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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_lavrresample.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_lavrresample.c')
-rw-r--r-- | audio/filter/af_lavrresample.c | 32 |
1 files changed, 18 insertions, 14 deletions
diff --git a/audio/filter/af_lavrresample.c b/audio/filter/af_lavrresample.c index 860e5a52d3..a6abd32aed 100644 --- a/audio/filter/af_lavrresample.c +++ b/audio/filter/af_lavrresample.c @@ -244,8 +244,10 @@ static int control(struct af_instance *af, int cmd, void *arg) if (af_to_avformat(in->format) == AV_SAMPLE_FMT_NONE) mp_audio_set_format(in, AF_FORMAT_FLOAT_NE); + mp_audio_force_interleaved_format(in); if (af_to_avformat(out->format) == AV_SAMPLE_FMT_NONE) mp_audio_set_format(out, in->format); + mp_audio_force_interleaved_format(out); af->mul = (double) (out->rate * out->nch) / (in->rate * in->nch); af->delay = out->nch * s->opts.filter_size / FFMIN(af->mul, 1); @@ -306,28 +308,30 @@ static struct mp_audio *play(struct af_instance *af, struct mp_audio *data) struct mp_audio *out = af->data; - int in_size = data->len; - int in_samples = in_size / (data->bps * data->nch); + int in_samples = data->samples; + int in_size = data->samples * data->sstride; int out_samples = avresample_available(s->avrctx) + av_rescale_rnd(get_delay(s) + in_samples, s->ctx.out_rate, s->ctx.in_rate, AV_ROUND_UP); - int out_size = out->bps * out_samples * out->nch; + int out_size = out_samples * out->sstride; - if (talloc_get_size(out->audio) < out_size) - out->audio = talloc_realloc_size(out, out->audio, out_size); + if (talloc_get_size(out->planes[0]) < out_size) + out->planes[0] = talloc_realloc_size(out, out->planes[0], out_size); af->delay = out->bps * av_rescale_rnd(get_delay(s), s->ctx.out_rate, s->ctx.in_rate, AV_ROUND_UP); -#if !USE_SET_CHANNEL_MAPPING - reorder_channels(data->audio, s->reorder_in, data->bps, data->nch, in_samples); +#if USE_SET_CHANNEL_MAPPING + (void)in_size; +#else + reorder_channels(data->planes[0], s->reorder_in, data->bps, data->nch, in_samples); #endif if (out_samples) { out_samples = avresample_convert(s->avrctx, - (uint8_t **) &out->audio, out_size, out_samples, - (uint8_t **) &in->audio, in_size, in_samples); + (uint8_t **) out->planes, out_size, out_samples, + (uint8_t **) in->planes, in_size, in_samples); if (out_samples < 0) return NULL; // error } @@ -338,16 +342,16 @@ static struct mp_audio *play(struct af_instance *af, struct mp_audio *data) if (needs_reorder(s->reorder_out, out->nch)) { if (talloc_get_size(s->reorder_buffer) < out_size) s->reorder_buffer = talloc_realloc_size(s, s->reorder_buffer, out_size); - data->audio = s->reorder_buffer; + data->planes[0] = s->reorder_buffer; out_samples = avresample_convert(s->avrctx_out, - (uint8_t **) &data->audio, out_size, out_samples, - (uint8_t **) &out->audio, out_size, out_samples); + (uint8_t **) data->planes, out_size, out_samples, + (uint8_t **) out->planes, out_size, out_samples); } #else - reorder_channels(data->audio, s->reorder_out, out->bps, out->nch, out_samples); + reorder_channels(data->planes[0], s->reorder_out, out->bps, out->nch, out_samples); #endif - data->len = out->bps * out_samples * out->nch; + data->samples = out_samples; return data; } |