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-rw-r--r--src/output/alsa_plugin.c257
1 files changed, 206 insertions, 51 deletions
diff --git a/src/output/alsa_plugin.c b/src/output/alsa_plugin.c
index 818c83ca2..8c36e32bd 100644
--- a/src/output/alsa_plugin.c
+++ b/src/output/alsa_plugin.c
@@ -1,5 +1,5 @@
/*
- * Copyright (C) 2003-2009 The Music Player Daemon Project
+ * Copyright (C) 2003-2010 The Music Player Daemon Project
* http://www.musicpd.org
*
* This program is free software; you can redistribute it and/or modify
@@ -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;
};
/**
@@ -172,15 +183,148 @@ alsa_test_default_device(void)
}
static snd_pcm_format_t
-get_bitformat(const struct audio_format *af)
+get_bitformat(enum sample_format sample_format)
+{
+ switch (sample_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_S24:
+ return G_BYTE_ORDER == G_BIG_ENDIAN
+ ? SND_PCM_FORMAT_S24_3BE
+ : SND_PCM_FORMAT_S24_3LE;
+
+ case SAMPLE_FORMAT_S32:
+ return SND_PCM_FORMAT_S32;
+
+ default:
+ return SND_PCM_FORMAT_UNKNOWN;
+ }
+}
+
+static snd_pcm_format_t
+byteswap_bitformat(snd_pcm_format_t fmt)
{
- 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(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_S24_3BE:
+ return SND_PCM_FORMAT_S24_3LE;
+
+ case SND_PCM_FORMAT_S24_3LE:
+ return SND_PCM_FORMAT_S24_3BE;
+
+ case SND_PCM_FORMAT_S32_BE: return SND_PCM_FORMAT_S32_LE;
+ default: return SND_PCM_FORMAT_UNKNOWN;
}
- return SND_PCM_FORMAT_UNKNOWN;
+}
+
+/**
+ * Attempts to configure the specified sample format.
+ */
+static int
+alsa_output_try_format(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
+ struct audio_format *audio_format,
+ enum sample_format sample_format)
+{
+ snd_pcm_format_t alsa_format = get_bitformat(sample_format);
+ if (alsa_format == SND_PCM_FORMAT_UNKNOWN)
+ return -EINVAL;
+
+ int err = snd_pcm_hw_params_set_format(pcm, hwparams, alsa_format);
+ if (err == 0)
+ audio_format->format = sample_format;
+
+ return err;
+}
+
+/**
+ * Attempts to configure the specified sample format with reversed
+ * host byte order.
+ */
+static int
+alsa_output_try_reverse(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
+ struct audio_format *audio_format,
+ enum sample_format sample_format)
+{
+ snd_pcm_format_t alsa_format =
+ byteswap_bitformat(get_bitformat(sample_format));
+ if (alsa_format == SND_PCM_FORMAT_UNKNOWN)
+ return -EINVAL;
+
+ int err = snd_pcm_hw_params_set_format(pcm, hwparams, alsa_format);
+ if (err == 0) {
+ audio_format->format = sample_format;
+ audio_format->reverse_endian = true;
+ }
+
+ return err;
+}
+
+/**
+ * Attempts to configure the specified sample format, and tries the
+ * reversed host byte order if was not supported.
+ */
+static int
+alsa_output_try_format_both(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
+ struct audio_format *audio_format,
+ enum sample_format sample_format)
+{
+ int err = alsa_output_try_format(pcm, hwparams, audio_format,
+ sample_format);
+ if (err == -EINVAL)
+ err = alsa_output_try_reverse(pcm, hwparams, audio_format,
+ sample_format);
+
+ return err;
+}
+
+/**
+ * Configure a sample format, and probe other formats if that fails.
+ */
+static int
+alsa_output_setup_format(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
+ struct audio_format *audio_format)
+{
+ /* try the input format first */
+
+ int err = alsa_output_try_format_both(pcm, hwparams, audio_format,
+ audio_format->format);
+ if (err != -EINVAL)
+ return err;
+
+ /* if unsupported by the hardware, try other formats */
+
+ static const enum sample_format probe_formats[] = {
+ SAMPLE_FORMAT_S24_P32,
+ SAMPLE_FORMAT_S32,
+ SAMPLE_FORMAT_S24,
+ SAMPLE_FORMAT_S16,
+ SAMPLE_FORMAT_S8,
+ SAMPLE_FORMAT_UNDEFINED,
+ };
+
+ for (unsigned i = 0; probe_formats[i] != SAMPLE_FORMAT_UNDEFINED; ++i) {
+ if (probe_formats[i] == audio_format->format)
+ continue;
+
+ err = alsa_output_try_format_both(pcm, hwparams, audio_format,
+ probe_formats[i]);
+ if (err != -EINVAL)
+ return err;
+ }
+
+ return -EINVAL;
}
/**
@@ -189,7 +333,6 @@ get_bitformat(const struct audio_format *af)
*/
static bool
alsa_setup(struct alsa_data *ad, struct audio_format *audio_format,
- snd_pcm_format_t bitformat,
GError **error)
{
snd_pcm_hw_params_t *hwparams;
@@ -208,7 +351,6 @@ 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)
@@ -235,31 +377,12 @@ configure_hw:
ad->writei = snd_pcm_writei;
}
- err = snd_pcm_hw_params_set_format(ad->pcm, hwparams, bitformat);
- if (err == -EINVAL && (audio_format->bits == 24 ||
- audio_format->bits == 16)) {
- /* 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 == -EINVAL && audio_format->bits != 16) {
- /* 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;
- }
- }
-
+ err = alsa_output_setup_format(ad->pcm, hwparams, audio_format);
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 +488,9 @@ 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:
@@ -378,19 +504,9 @@ static bool
alsa_open(void *data, struct audio_format *audio_format, GError **error)
{
struct alsa_data *ad = data;
- snd_pcm_format_t bitformat;
int err;
bool success;
- bitformat = get_bitformat(audio_format);
- if (bitformat == SND_PCM_FORMAT_UNKNOWN) {
- /* sample format is not supported by this plugin -
- fall back to 16 bit samples */
-
- audio_format->bits = 16;
- bitformat = SND_PCM_FORMAT_S16;
- }
-
err = snd_pcm_open(&ad->pcm, alsa_device(ad),
SND_PCM_STREAM_PLAYBACK, ad->mode);
if (err < 0) {
@@ -400,7 +516,7 @@ alsa_open(void *data, struct audio_format *audio_format, GError **error)
return false;
}
- success = alsa_setup(ad, audio_format, bitformat, error);
+ success = alsa_setup(ad, audio_format, error);
if (!success) {
snd_pcm_close(ad->pcm);
return false;
@@ -431,6 +547,7 @@ 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 +565,47 @@ 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 +613,6 @@ 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 +625,11 @@ 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 +647,9 @@ 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,
};