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-rw-r--r--src/output/alsa_plugin.c166
1 files changed, 141 insertions, 25 deletions
diff --git a/src/output/alsa_plugin.c b/src/output/alsa_plugin.c
index 818c83ca2..b7325de07 100644
--- a/src/output/alsa_plugin.c
+++ b/src/output/alsa_plugin.c
@@ -17,7 +17,8 @@
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
-#include "../output_api.h"
+#include "config.h"
+#include "output_api.h"
#include "mixer_list.h"
#include <glib.h>
@@ -69,6 +70,16 @@ struct alsa_data {
/** the size of one audio frame */
size_t frame_size;
+
+ /**
+ * The size of one period, in number of frames.
+ */
+ snd_pcm_uframes_t period_frames;
+
+ /**
+ * The number of frames written in the current period.
+ */
+ snd_pcm_uframes_t period_position;
};
/**
@@ -174,15 +185,37 @@ alsa_test_default_device(void)
static snd_pcm_format_t
get_bitformat(const struct audio_format *af)
{
- switch (af->bits) {
- case 8: return SND_PCM_FORMAT_S8;
- case 16: return SND_PCM_FORMAT_S16;
- case 24: return SND_PCM_FORMAT_S24;
- case 32: return SND_PCM_FORMAT_S32;
+ switch (af->format) {
+ case SAMPLE_FORMAT_S8:
+ return SND_PCM_FORMAT_S8;
+
+ case SAMPLE_FORMAT_S16:
+ return SND_PCM_FORMAT_S16;
+
+ case SAMPLE_FORMAT_S24_P32:
+ return SND_PCM_FORMAT_S24;
+
+ case SAMPLE_FORMAT_S32:
+ return SND_PCM_FORMAT_S32;
+
+ default:
+ return SND_PCM_FORMAT_UNKNOWN;
}
- return SND_PCM_FORMAT_UNKNOWN;
}
+static snd_pcm_format_t
+byteswap_bitformat(snd_pcm_format_t fmt)
+{
+ switch(fmt) {
+ case SND_PCM_FORMAT_S16_LE: return SND_PCM_FORMAT_S16_BE;
+ case SND_PCM_FORMAT_S24_LE: return SND_PCM_FORMAT_S24_BE;
+ case SND_PCM_FORMAT_S32_LE: return SND_PCM_FORMAT_S32_BE;
+ case SND_PCM_FORMAT_S16_BE: return SND_PCM_FORMAT_S16_LE;
+ case SND_PCM_FORMAT_S24_BE: return SND_PCM_FORMAT_S24_LE;
+ case SND_PCM_FORMAT_S32_BE: return SND_PCM_FORMAT_S32_LE;
+ default: return SND_PCM_FORMAT_UNKNOWN;
+ }
+}
/**
* Set up the snd_pcm_t object which was opened by the caller. Set up
* the configured settings and the audio format.
@@ -208,7 +241,6 @@ alsa_setup(struct alsa_data *ad, struct audio_format *audio_format,
configure_hw:
/* configure HW params */
snd_pcm_hw_params_alloca(&hwparams);
-
cmd = "snd_pcm_hw_params_any";
err = snd_pcm_hw_params_any(ad->pcm, hwparams);
if (err < 0)
@@ -236,30 +268,72 @@ configure_hw:
}
err = snd_pcm_hw_params_set_format(ad->pcm, hwparams, bitformat);
- if (err == -EINVAL && (audio_format->bits == 24 ||
- audio_format->bits == 16)) {
+ if (err == -EINVAL &&
+ byteswap_bitformat(bitformat) != SND_PCM_FORMAT_UNKNOWN) {
+ err = snd_pcm_hw_params_set_format(ad->pcm, hwparams,
+ byteswap_bitformat(bitformat));
+ if (err == 0) {
+ g_debug("ALSA device \"%s\": converting format %s to reverse-endian",
+ alsa_device(ad),
+ sample_format_to_string(audio_format->format));
+ audio_format->reverse_endian = 1;
+ }
+ }
+ if (err == -EINVAL && (audio_format->format == SAMPLE_FORMAT_S24_P32 ||
+ audio_format->format == SAMPLE_FORMAT_S16)) {
/* fall back to 32 bit, let pcm_convert.c do the conversion */
err = snd_pcm_hw_params_set_format(ad->pcm, hwparams,
SND_PCM_FORMAT_S32);
- if (err == 0)
- audio_format->bits = 32;
+ if (err == 0) {
+ g_debug("ALSA device \"%s\": converting format %s to 32 bit\n",
+ alsa_device(ad),
+ sample_format_to_string(audio_format->format));
+ audio_format->format = SAMPLE_FORMAT_S32;
+ }
+ }
+ if (err == -EINVAL && (audio_format->format == SAMPLE_FORMAT_S24_P32 ||
+ audio_format->format == SAMPLE_FORMAT_S16)) {
+ /* fall back to 32 bit, let pcm_convert.c do the conversion */
+ err = snd_pcm_hw_params_set_format(ad->pcm, hwparams,
+ byteswap_bitformat(SND_PCM_FORMAT_S32));
+ if (err == 0) {
+ g_debug("ALSA device \"%s\": converting format %s to 32 bit backward-endian\n",
+ alsa_device(ad),
+ sample_format_to_string(audio_format->format));
+ audio_format->format = SAMPLE_FORMAT_S32;
+ audio_format->reverse_endian = 1;
+ }
}
- if (err == -EINVAL && audio_format->bits != 16) {
+ if (err == -EINVAL && audio_format->format != SAMPLE_FORMAT_S16) {
/* fall back to 16 bit, let pcm_convert.c do the conversion */
err = snd_pcm_hw_params_set_format(ad->pcm, hwparams,
SND_PCM_FORMAT_S16);
if (err == 0) {
- g_debug("ALSA device \"%s\": converting %u bit to 16 bit\n",
- alsa_device(ad), audio_format->bits);
- audio_format->bits = 16;
+ g_debug("ALSA device \"%s\": converting format %s to 16 bit\n",
+ alsa_device(ad),
+ sample_format_to_string(audio_format->format));
+ audio_format->format = SAMPLE_FORMAT_S16;
+ }
+ }
+ if (err == -EINVAL && audio_format->format != SAMPLE_FORMAT_S16) {
+ /* fall back to 16 bit, let pcm_convert.c do the conversion */
+ err = snd_pcm_hw_params_set_format(ad->pcm, hwparams,
+ byteswap_bitformat(SND_PCM_FORMAT_S16));
+ if (err == 0) {
+ g_debug("ALSA device \"%s\": converting format %s to 16 bit backward-endian\n",
+ alsa_device(ad),
+ sample_format_to_string(audio_format->format));
+ audio_format->format = SAMPLE_FORMAT_S16;
+ audio_format->reverse_endian = 1;
}
}
if (err < 0) {
g_set_error(error, alsa_output_quark(), err,
- "ALSA device \"%s\" does not support %u bit audio: %s",
- alsa_device(ad), audio_format->bits,
+ "ALSA device \"%s\" does not support format %s: %s",
+ alsa_device(ad),
+ sample_format_to_string(audio_format->format),
snd_strerror(-err));
return false;
}
@@ -365,6 +439,9 @@ configure_hw:
g_debug("buffer_size=%u period_size=%u",
(unsigned)alsa_buffer_size, (unsigned)alsa_period_size);
+ ad->period_frames = alsa_period_size;
+ ad->period_position = 0;
+
return true;
error:
@@ -387,7 +464,7 @@ alsa_open(void *data, struct audio_format *audio_format, GError **error)
/* sample format is not supported by this plugin -
fall back to 16 bit samples */
- audio_format->bits = 16;
+ audio_format->format = SAMPLE_FORMAT_S16;
bitformat = SND_PCM_FORMAT_S16;
}
@@ -431,6 +508,7 @@ alsa_recover(struct alsa_data *ad, int err)
/* fall-through to snd_pcm_prepare: */
case SND_PCM_STATE_SETUP:
case SND_PCM_STATE_XRUN:
+ ad->period_position = 0;
err = snd_pcm_prepare(ad->pcm);
break;
case SND_PCM_STATE_DISCONNECTED:
@@ -448,11 +526,47 @@ alsa_recover(struct alsa_data *ad, int err)
}
static void
+alsa_drain(void *data)
+{
+ struct alsa_data *ad = data;
+
+ if (snd_pcm_state(ad->pcm) != SND_PCM_STATE_RUNNING)
+ return;
+
+ if (ad->period_position > 0) {
+ /* generate some silence to finish the partial
+ period */
+ snd_pcm_uframes_t nframes =
+ ad->period_frames - ad->period_position;
+ size_t nbytes = nframes * ad->frame_size;
+ void *buffer = g_malloc(nbytes);
+ snd_pcm_hw_params_t *params;
+ snd_pcm_format_t format;
+ unsigned channels;
+
+ snd_pcm_hw_params_alloca(&params);
+ snd_pcm_hw_params_current(ad->pcm, params);
+ snd_pcm_hw_params_get_format(params, &format);
+ snd_pcm_hw_params_get_channels(params, &channels);
+
+ snd_pcm_format_set_silence(format, buffer, nframes * channels);
+ ad->writei(ad->pcm, buffer, nframes);
+ g_free(buffer);
+ }
+
+ snd_pcm_drain(ad->pcm);
+
+ ad->period_position = 0;
+}
+
+static void
alsa_cancel(void *data)
{
struct alsa_data *ad = data;
- alsa_recover(ad, snd_pcm_drop(ad->pcm));
+ ad->period_position = 0;
+
+ snd_pcm_drop(ad->pcm);
}
static void
@@ -460,9 +574,6 @@ alsa_close(void *data)
{
struct alsa_data *ad = data;
- if (snd_pcm_state(ad->pcm) == SND_PCM_STATE_RUNNING)
- snd_pcm_drain(ad->pcm);
-
snd_pcm_close(ad->pcm);
}
@@ -475,8 +586,11 @@ alsa_play(void *data, const void *chunk, size_t size, GError **error)
while (true) {
snd_pcm_sframes_t ret = ad->writei(ad->pcm, chunk, size);
- if (ret > 0)
+ if (ret > 0) {
+ ad->period_position = (ad->period_position + ret)
+ % ad->period_frames;
return ret * ad->frame_size;
+ }
if (ret < 0 && ret != -EAGAIN && ret != -EINTR &&
alsa_recover(ad, ret) < 0) {
@@ -494,7 +608,9 @@ const struct audio_output_plugin alsaPlugin = {
.finish = alsa_finish,
.open = alsa_open,
.play = alsa_play,
+ .drain = alsa_drain,
.cancel = alsa_cancel,
.close = alsa_close,
- .mixer_plugin = &alsa_mixer,
+
+ .mixer_plugin = &alsa_mixer_plugin,
};
generated by cgit v1.2.3 (git 2.25.1) at 2024-08-21 19:50:49 +0000