af_lavcac3enc: automatically configure most encoder parameters

Instead of hardcoding what the libavcodec ac3 encoder expects, configure
it based on the AVCodec fields.

Unfortunately, it doesn't export the list of sample rates, so that is
done manually. This commit actually fixes the rate always to 48Khz. I
don't even know whether the other rates worked. (Possibly did, but
they'd still change the spdif parameters, and would work differently
from ad_spdif.c.)

1 files changed, 57 insertions, 29 deletions

diff --git a/audio/filter/af_lavcac3enc.c b/audio/filter/af_lavcac3enc.c
index de7e8d0090..a5c6a3632c 100644
--- a/audio/filter/af_lavcac3enc.c
+++ b/audio/filter/af_lavcac3enc.c
@@ -34,6 +34,7 @@
 #include "common/common.h"
 #include "af.h"
 #include "audio/audio_buffer.h"
+#include "audio/chmap_sel.h"
 #include "audio/fmt-conversion.h"
 
 
@@ -54,13 +55,49 @@ typedef struct af_ac3enc_s {
     struct mp_audio *pending;   // unconsumed input data
     int in_samples;     // samples of input per AC3 frame
     int out_samples;    // upper bound on encoded output per AC3 frame
-    int in_sampleformat;
 
     int cfg_add_iec61937_header;
     int cfg_bit_rate;
     int cfg_min_channel_num;
 } af_ac3enc_t;
 
+// fmt carries the input format. Change it to the best next-possible format
+// the encoder likely accepts.
+static void select_encode_format(AVCodecContext *c, struct mp_audio *fmt)
+{
+    int formats[AF_FORMAT_COUNT];
+    af_get_best_sample_formats(fmt->format, formats);
+
+    for (int n = 0; formats[n]; n++) {
+        const enum AVSampleFormat *lf = c->codec->sample_fmts;
+        for (int i = 0; lf && lf[i] != AV_SAMPLE_FMT_NONE; i++) {
+            int mpfmt = af_from_avformat(lf[i]);
+            if (mpfmt && mpfmt == formats[n]) {
+                mp_audio_set_format(fmt, mpfmt);
+                goto done_fmt;
+            }
+        }
+    }
+done_fmt: ;
+
+    int rate =
+        af_select_best_samplerate(fmt->rate, c->codec->supported_samplerates);
+    if (rate > 0)
+        fmt->rate = rate;
+
+    struct mp_chmap_sel sel = {0};
+    const uint64_t *lch = c->codec->channel_layouts;
+    for (int n = 0; lch && lch[n]; n++) {
+        struct mp_chmap chmap = {0};
+        mp_chmap_from_lavc(&chmap, lch[n]);
+        mp_chmap_sel_add_map(&sel, &chmap);
+    }
+    struct mp_chmap res = fmt->channels;
+    mp_chmap_sel_adjust(&sel, &res);
+    if (!mp_chmap_is_empty(&res))
+        mp_audio_set_channels(fmt, &res);
+}
+
 // Initialization and runtime control
 static int control(struct af_instance *af, int cmd, void *arg)
 {
@@ -76,23 +113,17 @@ static int control(struct af_instance *af, int cmd, void *arg)
         if (!af_fmt_is_pcm(in->format) || in->nch < s->cfg_min_channel_num)
             return AF_DETACH;
 
-        mp_audio_set_format(in, s->in_sampleformat);
+        // At least currently, the AC3 encoder doesn't export sample rates.
+        in->rate = 48000;
+        select_encode_format(s->lavc_actx, in);
 
-        if (in->rate != 48000 && in->rate != 44100 && in->rate != 32000)
-            in->rate = 48000;
         af->data->rate = in->rate;
-
-        mp_chmap_reorder_to_lavc(&in->channels);
-        if (in->nch > AC3_MAX_CHANNELS)
-            mp_audio_set_num_channels(in, AC3_MAX_CHANNELS);
-
         mp_audio_set_format(af->data, AF_FORMAT_S_AC3);
         mp_audio_set_num_channels(af->data, 2);
 
         if (!mp_audio_config_equals(in, &orig_in))
             return AF_FALSE;
 
-        s->in_samples = AC3_FRAME_SIZE;
         if (s->cfg_add_iec61937_header) {
             s->out_samples = AC3_FRAME_SIZE;
         } else {
@@ -100,8 +131,6 @@ static int control(struct af_instance *af, int cmd, void *arg)
         }
 
         mp_audio_copy_config(s->input, in);
-        mp_audio_realloc(s->input, s->in_samples);
-        s->input->samples = 0;
 
         talloc_free(s->pending);
         s->pending = NULL;
@@ -117,6 +146,7 @@ static int control(struct af_instance *af, int cmd, void *arg)
             avcodec_close(s->lavc_actx);
 
             // Put sample parameters
+            s->lavc_actx->sample_fmt = af_to_avformat(in->format);
             s->lavc_actx->channels = in->nch;
             s->lavc_actx->channel_layout = mp_chmap_to_lavc(&in->channels);
             s->lavc_actx->sample_rate = in->rate;
@@ -126,12 +156,15 @@ static int control(struct af_instance *af, int cmd, void *arg)
                 MP_ERR(af, "Couldn't open codec %s, br=%d.\n", "ac3", bit_rate);
                 return AF_ERROR;
             }
+
+            if (s->lavc_actx->frame_size < 1) {
+                MP_ERR(af, "encoder didn't specify input frame size\n");
+                return AF_ERROR;
+            }
         }
-        if (s->lavc_actx->frame_size != AC3_FRAME_SIZE) {
-            MP_ERR(af, "unexpected ac3 encoder frame size %d\n",
-                   s->lavc_actx->frame_size);
-            return AF_ERROR;
-        }
+        s->in_samples = s->lavc_actx->frame_size;
+        mp_audio_realloc(s->input, s->in_samples);
+        s->input->samples = 0;
         return AF_OK;
     }
     }
@@ -297,20 +330,15 @@ static int af_open(struct af_instance* af){
         MP_ERR(af, "Audio LAVC, couldn't allocate context!\n");
         return AF_ERROR;
     }
-    const enum AVSampleFormat *fmts = s->lavc_acodec->sample_fmts;
-    for (int i = 0; fmts[i] != AV_SAMPLE_FMT_NONE; i++) {
-        s->in_sampleformat = af_from_avformat(fmts[i]);
-        if (s->in_sampleformat) {
-            s->lavc_actx->sample_fmt = fmts[i];
-            break;
-        }
-    }
-    if (!s->in_sampleformat) {
-        MP_ERR(af, "Audio LAVC, encoder doesn't "
-               "support expected sample formats!\n");
+    // For this one, we require the decoder to expert lists of all supported
+    // parameters. (Not all decoders do that, but the ones we're interested
+    // in do.)
+    if (!s->lavc_acodec->sample_fmts ||
+        !s->lavc_acodec->channel_layouts)
+    {
+        MP_ERR(af, "Audio encoder doesn't list supported parameters.\n");
         return AF_ERROR;
     }
-    MP_VERBOSE(af, "in sample format: %s\n", af_fmt_to_str(s->in_sampleformat));
 
     s->input = talloc_zero(s, struct mp_audio);