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author | wm4 <wm4@nowhere> | 2015-01-15 20:10:52 +0100 |
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committer | wm4 <wm4@nowhere> | 2015-01-15 20:13:12 +0100 |
commit | 87fe7d878843688a0f530fc9ce5a1a0775c77b9e (patch) | |
tree | 185e92cf90cb51604161243f7e02eb977a05ae1d /audio/filter/af_extrastereo.c | |
parent | ba0e8b754c4282bae941aa5d30614b2a305f9f5f (diff) | |
download | mpv-87fe7d878843688a0f530fc9ce5a1a0775c77b9e.tar.bz2 mpv-87fe7d878843688a0f530fc9ce5a1a0775c77b9e.tar.xz |
audio/filter: switch remaining in-place filters to refcounting
Adds about 7 lines of boilerplate per filter. This could be avoided by
providing a different entrypoint (something like af->filter_inplace),
which would basically mirror the old interface exactly for this kind of
filter. But I feel like it would just be a hack to support all those
old, useless filters better. (The ideal solution would be using a
language that can do closures to provide a compat. wrapper, but
whatever.)
af_bs2b has terribly repetitious code for setting up filter functions
for each format (most of them useless, in addition to bs2b being
useless), so I did something terrible with macros.
af_sinesuppress had commented code for float filtering (maybe it was
broken; it has been commented every since it was added in 2006). Remove
this code.
Diffstat (limited to 'audio/filter/af_extrastereo.c')
-rw-r--r-- | audio/filter/af_extrastereo.c | 38 |
1 files changed, 20 insertions, 18 deletions
diff --git a/audio/filter/af_extrastereo.c b/audio/filter/af_extrastereo.c index 4ebe03a228..4b39793e09 100644 --- a/audio/filter/af_extrastereo.c +++ b/audio/filter/af_extrastereo.c @@ -35,9 +35,6 @@ typedef struct af_extrastereo_s float mul; }af_extrastereo_t; -static int play_s16(struct af_instance* af, struct mp_audio* data, int f); -static int play_float(struct af_instance* af, struct mp_audio* data, int f); - // Initialization and runtime control static int control(struct af_instance* af, int cmd, void* arg) { @@ -49,14 +46,8 @@ static int control(struct af_instance* af, int cmd, void* arg) mp_audio_copy_config(af->data, (struct mp_audio*)arg); mp_audio_force_interleaved_format(af->data); mp_audio_set_num_channels(af->data, 2); - if (af->data->format == AF_FORMAT_FLOAT) - { - af->filter = play_float; - }// else - { + if (af->data->format != AF_FORMAT_FLOAT) mp_audio_set_format(af->data, AF_FORMAT_S16); - af->filter = play_s16; - } return af_test_output(af,(struct mp_audio*)arg); } @@ -65,9 +56,8 @@ static int control(struct af_instance* af, int cmd, void* arg) } // Filter data through filter -static int play_s16(struct af_instance* af, struct mp_audio* data, int f) +static void play_s16(af_extrastereo_t *s, struct mp_audio* data) { - af_extrastereo_t *s = af->priv; register int i = 0; int16_t *a = (int16_t*)data->planes[0]; // Audio data int len = data->samples*data->nch; // Number of samples @@ -83,13 +73,10 @@ static int play_s16(struct af_instance* af, struct mp_audio* data, int f) a[i] = MPCLAMP(l, SHRT_MIN, SHRT_MAX); a[i + 1] = MPCLAMP(r, SHRT_MIN, SHRT_MAX); } - - return 0; } -static int play_float(struct af_instance* af, struct mp_audio* data, int f) +static void play_float(af_extrastereo_t *s, struct mp_audio* data) { - af_extrastereo_t *s = af->priv; register int i = 0; float *a = (float*)data->planes[0]; // Audio data int len = data->samples * data->nch; // Number of samples @@ -105,14 +92,29 @@ static int play_float(struct af_instance* af, struct mp_audio* data, int f) a[i] = af_softclip(l); a[i + 1] = af_softclip(r); } +} - return 0; +static int filter_frame(struct af_instance *af, struct mp_audio *data) +{ + if (!data) + return 0; + if (af_make_writeable(af, data) < 0) { + talloc_free(data); + return -1; + } + if (data->format == AF_FORMAT_FLOAT) { + play_float(af->priv, data); + } else { + play_s16(af->priv, data); + } + af_add_output_frame(af, data); + return 0; } // Allocate memory and set function pointers static int af_open(struct af_instance* af){ af->control=control; - af->filter=play_s16; + af->filter_frame = filter_frame; return AF_OK; } |