af_lavrresample: remove unnecessary indirections

Not sure why struct af_resample_opts even exists. It seems useful to
group the fields set by user options. But storing the current format
conversion parameters doesn't seem very elegant, and having a separate
instance in the "ctx" field isn't helpful either.

1 files changed, 30 insertions, 35 deletions

diff --git a/audio/filter/af_lavrresample.c b/audio/filter/af_lavrresample.c
index f4669af343..d651c10f4c 100644
--- a/audio/filter/af_lavrresample.c
+++ b/audio/filter/af_lavrresample.c
@@ -66,14 +66,6 @@ struct af_resample_opts {
     int linear;
     double cutoff;
     int normalize;
-
-    int in_rate_af; // filter input sample rate
-    int in_rate;    // actual rate (used by lavr), adjusted for playback speed
-    int in_format;
-    struct mp_chmap in_channels;
-    int out_rate;
-    int out_format;
-    struct mp_chmap out_channels;
 };
 
 struct af_resample {
@@ -85,19 +77,26 @@ struct af_resample {
     struct mp_audio pool_fmt; // format used to allocate frames for avrctx output
     struct mp_audio pre_out_fmt; // format before final conversion (S24)
     struct AVAudioResampleContext *avrctx_out; // for output channel reordering
-    struct af_resample_opts ctx;   // opts in the context
     struct af_resample_opts opts;  // opts requested by the user
     // At least libswresample keeps a pointer around for this:
     int reorder_in[MP_NUM_CHANNELS];
     int reorder_out[MP_NUM_CHANNELS];
     struct mp_audio_pool *reorder_buffer;
+
+    int in_rate_af; // filter input sample rate
+    int in_rate;    // actual rate (used by lavr), adjusted for playback speed
+    int in_format;
+    struct mp_chmap in_channels;
+    int out_rate;
+    int out_format;
+    struct mp_chmap out_channels;
 };
 
 #if HAVE_LIBAVRESAMPLE
 static double get_delay(struct af_resample *s)
 {
-    return avresample_get_delay(s->avrctx) / (double)s->ctx.in_rate +
-           avresample_available(s->avrctx) / (double)s->ctx.out_rate;
+    return avresample_get_delay(s->avrctx) / (double)s->in_rate +
+           avresample_available(s->avrctx) / (double)s->out_rate;
 }
 static void drop_all_output(struct af_resample *s)
 {
@@ -110,7 +109,7 @@ static int get_out_samples(struct af_resample *s, int in_samples)
 #else
 static double get_delay(struct af_resample *s)
 {
-    int64_t base = s->ctx.in_rate * (int64_t)s->ctx.out_rate;
+    int64_t base = s->in_rate * (int64_t)s->out_rate;
     return swr_get_delay(s->avrctx, base) / (double)base;
 }
 static void drop_all_output(struct af_resample *s)
@@ -122,8 +121,8 @@ static int get_out_samples(struct af_resample *s, int in_samples)
 #if LIBSWRESAMPLE_VERSION_MAJOR > 1 || LIBSWRESAMPLE_VERSION_MINOR >= 2
     return swr_get_out_samples(s->avrctx, in_samples);
 #else
-    return av_rescale_rnd(in_samples, s->ctx.out_rate, s->ctx.in_rate, AV_ROUND_UP)
-           + swr_get_delay(s->avrctx, s->ctx.out_rate);
+    return av_rescale_rnd(in_samples, s->out_rate, s->in_rate, AV_ROUND_UP)
+           + swr_get_delay(s->avrctx, s->out_rate);
 #endif
 }
 #endif
@@ -216,23 +215,19 @@ static int configure_lavrr(struct af_instance *af, struct mp_audio *in,
         out_samplefmtp == AV_SAMPLE_FMT_NONE)
         goto error;
 
-    s->ctx.out_rate    = out->rate;
-    s->ctx.in_rate_af  = in->rate;
-    s->ctx.in_rate     = rate_from_speed(in->rate, s->playback_speed);
-    s->ctx.out_format  = out->format;
-    s->ctx.in_format   = in->format;
-    s->ctx.out_channels= out->channels;
-    s->ctx.in_channels = in->channels;
-    s->ctx.filter_size = s->opts.filter_size;
-    s->ctx.phase_shift = s->opts.phase_shift;
-    s->ctx.linear      = s->opts.linear;
-    s->ctx.cutoff      = s->opts.cutoff;
+    s->out_rate    = out->rate;
+    s->in_rate_af  = in->rate;
+    s->in_rate     = rate_from_speed(in->rate, s->playback_speed);
+    s->out_format  = out->format;
+    s->in_format   = in->format;
+    s->out_channels= out->channels;
+    s->in_channels = in->channels;
 
-    av_opt_set_int(s->avrctx, "filter_size",        s->ctx.filter_size, 0);
-    av_opt_set_int(s->avrctx, "phase_shift",        s->ctx.phase_shift, 0);
-    av_opt_set_int(s->avrctx, "linear_interp",      s->ctx.linear, 0);
+    av_opt_set_int(s->avrctx, "filter_size",        s->opts.filter_size, 0);
+    av_opt_set_int(s->avrctx, "phase_shift",        s->opts.phase_shift, 0);
+    av_opt_set_int(s->avrctx, "linear_interp",      s->opts.linear, 0);
 
-    av_opt_set_double(s->avrctx, "cutoff",          s->ctx.cutoff, 0);
+    av_opt_set_double(s->avrctx, "cutoff",          s->opts.cutoff, 0);
 
 #if HAVE_LIBSWRESAMPLE
     av_opt_set_double(s->avrctx, "rematrix_maxval", s->opts.normalize ? 1 : 1000, 0);
@@ -304,8 +299,8 @@ static int configure_lavrr(struct af_instance *af, struct mp_audio *in,
     // Real conversion; output is input to avrctx_out.
     av_opt_set_int(s->avrctx, "in_channel_layout",  in_ch_layout, 0);
     av_opt_set_int(s->avrctx, "out_channel_layout", out_ch_layout, 0);
-    av_opt_set_int(s->avrctx, "in_sample_rate",     s->ctx.in_rate, 0);
-    av_opt_set_int(s->avrctx, "out_sample_rate",    s->ctx.out_rate, 0);
+    av_opt_set_int(s->avrctx, "in_sample_rate",     s->in_rate, 0);
+    av_opt_set_int(s->avrctx, "out_sample_rate",    s->out_rate, 0);
     av_opt_set_int(s->avrctx, "in_sample_fmt",      in_samplefmt, 0);
     av_opt_set_int(s->avrctx, "out_sample_fmt",     out_samplefmtp, 0);
 
@@ -319,8 +314,8 @@ static int configure_lavrr(struct af_instance *af, struct mp_audio *in,
     av_opt_set_int(s->avrctx_out, "out_channel_layout", fake_out_ch_layout, 0);
     av_opt_set_int(s->avrctx_out, "in_sample_fmt",      out_samplefmtp, 0);
     av_opt_set_int(s->avrctx_out, "out_sample_fmt",     out_samplefmt, 0);
-    av_opt_set_int(s->avrctx_out, "in_sample_rate",     s->ctx.out_rate, 0);
-    av_opt_set_int(s->avrctx_out, "out_sample_rate",    s->ctx.out_rate, 0);
+    av_opt_set_int(s->avrctx_out, "in_sample_rate",     s->out_rate, 0);
+    av_opt_set_int(s->avrctx_out, "out_sample_rate",    s->out_rate, 0);
 
     // API has weird requirements, quoting avresample.h:
     //  * This function can only be called when the allocated context is not open.
@@ -391,8 +386,8 @@ static int control(struct af_instance *af, int cmd, void *arg)
         return AF_OK;
     case AF_CONTROL_SET_PLAYBACK_SPEED_RESAMPLE: {
         s->playback_speed = *(double *)arg;
-        int new_rate = rate_from_speed(s->ctx.in_rate_af, s->playback_speed);
-        if (new_rate != s->ctx.in_rate && s->avrctx && af->fmt_out.format) {
+        int new_rate = rate_from_speed(s->in_rate_af, s->playback_speed);
+        if (new_rate != s->in_rate && s->avrctx && af->fmt_out.format) {
             // Before reconfiguring, drain the audio that is still buffered
             // in the resampler.
             af->filter_frame(af, NULL);