| Commit message (Collapse) | Author | Age | Files | Lines |
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Pass the input_stream object to decoder_data(). Without it, the MPD
core does not see stream tags.
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When you change the filesystem charset, discard the old database file
and create a new one. The old database file will most likely contain
stale or invalid information.
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Use WildMidi_SampledSeek() for seeking in a MIDI file.
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Currently, only the sidplay decoder plugin requires C++, and in all
other cases, MPD could build well without a C++ compiler.
Unfortunately, autoconf/automake are confused when we have a
conditional AC_PROG_CXX check. We could add lots of workarounds for
individual problems, but let's just always require a C++ compiler, and
forget about this autotools limitation.
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The _WM_Info struct provides all we need, it is obtained by
WildMidi_GetInfo().
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There are a few problems left in this plugin:
- fluidsynth decodes in real time, while MPD prefers to buffer as
quickly as possible; as a workaround, this plugin uses a timer
object to synchronize with real-time playback
- I don't know yet how fluidsynth tells me when the song has ended
- the "soundfont" configuration setting is not yet documented, and it
will likely change soon (in favor of a per-decoder configuration
block)
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When the sidplay plugin is disabled, "./configure" does not look for
the C++ compiler. This creates an odd situation: automake requires
the am__fastdepCXX conditional, although configure did not generate
it. Work around this autotools limitation by manually disabling
am__fastdepCXX.
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By accident, I committed a debug flag, which disallowed the decoder
thread to play files locally. Undo this hunk.
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The ffmpeg library supports the "True Audio Codec". The entry in
ffmpeg_suffixes was missing.
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When MPD is not playing, it may still remember which is the "current"
song. When you switch to "random" mode, MPD will always start playing
exactly this song. This defies the goal of "random" mode a little.
Clear the "current" song when MPD is not playing during the "random"
mode switch.
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The output_command library provides a command interface to the audio
outputs. It assumes the input comes from an untrusted source
(i.e. the client) and verifies all parameters.
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In addition to audio_format_valid(), provide functions which validate
only one attribute of an audio_format. These functions are reused by
audio_format_parse().
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Added audio_format_parse() in a separate library, with a modern
interface: return a GError instead of logging errors. This allows the
caller to deal with the error.
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port number
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When MPD explicitly starts playing, ignore the "REOPEN_AFTER" timeout.
This timeout was useful when MPD attempted to reopen a failed device
over and over, but it confuses users when they explicitly tell MPD to
start playing, while MPD insists to wait for the 10 seconds to pass.
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Merge some duplicate code into one function.
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When the pause() method fails, leave the pause loop, because calling
pause() on a closed device is not allowed.
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Fix a memory leak: it was not guaranteed that pcm_convert_deinit() was
called for each pcm_convert_init(). This patch always (de)initializes
the pcm_convert library when the audio_output.open flag is flipped.
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Renamed audio_output struct members.
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Use audio_format_frame_size() instead of
channels*audio_format_sample_size().
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Pass the music chunk as a "const void *" to the encoder, instead of a
"const char *". Actually, both encoders currently expect 16 bit
samples, passing a 8-bit character is rather pointless.
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For simplification, moved the PCM conversion code to
pcm16_to_ogg_buffer(). Work with a int16_t pointer instead of a char
pointer.
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writeSize is a memory size and its type should thus be size_t. This
allows us to remove two explicit casts.
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Nobody needs these debug messages anymore.
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audio_output_all_finished() returns bool, not int.
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Renamed functions and variables.
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Moved code which deals with all audio outputs at once into a separate
library.
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This function isn't used anymore.
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The crossfading code shouldn't depend on the audio output code. Pass
the current audio format to cross_fade_calc() and let it compare
directly, instead of using isCurrentAudioFormat().
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When MPD is stopped, but the last song is still the "current song",
and you delete it, playlist->current is not updated, and becomes an
invalid value. Fix this by catching "!playlist->playing &&
playlist->current == (int)songOrder".
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Added audio_output_get(), audio_output_find().
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audio_output_config_count() returns the number of audio outputs in the
configuration file. It is only used by initAudioDriver(). The public
function audio_output_count() now returns audioOutputArraySize.
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output_api.h is required for enum audio_output_command.
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Assertions on pc.command and pc.next_song.
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When we reset pc.next_song if there is no song queued, this might
cause a race condition: the next song to be played is cleared, while
pc.command was already set. Clear the "next_song" only if there is a
song queued for the current do_play() invocation.
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If a new song is queued before calling playerSeek(), then the player
and the playlist enter an inconsistent state, because the player
discards the playlist's "queued" song in favor of the seeked song.
Call playlist_update_queued_song() after playerSeek().
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After a player command (successful or not), reset pc.next_song,
because the queue is supposed to be empty then. Otherwise,
playlist.queued and pc.next_song may disagree, which triggers an
assertion failure.
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Commit f78cddb4 introduced a regression: after a song was moved, the
order array was not updated (in random mode). This caused MPD to
think the "current" song has changed when you moved something to the
position of the current song.
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Don't define HAVE_FFMPEG if the ffmpeg libraries were found via
pkg-config, but ffmpeg support was disabled because
avcodec_decode_audio2() is not available.
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Check if the "current+1" position is actually valid.
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Including "../config.h" breaks on some systems.
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Always assume the buffer is empty before calling the encoder. Always
flush the buffer immediately after there has been added something.
This reduces the risk of buffer overruns, because there will never be
a "rest" in the current buffer.
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Don't duplicate the tag received by the send_metadata() method - send
it to the shout server directly.
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Removed the manual timer synchronization from the shout plugin.
libshout's shout_sync() function does it for us.
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