Merge commit '6f243b17c537646b894857d43dfdac65f85ab377'

[ffmpeg] / libavresample / resample.c

diff --git a/libavresample/resample.c b/libavresample/resample.c
index 79275950b1fcd10de77e5ae6877240a2884fb4c5..dc87095bc398fad6e2c58472250f161dd0d907e1 100644 (file)
--- a/libavresample/resample.c
+++ b/libavresample/resample.c
@@ -42,7 +42,6 @@ struct ResampleContext {
     int linear;
     enum AVResampleFilterType filter_type;
     int kaiser_beta;
-    double factor;
     void (*set_filter)(void *filter, double *tab, int phase, int tap_count);
     void (*resample_one)(struct ResampleContext *c, void *dst0,
                          int dst_index, const void *src0,
@@ -52,6 +51,8 @@ struct ResampleContext {
     int padding_size;
     int initial_padding_filled;
     int initial_padding_samples;
+    int final_padding_filled;
+    int final_padding_samples;
 };
 
 
@@ -92,10 +93,10 @@ static double bessel(double x)
 }
 
 /* Build a polyphase filterbank. */
-static int build_filter(ResampleContext *c)
+static int build_filter(ResampleContext *c, double factor)
 {
     int ph, i;
-    double x, y, w, factor;
+    double x, y, w;
     double *tab;
     int tap_count    = c->filter_length;
     int phase_count  = 1 << c->phase_shift;
@@ -105,9 +106,6 @@ static int build_filter(ResampleContext *c)
     if (!tab)
         return AVERROR(ENOMEM);
 
-    /* if upsampling, only need to interpolate, no filter */
-    factor = FFMIN(c->factor, 1.0);
-
     for (ph = 0; ph < phase_count; ph++) {
         double norm = 0;
         for (i = 0; i < tap_count; i++) {
@@ -174,7 +172,6 @@ ResampleContext *ff_audio_resample_init(AVAudioResampleContext *avr)
     c->phase_shift   = avr->phase_shift;
     c->phase_mask    = phase_count - 1;
     c->linear        = avr->linear_interp;
-    c->factor        = factor;
     c->filter_length = FFMAX((int)ceil(avr->filter_size / factor), 1);
     c->filter_type   = avr->filter_type;
     c->kaiser_beta   = avr->kaiser_beta;
@@ -207,7 +204,7 @@ ResampleContext *ff_audio_resample_init(AVAudioResampleContext *avr)
     if (!c->filter_bank)
         goto error;
 
-    if (build_filter(c) < 0)
+    if (build_filter(c, factor) < 0)
         goto error;
 
     memcpy(&c->filter_bank[(c->filter_length * phase_count + 1) * felem_size],
@@ -433,27 +430,57 @@ int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src)
         ret = ff_audio_data_combine(c->buffer, in_leftover, src, 0, in_samples);
         if (ret < 0)
             return ret;
-    } else if (!in_leftover) {
+    } else if (in_leftover <= c->final_padding_samples) {
         /* no remaining samples to flush */
         return 0;
-    } else {
-        /* TODO: pad buffer to flush completely */
     }
 
     if (!c->initial_padding_filled) {
         int bps = av_get_bytes_per_sample(c->avr->internal_sample_fmt);
         int i;
 
-        if (c->buffer->nb_samples < 2 * c->padding_size)
+        if (src && c->buffer->nb_samples < 2 * c->padding_size)
             return 0;
 
         for (i = 0; i < c->padding_size; i++)
-            for (ch = 0; ch < c->buffer->channels; ch++)
-                memcpy(c->buffer->data[ch] + bps * i,
-                       c->buffer->data[ch] + bps * (2 * c->padding_size - i), bps);
+            for (ch = 0; ch < c->buffer->channels; ch++) {
+                if (c->buffer->nb_samples > 2 * c->padding_size - i) {
+                    memcpy(c->buffer->data[ch] + bps * i,
+                           c->buffer->data[ch] + bps * (2 * c->padding_size - i), bps);
+                } else {
+                    memset(c->buffer->data[ch] + bps * i, 0, bps);
+                }
+            }
         c->initial_padding_filled = 1;
     }
 
+    if (!src && !c->final_padding_filled) {
+        int bps = av_get_bytes_per_sample(c->avr->internal_sample_fmt);
+        int i;
+
+        ret = ff_audio_data_realloc(c->buffer, in_samples + c->padding_size);
+        if (ret < 0) {
+            av_log(c->avr, AV_LOG_ERROR, "Error reallocating resampling buffer\n");
+            return AVERROR(ENOMEM);
+        }
+
+        for (i = 0; i < c->padding_size; i++)
+            for (ch = 0; ch < c->buffer->channels; ch++) {
+                if (in_leftover > i) {
+                    memcpy(c->buffer->data[ch] + bps * (in_leftover + i),
+                           c->buffer->data[ch] + bps * (in_leftover - i - 1),
+                           bps);
+                } else {
+                    memset(c->buffer->data[ch] + bps * (in_leftover + i),
+                           0, bps);
+                }
+            }
+        c->buffer->nb_samples   += c->padding_size;
+        c->final_padding_samples = c->padding_size;
+        c->final_padding_filled  = 1;
+    }
+
+
     /* calculate output size and reallocate output buffer if needed */
     /* TODO: try to calculate this without the dummy resample() run */
     if (!dst->read_only && dst->allow_realloc) {