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author | wm4 <wm4@nowhere> | 2013-11-10 23:11:40 +0100 |
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committer | wm4 <wm4@nowhere> | 2013-11-12 23:16:31 +0100 |
commit | d2e7467eb203d3a34bc1111564c7058b5e9c6b12 (patch) | |
tree | 9285523821c8710a0609f47e3ee923a20d038826 /audio/filter/af_surround.c | |
parent | b2d4b5ee43206f8c4491b3af1c24fedd35dbdc31 (diff) | |
download | mpv-d2e7467eb203d3a34bc1111564c7058b5e9c6b12.tar.bz2 mpv-d2e7467eb203d3a34bc1111564c7058b5e9c6b12.tar.xz |
audio/filter: prepare filter chain for non-interleaved audio
Based on earlier work by Stefano Pigozzi.
There are 2 changes:
1. Instead of mp_audio.audio, mp_audio.planes[0] must be used.
2. mp_audio.len used to contain the size of the audio in bytes. Now
mp_audio.samples must be used. (Where 1 sample is the smallest unit
of audio that covers all channels.)
Also, some filters need changes to reject non-interleaved formats
properly.
Nothing uses the non-interleaved features yet, but this is needed so
that things don't just break when doing so.
Diffstat (limited to 'audio/filter/af_surround.c')
-rw-r--r-- | audio/filter/af_surround.c | 11 |
1 files changed, 5 insertions, 6 deletions
diff --git a/audio/filter/af_surround.c b/audio/filter/af_surround.c index e584e6505a..5ee45d126f 100644 --- a/audio/filter/af_surround.c +++ b/audio/filter/af_surround.c @@ -145,7 +145,7 @@ static int control(struct af_instance* af, int cmd, void* arg) static void uninit(struct af_instance* af) { if(af->data) - free(af->data->audio); + free(af->data->planes[0]); free(af->data); free(af->setup); } @@ -165,9 +165,9 @@ static float steering_matrix[][12] = { static struct mp_audio* play(struct af_instance* af, struct mp_audio* data){ af_surround_t* s = (af_surround_t*)af->setup; float* m = steering_matrix[0]; - float* in = data->audio; // Input audio data + float* in = data->planes[0]; // Input audio data float* out = NULL; // Output audio data - float* end = in + data->len / sizeof(float); // Loop end + float* end = in + data->samples * data->nch; int i = s->i; // Filter queue index int ri = s->ri; // Read index for delay queue int wi = s->wi; // Write index for delay queue @@ -175,7 +175,7 @@ static struct mp_audio* play(struct af_instance* af, struct mp_audio* data){ if (AF_OK != RESIZE_LOCAL_BUFFER(af, data)) return NULL; - out = af->data->audio; + out = af->data->planes[0]; while(in < end){ /* Dominance: @@ -237,8 +237,7 @@ static struct mp_audio* play(struct af_instance* af, struct mp_audio* data){ s->i = i; s->ri = ri; s->wi = wi; // Set output data - data->audio = af->data->audio; - data->len *= 2; + data->planes[0] = af->data->planes[0]; mp_audio_set_channels_old(data, af->data->nch); return data; |