| Commit message (Collapse) | Author | Files | Lines |
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Renamed all functions and variables. Also removed the mp3DecodeData
typedef.
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Yet another superfluous buffering layer. input_file was using FILE*,
but we're better off with unbuffered I/O using open(), read(), ...
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size_t and long aren't 64 bit safe (i.e. files larger than 2 GB on a
32 bit OS). Use off_t instead, which is a 64 bit integer if compiled
with large file support.
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Remove duplicated code from MPD.
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When the decoder failed to start, the function do_play() returned,
still having pc.command==PLAY. This is because pc.command was reset
only when the decoder started up successfully. Add another
player_command_finished() call in the error handler.
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Don't attempt to open a HTTP URL as a local file, and don't send a
local path to libcurl.
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Don't compile the sources of disabled output plugins at all.
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On some systems, string.h declares basename(). This emits a shadow
warning. Change the variable name.
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If the server sends the headers "icy-name", "ice-name",
"x-audiocast-name", set the stream title.
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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.
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g_free() allows passing the NULL pointer.
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For boolean values and success flags, use bool instead of integer (1/0
for true/false, 0/-1 for success/failure).
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close() shouldn't fail with read-only streams.
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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".
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Renamed all functions and variables.
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The global constructor is empty, and can be removed.
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The methods are only used in inputStream_fileOpen(), and should not be
exported.
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Everybody should use struct input_stream.
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Renamed inputStream.c and inputStream_file.c.
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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.
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The hook input_stream_global_finish() deinitializes global structures
of all input stream implementations.
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Cast playlist_max_length to off_t before comparing it to stat.st_size.
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check_bool() accepts only "0" or "1". The range check is superfluous.
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Again, no CamelCase in the directory name.
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These plugins are not input plugins, they are decoder plugins. No
CamelCase in the directory name.
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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.
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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.
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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().
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These are results from failed merges which I didn't notice.
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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.
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Move sample format dependent code to a separate function.
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Renamed all variables and functions. Add the prefix "mpd_jack_" to
function names.
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We must never pass partial frames. Added assertions to debug this.
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Merge the variables "avail_data" and "avail_frames" into "available".
Both variables are never used at the same time.
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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.
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Don't wait until there is room for the full data chunk passed to
jack_playAudio(). Try to incrementally send as much as possible into
the ring buffer.
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Don't hard-code a frame size of "4" (16 bit stereo), calculate the
sample size from sizeof(*buffer), and create the constant
"frame_size".
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Indent with tabs.
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libsamplerate 0.1.2 didn't have the 32 bit <-> float conversion
routines. Emulate them in case they aren't supported.
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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.
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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.
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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().
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pcm_convert() converted only to 16 bit. To be able to support other
sample sizes, move that 16 bit specific code to a separate function.
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The 24 bit implementation is mostly copy'n'paste of the 16 bit
version, except that the data type is int32_t instead of int16_t.
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Separate code from pcm_utils.c to keep it small and simple.
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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.
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A future patch will implement a 24 bit resampler. To unify code, move
code which can be shared to a separate function.
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Copy from source to destination buffer directly, don't use the
temporary variables "lsample" and "rsample".
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