| Commit message (Collapse) | Author | Age | Files | Lines |
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Case insensitivity isn't helpful, and comparing only the first 3 bytes
of a configured value may encourage users to supply wrong or
misleading values.
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If the sample format isn't supported by the device (i.e. 24 bit on
low-end sound chips), fall back to 16 bit output. There is code in
pcm_utils.c which converts PCM data to 16 bit.
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When calculating the conversion buffer size, don't hard-code the
formulas for only mono<->stereo.
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Convert any number of channels to stereo. In fact, this isn't really
stereo, it's rater mono blown up to stereo. This patch should only
make it possible to play 5.1 files at all; "real" conversion to stereo
should be implemented, but for now, this is better than nothing.
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In order to be able to deal with non-trivial conversions,
pcm_convertChannels() needs to know both the input and the output
channel count. Simplify buffer allocation in that function.
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Moved code from pcm_convertChannels() to pcm_convert_channels_1_to_2()
and pcm_convert_channels_2_to_1(). Improved the quality of
pcm_convert_channels_2_to_1() by calculating the arithmetic mean value
of both samples.
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Move code which loads configuration to alsa_configure(). This removes
one indent level.
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Pass int16_t pointers instead of char pointers to functions which can
deal with 16 bit audio only.
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buffered_before_play was copied to struct player because it was used
to disable buffering when seeking. Instead of mainaining a copy of
this number, move just the flag to the player struct.
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Renamed audio_configFormat to configured_audio_format. Renamed
audio_buffer.format to input_audio_format. Simplified its
initialization in openAudioDevice().
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All what's left from ad_state is a simple boolean flag (ENABLE /
DISABLE). Convert it to a C99 bool.
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audio.c maintained one of MPD's many layers of audio buffers. It was
without any benefit, since playAudio() can simply send the source
buffer directly to the audio output plugin.
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The function ob_is_empty() is called very often. It's worth it to
convert it to an inline function.
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QUEUE adds a new song to the player's queue. CANCEL clears the queue.
These two commands replace the old and complex queueState and
queueLockState code.
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Simplify and merge several if clauses before the clearPlayerQueue()
invocation. Call clearPlayerQueue() only if a song is actually
queued; add an assertion for that in clearPlayerQueue().
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The player struct holds the local variables which used to be passed to
all those helper functions in player_thread.c.
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set_current_song() is an internal function and consists of only one
assignment. Eliminate it.
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This variable is superfluous, it is only used to copy its value to
player_control.totalTime. Since the original source of this value
(song->tag->time) will still be available at this point, we can safely
remove fileTime.
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ALSA does a good job measuring its buffer_time and period_time. Don't
override its defaults, unless the user demands it.
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Revert e4f5d6bd "re-enable-nonblocking, but sleep if busy".
Non-blocking mode with manual sleeping doesn't help at all (by the
way, the patch should have used snd_pcm_wait() instead of
my_usleep()). ALSA knows much more about the hardware quirks, so we
just let it do the job.
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Leftover from the output API changes: oss_open_default() was changed
to return a void*, but it still returned "0" to report success.
Report the OssData pointer instead.
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The decoder was woken up after each chunk which had been played. That
caused a lot of superfluous context switches. Wake up the decoder
only when a certain amount of the buffer has been consumed. This
formula is somewhat arbitrary, and has to be proven experimentally.
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The mp3 plugin did not use the MAD_NCHANNELS() value correctly: when a
stream was not stereo, it was assumed to be mono, although the correct
number was passed to MPD. libmad doesn't support more than 2
channels, but this change allows gcc to optimize its inlining
strategy.
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The dithering function audio_linear_dither() worked for signed 16 bits
only anyway, having a variable "bits" just disables important gcc
optimizations.
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player_get_audio_format() replaces getPlayerSampleRate(),
getPlayerBits(), getPlayerChannels().
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This replaces the attributes bits, channels, sampleRate.
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Fix some gcc warnings by using unsigned where appropriate. Declare
numfrequencies as "const", and replaced the magic number 12 with a
sizeof.
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Again, a plugin which was disabled in my test environment and suffered
from compilation errors.
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This replaces the separate properties channels, sampleRate, bits.
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A frame contains one sample per channel, thus it is sample_size *
channels. This patch includes some cleanup for various locations
where the sample size for 24 bit audio was still 3 bytes (instead of
4).
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The last bit of CamelCase in audio_format.h. Additionally, rename a
bunch of local variables.
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"bits" and "channels" cannot be negative.
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There is only once update thread at a time. Make the "modified" flag
global and remove the return values of most functions. Propagating an
error is only useful for updateDirectory(), since updateInDirectory()
will delete failed subdirectories.
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When a song file was not modified, MPD printed the debug message "not
a directory or music", because the first "if" branch did not return.
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When the update queue is full, directory_update_init() did not free
the path argument.
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Since the return value cannot be -1 anymore, we can make it unsigned.
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The documentation for directory_update_init() was incorrect: a job ID
must be positive, not non-negative. If the update queue is full and
no job was created, it makes more sense to return 0 instead of -1,
because it is more consistent with the return value of isUpdatingDB().
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When the update task is idle, there is no need to check for deleted
songs. Return early from reap_update_task().
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pthread_join() expects a "pointer to a pointer" parameter, but it got
a "pointer to an enum". On AMD64, an enum is smaller than a pointer,
leading to a buffer overflow.
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Removed the local variable "was_empty": don't remember if the
directory is new. Always call removeDeletedFromDirectory().
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In updateInDirectory(), add new directories immediately and
delete them when they turn out to be empty. This simplifies the code
and allows us to eliminate addSubDirectoryToDirectory().
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The recursive checks were performed in several functions, and
sometimes a directory was checked twice.
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When reading a new directory, copy the stat data (which we have
anyway) to the directory struct. This may save a stat() in the
future.
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Pass a pointer to the stat struct to more functions.
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updatePath() duplicated a lot of code from the more generic
updateInDirectory(). Eliminate most of updatePath() and call
updateInDirectory().
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If the user requests database update during startup, call
directory_update_init(). This should be changed to fully asynchronous
update later.
For this to work, main_notify has to be initialized before db_init().
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The stat struct isn't going to be modified, make it const.
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