avfilter/asubprocess: split filter frame function into read/write part

Signed-off-by: Stefan Westerfeld <stefan@space.twc.de>

libavfilter/af_asubprocess.c

@@ -212,7 +212,6 @@ typedef struct AndioWMarkContext {
     int out_bit_depth;
     SP    *sp;
     void  *sample_buffer;
-    int    nb_samples;
     int    eof;
 } ASubProcessContext;
 
@@ -252,7 +251,7 @@ static av_cold void uninit(AVFilterContext *ctx)
 #endif
 }
 
-#define NB_BUFFER_SAMPLES 1024
+#define NB_BUFFER_SAMPLES 4096
 
 static void write_wav_header (ASubProcessContext *s, int nb_channels, int sample_rate)
 {
@@ -304,7 +303,6 @@ static int config_input(AVFilterLink *inlink)
     s->nb_samples = rubberband_get_samples_required(s->rbs);
     s->first_pts = AV_NOPTS_VALUE;
 #endif
-    s->nb_samples = 1024;
     s->state = STATE_EXPECT_RIFF;
     s->sample_buffer = av_malloc (NB_BUFFER_SAMPLES * inlink->ch_layout.nb_channels * 4 /* max 32 bit */); // TODO: leak
     return 0;
@@ -389,14 +387,9 @@ static void write_samples(ASubProcessContext *s, int32_t *data, int count)
     sp_write(s->sp, s->sample_buffer, sample_size * count);
 }
 
-static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+static int write_frame(ASubProcessContext *s, AVFilterLink *inlink, AVFrame *in)
 {
-    AVFilterContext *ctx = inlink->dst;
-    ASubProcessContext *s = ctx->priv;
-    AVFilterLink *outlink = ctx->outputs[0];
-    AVFrame *out;
-    int ret = 0, nb_samples;
-    bool gen_output;
+    int ret = 0;
 
     write_samples(s, (int32_t *)in->data[0], in->nb_samples * inlink->ch_layout.nb_channels);
     if (s->eof)
@@ -405,6 +398,15 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
             ;
         printf ("eof\n");
     }
+    av_frame_free(&in);
+    return ret;
+}
+
+static int try_read_frame(ASubProcessContext *s, AVFilterLink *outlink)
+{
+    AVFrame *out;
+    int ret;
+
     if (s->state == STATE_EXPECT_RIFF && sp_can_read (s->sp) >= 12)
     {
         char buffer[12];
@@ -442,56 +444,27 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
         }
         s->state = STATE_EXPECT_CHUNK;
     }
-    gen_output = s->state == STATE_IN_DATA_CHUNK;
-    while (gen_output)
+    if (s->state == STATE_IN_DATA_CHUNK)
     {
         int sample_size = s->out_bit_depth / 8;
-        int avail = sp_can_read (s->sp) / sample_size / inlink->ch_layout.nb_channels;
+        int avail = sp_can_read (s->sp) / sample_size / outlink->ch_layout.nb_channels;
         if (avail > NB_BUFFER_SAMPLES)
             avail = NB_BUFFER_SAMPLES;
-        if (avail > 0)
+        if (avail == NB_BUFFER_SAMPLES || (s->eof && avail > 0))
         {
             out = ff_get_audio_buffer(outlink, avail);
             if (!out) {
-                av_frame_free(&in);
                 return AVERROR(ENOMEM);
             }
-            read_samples(s, (int32_t *)out->data[0], avail * inlink->ch_layout.nb_channels);
+            read_samples(s, (int32_t *)out->data[0], avail * outlink->ch_layout.nb_channels);
             ret = ff_filter_frame(outlink, out);
-        }
-        else {
-            gen_output = false;
+            if (ret < 0)
+                return ret;
+            return avail;
         }
     }
-#if 0
-    if (s->first_pts == AV_NOPTS_VALUE)
-        s->first_pts = in->pts;
-
-    rubberband_process(s->rbs, (const float *const *)in->extended_data, in->nb_samples, s->eof);
-    s->nb_samples_in += in->nb_samples;
 
-    nb_samples = rubberband_available(s->rbs);
-    if (nb_samples > 0) {
-        out = ff_get_audio_buffer(outlink, nb_samples);
-        if (!out) {
-            av_frame_free(&in);
-            return AVERROR(ENOMEM);
-        }
-        out->pts = s->first_pts + av_rescale_q(s->nb_samples_out,
-                     (AVRational){ 1, outlink->sample_rate },
-                     outlink->time_base);
-        s->last_pts = out->pts;
-        nb_samples = rubberband_retrieve(s->rbs, (float *const *)out->extended_data, nb_samples);
-        out->nb_samples = nb_samples;
-        ret = ff_filter_frame(outlink, out);
-        s->nb_samples_out += nb_samples;
-    }
-#endif
-
-    av_frame_free(&in);
-    if (ff_inlink_queued_samples(inlink) >= 0)
-        ff_filter_set_ready(ctx, 100);
-    return ret < 0 ? ret : nb_samples;
+    return 0;
 }
 
 static int activate(AVFilterContext *ctx)
@@ -505,7 +478,11 @@ static int activate(AVFilterContext *ctx)
 
     FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
 
-    ret = ff_inlink_consume_samples(inlink, s->nb_samples, s->nb_samples, &in);
+    ret = try_read_frame(s, outlink);
+    if (ret != 0)
+        return ret;
+    ret = ff_inlink_consume_samples(inlink, NB_BUFFER_SAMPLES, NB_BUFFER_SAMPLES, &in);
+    //ret = ff_inlink_consume_frame(inlink, &in);
 
     if (ff_inlink_acknowledge_status(inlink, &status, &pts))
         s->eof |= status == AVERROR_EOF;
@@ -513,10 +490,17 @@ static int activate(AVFilterContext *ctx)
     if (ret < 0)
         return ret;
     if (ret > 0) {
-        ret = filter_frame(inlink, in);
+        ret = write_frame(s, inlink, in);
+
+        if (ff_inlink_queued_samples(inlink) >= 0)
+            ff_filter_set_ready(ctx, 100);
+
         if (ret != 0)
             return ret;
     }
+    ret = try_read_frame(s, outlink);
+    if (ret != 0)
+        return ret;
 
     if (s->eof) {
         // s->last_pts
@@ -529,24 +513,6 @@ static int activate(AVFilterContext *ctx)
     return FFERROR_NOT_READY;
 }
 
-static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
-                           char *res, int res_len, int flags)
-{
-  printf ("TODO: process\n");
-#if 0
-    RubberBandContext *s = ctx->priv;
-    int ret;
-
-    ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
-    if (ret < 0)
-        return ret;
-
-    rubberband_set_time_ratio(s->rbs, 1. / s->tempo);
-    rubberband_set_pitch_scale(s->rbs, s->pitch);
-    s->nb_samples = rubberband_get_samples_required(s->rbs);
-#endif
-    return 0;
-}
 static const AVFilterPad asubprocess_inputs[] = {
     {
         .name          = "default",
@@ -566,5 +532,4 @@ const AVFilter ff_af_asubprocess = {
     FILTER_INPUTS(asubprocess_inputs),
     FILTER_OUTPUTS(ff_audio_default_filterpad),
     FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_S32),
-    .process_command = process_command,
 };