pthreads-based audio and miscellaneous audio clean-up (malc).
ESD support (malc, Frederick Reeve).
git-svn-id: svn://svn.savannah.nongnu.org/qemu/trunk@3917 c046a42c-6fe2-441c-8c8c-71466251a162
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index 77a08a1..43cfa25 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -86,9 +86,9 @@
};
struct alsa_params_req {
- unsigned int freq;
- audfmt_e fmt;
- unsigned int nchannels;
+ int freq;
+ snd_pcm_format_t fmt;
+ int nchannels;
unsigned int buffer_size;
unsigned int period_size;
};
@@ -96,6 +96,7 @@
struct alsa_params_obt {
int freq;
audfmt_e fmt;
+ int endianness;
int nchannels;
snd_pcm_uframes_t samples;
};
@@ -143,7 +144,7 @@
return audio_pcm_sw_write (sw, buf, len);
}
-static int aud_to_alsafmt (audfmt_e fmt)
+static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt)
{
switch (fmt) {
case AUD_FMT_S8:
@@ -173,7 +174,8 @@
}
}
-static int alsa_to_audfmt (int alsafmt, audfmt_e *fmt, int *endianness)
+static int alsa_to_audfmt (snd_pcm_format_t alsafmt, audfmt_e *fmt,
+ int *endianness)
{
switch (alsafmt) {
case SND_PCM_FORMAT_S8:
@@ -234,7 +236,6 @@
return 0;
}
-#if defined DEBUG_MISMATCHES || defined DEBUG
static void alsa_dump_info (struct alsa_params_req *req,
struct alsa_params_obt *obt)
{
@@ -248,7 +249,6 @@
req->buffer_size, req->period_size);
dolog ("obtained: samples %ld\n", obt->samples);
}
-#endif
static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
{
@@ -291,6 +291,7 @@
unsigned int period_size, buffer_size;
snd_pcm_uframes_t obt_buffer_size;
const char *typ = in ? "ADC" : "DAC";
+ snd_pcm_format_t obtfmt;
freq = req->freq;
period_size = req->period_size;
@@ -327,9 +328,8 @@
}
err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt);
- if (err < 0) {
+ if (err < 0 && conf.verbose) {
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);
@@ -494,6 +494,17 @@
goto err;
}
+ err = snd_pcm_hw_params_get_format (hw_params, &obtfmt);
+ if (err < 0) {
+ alsa_logerr2 (err, typ, "Failed to get format\n");
+ goto err;
+ }
+
+ if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) {
+ dolog ("Invalid format was returned %d\n", obtfmt);
+ goto err;
+ }
+
err = snd_pcm_prepare (handle);
if (err < 0) {
alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle);
@@ -504,28 +515,41 @@
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);
+ bytes_per_sec = freq << (nchannels == 2);
+
+ switch (obt->fmt) {
+ case AUD_FMT_S8:
+ case AUD_FMT_U8:
+ break;
+
+ case AUD_FMT_S16:
+ case AUD_FMT_U16:
+ bytes_per_sec <<= 1;
+ break;
+
+ case AUD_FMT_S32:
+ case AUD_FMT_U32:
+ bytes_per_sec <<= 2;
+ break;
+ }
threshold = (conf.threshold * bytes_per_sec) / 1000;
alsa_set_threshold (handle, threshold);
}
- obt->fmt = req->fmt;
obt->nchannels = nchannels;
obt->freq = freq;
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) {
- dolog ("Audio paramters mismatch for %s\n", typ);
+ if (conf.verbose &&
+ (obt->fmt != req->fmt ||
+ obt->nchannels != req->nchannels ||
+ obt->freq != req->freq)) {
+ dolog ("Audio paramters for %s\n", typ);
alsa_dump_info (req, obt);
}
-#endif
#ifdef DEBUG
alsa_dump_info (req, obt);
@@ -665,9 +689,6 @@
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;
audsettings_t obt_as;
@@ -681,16 +702,10 @@
return -1;
}
- err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness);
- if (err) {
- alsa_anal_close (&handle);
- return -1;
- }
-
obt_as.freq = obt.freq;
obt_as.nchannels = obt.nchannels;
- obt_as.fmt = effective_fmt;
- obt_as.endianness = endianness;
+ obt_as.fmt = obt.fmt;
+ obt_as.endianness = obt.endianness;
audio_pcm_init_info (&hw->info, &obt_as);
hw->samples = obt.samples;
@@ -751,9 +766,6 @@
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;
audsettings_t obt_as;
@@ -767,16 +779,10 @@
return -1;
}
- err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness);
- if (err) {
- alsa_anal_close (&handle);
- return -1;
- }
-
obt_as.freq = obt.freq;
obt_as.nchannels = obt.nchannels;
- obt_as.fmt = effective_fmt;
- obt_as.endianness = endianness;
+ obt_as.fmt = obt.fmt;
+ obt_as.endianness = obt.endianness;
audio_pcm_init_info (&hw->info, &obt_as);
hw->samples = obt.samples;