There was a discrepancy between what was written to the buffer and the
size returned by pcm_dsd_to_dop(): the "for" loop uses num_frames/2,
rounding down, while the return value is num_samples which is
num_frames*channels, without rounding. This could cause undefined
data at the end of the destination buffer if the source buffer size
was not aligned to multiples of 8 bytes (4 DSD bytes per channel).
The latter however can occur in the 0.21 branch after commit
a06bf388d9Closes#233
Instead of stopping playback (due to seek time overflow), reject the
seek command. Closes#240
Relative negative values (with "seekcur") are still allowed, and MPD
will fix the resulting position if it turns out to be negative. But
the "seek" and "seekid" commands use an unsigned time stamp which must
not be negative.
With Grand Central Dispatch used in Main.cxx, debug builds on macOS
crash as the IsInside() assertion gets triggered in the event loop. As
a simple fix, usage of GCD is removed. Plugging and unplugging
headphones or changes of the default output device was tested without
issues. Whatever the original commit tried to fix by GCD probably does
not need fixing anymore.
From: Christian Kröner <ckroener@gmx.net>
This just copies the necessary bits and pieces from the ALSA plugin and applies them to OSXOutput based on dop config setting. It only changes the OSXOutput plugin as needed for DoP (further changes to support additionally e.g. integer mode or setting the physical device mode require rather a complete rewrite of the output plugin).
Fortunately the Core Audio API is by default bit perfect and supports DoP with minimal changes (setting the sampling rate accordingly after ensuring that the physical mode supports at least 24 bits per channel seems to be enough). This was tested on an Amanero Combo384 device hooked up to a ES9018 DAC.
USAGE (try only on DACs that support DoP):
- Add dop "yes" option to mpdconf
- Be sure to set at least 24bits per channel before playing some DSD file (using Audio-MIDI-Setup)
- Based on the dop setting, MPD will change the sample rate as required and output DoP signal to the DAC
- Hog mode is recommended to ensure that no other program will try to mix some output with the DoP stream (resulting in bad noise)
- Alternatively set the default output device to another device (e.g. the built-in output) to avoid having other audio interfere with DSD playback
support for chaining ogg opus streams to enable changing stream' metadata on the fly.
currently support is opt-in (enabled by additional option) because lots of clients can't handle this properly yet.
read_stream_art uses PRIu64 unconditionally with the Format
method of a Respone instance to output a size_t typed value.
If size_t is 32bit the output is garbeled. This patch uses
offset_type and PRIoffset to make sure the format string
and the type of the output value always match.
This addresses two problems:
1. the libFLAC write callback had to send an error status to its
caller when SubmitData() returned a command; this disrupted libFLAC
and the resulting command could not be used for anything;
2. the libFLAC function FLAC__stream_decoder_seek_absolute() also
calls the write callback, but its result cannot be used, because
seeking is still in progress, so we lose all data from one FLAC frame.
By moving the SubmitData() call until after CommandFinished(), we
avoid losing this data. This fixes another part of #113
Instead of passing whole chunks to the MusicPipe and checking the
end_time after each chunk, truncate the last chunk if it would exceed
the end_time. This requires keeping track of the absolute PCM frame
number.
This fixes a problem with gapless CUE song transitions: a small part
of the following song was always played twice.
Closes#113
The normal I/O event thread can have a large latency, e.g. when
libgnutls loads all TLS CA certificates for a https connect. This
makes it unreliable for the ALSA I/O notifications, and causes ring
buffer xruns. To avoid interfering with high latency events such as
CURL's, we move the ALSA I/O events to a separate I/O thread which
also obtains real-time scheduling (if possible).
Closes#221
Due to rounding errors, a slightly negative value can be passed to
set_normalized_volume(), which will make the log10() call fail.
Actually, volume 0 is already failing because log10(0) is illegal. So
let's fix this by implementing two corner cases: <=0 and >=100.
Closes#212
