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authorwm4 <wm4@nowhere>2015-01-15 20:10:52 +0100
committerwm4 <wm4@nowhere>2015-01-15 20:13:12 +0100
commit87fe7d878843688a0f530fc9ce5a1a0775c77b9e (patch)
tree185e92cf90cb51604161243f7e02eb977a05ae1d /audio/filter/af_extrastereo.c
parentba0e8b754c4282bae941aa5d30614b2a305f9f5f (diff)
downloadmpv-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.c38
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;
}