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-rw-r--r--Makefile.am9
-rw-r--r--NEWS21
-rw-r--r--configure.ac12
-rw-r--r--src/AudioCompress/compress.c28
-rw-r--r--src/audio_format.h6
-rw-r--r--src/audio_parser.c1
-rw-r--r--src/command.c2
-rw-r--r--src/decoder/audiofile_decoder_plugin.c2
-rw-r--r--src/decoder/gme_decoder_plugin.c3
-rw-r--r--src/encoder/flac_encoder.c6
-rw-r--r--src/encoder/vorbis_encoder.c2
-rw-r--r--src/encoder/wave_encoder.c4
-rw-r--r--src/mixer/winmm_mixer_plugin.c10
-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
-rw-r--r--src/pcm_byteswap.c2
-rw-r--r--src/pipe.h2
-rw-r--r--src/update_walk.c7
24 files changed, 128 insertions, 43 deletions
diff --git a/Makefile.am b/Makefile.am
index 7b41d1c75..64c474358 100644
--- a/Makefile.am
+++ b/Makefile.am
@@ -886,6 +886,7 @@ test_run_input_LDADD = $(MPD_LIBS) \
$(INPUT_LIBS) \
$(GLIB_LIBS)
test_run_input_SOURCES = test/run_input.c \
+ test/stdbin.h \
src/conf.c src/tokenizer.c src/utils.c src/string_util.c\
src/tag.c src/tag_pool.c src/tag_save.c \
src/fd_util.c \
@@ -933,6 +934,7 @@ test_run_decoder_LDADD = $(MPD_LIBS) \
$(INPUT_LIBS) $(DECODER_LIBS) \
$(GLIB_LIBS)
test_run_decoder_SOURCES = test/run_decoder.c \
+ test/stdbin.h \
src/conf.c src/tokenizer.c src/utils.c src/string_util.c src/log.c \
src/tag.c src/tag_pool.c \
src/replay_gain_info.c \
@@ -973,6 +975,7 @@ test_run_filter_LDADD = $(MPD_LIBS) \
$(SAMPLERATE_LIBS) \
$(GLIB_LIBS)
test_run_filter_SOURCES = test/run_filter.c \
+ test/stdbin.h \
src/filter_plugin.c \
src/filter_registry.c \
src/conf.c src/tokenizer.c src/utils.c src/string_util.c \
@@ -995,6 +998,7 @@ endif
if ENABLE_ENCODER
noinst_PROGRAMS += test/run_encoder
test_run_encoder_SOURCES = test/run_encoder.c \
+ test/stdbin.h \
src/conf.c src/tokenizer.c \
src/utils.c src/string_util.c \
src/tag.c src/tag_pool.c \
@@ -1002,12 +1006,15 @@ test_run_encoder_SOURCES = test/run_encoder.c \
src/audio_format.c \
src/audio_parser.c \
$(ENCODER_SRC)
+test_run_encoder_CPPFLAGS = $(AM_CPPFLAGS) \
+ $(ENCODER_CFLAGS)
test_run_encoder_LDADD = $(MPD_LIBS) \
$(ENCODER_LIBS) \
$(GLIB_LIBS)
endif
test_software_volume_SOURCES = test/software_volume.c \
+ test/stdbin.h \
src/audio_check.c \
src/audio_parser.c \
src/pcm_volume.c
@@ -1015,6 +1022,7 @@ test_software_volume_LDADD = \
$(GLIB_LIBS)
test_run_normalize_SOURCES = test/run_normalize.c \
+ test/stdbin.h \
src/audio_check.c \
src/audio_parser.c \
src/AudioCompress/compress.c
@@ -1052,6 +1060,7 @@ test_run_output_LDADD = $(MPD_LIBS) \
$(OUTPUT_LIBS) \
$(GLIB_LIBS)
test_run_output_SOURCES = test/run_output.c \
+ test/stdbin.h \
src/conf.c src/tokenizer.c src/utils.c src/string_util.c src/log.c \
src/audio_check.c \
src/audio_format.c \
diff --git a/NEWS b/NEWS
index 4bf0dc882..5b06a39a9 100644
--- a/NEWS
+++ b/NEWS
@@ -15,6 +15,21 @@ ver 0.17 (2011/??/??)
* state_file: add option "restore_paused"
+ver 0.16.2 (2011/03/18)
+* configure.ac:
+ - fix bashism in tremor test
+* decoder:
+ - tremor: fix configure test
+ - gme: detect end of song
+* encoder:
+ - vorbis: reset the Ogg stream after flush
+* output:
+ - httpd: fix uninitialized variable
+ - httpd: include sys/socket.h
+ - oss: AFMT_S24_PACKED is little-endian
+ - oss: disable 24 bit playback on FreeBSD
+
+
ver 0.16.1 (2011/01/09)
* audio_check: fix parameter in prototype
* add void casts to suppress "result unused" warnings (clang)
@@ -145,9 +160,13 @@ ver 0.16 (2010/12/11)
* make single mode 'sticky'
-ver 0.15.16 (2010/??/??)
+ver 0.15.16 (2011/03/13)
+* output:
+ - ao: initialize the ao_sample_format struct
+ - jack: fix crash with mono playback
* encoders:
- lame: explicitly configure the output sample rate
+* update: log all file permission problems
ver 0.15.15 (2010/11/08)
diff --git a/configure.ac b/configure.ac
index 0d07555e0..2142bcfc1 100644
--- a/configure.ac
+++ b/configure.ac
@@ -660,7 +660,7 @@ fi
AM_CONDITIONAL(ENABLE_CDIO_PARANOIA, test x$enable_cdio_paranoia = xyes)
dnl ---------------------------------- libogg ---------------------------------
-if test x$with_tremor == xno || test -z $with_tremor; then
+if test x$with_tremor = xno || test -z $with_tremor; then
PKG_CHECK_MODULES(OGG, [ogg], enable_ogg=yes, enable_ogg=no)
fi
@@ -959,13 +959,19 @@ if test x$enable_tremor = xyes; then
ac_save_LIBS="$LIBS"
CFLAGS="$CFLAGS $TREMOR_CFLAGS"
LIBS="$LIBS $TREMOR_LIBS"
- AC_CHECK_LIB(vorbisidec,ov_read,enable_vorbis=yes,enable_vorbis=no;
+ AC_CHECK_LIB(vorbisidec,ov_read,,enable_tremor=no;
AC_MSG_WARN([vorbisidec lib needed for ogg support with tremor -- disabling ogg support]))
CFLAGS="$ac_save_CFLAGS"
LIBS="$ac_save_LIBS"
+fi
+if test x$enable_tremor = xyes; then
AC_DEFINE(HAVE_TREMOR,1,
[Define to use tremor (libvorbisidec) for ogg support])
+ AC_DEFINE(ENABLE_VORBIS_DECODER, 1, [Define for Ogg Vorbis support]),
+else
+ TREMOR_CFLAGS=
+ TREMOR_LIBS=
fi
AC_SUBST(TREMOR_CFLAGS)
@@ -1005,7 +1011,7 @@ if test x$enable_vorbis = xyes; then
fi
fi
-AM_CONDITIONAL(ENABLE_VORBIS_DECODER, test x$enable_vorbis = xyes)
+AM_CONDITIONAL(ENABLE_VORBIS_DECODER, test x$enable_vorbis = xyes || test x$enable_tremor = xyes)
dnl --------------------------------- sidplay ---------------------------------
found_sidplay=$HAVE_CXX
diff --git a/src/AudioCompress/compress.c b/src/AudioCompress/compress.c
index d5c08372c..36cdfd8dd 100644
--- a/src/AudioCompress/compress.c
+++ b/src/AudioCompress/compress.c
@@ -16,16 +16,16 @@
struct Compressor {
//! The compressor's preferences
struct CompressorConfig prefs;
-
+
//! History of the peak values
int *peaks;
-
+
//! History of the gain values
int *gain;
-
+
//! History of clip amounts
int *clipped;
-
+
unsigned int pos;
unsigned int bufsz;
};
@@ -41,9 +41,9 @@ struct Compressor *Compressor_new(unsigned int history)
obj->peaks = obj->gain = obj->clipped = NULL;
obj->bufsz = 0;
obj->pos = 0;
-
+
Compressor_setHistory(obj, history);
-
+
return obj;
}
@@ -70,7 +70,7 @@ void Compressor_setHistory(struct Compressor *obj, unsigned int history)
{
if (!history)
history = BUCKETS;
-
+
obj->peaks = resizeArray(obj->peaks, history, obj->bufsz);
obj->gain = resizeArray(obj->gain, history, obj->bufsz);
obj->clipped = resizeArray(obj->clipped, history, obj->bufsz);
@@ -82,7 +82,7 @@ struct CompressorConfig *Compressor_getConfig(struct Compressor *obj)
return &obj->prefs;
}
-void Compressor_Process_int16(struct Compressor *obj, int16_t *audio,
+void Compressor_Process_int16(struct Compressor *obj, int16_t *audio,
unsigned int count)
{
struct CompressorConfig *prefs = Compressor_getConfig(obj);
@@ -97,7 +97,7 @@ void Compressor_Process_int16(struct Compressor *obj, int16_t *audio,
int *clipped = obj->clipped + slot;
unsigned int ramp = count;
int delta;
-
+
ap = audio;
for (i = 0; i < count; i++)
{
@@ -124,15 +124,15 @@ void Compressor_Process_int16(struct Compressor *obj, int16_t *audio,
//! Determine target gain
newGain = (1 << 10)*prefs->target/peakVal;
-
+
//! Adjust the gain with inertia from the previous gain value
- newGain = (curGain*((1 << prefs->smooth) - 1) + newGain)
+ newGain = (curGain*((1 << prefs->smooth) - 1) + newGain)
>> prefs->smooth;
-
+
//! Make sure it's no more than the maximum gain value
if (newGain > (prefs->maxgain << 10))
newGain = prefs->maxgain << 10;
-
+
//! Make sure it's no less than 1:1
if (newGain < (1 << 10))
newGain = 1 << 10;
@@ -144,7 +144,7 @@ void Compressor_Process_int16(struct Compressor *obj, int16_t *audio,
//! Truncate the ramp time
ramp = peakPos;
}
-
+
//! Record the new gain
obj->gain[slot] = newGain;
diff --git a/src/audio_format.h b/src/audio_format.h
index 340e4987d..1a54a092a 100644
--- a/src/audio_format.h
+++ b/src/audio_format.h
@@ -22,6 +22,7 @@
#include <stdint.h>
#include <stdbool.h>
+#include <assert.h>
enum sample_format {
SAMPLE_FORMAT_UNDEFINED = 0,
@@ -219,6 +220,9 @@ static inline void
audio_format_mask_apply(struct audio_format *af,
const struct audio_format *mask)
{
+ assert(audio_format_valid(af));
+ assert(audio_format_mask_valid(mask));
+
if (mask->sample_rate != 0)
af->sample_rate = mask->sample_rate;
@@ -227,6 +231,8 @@ audio_format_mask_apply(struct audio_format *af,
if (mask->channels != 0)
af->channels = mask->channels;
+
+ assert(audio_format_valid(af));
}
/**
diff --git a/src/audio_parser.c b/src/audio_parser.c
index bde48828e..1138c9588 100644
--- a/src/audio_parser.c
+++ b/src/audio_parser.c
@@ -192,6 +192,7 @@ audio_format_parse(struct audio_format *dest, const char *src,
}
audio_format_init(dest, rate, sample_format, channels);
+ assert(audio_format_valid(dest));
return true;
}
diff --git a/src/command.c b/src/command.c
index 03d48a38a..bbfc84359 100644
--- a/src/command.c
+++ b/src/command.c
@@ -763,7 +763,7 @@ handle_load(struct client *client, G_GNUC_UNUSED int argc, char *argv[])
result = playlist_open_into_queue(argv[1], &g_playlist,
client->player_control, true);
if (result != PLAYLIST_RESULT_NO_SUCH_LIST)
- return result;
+ return print_playlist_result(client, result);
result = playlist_load_spl(&g_playlist, client->player_control,
argv[1]);
diff --git a/src/decoder/audiofile_decoder_plugin.c b/src/decoder/audiofile_decoder_plugin.c
index 6f9d0dd20..c862168f8 100644
--- a/src/decoder/audiofile_decoder_plugin.c
+++ b/src/decoder/audiofile_decoder_plugin.c
@@ -244,7 +244,7 @@ static const char *const audiofile_suffixes[] = {
static const char *const audiofile_mime_types[] = {
"audio/x-wav",
"audio/x-aiff",
- NULL
+ NULL
};
const struct decoder_plugin audiofile_decoder_plugin = {
diff --git a/src/decoder/gme_decoder_plugin.c b/src/decoder/gme_decoder_plugin.c
index 4a5220a3f..e14a52d32 100644
--- a/src/decoder/gme_decoder_plugin.c
+++ b/src/decoder/gme_decoder_plugin.c
@@ -153,6 +153,9 @@ gme_file_decode(struct decoder *decoder, const char *path_fs)
if((gme_err = gme_start_track(emu, song_num)) != NULL)
g_warning("%s", gme_err);
+ if(ti->length > 0)
+ gme_set_fade(emu, ti->length);
+
/* play */
do {
gme_err = gme_play(emu, GME_BUFFER_SAMPLES, buf);
diff --git a/src/encoder/flac_encoder.c b/src/encoder/flac_encoder.c
index 4312fedb2..a7d35597e 100644
--- a/src/encoder/flac_encoder.c
+++ b/src/encoder/flac_encoder.c
@@ -55,7 +55,7 @@ static bool
flac_encoder_configure(struct flac_encoder *encoder,
const struct config_param *param, G_GNUC_UNUSED GError **error)
{
- encoder->compression = config_get_block_unsigned(param,
+ encoder->compression = config_get_block_unsigned(param,
"compression", 5);
return true;
@@ -218,7 +218,7 @@ flac_encoder_open(struct encoder *_encoder, struct audio_format *audio_format,
if (init_status != FLAC__STREAM_ENCODER_OK) {
g_set_error(error, flac_encoder_quark(), 0,
- "failed to initialize encoder: %s\n",
+ "failed to initialize encoder: %s\n",
FLAC__StreamEncoderStateString[init_status]);
flac_encoder_close(_encoder);
return false;
@@ -234,7 +234,7 @@ flac_encoder_open(struct encoder *_encoder, struct audio_format *audio_format,
if(init_status != FLAC__STREAM_ENCODER_INIT_STATUS_OK) {
g_set_error(error, flac_encoder_quark(), 0,
- "failed to initialize encoder: %s\n",
+ "failed to initialize encoder: %s\n",
FLAC__StreamEncoderInitStatusString[init_status]);
flac_encoder_close(_encoder);
return false;
diff --git a/src/encoder/vorbis_encoder.c b/src/encoder/vorbis_encoder.c
index 94816a7ac..5b3110988 100644
--- a/src/encoder/vorbis_encoder.c
+++ b/src/encoder/vorbis_encoder.c
@@ -276,6 +276,8 @@ vorbis_encoder_flush(struct encoder *_encoder, G_GNUC_UNUSED GError **error)
vorbis_analysis_init(&encoder->vd, &encoder->vi);
vorbis_block_init(&encoder->vd, &encoder->vb);
+ ogg_stream_reset(&encoder->os);
+
encoder->flush = true;
return true;
}
diff --git a/src/encoder/wave_encoder.c b/src/encoder/wave_encoder.c
index 7f1d4fcdf..6ebacab7d 100644
--- a/src/encoder/wave_encoder.c
+++ b/src/encoder/wave_encoder.c
@@ -58,7 +58,7 @@ wave_encoder_quark(void)
}
static void
-fill_wave_header(struct wave_header *header, int channels, int bits,
+fill_wave_header(struct wave_header *header, int channels, int bits,
int freq, int block_size)
{
int data_size = 0x0FFFFFFF;
@@ -142,7 +142,7 @@ wave_encoder_open(struct encoder *_encoder,
buffer = pcm_buffer_get(&encoder->buffer, sizeof(struct wave_header) );
/* create PCM wave header in initial buffer */
- fill_wave_header((struct wave_header *) buffer,
+ fill_wave_header((struct wave_header *) buffer,
audio_format->channels,
encoder->bits,
audio_format->sample_rate,
diff --git a/src/mixer/winmm_mixer_plugin.c b/src/mixer/winmm_mixer_plugin.c
index 174c1ec04..ceddf6afd 100644
--- a/src/mixer/winmm_mixer_plugin.c
+++ b/src/mixer/winmm_mixer_plugin.c
@@ -58,11 +58,11 @@ winmm_mixer_init(void *ao, G_GNUC_UNUSED const struct config_param *param,
G_GNUC_UNUSED GError **error_r)
{
assert(ao != NULL);
-
+
struct winmm_mixer *wm = g_new(struct winmm_mixer, 1);
mixer_init(&wm->base, &winmm_mixer_plugin);
wm->output = (struct winmm_output *) ao;
-
+
return &wm->base;
}
@@ -79,13 +79,13 @@ winmm_mixer_get_volume(struct mixer *mixer, GError **error_r)
DWORD volume;
HWAVEOUT handle = winmm_output_get_handle(wm->output);
MMRESULT result = waveOutGetVolume(handle, &volume);
-
+
if (result != MMSYSERR_NOERROR) {
g_set_error(error_r, 0, winmm_mixer_quark(),
"Failed to get winmm volume");
return -1;
}
-
+
return winmm_volume_decode(volume);
}
@@ -102,7 +102,7 @@ winmm_mixer_set_volume(struct mixer *mixer, unsigned volume, GError **error_r)
"Failed to set winmm volume");
return false;
}
-
+
return true;
}
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);
diff --git a/src/pcm_byteswap.c b/src/pcm_byteswap.c
index 9aabe592f..4e604c79a 100644
--- a/src/pcm_byteswap.c
+++ b/src/pcm_byteswap.c
@@ -49,7 +49,7 @@ const int16_t *pcm_byteswap_16(struct pcm_buffer *buffer,
static inline uint32_t swab32(uint32_t x)
{
- return (x << 24) |
+ return (x << 24) |
((x & 0xff00) << 8) |
((x & 0xff0000) >> 8) |
(x >> 24);
diff --git a/src/pipe.h b/src/pipe.h
index 512e7bf4f..785c1b943 100644
--- a/src/pipe.h
+++ b/src/pipe.h
@@ -20,9 +20,9 @@
#ifndef MPD_PIPE_H
#define MPD_PIPE_H
-#ifndef NDEBUG
#include <stdbool.h>
+#ifndef NDEBUG
struct audio_format;
#endif
diff --git a/src/update_walk.c b/src/update_walk.c
index a3272e618..e5ab4fc37 100644
--- a/src/update_walk.c
+++ b/src/update_walk.c
@@ -300,6 +300,9 @@ stat_directory(const struct directory *directory, struct stat *st)
if (path_fs == NULL)
return -1;
ret = stat(path_fs, st);
+ if (ret < 0)
+ g_warning("Failed to stat %s: %s", path_fs, g_strerror(errno));
+
g_free(path_fs);
return ret;
}
@@ -316,6 +319,9 @@ stat_directory_child(const struct directory *parent, const char *name,
return -1;
ret = stat(path_fs, st);
+ if (ret < 0)
+ g_warning("Failed to stat %s: %s", path_fs, g_strerror(errno));
+
g_free(path_fs);
return ret;
}
@@ -557,6 +563,7 @@ directory_child_access(const struct directory *directory,
/* access() is useless on WIN32 */
(void)directory;
(void)name;
+ (void)mode;
return true;
#else
char *path = map_directory_child_fs(directory, name);
generated by cgit v1.2.3 (git 2.25.1) at 2024-12-23 09:06:50 +0000