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Merge remote-tracking branch 'remotes/kraxel/tags/audio-20190821-pull…
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…-request' into staging

audio: second batch of -audiodev support, adding support for multiple backends.

# gpg: Signature made Wed 21 Aug 2019 09:40:37 BST
# gpg:                using RSA key 4CB6D8EED3E87138
# gpg: Good signature from "Gerd Hoffmann (work) <[email protected]>" [full]
# gpg:                 aka "Gerd Hoffmann <[email protected]>" [full]
# gpg:                 aka "Gerd Hoffmann (private) <[email protected]>" [full]
# Primary key fingerprint: A032 8CFF B93A 17A7 9901  FE7D 4CB6 D8EE D3E8 7138

* remotes/kraxel/tags/audio-20190821-pull-request:
  audio: fix memory leak reported by ASAN
  audio: use size_t where makes sense
  audio: remove read and write pcm_ops
  paaudio: fix playback glitches
  audio: do not run each backend in audio_run
  audio: remove audio_MIN, audio_MAX
  paaudio: properly disconnect streams in fini_*
  paaudio: do not move stream when sink/source name is specified
  audio: audiodev= parameters no longer optional when -audiodev present
  paaudio: prepare for multiple audiodev
  audio: add audiodev properties to frontends
  audio: add audiodev property to vnc and wav_capture
  audio: basic support for multi backend audio
  audio: reduce glob_audio_state usage
  audio: Add missing fall through comments

Signed-off-by: Peter Maydell <[email protected]>
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@pm215
pm215 committed Aug 21, 2019
2 parents e65472c + e76ba19 commit 33f18cf
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Showing 38 changed files with 790 additions and 651 deletions.
49 changes: 20 additions & 29 deletions audio/alsaaudio.c
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Original file line number Diff line number Diff line change
Expand Up @@ -39,6 +39,7 @@ struct pollhlp {
struct pollfd *pfds;
int count;
int mask;
AudioState *s;
};

typedef struct ALSAVoiceOut {
Expand Down Expand Up @@ -199,11 +200,11 @@ static void alsa_poll_handler (void *opaque)
break;

case SND_PCM_STATE_PREPARED:
audio_run ("alsa run (prepared)");
audio_run(hlp->s, "alsa run (prepared)");
break;

case SND_PCM_STATE_RUNNING:
audio_run ("alsa run (running)");
audio_run(hlp->s, "alsa run (running)");
break;

default:
Expand Down Expand Up @@ -269,11 +270,6 @@ static int alsa_poll_in (HWVoiceIn *hw)
return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLIN);
}

static int alsa_write (SWVoiceOut *sw, void *buf, int len)
{
return audio_pcm_sw_write (sw, buf, len);
}

static snd_pcm_format_t aud_to_alsafmt (AudioFormat fmt, int endianness)
{
switch (fmt) {
Expand Down Expand Up @@ -634,7 +630,7 @@ static void alsa_write_pending (ALSAVoiceOut *alsa)

while (alsa->pending) {
int left_till_end_samples = hw->samples - alsa->wpos;
int len = audio_MIN (alsa->pending, left_till_end_samples);
int len = MIN (alsa->pending, left_till_end_samples);
char *src = advance (alsa->pcm_buf, alsa->wpos << hw->info.shift);

while (len) {
Expand Down Expand Up @@ -685,10 +681,10 @@ static void alsa_write_pending (ALSAVoiceOut *alsa)
}
}

static int alsa_run_out (HWVoiceOut *hw, int live)
static size_t alsa_run_out(HWVoiceOut *hw, size_t live)
{
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
int decr;
size_t decr;
snd_pcm_sframes_t avail;

avail = alsa_get_avail (alsa->handle);
Expand All @@ -697,7 +693,7 @@ static int alsa_run_out (HWVoiceOut *hw, int live)
return 0;
}

decr = audio_MIN (live, avail);
decr = MIN (live, avail);
decr = audio_pcm_hw_clip_out (hw, alsa->pcm_buf, decr, alsa->pending);
alsa->pending += decr;
alsa_write_pending (alsa);
Expand Down Expand Up @@ -743,12 +739,13 @@ static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,

alsa->pcm_buf = audio_calloc(__func__, obt.samples, 1 << hw->info.shift);
if (!alsa->pcm_buf) {
dolog ("Could not allocate DAC buffer (%d samples, each %d bytes)\n",
hw->samples, 1 << hw->info.shift);
dolog("Could not allocate DAC buffer (%zu samples, each %d bytes)\n",
hw->samples, 1 << hw->info.shift);
alsa_anal_close1 (&handle);
return -1;
}

alsa->pollhlp.s = hw->s;
alsa->handle = handle;
alsa->dev = dev;
return 0;
Expand Down Expand Up @@ -844,12 +841,13 @@ static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)

alsa->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
if (!alsa->pcm_buf) {
dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n",
hw->samples, 1 << hw->info.shift);
dolog("Could not allocate ADC buffer (%zu samples, each %d bytes)\n",
hw->samples, 1 << hw->info.shift);
alsa_anal_close1 (&handle);
return -1;
}

alsa->pollhlp.s = hw->s;
alsa->handle = handle;
alsa->dev = dev;
return 0;
Expand All @@ -865,17 +863,17 @@ static void alsa_fini_in (HWVoiceIn *hw)
alsa->pcm_buf = NULL;
}

static int alsa_run_in (HWVoiceIn *hw)
static size_t 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;
int decr;
size_t live = audio_pcm_hw_get_live_in (hw);
size_t dead = hw->samples - live;
size_t decr;
struct {
int add;
int len;
size_t add;
size_t len;
} bufs[2] = {
{ .add = hw->wpos, .len = 0 },
{ .add = 0, .len = 0 }
Expand Down Expand Up @@ -915,7 +913,7 @@ static int alsa_run_in (HWVoiceIn *hw)
}
}

decr = audio_MIN (dead, avail);
decr = MIN(dead, avail);
if (!decr) {
return 0;
}
Expand Down Expand Up @@ -985,11 +983,6 @@ static int alsa_run_in (HWVoiceIn *hw)
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, ...)
{
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
Expand Down Expand Up @@ -1073,13 +1066,11 @@ static struct audio_pcm_ops alsa_pcm_ops = {
.init_out = alsa_init_out,
.fini_out = alsa_fini_out,
.run_out = alsa_run_out,
.write = alsa_write,
.ctl_out = alsa_ctl_out,

.init_in = alsa_init_in,
.fini_in = alsa_fini_in,
.run_in = alsa_run_in,
.read = alsa_read,
.ctl_in = alsa_ctl_in,
};

Expand Down
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