Commit Graph

7 Commits

Author SHA1 Message Date
Stefano Pigozzi c8c70dce57 audio: fix af_fmt_seconds_to_bytes
Was missing samplerate
2013-06-16 19:28:04 +02:00
Stefano Pigozzi bff03a181f core: add a spsc ringbuffer implementation
Currently every single AO was implementing it's own ringbuffer, many times
with slightly different semantics. This is an attempt to fix the problem.

I stole some good ideas from ao_portaudio's ringbuffer and went from there.
The main difference is this one stores wpos and rpos which are absolute
positions in an "infinite" buffer. To find the actual position for writing /
reading just apply modulo size.

The producer only modifies wpos while the consumer only modifies rpos. This
makes it pretty easy to reason about and make the operations thread safe by
using barriers (thread safety is guaranteed only in the Single-Producer/Single-
Consumer case).

Also adapted ao_coreaudio to use this ringbuffer.
2013-06-16 18:20:39 +02:00
wm4 9dd9ccbd8d audio: add double sample format
To make this easier, get rid of the direct mapping of the
AF_FORMAT_BITS_MASK bit field to number of bytes. This way we can throw
away the unused AF_FORMAT_48BIT and don't have to add ..._56BIT.
2013-05-12 21:24:57 +02:00
wm4 f7a427676c audio: add some setters for mp_audio, and require filters to use them
mp_audio has some redundant fields. Setters like mp_audio_set_format()
initialize these properly.

Also move the mp_audio struct to a the file audio.c.

We can remove a mysterious line of code from af.c:

    in.format |= af_bits2fmt(in.bps * 8);

I'm not sure if this was ever actually needed, or if it was some kind of
"make it work" quick-fix that works against the way things were supposed
to work. All filters etc. now set the format correctly, so if there ever
was a need for this code, it's definitely gone.
2013-05-12 21:24:54 +02:00
wm4 74ab902dea audio: remove support for native alaw/mulaw/adpcm output
This is considered a worthless feature. Note that alaw/mulaw/adpcm input
is unaffected: such data is handed to libavcodec and "decoded" to linear
PCM.
2012-12-11 00:37:54 +01:00
wm4 4873b32c59 Rename directories, move files (step 2 of 2)
Finish renaming directories and moving files. Adjust all include
statements to make the previous commit compile.

The two commits are separate, because git is bad at tracking renames
and content changes at the same time.

Also take this as an opportunity to remove the separation between
"common" and "mplayer" sources in the Makefile. ("common" used to be
shared between mplayer and mencoder.)
2012-11-12 20:08:18 +01:00
wm4 d4bdd0473d Rename directories, move files (step 1 of 2) (does not compile)
Tis drops the silly lib prefixes, and attempts to organize the tree in
a more logical way. Make the top-level directory less cluttered as
well.

Renames the following directories:
    libaf -> audio/filter
    libao2 -> audio/out
    libvo -> video/out
    libmpdemux -> demux

Split libmpcodecs:
    vf* -> video/filter
    vd*, dec_video.* -> video/decode
    mp_image*, img_format*, ... -> video/
    ad*, dec_audio.* -> audio/decode

libaf/format.* is moved to audio/ - this is similar to how mp_image.*
is located in video/.

Move most top-level .c/.h files to core. (talloc.c/.h is left on top-
level, because it's external.) Park some of the more annoying files
in compat/. Some of these are relicts from the time mplayer used
ffmpeg internals.

sub/ is not split, because it's too much of a mess (subtitle code is
mixed with OSD display and rendering).

Maybe the organization of core is not ideal: it mixes playback core
(like mplayer.c) and utility helpers (like bstr.c/h). Should the need
arise, the playback core will be moved somewhere else, while core
contains all helper and common code.
2012-11-12 20:06:14 +01:00