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af_lavrresample: don't call swr_set_compensation() unless necessary 

This was _always_ called, even if the resampling was static, or the
filter was inserted for format conversion only. This should have been
fine, as I expected the function not to enable resampling when the
compensation is unset, and the source/target rates are the same. But
this is not the case, and it always enables resampling.

So explicitly avoid the call. If we have already called it successfully,
it's better not do avoid it (to overwrite the previous compensation
value), but it will also be cheap/no-op then.

Probably fixes #4716.
This commit is contained in:
wm4 2017-08-12 12:12:52 +02:00
parent c20df5b3e1
commit baead23ea0
1 changed files with 9 additions and 5 deletions
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14
audio/filter/af_lavrresample.c
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@ -76,6 +76,7 @@ struct af_resample {
int allow_detach; int allow_detach;
char **avopts; char **avopts;
double playback_speed; double playback_speed;
bool is_resampling;
struct AVAudioResampleContext *avrctx; struct AVAudioResampleContext *avrctx;
struct mp_audio avrctx_fmt; // output format of avrctx struct mp_audio avrctx_fmt; // output format of avrctx
struct mp_audio pool_fmt; // format used to allocate frames for avrctx output struct mp_audio pool_fmt; // format used to allocate frames for avrctx output
@ -332,6 +333,8 @@ static int configure_lavrr(struct af_instance *af, struct mp_audio *in,
// * Also, the input channel layout must have already been set. // * Also, the input channel layout must have already been set.
avresample_set_channel_mapping(s->avrctx, s->reorder_in); avresample_set_channel_mapping(s->avrctx, s->reorder_in);
s->is_resampling = false;
if (avresample_open(s->avrctx) < 0 || avresample_open(s->avrctx_out) < 0) { if (avresample_open(s->avrctx) < 0 || avresample_open(s->avrctx_out) < 0) {
MP_ERR(af, "Cannot open Libavresample Context. \n"); MP_ERR(af, "Cannot open Libavresample Context. \n");
goto error; goto error;
@ -518,9 +521,7 @@ static int filter(struct af_instance *af, struct mp_audio *in)
struct af_resample *s = af->priv; struct af_resample *s = af->priv;
int new_rate = rate_from_speed(s->in_rate_af, s->playback_speed); int new_rate = rate_from_speed(s->in_rate_af, s->playback_speed);
bool need_reinit = fabs(new_rate / (double)s->in_rate - 1) > 0.01; if (s->avrctx && !(!s->is_resampling && new_rate == s->in_rate)) {
if (s->avrctx) {
AVRational r = av_d2q(s->playback_speed * s->in_rate_af / s->in_rate, AVRational r = av_d2q(s->playback_speed * s->in_rate_af / s->in_rate,
INT_MAX / 2); INT_MAX / 2);
// Essentially, swr/avresample_set_compensation() does 2 things: // Essentially, swr/avresample_set_compensation() does 2 things:
@ -531,10 +532,13 @@ static int filter(struct af_instance *af, struct mp_audio *in)
// feeding it, until the next filter() call. // feeding it, until the next filter() call.
int mult = INT_MAX / 2 / MPMAX(MPMAX(abs(r.num), abs(r.den)), 1); int mult = INT_MAX / 2 / MPMAX(MPMAX(abs(r.num), abs(r.den)), 1);
r = (AVRational){ r.num * mult, r.den * mult }; r = (AVRational){ r.num * mult, r.den * mult };
if (avresample_set_compensation(s->avrctx, r.den - r.num, r.den) < 0) if (avresample_set_compensation(s->avrctx, r.den - r.num, r.den) >= 0) {
need_reinit = true; new_rate = s->in_rate;
s->is_resampling = true;
}
} }
bool need_reinit = fabs(new_rate / (double)s->in_rate - 1) > 0.01;
if (need_reinit && new_rate != s->in_rate) { if (need_reinit && new_rate != s->in_rate) {
// Before reconfiguring, drain the audio that is still buffered // Before reconfiguring, drain the audio that is still buffered
// in the resampler. // in the resampler.