Replaced the local variable "colon" (which had only temporary meaning)
with the variable "value". It is a pointer to the first byte of the
header value.
Instead of managing a set of method pointers in each input_stream
struct, move these into the new input_plugin struct. Each
input_stream has only a pointer to the plugin struct. Pointers to all
implementations are kept in the array "input_plugins".
MPD's HTTP client code has always been broken, no matter how effort
was put into fixing it. Replace it with libcurl, which is known to be
quite stable. This adds a fat library dependency, but only for people
who need streaming.
Several clients refuse to accept the protocol version "0.14~git",
because they think it is malformed. This is clearly a client bug, but
we cannot wait for all clients to fix this bug right now. For now,
change the version back to "0.14.0".
For testers, it should be clear that they're not using version 0.14.0
final, but an inofficial intermediate version from the git repository.
The protocol version is set to the same string, since the protocol is
subject to change during MPD development.
MPD shouldn't integrate sources of other libraries. Since libmp4ff is
part of libfaad, we should remove the old copy from src/mp4ff and link
with the current version from libfaad instead.
PA_SAMPLE_S16NE is the only sample format which is suported by both
MPD and pulseaudio. Unfortunately, pulse does not accept 24 bit
samples.
Instead of bailing out with an error message, we should tell the MPD
core to convert all samples to 16 bit for pulse.
This bug caused the audio output devices to stay open, although MPD
wasn't playing: quitDecode() resetted player_control.command, assuming
that the command was STOP. This way, player_task() didn't see the
CLOSE_AUDIO command, and the device was kept open.
Don't clear player_control.command in quitDecode().
When the audio source provides 24 bit samples, don't bother to convert
(lossily) them to 16 bit before jack's floating point conversion - go
directly from 24 bit to float.
The JACK documentation postulates that the process() callback must not
block, therefore locking is forbidden. Anyway, the old code was racy.
Remove all locks, and don't wait for more data to become available -
just send to the port what is already in the buffer.
Another partial frame fix: the silence buffer was 1020 bytes, which
had room for 127.5 24 bit stereo frames. Don't send the partial last
frame in this case.
24 bit output is as important as 16 bit output. Provide a
pcm_convert() implementation which can convert to 24 bit with as
little quality loss as possible.
The old pcm_convert_size() ignored most of the destination format,
e.g. it did not check its sample size, and assumed it is 16 bit.
Simplify and universalize it by using audio_format_frame_size().
Similar to pcm_resample_16(), implement pcm_resample_24(). The 24 bit
implementation is very similar, but it uses src_int_to_float_array()
instead of src_short_to_float_array() before sending data to
libsamplerate.
