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Merge commit 'release-0.16.2'

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Conflicts:
	Makefile.am
	NEWS
	configure.ac
This commit is contained in:
Max Kellermann 2011-03-19 09:58:07 +01:00
commit 0c9fc2f809
24 changed files with 128 additions and 43 deletions
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9
Makefile.am
View File

@ -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 \

21
NEWS
View File

@ -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)

12
configure.ac
View File

@ -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

28
src/AudioCompress/compress.c
View File

@ -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;

6
src/audio_format.h
View File

@ -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));
}
/**

1
src/audio_parser.c
View File

@ -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;
}

2
src/command.c
View File

@ -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]);

2
src/decoder/audiofile_decoder_plugin.c
View File

@ -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 = {

3
src/decoder/gme_decoder_plugin.c
View File

@ -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);

6
src/encoder/flac_encoder.c
View File

@ -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;

2
src/encoder/vorbis_encoder.c
View File

@ -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;
}

4
src/encoder/wave_encoder.c
View File

@ -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,

10
src/mixer/winmm_mixer_plugin.c
View File

@ -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;
}

5
src/output/ao_plugin.c
View File

@ -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) {

2
src/output/httpd_internal.h
View File

@ -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;

2
src/output/httpd_output_plugin.c
View File

@ -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 */

14
src/output/jack_output_plugin.c
View File

@ -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;

2
src/output/mvp_plugin.c
View File

@ -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/
*/

23
src/output/oss_plugin.c
View File

@ -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:

1
src/output_control.c
View File

@ -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) {

5
src/output_thread.c
View File

@ -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);

2
src/pcm_byteswap.c
View File

@ -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);

2
src/pipe.h
View File

@ -20,9 +20,9 @@
#ifndef MPD_PIPE_H
#define MPD_PIPE_H
#ifndef NDEBUG
#include <stdbool.h>
#ifndef NDEBUG
struct audio_format;
#endif

7
src/update_walk.c
View File

@ -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);