Merge tag 'q800-for-8.2-pull-request' of https://github.com/vivier/qemu-m68k into...

[mirror_qemu.git] / audio / alsaaudio.c

diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index a8e62542f97e01ef9290ac184c71ad0c30790cca..cacae1ea59ce82db43884a185f476c6749d4d5fc 100644 (file)
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -34,6 +34,8 @@
 #define AUDIO_CAP "alsa"
 #include "audio_int.h"
 
+#define DEBUG_ALSA 0
+
 struct pollhlp {
     snd_pcm_t *handle;
     struct pollfd *pfds;
@@ -70,7 +72,7 @@ struct alsa_params_obt {
     snd_pcm_uframes_t samples;
 };
 
-static void GCC_FMT_ATTR (2, 3) alsa_logerr (int err, const char *fmt, ...)
+static void G_GNUC_PRINTF (2, 3) alsa_logerr (int err, const char *fmt, ...)
 {
     va_list ap;
 
@@ -81,7 +83,7 @@ static void GCC_FMT_ATTR (2, 3) alsa_logerr (int err, const char *fmt, ...)
     AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
 }
 
-static void GCC_FMT_ATTR (3, 4) alsa_logerr2 (
+static void G_GNUC_PRINTF (3, 4) alsa_logerr2 (
     int err,
     const char *typ,
     const char *fmt,
@@ -220,11 +222,7 @@ static int alsa_poll_helper (snd_pcm_t *handle, struct pollhlp *hlp, int mask)
         return -1;
     }
 
-    pfds = audio_calloc ("alsa_poll_helper", count, sizeof (*pfds));
-    if (!pfds) {
-        dolog ("Could not initialize poll mode\n");
-        return -1;
-    }
+    pfds = g_new0(struct pollfd, count);
 
     err = snd_pcm_poll_descriptors (handle, pfds, count);
     if (err < 0) {
@@ -278,32 +276,28 @@ static snd_pcm_format_t aud_to_alsafmt (AudioFormat fmt, int endianness)
     case AUDIO_FORMAT_S16:
         if (endianness) {
             return SND_PCM_FORMAT_S16_BE;
-        }
-        else {
+        } else {
             return SND_PCM_FORMAT_S16_LE;
         }
 
     case AUDIO_FORMAT_U16:
         if (endianness) {
             return SND_PCM_FORMAT_U16_BE;
-        }
-        else {
+        } else {
             return SND_PCM_FORMAT_U16_LE;
         }
 
     case AUDIO_FORMAT_S32:
         if (endianness) {
             return SND_PCM_FORMAT_S32_BE;
-        }
-        else {
+        } else {
             return SND_PCM_FORMAT_S32_LE;
         }
 
     case AUDIO_FORMAT_U32:
         if (endianness) {
             return SND_PCM_FORMAT_U32_BE;
-        }
-        else {
+        } else {
             return SND_PCM_FORMAT_U32_LE;
         }
 
@@ -451,7 +445,7 @@ static int alsa_open(bool in, struct alsa_params_req *req,
     snd_pcm_hw_params_t *hw_params;
     int err;
     unsigned int freq, nchannels;
-    const char *pcm_name = apdo->has_dev ? apdo->dev : "default";
+    const char *pcm_name = apdo->dev ?: "default";
     snd_pcm_uframes_t obt_buffer_size;
     const char *typ = in ? "ADC" : "DAC";
     snd_pcm_format_t obtfmt;
@@ -591,16 +585,12 @@ static int alsa_open(bool in, struct alsa_params_req *req,
 
     *handlep = handle;
 
-    if (obtfmt != req->fmt ||
-         obt->nchannels != req->nchannels ||
-         obt->freq != req->freq) {
+    if (DEBUG_ALSA || obtfmt != req->fmt ||
+        obt->nchannels != req->nchannels || obt->freq != req->freq) {
         dolog ("Audio parameters for %s\n", typ);
         alsa_dump_info(req, obt, obtfmt, apdo);
     }
 
-#ifdef DEBUG
-    alsa_dump_info(req, obt, obtfmt, pdo);
-#endif
     return 0;
 
  err:
@@ -608,6 +598,42 @@ static int alsa_open(bool in, struct alsa_params_req *req,
     return -1;
 }
 
+static size_t alsa_buffer_get_free(HWVoiceOut *hw)
+{
+    ALSAVoiceOut *alsa = (ALSAVoiceOut *)hw;
+    snd_pcm_sframes_t avail;
+    size_t alsa_free, generic_free, generic_in_use;
+
+    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) {
+            alsa_logerr(avail,
+                        "Could not obtain number of available frames\n");
+            avail = 0;
+        }
+    }
+
+    alsa_free = avail * hw->info.bytes_per_frame;
+    generic_free = audio_generic_buffer_get_free(hw);
+    generic_in_use = hw->samples * hw->info.bytes_per_frame - generic_free;
+    if (generic_in_use) {
+        /*
+         * This code can only be reached in the unlikely case that
+         * snd_pcm_avail_update() returned a larger number of frames
+         * than snd_pcm_writei() could write. Make sure that all
+         * remaining bytes in the generic buffer can be written.
+         */
+        alsa_free = alsa_free > generic_in_use ? alsa_free - generic_in_use : 0;
+    }
+
+    return alsa_free;
+}
+
 static size_t alsa_write(HWVoiceOut *hw, void *buf, size_t len)
 {
     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
@@ -722,8 +748,7 @@ static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int ctl)
             alsa_logerr (err, "Could not stop %s\n", typ);
             return -1;
         }
-    }
-    else {
+    } else {
         err = snd_pcm_prepare (handle);
         if (err < 0) {
             alsa_logerr (err, "Could not prepare handle for %s\n", typ);
@@ -879,7 +904,7 @@ static void alsa_init_per_direction(AudiodevAlsaPerDirectionOptions *apdo)
     }
 }
 
-static void *alsa_audio_init(Audiodev *dev)
+static void *alsa_audio_init(Audiodev *dev, Error **errp)
 {
     AudiodevAlsaOptions *aopts;
     assert(dev->driver == AUDIODEV_DRIVER_ALSA);
@@ -888,28 +913,23 @@ static void *alsa_audio_init(Audiodev *dev)
     alsa_init_per_direction(aopts->in);
     alsa_init_per_direction(aopts->out);
 
-    /*
-     * need to define them, as otherwise alsa produces no sound
-     * doesn't set has_* so alsa_open can identify it wasn't set by the user
-     */
+    /* don't set has_* so alsa_open can identify it wasn't set by the user */
     if (!dev->u.alsa.out->has_period_length) {
-        /* 1024 frames assuming 44100Hz */
-        dev->u.alsa.out->period_length = 1024 * 1000000 / 44100;
+        /* 256 frames assuming 44100Hz */
+        dev->u.alsa.out->period_length = 5805;
     }
     if (!dev->u.alsa.out->has_buffer_length) {
         /* 4096 frames assuming 44100Hz */
-        dev->u.alsa.out->buffer_length = 4096ll * 1000000 / 44100;
+        dev->u.alsa.out->buffer_length = 92880;
     }
 
-    /*
-     * OptsVisitor sets unspecified optional fields to zero, but do not depend
-     * on it...
-     */
     if (!dev->u.alsa.in->has_period_length) {
-        dev->u.alsa.in->period_length = 0;
+        /* 256 frames assuming 44100Hz */
+        dev->u.alsa.in->period_length = 5805;
     }
     if (!dev->u.alsa.in->has_buffer_length) {
-        dev->u.alsa.in->buffer_length = 0;
+        /* 4096 frames assuming 44100Hz */
+        dev->u.alsa.in->buffer_length = 92880;
     }
 
     return dev;
@@ -923,12 +943,14 @@ static struct audio_pcm_ops alsa_pcm_ops = {
     .init_out = alsa_init_out,
     .fini_out = alsa_fini_out,
     .write    = alsa_write,
+    .buffer_get_free = alsa_buffer_get_free,
     .run_buffer_out = audio_generic_run_buffer_out,
     .enable_out = alsa_enable_out,
 
     .init_in  = alsa_init_in,
     .fini_in  = alsa_fini_in,
     .read     = alsa_read,
+    .run_buffer_in = audio_generic_run_buffer_in,
     .enable_in = alsa_enable_in,
 };
 
@@ -938,7 +960,6 @@ static struct audio_driver alsa_audio_driver = {
     .init           = alsa_audio_init,
     .fini           = alsa_audio_fini,
     .pcm_ops        = &alsa_pcm_ops,
-    .can_be_default = 1,
     .max_voices_out = INT_MAX,
     .max_voices_in  = INT_MAX,
     .voice_size_out = sizeof (ALSAVoiceOut),