static int write_audio (SB16State *s, int nchan, int dma_pos,
         dma_pos = (dma_pos + copied) % dma_len;
         net += copied;
 
-        if (!copied)
+        if (!copied) {
             break;
         }
+    }
 
     return net;
 }
@@ -1030,27 +1136,28 @@ static int write_audio (SB16State *s, int nchan, int dma_pos,
 static int SB_read_DMA (void *opaque, int nchan, int dma_pos, int dma_len)
 {
     SB16State *s = opaque;
-    int free, rfree, till, copy, written, elapsed;
+    int till, copy, written, free;
 
     if (s->left_till_irq < 0) {
         s->left_till_irq = s->block_size;
     }
 
-    elapsed = AUD_calc_elapsed (s->voice);
+    if (s->voice) {
-    free = elapsed;/* AUD_get_free (s->voice); */
+        free = s->audio_free & ~s->align;
-    rfree = free;
-    free = audio_MIN (free, elapsed) & ~s->align;
-
         if ((free <= 0) || !dma_len) {
             return dma_pos;
         }
+    }
+    else {
+        free = dma_len;
+    }
 
     copy = free;
     till = s->left_till_irq;
 
 #ifdef DEBUG_SB16_MOST
-    dolog ("pos:%06d free:%d,%d till:%d len:%d\n",
+    dolog ("pos:%06d %d till:%d len:%d\n",
-           dma_pos, free, AUD_get_free (s->voice), till, dma_len);
+           dma_pos, free, till, dma_len);
 #endif
 
     if (till <= copy) {
@@ -1082,15 +1189,13 @@ static int SB_read_DMA (void *opaque, int nchan, int dma_pos, int dma_len)
         s->left_till_irq = s->block_size + s->left_till_irq;
     }
 
-    AUD_adjust (s->voice, written);
     return dma_pos;
 }
 
-void SB_timer (void *opaque)
+static void SB_audio_callback (void *opaque, int free)
 {
     SB16State *s = opaque;
-    AUD_run ();
+    s->audio_free = free;
-    qemu_mod_timer (s->ts, qemu_get_clock (vm_clock) + 1);
 }
 
 static void SB_save (QEMUFile *f, void *opaque)
@@ -1150,8 +1255,9 @@ static int SB_load (QEMUFile *f, void *opaque, int version_id)
 {
     SB16State *s = opaque;
 
-    if (version_id != 1)
+    if (version_id != 1) {
         return -EINVAL;
+    }
 
     qemu_get_be32s (f, &s->irq);
     qemu_get_be32s (f, &s->dma);
@@ -1202,14 +1308,23 @@ static int SB_load (QEMUFile *f, void *opaque, int version_id)
     qemu_get_buffer (f, s->mixer_regs, 256);
 
     if (s->voice) {
-        AUD_close (s->voice);
+        AUD_close_out (s->voice);
         s->voice = NULL;
     }
 
     if (s->dma_running) {
-        if (s->freq)
+        if (s->freq) {
-            s->voice = AUD_open (s->voice, "sb16", s->freq,
+            s->audio_free = 0;
-                                 1 << s->fmt_stereo, s->fmt);
+            s->voice = AUD_open_out (
+                s->voice,
+                "sb16",
+                s,
+                SB_audio_callback,
+                s->freq,
+                1 << s->fmt_stereo,
+                s->fmt
+                );
+        }
 
         control (s, 1);
         speaker (s, s->speaker);
@@ -1224,10 +1339,7 @@ void SB16_init (void)
     static const uint8_t dsp_write_ports[] = {0x6, 0xc};
     static const uint8_t dsp_read_ports[] = {0x6, 0xa, 0xc, 0xd, 0xe, 0xf};
 
-    s->ts = qemu_new_timer (vm_clock, SB_timer, s);
+    s->cmd = -1;
-    if (!s->ts)
-        return;
-
     s->irq = conf.irq;
     s->dma = conf.dma;
     s->hdma = conf.hdma;
@@ -1243,8 +1355,9 @@ void SB16_init (void)
 
     reset_mixer (s);
     s->aux_ts = qemu_new_timer (vm_clock, aux_timer, s);
-    if (!s->aux_ts)
+    if (!s->aux_ts) {
-        return;
+        dolog ("Can not create auxiliary timer\n");
+    }
 
     for (i = 0; i < LENOFA (dsp_write_ports); i++) {
         register_ioport_write (s->port + dsp_write_ports[i], 1, 1, dsp_write, s);
@@ -1263,6 +1376,5 @@ void SB16_init (void)
     DMA_register_channel (s->dma, SB_read_DMA, s);
     s->can_write = 1;
 
-    qemu_mod_timer (s->ts, qemu_get_clock (vm_clock) + 1);
     register_savevm ("sb16", 0, 1, SB_save, SB_load, s);
 }

qemu-doc.texi

@@ -181,8 +181,23 @@ The default is @code{en-us}.
 
 @item -enable-audio
 
-The SB16 emulation is disabled by default as it may give problems with
+Will enable audio and all the sound hardware QEMU was built with.
-Windows. You can enable it manually with this option.
+
+@item -audio-help
+
+Will show the audio subsystem help: list of drivers, tunable
+parameters.
+
+@item -soundhw card1,card2,...
+
+Enable audio and selected sound hardware. Use ? to print all
+available sound hardware.
+
+@example
+qemu -soundhw sb16,adlib hda
+qemu -soundhw es1370 hda
+qemu -soundhw ?
+@end example
 
 @item -localtime
 Set the real time clock to local time (the default is to UTC

vl.c

@@ -127,10 +127,13 @@ int nb_nics;
 NetDriverState nd_table[MAX_NICS];
 QEMUTimer *gui_timer;
 int vm_running;
+#ifdef HAS_AUDIO
 int audio_enabled = 0;
-int sb16_enabled = 1;
+int sb16_enabled = 0;
-int adlib_enabled = 1;
+int adlib_enabled = 0;
-int gus_enabled = 1;
+int gus_enabled = 0;
+int es1370_enabled = 0;
+#endif
 int pci_enabled = 1;
 int prep_enabled = 0;
 int rtc_utc = 1;
@@ -930,7 +933,7 @@ static void init_timers(void)
 #ifdef _WIN32
     {
         int count=0;
-        timerID = timeSetEvent(10,    // interval (ms)
+        timerID = timeSetEvent(1,     // interval (ms)
                                0,     // resolution
                                host_alarm_handler, // function
                                (DWORD)&count,  // user parameter
@@ -2837,7 +2840,12 @@ void help(void)
 #ifndef _WIN32
 	   "-k language     use keyboard layout (for example \"fr\" for French)\n"
 #endif
+#ifdef HAS_AUDIO
            "-enable-audio   enable audio support\n"
+           "-audio-help     print list of audio drivers and their options\n"
+           "-soundhw c1,... comma separated list of sound card names\n"
+           "                use -soundhw ? to get the list of supported sound cards\n"
+#endif
            "-localtime      set the real time clock to local time [default=utc]\n"
            "-full-screen    start in full screen\n"
 #ifdef TARGET_I386
@@ -2935,7 +2943,11 @@ enum {
     QEMU_OPTION_snapshot,
     QEMU_OPTION_m,
     QEMU_OPTION_nographic,
+#ifdef HAS_AUDIO
     QEMU_OPTION_enable_audio,
+    QEMU_OPTION_audio_help,
+    QEMU_OPTION_soundhw,
+#endif
 
     QEMU_OPTION_nics,
     QEMU_OPTION_macaddr,
@@ -2998,7 +3010,11 @@ const QEMUOption qemu_options[] = {
     { "m", HAS_ARG, QEMU_OPTION_m },
     { "nographic", 0, QEMU_OPTION_nographic },
     { "k", HAS_ARG, QEMU_OPTION_k },
+#ifdef HAS_AUDIO
     { "enable-audio", 0, QEMU_OPTION_enable_audio },
+    { "audio-help", 0, QEMU_OPTION_audio_help },
+    { "soundhw", HAS_ARG, QEMU_OPTION_soundhw },
+#endif
 
     { "nics", HAS_ARG, QEMU_OPTION_nics},
     { "macaddr", HAS_ARG, QEMU_OPTION_macaddr},
@@ -3117,6 +3133,78 @@ void register_machines(void)
 #endif
 }
 
+#ifdef HAS_AUDIO
+static void select_soundhw (const char *optarg)
+{
+    if (*optarg == '?') {
+    show_valid_cards:
+        printf ("Valid sound card names (comma separated):\n");
+        printf ("sb16       Creative Sound Blaster 16\n");
+#ifdef CONFIG_ADLIB
+#ifdef HAS_YMF262
+        printf ("adlib      Ymaha YMF262 (OPL3)\n");
+#else
+        printf ("adlib      Ymaha YM3812 (OPL2)\n");
+#endif
+#endif
+#ifdef CONFIG_GUS
+        printf ("gus        Gravis Ultrasound GF1\n");
+#endif
+        printf ("es1370     ENSONIQ AudioPCI ES1370\n");
+        exit (*optarg != '?');
+    }
+    else {
+        struct {
+            char *name;
+            int *enabledp;
+        } soundhw_tab[] = {
+            { "sb16", &sb16_enabled },
+#ifdef CONFIG_ADLIB
+            { "adlib", &adlib_enabled },
+#endif
+#ifdef CONFIG_GUS
+            { "gus", &gus_enabled },
+#endif
+            { "es1370", &es1370_enabled },
+        };
+        size_t tablen, l, i;
+        const char *p;
+        char *e;
+        int bad_card = 0;
+
+        p = optarg;
+        tablen = sizeof (soundhw_tab) / sizeof (soundhw_tab[0]);
+
+        while (*p) {
+            e = strchr (p, ',');
+            l = !e ? strlen (p) : (size_t) (e - p);
+            for (i = 0; i < tablen; ++i) {
+                if (!strncmp (soundhw_tab[i].name, p, l)) {
+                    audio_enabled = 1;
+                    *soundhw_tab[i].enabledp = 1;
+                    break;
+                }
+            }
+            if (i == tablen) {
+                if (l > 80) {
+                    fprintf (stderr,
+                             "Unknown sound card name (too big to show)\n");
+                }
+                else {
+                    fprintf (stderr, "Unknown sound card name `%.*s'\n",
+                             (int) l, p);
+                }
+                bad_card = 1;
+            }
+            p += l + (e != NULL);
+        }
+
+        if (bad_card)
+            goto show_valid_cards;
+    }
+}
+#endif
+
 #define NET_IF_TUN   0
 #define NET_IF_USER  1
 #define NET_IF_DUMMY 2
@@ -3401,9 +3489,22 @@ int main(int argc, char **argv)
             case QEMU_OPTION_dummy_net:
                 net_if_type = NET_IF_DUMMY;
                 break;
+#ifdef HAS_AUDIO
             case QEMU_OPTION_enable_audio:
                 audio_enabled = 1;
+                sb16_enabled = 1;
+                adlib_enabled = 1;
+                gus_enabled = 1;
+                es1370_enabled = 1;
                 break;
+            case QEMU_OPTION_audio_help:
+                AUD_help ();
+                exit (0);
+                break;
+            case QEMU_OPTION_soundhw:
+                select_soundhw (optarg);
+                break;
+#endif
             case QEMU_OPTION_h:
                 help();
                 break;

vl.h

@@ -124,6 +124,7 @@ extern int audio_enabled;
 extern int sb16_enabled;
 extern int adlib_enabled;
 extern int gus_enabled;
+extern int es1370_enabled;
 extern int ram_size;
 extern int bios_size;
 extern int rtc_utc;
@@ -628,6 +629,9 @@ void pci_piix3_ide_init(PCIBus *bus, BlockDriverState **hd_table);
 int pmac_ide_init (BlockDriverState **hd_table,
                    SetIRQFunc *set_irq, void *irq_opaque, int irq);
 
+/* es1370.c */
+int es1370_init (PCIBus *bus);
+
 /* sb16.c */
 void SB16_init (void);