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40 Commits

Author SHA1 Message Date
Avuton Olrich
4d4b7e3de0 mpd version 0.15.16 2011-03-13 20:27:33 -07:00
Max Kellermann
1674a4ec82 output/jack: fix crash with mono playback
With mono sound, jack_sample_size is smaller than frame_size (4 vs 2
bytes), and "space/jack_sample_size==0".  That means mpd_jack_play()
will return 0, although no error has occurred.
2011-02-27 23:26:50 +01:00
Max Kellermann
ce370bee60 output/jack: rename variable sample_size to jack_sample_size 2011-02-25 10:46:44 +01:00
Christopher Brannon
2a1f4539f6 Insure proper initialization of stack-allocated struct.
Version 1.0.0 of the libao library added a new field to the
ao_sample_format struct.  It is a char * named matrix.  When
an ao_sample_format is allocated on the stack, this field contains
garbage.  The proper course is to insure that is initialized to NULL.
NULL indicates that we do not want any mapping.
The struct is now initialized using a static initializer, and this
technique is compatible with all known versions of libao.
2011-02-15 12:16:25 +01:00
Max Kellermann
03018611f8 update: log all file permission problems 2011-01-31 09:39:24 +01:00
Andreas Wiese
e6c3acaa6f Fix NDEBUG test
<stdbool.h> needs to be included unconditionally from definition of
NDEBUG, since »bool« doesn't get defined otherwise.

Signed-off-by: Andreas Wiese <aw-devel@meterriblecrew.net>
2011-01-14 16:22:25 +01:00
Max Kellermann
0022fb100b encoder/lame: explicitly configure the output sample rate
When you don't explicitly set an output sample rate, liblame tries to
guess an output sample rate from the input sample rate.  You would
think that this "guessing" consists of just setting both equal, but
that is not the case.  For 44.1kHz at 96kbit/s, liblame chooses
32kHz.  This patch explicitly configures the output sample rate, to
stop the bad guessing.
2011-01-07 19:37:39 +01:00
Max Kellermann
4f2d67dfb0 output/httpd: define G_LOG_DOMAIN in httpd_client.c 2011-01-07 18:00:12 +01:00
Max Kellermann
b75d53413d configure.ac: use AC_LANG_SOURCE
Fixes autotools warnings.
2011-01-07 17:25:52 +01:00
Max Kellermann
c44a744c0b fix version number in NEWS 2011-01-07 17:25:25 +01:00
Max Kellermann
60b4f6b3eb directory: fix warning "comparison between signed and unsigned"
Cast the constant to dev_t, not to unsigned.
2010-12-21 20:21:22 +01:00
Max Kellermann
546232b1c0 zeroconf-bonjour: use g_htons() instead of htons()
Fixes the gcc warning "implicit declaration of function 'htons'".
2010-12-21 20:21:20 +01:00
Max Kellermann
42c5788de3 Modify version string to post-release version 0.15.16~git 2010-12-21 20:19:49 +01:00
Max Kellermann
23cd8a74be mpd version 0.15.15 2010-11-08 18:48:28 +01:00
Max Kellermann
cc1debc948 output/shout: artist comes first in stream title
After popular demand, I've switched the order of "artist" and "title"
in the stream title.  There is no standard, and there is no reliable
way to parse those from the stream title.
2010-11-08 18:46:14 +01:00
Max Kellermann
ad52eb236d input/rewind: fix assertion failure
The assertion added in MPD 0.15.14 was too much, it failed when the
MIME type of a stream was NULL.
2010-11-08 10:37:09 +01:00
Avuton Olrich
462bba8e2f Modify version string to post-release version 0.15.15~git 2010-11-06 14:42:03 -07:00
Avuton Olrich
dec7090198 mpd version 0.15.14 2010-11-06 14:42:02 -07:00
Max Kellermann
83ec0e5552 player_thread: fix assertion failure due to wrong music pipe on seek
When one song is played twice, and the decoder is working on the
second "instance", but the first should be seeked, the check in
player_seek_decoder() may assume that it can reuse the decoder without
exchanging pipes.  The last thing was the mistake: the pipe pointer
was different, which led to an assertion failure.  This patch adds
another check which exchanges the player pipe.
2010-11-05 19:24:42 +01:00
Max Kellermann
cc261872c2 decoder_control: pass music_pipe to dc_start()
More abstraction for decoder_control.pipe.
2010-11-05 19:18:44 +01:00
Max Kellermann
5223261f19 player_thread: add helper function player_dc_at_next_song()
Some abstraction for decoder_control.pipe access.
2010-11-05 19:08:59 +01:00
Max Kellermann
c594afeee7 pipe: add helper function music_pipe_empty() 2010-11-05 18:40:23 +01:00
Max Kellermann
32d10eedbd input/rewind: remove redundant NULL check before g_free() call 2010-11-05 18:40:14 +01:00
Max Kellermann
dfd98eede7 input/rewind: add two assertions 2010-11-05 18:40:07 +01:00
Max Kellermann
a728d7a026 input/rewind: fix double free bug
Duplicate the "mime" attribute of the inner input_stream object,
instead of copying the pointer.
2010-11-05 18:39:40 +01:00
Max Kellermann
e8d8bd4c0d decoder/{mp4ff,ffmpeg}: add extension ".m4b" (audio book)
Same as ".m4a".
2010-11-05 02:01:35 +01:00
Max Kellermann
8d5fa754e8 output_thread: fix assertion failure due to race condition in OPEN
Change the assertion on "fail_timer==NULL" in OPEN to a runtime check.
This assertion crashed when the output thread failed while the player
thread was calling audio_output_open().
2010-11-04 23:44:23 +01:00
Max Kellermann
2ee047a1dd output_internal: protect attribute "fail_timer" with mutex 2010-11-04 23:40:43 +01:00
Max Kellermann
9562f66741 output_control: lock object in audio_output_open()
Protect the attributes "open" and "fail_timer".
2010-11-04 23:28:18 +01:00
Max Kellermann
21223154aa output_control: lock object in audio_output_close()
Protect the attributes "open" and "fail_timer".
2010-11-04 21:51:02 +01:00
Avuton Olrich
a549d871f3 Modify version string to post-release version 0.15.14~git 2010-10-10 09:57:57 -07:00
Avuton Olrich
b552e9a120 mpd version 0.15.13 2010-10-10 09:57:52 -07:00
Max Kellermann
5923cfcde3 output/httpd: MIME type audio/ogg for Ogg Vorbis
RFC 5334 10.3 defines the MIME type "audio/ogg".  We could use
"application/ogg" as well, but we know for sure that we only emit
audio data.
2010-10-03 16:22:03 +02:00
Thomas Jansen
e3f4c7b91c input/rewind: enable for MMS 2010-09-28 12:56:47 +02:00
Thomas Jansen
54294366d5 rewind_input_plugin: Update MIME not only once
The assumption that MIME type is set only once is not valid with CURL,
as URL redirections may update the MIME type.

This fixes bug .
2010-09-23 20:39:13 +02:00
Qball Cow
4a7abc9d44 Correctly terminate stream_title.
This caused random data to be send via icy-server if the played
song had no tags.
2010-09-08 13:19:59 +02:00
Max Kellermann
589bb54111 input/curl: fix version check for curl_multi_timeout()
According to the CURL web site, curl_multi_timeout() was added in
version 7.15.4:

 http://curl.haxx.se/libcurl/c/curl_multi_timeout.html
2010-09-07 21:40:56 +02:00
Max Kellermann
64dacd175a output_thread: fix race condition after CANCEL command
Clear the notification before finishing the CANCEL command, so the
notify_wait() after that will always wait for the right notification,
sent by audio_output_all_cancel().
2010-08-19 11:05:24 +02:00
Max Kellermann
625e4755d1 notify: add function notify_clear() 2010-08-19 11:03:53 +02:00
Avuton Olrich
676739c426 Modify version string to post-release version 0.15.13~git 2010-07-21 06:40:33 +02:00
28 changed files with 260 additions and 62 deletions

35
NEWS

@@ -1,3 +1,38 @@
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)
* input:
- rewind: fix assertion failure
* output:
- shout: artist comes first in stream title
ver 0.15.14 (2010/11/06)
* player_thread: fix assertion failure due to wrong music pipe on seek
* output_thread: fix assertion failure due to race condition in OPEN
* input:
- rewind: fix double free bug
* decoders:
- mp4ff, ffmpeg: add extension ".m4b" (audio book)
ver 0.15.13 (2010/10/10)
* output_thread: fix race condition after CANCEL command
* output:
- httpd: fix random data in stream title
- httpd: MIME type audio/ogg for Ogg Vorbis
* input:
- rewind: update MIME not only once
- rewind: enable for MMS
ver 0.15.12 (2010/07/20)
* input:
- curl: remove assertion after curl_multi_fdset()

@@ -1,5 +1,5 @@
AC_PREREQ(2.60)
AC_INIT(mpd, 0.15.12, musicpd-dev-team@lists.sourceforge.net)
AC_INIT(mpd, 0.15.16, musicpd-dev-team@lists.sourceforge.net)
AC_CONFIG_SRCDIR([src/main.c])
AM_INIT_AUTOMAKE([foreign 1.9 dist-bzip2])
AM_CONFIG_HEADER(config.h)

@@ -58,7 +58,7 @@ if test x$enable_aac = xyes; then
fi
if test x$enable_aac = xyes; then
AC_MSG_CHECKING(that FAAD2 uses buffer and bufferlen)
AC_COMPILE_IFELSE([
AC_COMPILE_IFELSE([AC_LANG_SOURCE([
#include <faad.h>
int main() {
@@ -82,9 +82,9 @@ int main() {
return 0;
}
],[AC_MSG_RESULT(yes);AC_DEFINE(HAVE_FAAD_BUFLEN_FUNCS,1,[Define if FAAD2 uses buflen in function calls])],[AC_MSG_RESULT(no);
])],[AC_MSG_RESULT(yes);AC_DEFINE(HAVE_FAAD_BUFLEN_FUNCS,1,[Define if FAAD2 uses buflen in function calls])],[AC_MSG_RESULT(no);
AC_MSG_CHECKING(that FAAD2 can even be used)
AC_COMPILE_IFELSE([
AC_COMPILE_IFELSE([AC_LANG_SOURCE([
#include <faad.h>
int main() {
@@ -113,7 +113,7 @@ int main() {
return 0;
}
],AC_MSG_RESULT(yes),[AC_MSG_RESULT(no);enable_aac=no])
])],AC_MSG_RESULT(yes),[AC_MSG_RESULT(no);enable_aac=no])
])
fi
if test x$enable_aac = xyes; then
@@ -136,7 +136,7 @@ if test x$enable_aac = xyes; then
CPPFLAGS=$CFLAGS
AC_MSG_CHECKING(for broken libfaad headers)
AC_COMPILE_IFELSE([
AC_COMPILE_IFELSE([AC_LANG_SOURCE([
#include <faad.h>
#include <stddef.h>
#include <stdint.h>
@@ -148,7 +148,7 @@ int main() {
faacDecInit2(NULL, NULL, 0, &sample_rate, &channels);
return 0;
}
],
])],
[AC_MSG_RESULT(correct)],
[AC_MSG_RESULT(broken);
AC_DEFINE(HAVE_FAAD_LONG, 1, [Define if faad.h uses the broken "unsigned long" pointers])])

@@ -4,9 +4,9 @@ AC_DEFUN([MPD_CHECK_FLAG],[
[mpd_check_cflag_$var],[
save_CFLAGS="$CFLAGS"
CFLAGS="$CFLAGS $1"
AC_COMPILE_IFELSE([
AC_COMPILE_IFELSE([AC_LANG_SOURCE([
int main(void) { return 0; }
], [ eval "mpd_check_cflag_$var=yes"
])], [ eval "mpd_check_cflag_$var=yes"
], [ eval "mpd_check_cflag_$var=no" ])
CFLAGS="$save_CFLAGS"
])

@@ -501,7 +501,9 @@ static const char *const ffmpeg_suffixes[] = {
"atrac", "au", "aud", "avi", "avm2", "avs", "bap", "bfi", "c93", "cak",
"cin", "cmv", "cpk", "daud", "dct", "divx", "dts", "dv", "dvd", "dxa",
"eac3", "film", "flac", "flc", "fli", "fll", "flx", "flv", "g726",
"gsm", "gxf", "iss", "m1v", "m2v", "m2t", "m2ts", "m4a", "m4v", "mad",
"gsm", "gxf", "iss", "m1v", "m2v", "m2t", "m2ts",
"m4a", "m4b", "m4v",
"mad",
"mj2", "mjpeg", "mjpg", "mka", "mkv", "mlp", "mm", "mmf", "mov", "mp+",
"mp1", "mp2", "mp3", "mp4", "mpc", "mpeg", "mpg", "mpga", "mpp", "mpu",
"mve", "mvi", "mxf", "nc", "nsv", "nut", "nuv", "oga", "ogm", "ogv",

@@ -425,7 +425,13 @@ mp4_tag_dup(const char *file)
return ret;
}
static const char *const mp4_suffixes[] = { "m4a", "mp4", NULL };
static const char *const mp4_suffixes[] = {
"m4a",
"m4b",
"mp4",
NULL
};
static const char *const mp4_mime_types[] = { "audio/mp4", "audio/m4a", NULL };
const struct decoder_plugin mp4ff_decoder_plugin = {

@@ -18,6 +18,7 @@
*/
#include "decoder_control.h"
#include "pipe.h"
#include <assert.h>
@@ -58,22 +59,28 @@ static void dc_command_async(enum decoder_command cmd)
}
void
dc_start(struct notify *notify, struct song *song)
dc_start(struct notify *notify, struct song *song, struct music_pipe *pipe)
{
assert(dc.pipe != NULL);
assert(dc.pipe == NULL);
assert(song != NULL);
assert(pipe != NULL);
assert(music_pipe_empty(pipe));
dc.next_song = song;
dc.pipe = pipe;
dc_command(notify, DECODE_COMMAND_START);
}
void
dc_start_async(struct song *song)
dc_start_async(struct song *song, struct music_pipe *pipe)
{
assert(dc.pipe != NULL);
assert(dc.pipe == NULL);
assert(song != NULL);
assert(pipe != NULL);
assert(music_pipe_empty(pipe));
dc.next_song = song;
dc.pipe = pipe;
dc_command_async(DECODE_COMMAND_START);
}

@@ -118,10 +118,10 @@ void
dc_command_wait(struct notify *notify);
void
dc_start(struct notify *notify, struct song *song);
dc_start(struct notify *notify, struct song *song, struct music_pipe *pipe);
void
dc_start_async(struct song *song);
dc_start_async(struct song *song, struct music_pipe *pipe);
void
dc_stop(struct notify *notify);

@@ -28,8 +28,8 @@
#define DIRECTORY_DIR "directory: "
#define DEVICE_INARCHIVE (unsigned)(-1)
#define DEVICE_CONTAINER (unsigned)(-2)
#define DEVICE_INARCHIVE (dev_t)(-1)
#define DEVICE_CONTAINER (dev_t)(-2)
struct directory {
struct dirvec children;

@@ -169,6 +169,13 @@ lame_encoder_setup(struct lame_encoder *encoder, GError **error)
return false;
}
if (0 != lame_set_out_samplerate(encoder->gfp,
encoder->audio_format.sample_rate)) {
g_set_error(error, lame_encoder_quark(), 0,
"error setting lame out sample rate");
return false;
}
if (0 > lame_init_params(encoder->gfp)) {
g_set_error(error, lame_encoder_quark(), 0,
"error initializing lame params");

@@ -95,6 +95,7 @@ icy_server_metadata_page(const struct tag *tag, ...)
gchar stream_title[(1 + 255 - 28) * 16]; // Length + Metadata -
// "StreamTitle='';StreamUrl='';"
// = 4081 - 28
stream_title[0] = '\0';
last_item = -1;

@@ -259,7 +259,7 @@ input_curl_select(struct input_curl *c)
return -1;
}
#if LIBCURL_VERSION_NUM >= 0x070f00
#if LIBCURL_VERSION_NUM >= 0x070f04
long timeout2;
mcode = curl_multi_timeout(c->multi, &timeout2);
if (mcode != CURLM_OK) {

@@ -20,6 +20,9 @@
#include "config.h"
#include "input/rewind_input_plugin.h"
#include "input/curl_input_plugin.h"
#ifdef ENABLE_MMS
#include "input/mms_input_plugin.h"
#endif
#include "input_plugin.h"
#include "tag.h"
@@ -80,16 +83,19 @@ copy_attributes(struct input_stream *dest)
const struct input_rewind *r = dest->data;
const struct input_stream *src = &r->input;
assert(dest != src);
assert(src->mime == NULL || dest->mime != src->mime);
dest->ready = src->ready;
dest->seekable = src->seekable;
dest->error = src->error;
dest->size = src->size;
dest->offset = src->offset;
if (dest->mime == NULL && src->mime != NULL)
/* this is set only once, and the duplicated pointer
is freed by input_stream_close() */
if (src->mime != NULL) {
g_free(dest->mime);
dest->mime = g_strdup(src->mime);
}
}
static void
@@ -219,7 +225,11 @@ input_rewind_open(struct input_stream *is)
assert(is != NULL);
assert(is->offset == 0);
if (is->plugin != &input_plugin_curl)
if (is->plugin != &input_plugin_curl
#ifdef ENABLE_MMS
&& is->plugin != &input_plugin_mms
#endif
)
/* due to limitations in the input_plugin API, we only
(explicitly) support the CURL input plugin */
return;
@@ -229,10 +239,12 @@ input_rewind_open(struct input_stream *is)
/* move the CURL input stream to c->input */
c->input = *is;
input_curl_reinit(&c->input);
if (is->plugin == &input_plugin_curl)
input_curl_reinit(&c->input);
/* convert the existing input_stream pointer to a "rewind"
input stream */
is->plugin = &rewind_input_plugin;
is->data = c;
is->mime = g_strdup(c->input.mime);
}

@@ -48,3 +48,10 @@ void notify_signal(struct notify *notify)
g_cond_signal(notify->cond);
g_mutex_unlock(notify->mutex);
}
void notify_clear(struct notify *notify)
{
g_mutex_lock(notify->mutex);
notify->pending = false;
g_mutex_unlock(notify->mutex);
}

@@ -45,4 +45,9 @@ void notify_wait(struct notify *notify);
*/
void notify_signal(struct notify *notify);
/**
* Clears a pending notification.
*/
void notify_clear(struct notify *notify);
#endif

@@ -25,6 +25,9 @@
#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 {
@@ -166,7 +169,7 @@ 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;
/* support for 24 bit samples in libao is currently dubious,

@@ -27,6 +27,9 @@
#include <assert.h>
#include <string.h>
#undef G_LOG_DOMAIN
#define G_LOG_DOMAIN "httpd_output"
struct httpd_client {
/**
* The httpd output object this client is connected to.

@@ -70,7 +70,7 @@ httpd_output_init(G_GNUC_UNUSED const struct audio_format *audio_format,
}
if (strcmp(encoder_name, "vorbis") == 0)
httpd->content_type = "application/x-ogg";
httpd->content_type = "audio/ogg";
else if (strcmp(encoder_name, "lame") == 0)
httpd->content_type = "audio/mpeg";
else

@@ -36,7 +36,7 @@
#undef G_LOG_DOMAIN
#define G_LOG_DOMAIN "jack"
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);
static const char *const port_names[2] = {
"left", "right",
@@ -118,14 +118,15 @@ mpd_jack_process(jack_nframes_t nframes, void *arg)
for (unsigned i = 0; i < G_N_ELEMENTS(jd->ringbuffer); ++i) {
available = jack_ringbuffer_read_space(jd->ringbuffer[i]);
assert(available % sample_size == 0);
available /= sample_size;
assert(available % jack_sample_size == 0);
available /= jack_sample_size;
if (available > nframes)
available = nframes;
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);
while (available < nframes)
/* ringbuffer underrun, fill with silence */
@@ -422,7 +423,7 @@ mpd_jack_play(void *data, const void *chunk, size_t size, GError **error)
/* send data symmetrically */
space = space1;
if (space >= frame_size)
if (space >= jack_sample_size)
break;
/* XXX do something more intelligent to
@@ -430,7 +431,7 @@ mpd_jack_play(void *data, const void *chunk, size_t size, GError **error)
g_usleep(1000);
}
space /= sample_size;
space /= jack_sample_size;
if (space < size)
size = space;

@@ -483,7 +483,7 @@ shout_tag_to_metadata(const struct tag *tag, char *dest, size_t size)
}
}
snprintf(dest, size, "%s - %s", title, artist);
snprintf(dest, size, "%s - %s", artist, title);
}
static void my_shout_set_tag(void *data,

@@ -266,7 +266,7 @@ audio_output_all_open(const struct audio_format *audio_format,
else
/* if the pipe hasn't been cleared, the the audio
format must not have changed */
assert(music_pipe_size(g_mp) == 0 ||
assert(music_pipe_empty(g_mp) ||
audio_format_equals(audio_format,
&input_audio_format));
@@ -378,7 +378,7 @@ audio_output_all_check(void)
assert(g_mp != NULL);
while ((chunk = music_pipe_peek(g_mp)) != NULL) {
assert(music_pipe_size(g_mp) > 0);
assert(!music_pipe_empty(g_mp));
if (!chunk_is_consumed(chunk))
/* at least one output is not finished playing

@@ -50,6 +50,20 @@ static void ao_command(struct audio_output *ao, enum audio_output_command cmd)
ao_command_wait(ao);
}
/**
* Like ao_command(), but assumes the object is locked by the caller.
*/
static void
ao_command_locked(struct audio_output *ao, enum audio_output_command cmd)
{
assert(ao->command == AO_COMMAND_NONE);
ao->command = cmd;
g_mutex_unlock(ao->mutex);
ao_command_wait(ao);
g_mutex_lock(ao->mutex);
}
static void ao_command_async(struct audio_output *ao,
enum audio_output_command cmd)
{
@@ -58,6 +72,12 @@ static void ao_command_async(struct audio_output *ao,
notify_signal(&ao->notify);
}
static void
audio_output_close_locked(struct audio_output *ao);
/**
* Object must be locked (and unlocked) by the caller.
*/
static bool
audio_output_open(struct audio_output *ao,
const struct audio_format *audio_format,
@@ -84,7 +104,7 @@ audio_output_open(struct audio_output *ao,
/* we're not using audio_output_cancel() here,
because that function is asynchronous */
ao_command(ao, AO_COMMAND_CANCEL);
ao_command_locked(ao, AO_COMMAND_CANCEL);
}
return true;
@@ -95,7 +115,7 @@ audio_output_open(struct audio_output *ao,
if (!ao->config_audio_format) {
if (ao->open)
audio_output_close(ao);
audio_output_close_locked(ao);
/* no audio format is configured: copy in->out, let
the output's open() method determine the effective
@@ -110,7 +130,7 @@ audio_output_open(struct audio_output *ao,
open = ao->open;
if (!open) {
ao_command(ao, AO_COMMAND_OPEN);
ao_command_locked(ao, AO_COMMAND_OPEN);
open = ao->open;
}
@@ -127,12 +147,19 @@ audio_output_update(struct audio_output *ao,
{
assert(mp != NULL);
g_mutex_lock(ao->mutex);
if (ao->enabled) {
if (ao->fail_timer == NULL ||
g_timer_elapsed(ao->fail_timer, NULL) > REOPEN_AFTER)
return audio_output_open(ao, audio_format, mp);
g_timer_elapsed(ao->fail_timer, NULL) > REOPEN_AFTER) {
bool ret = audio_output_open(ao, audio_format, mp);
g_mutex_unlock(ao->mutex);
return ret;
}
} else if (audio_output_is_open(ao))
audio_output_close(ao);
audio_output_close_locked(ao);
g_mutex_unlock(ao->mutex);
return false;
}
@@ -162,21 +189,33 @@ void audio_output_cancel(struct audio_output *ao)
ao_command_async(ao, AO_COMMAND_CANCEL);
}
void audio_output_close(struct audio_output *ao)
static void
audio_output_close_locked(struct audio_output *ao)
{
assert(ao != NULL);
assert(!ao->open || ao->fail_timer == NULL);
if (ao->mixer != NULL)
mixer_auto_close(ao->mixer);
if (ao->open)
ao_command(ao, AO_COMMAND_CLOSE);
ao_command_locked(ao, AO_COMMAND_CLOSE);
else if (ao->fail_timer != NULL) {
g_timer_destroy(ao->fail_timer);
ao->fail_timer = NULL;
}
}
void audio_output_close(struct audio_output *ao)
{
assert(ao != NULL);
assert(!ao->open || ao->fail_timer == NULL);
g_mutex_lock(ao->mutex);
audio_output_close_locked(ao);
g_mutex_unlock(ao->mutex);
}
void audio_output_finish(struct audio_output *ao)
{
audio_output_close(ao);

@@ -131,7 +131,8 @@ struct audio_output {
const struct music_pipe *pipe;
/**
* This mutex protects #open, #chunk and #chunk_finished.
* This mutex protects #open, #fail_timer, #chunk and
* #chunk_finished.
*/
GMutex *mutex;

@@ -105,7 +105,12 @@ ao_play_chunk(struct audio_output *ao, const struct music_chunk *chunk)
/* don't automatically reopen this device for
10 seconds */
g_mutex_lock(ao->mutex);
assert(ao->fail_timer == NULL);
ao->fail_timer = g_timer_new();
g_mutex_unlock(ao->mutex);
return false;
}
@@ -192,10 +197,18 @@ static gpointer audio_output_task(gpointer arg)
case AO_COMMAND_OPEN:
assert(!ao->open);
assert(ao->fail_timer == NULL);
assert(ao->pipe != NULL);
assert(ao->chunk == NULL);
if (ao->fail_timer != NULL) {
/* this can only happen when this
output thread fails while
audio_output_open() is run in the
player thread */
g_timer_destroy(ao->fail_timer);
ao->fail_timer = NULL;
}
error = NULL;
ret = ao_plugin_open(ao->plugin, ao->data,
&ao->out_audio_format,
@@ -268,6 +281,16 @@ static gpointer audio_output_task(gpointer arg)
ao->chunk = NULL;
if (ao->open)
ao_plugin_cancel(ao->plugin, ao->data);
/* we must clear the notification now, because
the notify_wait() call below must wait
until audio_output_all_cancel() has cleared
the pipe; if another notification happens
to be still pending, we get a race
condition with a crash or an assertion
failure */
notify_clear(&ao->notify);
ao_command_finished(ao);
/* the player thread will now clear our music

@@ -20,9 +20,9 @@
#ifndef MPD_PIPE_H
#define MPD_PIPE_H
#ifndef NDEBUG
#include <stdbool.h>
#ifndef NDEBUG
struct audio_format;
#endif
@@ -99,4 +99,10 @@ music_pipe_push(struct music_pipe *mp, struct music_chunk *chunk);
unsigned
music_pipe_size(const struct music_pipe *mp);
static inline bool
music_pipe_empty(const struct music_pipe *mp)
{
return music_pipe_size(mp) == 0;
}
#endif

@@ -118,6 +118,33 @@ static void player_command_finished(void)
notify_signal(&main_notify);
}
/**
* Is the decoder still busy on the same song as the player?
*
* Note: this function does not check if the decoder is already
* finished.
*/
static bool
player_dc_at_current_song(const struct player *player)
{
assert(player != NULL);
assert(player->pipe != NULL);
return dc.pipe == player->pipe;
}
/**
* Has the decoder already begun decoding the next song?
*
* Note: this function does not check if the decoder is already
* finished.
*/
static bool
player_dc_at_next_song(const struct player *player)
{
return dc.pipe != NULL && !player_dc_at_current_song(player);
}
/**
* Stop the decoder and clears (and frees) its music pipe.
*/
@@ -297,10 +324,9 @@ static bool player_seek_decoder(struct player *player)
/* clear music chunks which might still reside in the
pipe */
music_pipe_clear(player->pipe, player_buffer);
dc.pipe = player->pipe;
/* re-start the decoder */
dc_start_async(pc.next_song);
dc_start_async(pc.next_song, player->pipe);
ret = player_wait_for_decoder(player);
if (!ret) {
/* decoder failure */
@@ -308,6 +334,14 @@ static bool player_seek_decoder(struct player *player)
return false;
}
} else {
if (!player_dc_at_current_song(player)) {
/* the decoder is already decoding the "next" song,
but it is the same song file; exchange the pipe */
music_pipe_clear(player->pipe, player_buffer);
music_pipe_free(player->pipe);
player->pipe = dc.pipe;
}
pc.next_song = NULL;
player->queued = false;
}
@@ -364,7 +398,7 @@ static void player_process_command(struct player *player)
case PLAYER_COMMAND_QUEUE:
assert(pc.next_song != NULL);
assert(!player->queued);
assert(dc.pipe == NULL || dc.pipe == player->pipe);
assert(!player_dc_at_next_song(player));
player->queued = true;
player_command_finished();
@@ -409,7 +443,7 @@ static void player_process_command(struct player *player)
return;
}
if (dc.pipe != NULL && dc.pipe != player->pipe)
if (player_dc_at_next_song(player))
/* the decoder is already decoding the song -
stop it and reset the position */
player_dc_stop(player);
@@ -505,7 +539,7 @@ play_next_chunk(struct player *player)
return true;
if (player->xfade == XFADE_ENABLED &&
dc.pipe != NULL && dc.pipe != player->pipe &&
player_dc_at_next_song(player) &&
(cross_fade_position = music_pipe_size(player->pipe))
<= player->cross_fade_chunks) {
/* perform cross fade */
@@ -638,8 +672,7 @@ static void do_play(void)
player.pipe = music_pipe_new();
dc.buffer = player_buffer;
dc.pipe = player.pipe;
dc_start(&pc.notify, pc.next_song);
dc_start(&pc.notify, pc.next_song, player.pipe);
if (!player_wait_for_decoder(&player)) {
player_dc_stop(&player);
player_command_finished();
@@ -706,14 +739,15 @@ static void do_play(void)
/* the decoder has finished the current song;
make it decode the next song */
assert(pc.next_song != NULL);
assert(dc.pipe == NULL || dc.pipe == player.pipe);
assert(!player_dc_at_next_song(&player));
dc.pipe = NULL;
player.queued = false;
dc.pipe = music_pipe_new();
dc_start_async(pc.next_song);
dc_start_async(pc.next_song, music_pipe_new());
}
if (dc.pipe != NULL && dc.pipe != player.pipe &&
if (player_dc_at_next_song(&player) &&
player.xfade == XFADE_UNKNOWN &&
!decoder_is_starting()) {
/* enable cross fading in this song? if yes,
@@ -736,7 +770,7 @@ static void do_play(void)
if (player.paused)
notify_wait(&pc.notify);
else if (music_pipe_size(player.pipe) > 0) {
else if (!music_pipe_empty(player.pipe)) {
/* at least one music chunk is ready - send it
to the audio output */
@@ -748,7 +782,7 @@ static void do_play(void)
/* XXX synchronize in a better way */
g_usleep(10000);
} else if (dc.pipe != NULL && dc.pipe != player.pipe) {
} else if (player_dc_at_next_song(&player)) {
/* at the beginning of a new song */
if (!player_song_border(&player))
@@ -757,7 +791,7 @@ static void do_play(void)
/* check the size of the pipe again, because
the decoder thread may have added something
since we last checked */
if (music_pipe_size(player.pipe) == 0)
if (music_pipe_empty(player.pipe))
break;
} else {
/* the decoder is too busy and hasn't provided

@@ -254,6 +254,9 @@ 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;
}
@@ -270,6 +273,9 @@ 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;
}

@@ -62,7 +62,7 @@ void init_zeroconf_osx(const char *serviceName)
DNSServiceErrorType error = DNSServiceRegister(&dnsReference,
0, 0, serviceName,
SERVICE_TYPE, NULL, NULL,
htons(listen_port), 0,
g_htons(listen_port), 0,
NULL,
dnsRegisterCallback,
NULL);