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-rw-r--r--src/output/ao_plugin.c5
-rw-r--r--src/output/httpd_internal.h2
-rw-r--r--src/output/httpd_output_plugin.c2
-rw-r--r--src/output/jack_output_plugin.c14
-rw-r--r--src/output/mvp_plugin.c2
-rw-r--r--src/output/oss_plugin.c23
-rw-r--r--src/output_control.c1
-rw-r--r--src/output_thread.c5
8 files changed, 43 insertions, 11 deletions
diff --git a/src/output/ao_plugin.c b/src/output/ao_plugin.c
index 21d60eb58..71e06bad9 100644
--- a/src/output/ao_plugin.c
+++ b/src/output/ao_plugin.c
@@ -26,6 +26,9 @@
#undef G_LOG_DOMAIN
#define G_LOG_DOMAIN "ao"
+/* An ao_sample_format, with all fields set to zero: */
+static const ao_sample_format OUR_AO_FORMAT_INITIALIZER;
+
static unsigned ao_output_ref;
struct ao_data {
@@ -167,7 +170,7 @@ static bool
ao_output_open(void *data, struct audio_format *audio_format,
GError **error)
{
- ao_sample_format format;
+ ao_sample_format format = OUR_AO_FORMAT_INITIALIZER;
struct ao_data *ad = (struct ao_data *)data;
switch (audio_format->format) {
diff --git a/src/output/httpd_internal.h b/src/output/httpd_internal.h
index d9c764bfd..67396b667 100644
--- a/src/output/httpd_internal.h
+++ b/src/output/httpd_internal.h
@@ -111,7 +111,7 @@ struct httpd_output {
char buffer[32768];
/**
- * The maximum and current number of clients connected
+ * The maximum and current number of clients connected
* at the same time.
*/
guint clients_max, clients_cnt;
diff --git a/src/output/httpd_output_plugin.c b/src/output/httpd_output_plugin.c
index bcc27f884..7fde676c5 100644
--- a/src/output/httpd_output_plugin.c
+++ b/src/output/httpd_output_plugin.c
@@ -36,6 +36,7 @@
#include <errno.h>
#ifdef HAVE_LIBWRAP
+#include <sys/socket.h> /* needed for AF_UNIX */
#include <tcpd.h>
#endif
@@ -123,6 +124,7 @@ httpd_output_init(G_GNUC_UNUSED const struct audio_format *audio_format,
/* initialize metadata */
httpd->metadata = NULL;
+ httpd->unflushed_input = 0;
/* initialize encoder */
diff --git a/src/output/jack_output_plugin.c b/src/output/jack_output_plugin.c
index 8ea5cb2fd..4df84fd23 100644
--- a/src/output/jack_output_plugin.c
+++ b/src/output/jack_output_plugin.c
@@ -40,7 +40,7 @@ enum {
MAX_PORTS = 16,
};
-static const size_t sample_size = sizeof(jack_default_audio_sample_t);
+static const size_t jack_sample_size = sizeof(jack_default_audio_sample_t);
struct jack_data {
/**
@@ -103,9 +103,9 @@ mpd_jack_available(const struct jack_data *jd)
min = current;
}
- assert(min % sample_size == 0);
+ assert(min % jack_sample_size == 0);
- return min / sample_size;
+ return min / jack_sample_size;
}
static int
@@ -123,7 +123,7 @@ mpd_jack_process(jack_nframes_t nframes, void *arg)
const jack_nframes_t available = mpd_jack_available(jd);
for (unsigned i = 0; i < jd->audio_format.channels; ++i)
jack_ringbuffer_read_advance(jd->ringbuffer[i],
- available * sample_size);
+ available * jack_sample_size);
/* generate silence while MPD is paused */
@@ -144,7 +144,7 @@ mpd_jack_process(jack_nframes_t nframes, void *arg)
for (unsigned i = 0; i < jd->audio_format.channels; ++i) {
out = jack_port_get_buffer(jd->ports[i], nframes);
jack_ringbuffer_read(jd->ringbuffer[i],
- (char *)out, available * sample_size);
+ (char *)out, available * jack_sample_size);
for (jack_nframes_t f = available; f < nframes; ++f)
/* ringbuffer underrun, fill with silence */
@@ -675,7 +675,7 @@ mpd_jack_play(void *data, const void *chunk, size_t size, GError **error_r)
space = space1;
}
- if (space >= frame_size)
+ if (space >= jack_sample_size)
break;
/* XXX do something more intelligent to
@@ -683,7 +683,7 @@ mpd_jack_play(void *data, const void *chunk, size_t size, GError **error_r)
g_usleep(1000);
}
- space /= sample_size;
+ space /= jack_sample_size;
if (space < size)
size = space;
diff --git a/src/output/mvp_plugin.c b/src/output/mvp_plugin.c
index 64fc77016..be4c8dbc0 100644
--- a/src/output/mvp_plugin.c
+++ b/src/output/mvp_plugin.c
@@ -17,7 +17,7 @@
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
-/*
+/*
* Media MVP audio output based on code from MVPMC project:
* http://mvpmc.sourceforge.net/
*/
diff --git a/src/output/oss_plugin.c b/src/output/oss_plugin.c
index bd3ccb774..d7df594d3 100644
--- a/src/output/oss_plugin.c
+++ b/src/output/oss_plugin.c
@@ -41,6 +41,15 @@
# include <sys/soundcard.h>
#endif /* !(defined(__OpenBSD__) || defined(__NetBSD__) */
+/* We got bug reports from FreeBSD users who said that the two 24 bit
+ formats generate white noise on FreeBSD, but 32 bit works. This is
+ a workaround until we know what exactly is expected by the kernel
+ audio drivers. */
+#ifndef __linux__
+#undef AFMT_S24_PACKED
+#undef AFMT_S24_NE
+#endif
+
struct oss_data {
int fd;
const char *device;
@@ -347,7 +356,7 @@ oss_setup_sample_rate(int fd, struct audio_format *audio_format,
case SUCCESS:
if (!audio_valid_sample_rate(sample_rate))
break;
-
+
audio_format->sample_rate = sample_rate;
return true;
@@ -461,6 +470,12 @@ oss_setup_sample_format(int fd, struct audio_format *audio_format,
break;
audio_format->format = mpd_format;
+
+#ifdef AFMT_S24_PACKED
+ if (oss_format == AFMT_S24_PACKED)
+ audio_format->reverse_endian =
+ G_BYTE_ORDER != G_LITTLE_ENDIAN;
+#endif
return true;
case ERROR:
@@ -502,6 +517,12 @@ oss_setup_sample_format(int fd, struct audio_format *audio_format,
break;
audio_format->format = mpd_format;
+
+#ifdef AFMT_S24_PACKED
+ if (oss_format == AFMT_S24_PACKED)
+ audio_format->reverse_endian =
+ G_BYTE_ORDER != G_LITTLE_ENDIAN;
+#endif
return true;
case ERROR:
diff --git a/src/output_control.c b/src/output_control.c
index d4bfc241c..8efe08f05 100644
--- a/src/output_control.c
+++ b/src/output_control.c
@@ -139,6 +139,7 @@ audio_output_open(struct audio_output *ao,
{
bool open;
+ assert(audio_format_valid(audio_format));
assert(mp != NULL);
if (ao->fail_timer != NULL) {
diff --git a/src/output_thread.c b/src/output_thread.c
index 7d47baf46..21096eb04 100644
--- a/src/output_thread.c
+++ b/src/output_thread.c
@@ -96,6 +96,8 @@ ao_filter_open(struct audio_output *ao,
struct audio_format *audio_format,
GError **error_r)
{
+ assert(audio_format_valid(audio_format));
+
/* the replay_gain filter cannot fail here */
if (ao->replay_gain_filter != NULL)
filter_open(ao->replay_gain_filter, audio_format, error_r);
@@ -137,6 +139,7 @@ ao_open(struct audio_output *ao)
assert(!ao->open);
assert(ao->pipe != NULL);
assert(ao->chunk == NULL);
+ assert(audio_format_valid(&ao->in_audio_format));
if (ao->fail_timer != NULL) {
/* this can only happen when this
@@ -165,6 +168,8 @@ ao_open(struct audio_output *ao)
return;
}
+ assert(audio_format_valid(filter_audio_format));
+
ao->out_audio_format = *filter_audio_format;
audio_format_mask_apply(&ao->out_audio_format,
&ao->config_audio_format);
generated by cgit v1.2.3 (git 2.25.1) at 2024-12-23 15:51:38 +0000