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Diffstat (limited to 'audio/alsaaudio.c')
-rw-r--r--audio/alsaaudio.c86
1 files changed, 49 insertions, 37 deletions
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index 133690576e..65a0a0d2eb 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -98,7 +98,7 @@ struct alsa_params_obt {
audfmt_e fmt;
int nchannels;
int can_pause;
- snd_pcm_uframes_t buffer_size;
+ snd_pcm_uframes_t samples;
};
static void GCC_FMT_ATTR (2, 3) alsa_logerr (int err, const char *fmt, ...)
@@ -121,7 +121,7 @@ static void GCC_FMT_ATTR (3, 4) alsa_logerr2 (
{
va_list ap;
- AUD_log (AUDIO_CAP, "Can not initialize %s\n", typ);
+ AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ);
va_start (ap, fmt);
AUD_vlog (AUDIO_CAP, fmt, ap);
@@ -209,7 +209,7 @@ static int alsa_to_audfmt (int alsafmt, audfmt_e *fmt, int *endianness)
return 0;
}
-#ifdef DEBUG_MISMATCHES
+#if defined DEBUG_MISMATCHES || defined DEBUG
static void alsa_dump_info (struct alsa_params_req *req,
struct alsa_params_obt *obt)
{
@@ -221,7 +221,7 @@ static void alsa_dump_info (struct alsa_params_req *req,
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);
+ dolog ("obtained: samples %ld\n", obt->samples);
}
#endif
@@ -234,14 +234,14 @@ static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
err = snd_pcm_sw_params_current (handle, sw_params);
if (err < 0) {
- dolog ("Can not fully initialize DAC\n");
+ dolog ("Could not fully initialize DAC\n");
alsa_logerr (err, "Failed to get current software parameters\n");
return;
}
err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold);
if (err < 0) {
- dolog ("Can not fully initialize DAC\n");
+ dolog ("Could not fully initialize DAC\n");
alsa_logerr (err, "Failed to set software threshold to %ld\n",
threshold);
return;
@@ -249,7 +249,7 @@ static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
err = snd_pcm_sw_params (handle, sw_params);
if (err < 0) {
- dolog ("Can not fully initialize DAC\n");
+ dolog ("Could not fully initialize DAC\n");
alsa_logerr (err, "Failed to set software parameters\n");
return;
}
@@ -344,7 +344,8 @@ static int alsa_open (int in, struct alsa_params_req *req,
handle,
hw_params,
&period_size,
- 0);
+ 0
+ );
if (err < 0) {
alsa_logerr2 (err, typ,
"Failed to set period time %d\n",
@@ -357,7 +358,8 @@ static int alsa_open (int in, struct alsa_params_req *req,
handle,
hw_params,
&buffer_size,
- 0);
+ 0
+ );
if (err < 0) {
alsa_logerr2 (err, typ,
@@ -382,7 +384,7 @@ static int alsa_open (int in, struct alsa_params_req *req,
if (err < 0) {
alsa_logerr (
err,
- "Can not get minmal period size for %s\n",
+ "Could not get minmal period size for %s\n",
typ
);
}
@@ -419,7 +421,7 @@ static int alsa_open (int in, struct alsa_params_req *req,
&minval
);
if (err < 0) {
- alsa_logerr (err, "Can not get minmal buffer size for %s\n",
+ alsa_logerr (err, "Could not get minmal buffer size for %s\n",
typ);
}
else {
@@ -451,7 +453,7 @@ static int alsa_open (int in, struct alsa_params_req *req,
}
}
else {
- dolog ("warning: buffer size is not set\n");
+ dolog ("warning: Buffer size is not set\n");
}
err = snd_pcm_hw_params (handle, hw_params);
@@ -468,13 +470,13 @@ static int alsa_open (int in, struct alsa_params_req *req,
err = snd_pcm_prepare (handle);
if (err < 0) {
- alsa_logerr2 (err, typ, "Can not prepare handle %p\n", handle);
+ alsa_logerr2 (err, typ, "Could 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);
+ alsa_logerr (err, "Could not get pause capability for %s\n", typ);
obt->can_pause = 0;
}
@@ -493,17 +495,17 @@ static int alsa_open (int in, struct alsa_params_req *req,
obt->fmt = req->fmt;
obt->nchannels = nchannels;
obt->freq = freq;
- obt->buffer_size = snd_pcm_frames_to_bytes (handle, obt_buffer_size);
+ obt->samples = obt_buffer_size;
*handlep = handle;
+#if defined DEBUG_MISMATCHES || defined DEBUG
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
}
+#endif
#ifdef DEBUG
alsa_dump_info (req, obt);
@@ -550,7 +552,7 @@ static int alsa_run_out (HWVoiceOut *hw)
}
}
- alsa_logerr (avail, "Can not get amount free space\n");
+ alsa_logerr (avail, "Could not get amount free space\n");
return 0;
}
@@ -618,7 +620,7 @@ static void alsa_fini_out (HWVoiceOut *hw)
}
}
-static int alsa_init_out (HWVoiceOut *hw, int freq, int nchannels, audfmt_e fmt)
+static int alsa_init_out (HWVoiceOut *hw, audsettings_t *as)
{
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
struct alsa_params_req req;
@@ -627,10 +629,11 @@ static int alsa_init_out (HWVoiceOut *hw, int freq, int nchannels, audfmt_e fmt)
int endianness;
int err;
snd_pcm_t *handle;
+ audsettings_t obt_as;
- req.fmt = aud_to_alsafmt (fmt);
- req.freq = freq;
- req.nchannels = nchannels;
+ req.fmt = aud_to_alsafmt (as->fmt);
+ req.freq = as->freq;
+ req.nchannels = as->nchannels;
req.period_size = conf.period_size_out;
req.buffer_size = conf.buffer_size_out;
@@ -644,18 +647,22 @@ static int alsa_init_out (HWVoiceOut *hw, int freq, int nchannels, audfmt_e fmt)
return -1;
}
+ obt_as.freq = obt.freq;
+ obt_as.nchannels = obt.nchannels;
+ obt_as.fmt = effective_fmt;
+
audio_pcm_init_info (
&hw->info,
- obt.freq,
- obt.nchannels,
- effective_fmt,
+ &obt_as,
audio_need_to_swap_endian (endianness)
);
alsa->can_pause = obt.can_pause;
- hw->bufsize = obt.buffer_size;
+ hw->samples = obt.samples;
- alsa->pcm_buf = qemu_mallocz (hw->bufsize);
+ alsa->pcm_buf = audio_calloc (AUDIO_FUNC, obt.samples, 1 << hw->info.shift);
if (!alsa->pcm_buf) {
+ dolog ("Could not allocate DAC buffer (%d bytes)\n",
+ hw->samples << hw->info.shift);
alsa_anal_close (&handle);
return -1;
}
@@ -703,8 +710,7 @@ static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...)
return 0;
}
-static int alsa_init_in (HWVoiceIn *hw,
- int freq, int nchannels, audfmt_e fmt)
+static int alsa_init_in (HWVoiceIn *hw, audsettings_t *as)
{
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
struct alsa_params_req req;
@@ -713,10 +719,11 @@ static int alsa_init_in (HWVoiceIn *hw,
int err;
audfmt_e effective_fmt;
snd_pcm_t *handle;
+ audsettings_t obt_as;
- req.fmt = aud_to_alsafmt (fmt);
- req.freq = freq;
- req.nchannels = nchannels;
+ req.fmt = aud_to_alsafmt (as->fmt);
+ req.freq = as->freq;
+ req.nchannels = as->nchannels;
req.period_size = conf.period_size_in;
req.buffer_size = conf.buffer_size_in;
@@ -730,17 +737,22 @@ static int alsa_init_in (HWVoiceIn *hw,
return -1;
}
+ obt_as.freq = obt.freq;
+ obt_as.nchannels = obt.nchannels;
+ obt_as.fmt = effective_fmt;
+
audio_pcm_init_info (
&hw->info,
- obt.freq,
- obt.nchannels,
- effective_fmt,
+ &obt_as,
audio_need_to_swap_endian (endianness)
);
alsa->can_pause = obt.can_pause;
- hw->bufsize = obt.buffer_size;
- alsa->pcm_buf = qemu_mallocz (hw->bufsize);
+ hw->samples = obt.samples;
+
+ alsa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
if (!alsa->pcm_buf) {
+ dolog ("Could not allocate ADC buffer (%d bytes)\n",
+ hw->samples << hw->info.shift);
alsa_anal_close (&handle);
return -1;
}
generated by cgit v1.2.3 (git 2.26.2) at 2024-12-02 20:14:22 +0000