Merge remote-tracking branch 'qatar/master'

* qatar/master:
  mss3: use standard zigzag table
  mss3: split DSP functions that are used in MTS2(MSS4) into separate file
  motion-test: do not use getopt()
  tcp: add initial timeout limit for incoming connections
  configure: Change the rdtsc check to a linker check
  avconv: propagate fatal errors from lavfi.
  lavfi: add error handling to filter_samples().
  fate-run: make avconv() properly deal with multiple inputs.
  asplit: don't leak the input buffer.
  af_resample: fix request_frame() behavior.
  af_asyncts: fix request_frame() behavior.
  libx264: support aspect ratio switching
  matroskadec: honor error_recognition when encountering unknown elements.
  lavr: resampling: add support for s32p, fltp, and dblp internal sample formats
  lavr: resampling: add filter type and Kaiser window beta to AVOptions
  lavr: Use AV_SAMPLE_FMT_NONE to auto-select the internal sample format
  lavr: mix: validate internal sample format in ff_audio_mix_init()

Conflicts:
	ffmpeg.c
	ffplay.c
	libavcodec/libx264.c
	libavfilter/audio.c
	libavfilter/split.c
	libavformat/tcp.c
	tests/fate-run.sh

Merged-by: Michael Niedermayer <michaelni@gmx.at>

libavcodec/Makefile

@@ -327,7 +327,7 @@ OBJS-$(CONFIG_MSMPEG4V3_ENCODER)       += msmpeg4.o msmpeg4enc.o msmpeg4data.o \
                                           h263dec.o h263.o ituh263dec.o        \
                                           mpeg4videodec.o
 OBJS-$(CONFIG_MSRLE_DECODER)           += msrle.o msrledec.o
-OBJS-$(CONFIG_MSA1_DECODER)            += mss3.o
+OBJS-$(CONFIG_MSA1_DECODER)            += mss3.o mss34dsp.o
 OBJS-$(CONFIG_MSS1_DECODER)            += mss1.o
 OBJS-$(CONFIG_MSVIDEO1_DECODER)        += msvideo1.o
 OBJS-$(CONFIG_MSVIDEO1_ENCODER)        += msvideo1enc.o elbg.o

libavcodec/libx264.c

@@ -175,10 +175,10 @@ static int X264_frame(AVCodecContext *ctx, AVPacket *pkt, const AVFrame *frame,
             x4->params.b_tff = frame->top_field_first;
             x264_encoder_reconfig(x4->enc, &x4->params);
         }
-        if (x4->params.vui.i_sar_height != ctx->sample_aspect_ratio.den
+        if (x4->params.vui.i_sar_height != ctx->sample_aspect_ratio.den ||
-         || x4->params.vui.i_sar_width != ctx->sample_aspect_ratio.num) {
+            x4->params.vui.i_sar_width  != ctx->sample_aspect_ratio.num) {
             x4->params.vui.i_sar_height = ctx->sample_aspect_ratio.den;
-            x4->params.vui.i_sar_width = ctx->sample_aspect_ratio.num;
+            x4->params.vui.i_sar_width  = ctx->sample_aspect_ratio.num;
             x264_encoder_reconfig(x4->enc, &x4->params);
         }
     }

libavcodec/motion-test.c

@@ -119,15 +119,9 @@ int main(int argc, char **argv)
     int flags[2] = { AV_CPU_FLAG_MMX, AV_CPU_FLAG_MMX2 };
     int flags_size = HAVE_MMX2 ? 2 : 1;
 
-    for(;;) {
+    if (argc > 1) {
-        c = getopt(argc, argv, "h");
+        help();
-        if (c == -1)
+        return 1;
-            break;
-        switch(c) {
-        case 'h':
-            help();
-            return 1;
-        }
     }
 
     printf("ffmpeg motion test\n");

libavcodec/mss3.c

@@ -26,6 +26,8 @@
 
 #include "avcodec.h"
 #include "bytestream.h"
+#include "dsputil.h"
+#include "mss34dsp.h"
 
 #define HEADER_SIZE 27
 
@@ -119,39 +121,6 @@ typedef struct MSS3Context {
     int              hblock[16 * 16];
 } MSS3Context;
 
-static const uint8_t mss3_luma_quant[64] = {
-    16,  11,  10,  16,  24,  40,  51,  61,
-    12,  12,  14,  19,  26,  58,  60,  55,
-    14,  13,  16,  24,  40,  57,  69,  56,
-    14,  17,  22,  29,  51,  87,  80,  62,
-    18,  22,  37,  56,  68, 109, 103,  77,
-    24,  35,  55,  64,  81, 104, 113,  92,
-    49,  64,  78,  87, 103, 121, 120, 101,
-    72,  92,  95,  98, 112, 100, 103,  99
-};
-
-static const uint8_t mss3_chroma_quant[64] = {
-    17, 18, 24, 47, 99, 99, 99, 99,
-    18, 21, 26, 66, 99, 99, 99, 99,
-    24, 26, 56, 99, 99, 99, 99, 99,
-    47, 66, 99, 99, 99, 99, 99, 99,
-    99, 99, 99, 99, 99, 99, 99, 99,
-    99, 99, 99, 99, 99, 99, 99, 99,
-    99, 99, 99, 99, 99, 99, 99, 99,
-    99, 99, 99, 99, 99, 99, 99, 99
-};
-
-static const uint8_t zigzag_scan[64] = {
-    0,   1,  8, 16,  9,  2,  3, 10,
-    17, 24, 32, 25, 18, 11,  4,  5,
-    12, 19, 26, 33, 40, 48, 41, 34,
-    27, 20, 13,  6,  7, 14, 21, 28,
-    35, 42, 49, 56, 57, 50, 43, 36,
-    29, 22, 15, 23, 30, 37, 44, 51,
-    58, 59, 52, 45, 38, 31, 39, 46,
-    53, 60, 61, 54, 47, 55, 62, 63
-};
-
 
 static void model2_reset(Model2 *m)
 {
@@ -578,7 +547,7 @@ static int decode_dct(RangeCoder *c, DCTBlockCoder *bc, int *block,
         if (!sign)
             val = -val;
 
-        zz_pos = zigzag_scan[pos];
+        zz_pos = ff_zigzag_direct[pos];
         block[zz_pos] = val * bc->qmat[zz_pos];
         pos++;
     }
@@ -586,58 +555,6 @@ static int decode_dct(RangeCoder *c, DCTBlockCoder *bc, int *block,
     return pos == 64 ? 0 : -1;
 }
 
-#define DCT_TEMPLATE(blk, step, SOP, shift)                         \
-    const int t0 = -39409 * blk[7 * step] -  58980 * blk[1 * step]; \
-    const int t1 =  39410 * blk[1 * step] -  58980 * blk[7 * step]; \
-    const int t2 = -33410 * blk[5 * step] - 167963 * blk[3 * step]; \
-    const int t3 =  33410 * blk[3 * step] - 167963 * blk[5 * step]; \
-    const int t4 =          blk[3 * step] +          blk[7 * step]; \
-    const int t5 =          blk[1 * step] +          blk[5 * step]; \
-    const int t6 =  77062 * t4            +  51491 * t5;            \
-    const int t7 =  77062 * t5            -  51491 * t4;            \
-    const int t8 =  35470 * blk[2 * step] -  85623 * blk[6 * step]; \
-    const int t9 =  35470 * blk[6 * step] +  85623 * blk[2 * step]; \
-    const int tA = SOP(blk[0 * step] - blk[4 * step]);              \
-    const int tB = SOP(blk[0 * step] + blk[4 * step]);              \
-                                                                    \
-    blk[0 * step] = (  t1 + t6  + t9 + tB) >> shift;                \
-    blk[1 * step] = (  t3 + t7  + t8 + tA) >> shift;                \
-    blk[2 * step] = (  t2 + t6  - t8 + tA) >> shift;                \
-    blk[3 * step] = (  t0 + t7  - t9 + tB) >> shift;                \
-    blk[4 * step] = (-(t0 + t7) - t9 + tB) >> shift;                \
-    blk[5 * step] = (-(t2 + t6) - t8 + tA) >> shift;                \
-    blk[6 * step] = (-(t3 + t7) + t8 + tA) >> shift;                \
-    blk[7 * step] = (-(t1 + t6) + t9 + tB) >> shift;                \
-
-#define SOP_ROW(a) ((a) << 16) + 0x2000
-#define SOP_COL(a) ((a + 32) << 16)
-
-static void dct_put(uint8_t *dst, int stride, int *block)
-{
-    int i, j;
-    int *ptr;
-
-    ptr = block;
-    for (i = 0; i < 8; i++) {
-        DCT_TEMPLATE(ptr, 1, SOP_ROW, 13);
-        ptr += 8;
-    }
-
-    ptr = block;
-    for (i = 0; i < 8; i++) {
-        DCT_TEMPLATE(ptr, 8, SOP_COL, 22);
-        ptr++;
-    }
-
-    ptr = block;
-    for (j = 0; j < 8; j++) {
-        for (i = 0; i < 8; i++)
-            dst[i] = av_clip_uint8(ptr[i] + 128);
-        dst += stride;
-        ptr += 8;
-    }
-}
-
 static void decode_dct_block(RangeCoder *c, DCTBlockCoder *bc,
                              uint8_t *dst, int stride, int block_size,
                              int *block, int mb_x, int mb_y)
@@ -655,7 +572,7 @@ static void decode_dct_block(RangeCoder *c, DCTBlockCoder *bc,
                 c->got_error = 1;
                 return;
             }
-            dct_put(dst + i * 8, stride, block);
+            ff_mss34_dct_put(dst + i * 8, stride, block);
         }
         dst += 8 * stride;
     }
@@ -702,14 +619,6 @@ static void decode_haar_block(RangeCoder *c, HaarBlockCoder *hc,
     }
 }
 
-static void gen_quant_mat(uint16_t *qmat, const uint8_t *ref, float scale)
-{
-    int i;
-
-    for (i = 0; i < 64; i++)
-        qmat[i] = (uint16_t)(ref[i] * scale + 50.0) / 100;
-}
-
 static void reset_coders(MSS3Context *ctx, int quality)
 {
     int i, j;
@@ -726,15 +635,8 @@ static void reset_coders(MSS3Context *ctx, int quality)
         for (j = 0; j < 125; j++)
             model_reset(&ctx->image_coder[i].vq_model[j]);
         if (ctx->dct_coder[i].quality != quality) {
-            float scale;
             ctx->dct_coder[i].quality = quality;
-            if (quality > 50)
+            ff_mss34_gen_quant_mat(ctx->dct_coder[i].qmat, quality, !i);
-                scale = 200.0f - 2 * quality;
-            else
-                scale = 5000.0f / quality;
-            gen_quant_mat(ctx->dct_coder[i].qmat,
-                          i ? mss3_chroma_quant : mss3_luma_quant,
-                          scale);
         }
         memset(ctx->dct_coder[i].prev_dc, 0,
                sizeof(*ctx->dct_coder[i].prev_dc) *

libavcodec/mss34dsp.c

@@ -0,0 +1,114 @@
+/*
+ * Common stuff for some Microsoft Screen codecs
+ * Copyright (C) 2012 Konstantin Shishkov
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include <stdint.h>
+#include "libavutil/common.h"
+#include "mss34dsp.h"
+
+static const uint8_t luma_quant[64] = {
+    16,  11,  10,  16,  24,  40,  51,  61,
+    12,  12,  14,  19,  26,  58,  60,  55,
+    14,  13,  16,  24,  40,  57,  69,  56,
+    14,  17,  22,  29,  51,  87,  80,  62,
+    18,  22,  37,  56,  68, 109, 103,  77,
+    24,  35,  55,  64,  81, 104, 113,  92,
+    49,  64,  78,  87, 103, 121, 120, 101,
+    72,  92,  95,  98, 112, 100, 103,  99
+};
+
+static const uint8_t chroma_quant[64] = {
+    17, 18, 24, 47, 99, 99, 99, 99,
+    18, 21, 26, 66, 99, 99, 99, 99,
+    24, 26, 56, 99, 99, 99, 99, 99,
+    47, 66, 99, 99, 99, 99, 99, 99,
+    99, 99, 99, 99, 99, 99, 99, 99,
+    99, 99, 99, 99, 99, 99, 99, 99,
+    99, 99, 99, 99, 99, 99, 99, 99,
+    99, 99, 99, 99, 99, 99, 99, 99
+};
+
+void ff_mss34_gen_quant_mat(uint16_t *qmat, int quality, int luma)
+{
+    int i;
+    const uint8_t *qsrc = luma ? luma_quant : chroma_quant;
+
+    if (quality >= 50) {
+        int scale = 200 - 2 * quality;
+
+        for (i = 0; i < 64; i++)
+            qmat[i] = (qsrc[i] * scale + 50) / 100;
+    } else {
+        for (i = 0; i < 64; i++)
+            qmat[i] = (5000 * qsrc[i] / quality + 50) / 100;
+    }
+}
+
+#define DCT_TEMPLATE(blk, step, SOP, shift)                         \
+    const int t0 = -39409 * blk[7 * step] -  58980 * blk[1 * step]; \
+    const int t1 =  39410 * blk[1 * step] -  58980 * blk[7 * step]; \
+    const int t2 = -33410 * blk[5 * step] - 167963 * blk[3 * step]; \
+    const int t3 =  33410 * blk[3 * step] - 167963 * blk[5 * step]; \
+    const int t4 =          blk[3 * step] +          blk[7 * step]; \
+    const int t5 =          blk[1 * step] +          blk[5 * step]; \
+    const int t6 =  77062 * t4            +  51491 * t5;            \
+    const int t7 =  77062 * t5            -  51491 * t4;            \
+    const int t8 =  35470 * blk[2 * step] -  85623 * blk[6 * step]; \
+    const int t9 =  35470 * blk[6 * step] +  85623 * blk[2 * step]; \
+    const int tA = SOP(blk[0 * step] - blk[4 * step]);              \
+    const int tB = SOP(blk[0 * step] + blk[4 * step]);              \
+                                                                    \
+    blk[0 * step] = (  t1 + t6  + t9 + tB) >> shift;                \
+    blk[1 * step] = (  t3 + t7  + t8 + tA) >> shift;                \
+    blk[2 * step] = (  t2 + t6  - t8 + tA) >> shift;                \
+    blk[3 * step] = (  t0 + t7  - t9 + tB) >> shift;                \
+    blk[4 * step] = (-(t0 + t7) - t9 + tB) >> shift;                \
+    blk[5 * step] = (-(t2 + t6) - t8 + tA) >> shift;                \
+    blk[6 * step] = (-(t3 + t7) + t8 + tA) >> shift;                \
+    blk[7 * step] = (-(t1 + t6) + t9 + tB) >> shift;                \
+
+#define SOP_ROW(a) ((a) << 16) + 0x2000
+#define SOP_COL(a) ((a + 32) << 16)
+
+void ff_mss34_dct_put(uint8_t *dst, int stride, int *block)
+{
+    int i, j;
+    int *ptr;
+
+    ptr = block;
+    for (i = 0; i < 8; i++) {
+        DCT_TEMPLATE(ptr, 1, SOP_ROW, 13);
+        ptr += 8;
+    }
+
+    ptr = block;
+    for (i = 0; i < 8; i++) {
+        DCT_TEMPLATE(ptr, 8, SOP_COL, 22);
+        ptr++;
+    }
+
+    ptr = block;
+    for (j = 0; j < 8; j++) {
+        for (i = 0; i < 8; i++)
+            dst[i] = av_clip_uint8(ptr[i] + 128);
+        dst += stride;
+        ptr += 8;
+    }
+}

libavcodec/mss34dsp.h

@@ -0,0 +1,45 @@
+/*
+ * Common stuff for some Microsoft Screen codecs
+ * Copyright (C) 2012 Konstantin Shishkov
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_MSS34DSP_H
+#define AVCODEC_MSS34DSP_H
+
+#include <stdint.h>
+
+/**
+ * Generate quantisation matrix for given quality.
+ *
+ * @param qmat    destination matrix
+ * @param quality quality setting (1-100)
+ * @param luma    generate quantisation matrix for luma or chroma
+ */
+void ff_mss34_gen_quant_mat(uint16_t *qmat, int quality, int luma);
+
+/**
+ * Transform and output DCT block.
+ *
+ * @param dst     output plane
+ * @param stride  output plane stride
+ * @param block   block to transform and output
+ */
+void ff_mss34_dct_put(uint8_t *dst, int stride, int *block);
+
+#endif /* AVCODEC_MSS34DSP_H */

libavfilter/af_aconvert.c

@@ -135,12 +135,13 @@ static int config_output(AVFilterLink *outlink)
     return 0;
 }
 
-static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
+static int  filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
 {
     AConvertContext *aconvert = inlink->dst->priv;
     const int n = insamplesref->audio->nb_samples;
     AVFilterLink *const outlink = inlink->dst->outputs[0];
     AVFilterBufferRef *outsamplesref = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n);
+    int ret;
 
     swr_convert(aconvert->swr, outsamplesref->data, n,
                         (void *)insamplesref->data, n);
@@ -148,8 +149,9 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref
     avfilter_copy_buffer_ref_props(outsamplesref, insamplesref);
     outsamplesref->audio->channel_layout = outlink->channel_layout;
 
-    ff_filter_samples(outlink, outsamplesref);
+    ret = ff_filter_samples(outlink, outsamplesref);
     avfilter_unref_buffer(insamplesref);
+    return ret;
 }
 
 AVFilter avfilter_af_aconvert = {

libavfilter/af_amerge.c

@@ -212,7 +212,7 @@ static inline void copy_samples(int nb_inputs, struct amerge_input in[],
     }
 }
 
-static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
+static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
 {
     AVFilterContext *ctx = inlink->dst;
     AMergeContext *am = ctx->priv;
@@ -232,7 +232,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
     for (i = 1; i < am->nb_inputs; i++)
         nb_samples = FFMIN(nb_samples, am->in[i].nb_samples);
     if (!nb_samples)
-        return;
+        return 0;
 
     outbuf = ff_get_audio_buffer(ctx->outputs[0], AV_PERM_WRITE, nb_samples);
     outs = outbuf->data[0];
@@ -285,7 +285,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
             }
         }
     }
-    ff_filter_samples(ctx->outputs[0], outbuf);
+    return ff_filter_samples(ctx->outputs[0], outbuf);
 }
 
 static av_cold int init(AVFilterContext *ctx, const char *args)

libavfilter/af_amix.c

@@ -305,9 +305,7 @@ static int output_frame(AVFilterLink *outlink, int nb_samples)
     if (s->next_pts != AV_NOPTS_VALUE)
         s->next_pts += nb_samples;
 
-    ff_filter_samples(outlink, out_buf);
+    return ff_filter_samples(outlink, out_buf);
-
-    return 0;
 }
 
 /**
@@ -448,31 +446,37 @@ static int request_frame(AVFilterLink *outlink)
     return output_frame(outlink, available_samples);
 }
 
-static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
+static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
 {
     AVFilterContext  *ctx = inlink->dst;
     MixContext       *s = ctx->priv;
     AVFilterLink *outlink = ctx->outputs[0];
-    int i;
+    int i, ret = 0;
 
     for (i = 0; i < ctx->nb_inputs; i++)
         if (ctx->inputs[i] == inlink)
             break;
     if (i >= ctx->nb_inputs) {
         av_log(ctx, AV_LOG_ERROR, "unknown input link\n");
-        return;
+        ret = AVERROR(EINVAL);
+        goto fail;
     }
 
     if (i == 0) {
         int64_t pts = av_rescale_q(buf->pts, inlink->time_base,
                                    outlink->time_base);
-        frame_list_add_frame(s->frame_list, buf->audio->nb_samples, pts);
+        ret = frame_list_add_frame(s->frame_list, buf->audio->nb_samples, pts);
+        if (ret < 0)
+            goto fail;
     }
 
-    av_audio_fifo_write(s->fifos[i], (void **)buf->extended_data,
+    ret = av_audio_fifo_write(s->fifos[i], (void **)buf->extended_data,
-                        buf->audio->nb_samples);
+                              buf->audio->nb_samples);
 
+fail:
     avfilter_unref_buffer(buf);
+
+    return ret;
 }
 
 static int init(AVFilterContext *ctx, const char *args)

libavfilter/af_aresample.c

@@ -168,13 +168,14 @@ static int config_output(AVFilterLink *outlink)
     return 0;
 }
 
-static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
+static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
 {
     AResampleContext *aresample = inlink->dst->priv;
     const int n_in  = insamplesref->audio->nb_samples;
     int n_out       = n_in * aresample->ratio * 2 ;
     AVFilterLink *const outlink = inlink->dst->outputs[0];
     AVFilterBufferRef *outsamplesref = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n_out);
+    int ret;
 
 
     avfilter_copy_buffer_ref_props(outsamplesref, insamplesref);
@@ -193,15 +194,16 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref
     if (n_out <= 0) {
         avfilter_unref_buffer(outsamplesref);
         avfilter_unref_buffer(insamplesref);
-        return;
+        return 0;
     }
 
     outsamplesref->audio->sample_rate = outlink->sample_rate;
     outsamplesref->audio->nb_samples  = n_out;
 
-    ff_filter_samples(outlink, outsamplesref);
+    ret = ff_filter_samples(outlink, outsamplesref);
     aresample->req_fullfilled= 1;
     avfilter_unref_buffer(insamplesref);
+    return ret;
 }
 
 static int request_frame(AVFilterLink *outlink)

libavfilter/af_asetnsamples.c

@@ -131,7 +131,7 @@ static int push_samples(AVFilterLink *outlink)
     return nb_out_samples;
 }
 
-static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
+static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
 {
     AVFilterContext *ctx = inlink->dst;
     ASNSContext *asns = ctx->priv;
@@ -145,7 +145,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
         if (ret < 0) {
             av_log(ctx, AV_LOG_ERROR,
                    "Stretching audio fifo failed, discarded %d samples\n", nb_samples);
-            return;
+            return -1;
         }
     }
     av_audio_fifo_write(asns->fifo, (void **)insamples->extended_data, nb_samples);
@@ -155,6 +155,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
 
     if (av_audio_fifo_size(asns->fifo) >= asns->nb_out_samples)
         push_samples(outlink);
+    return 0;
 }
 
 static int request_frame(AVFilterLink *outlink)

libavfilter/af_ashowinfo.c

@@ -40,7 +40,7 @@ static av_cold int init(AVFilterContext *ctx, const char *args)
     return 0;
 }
 
-static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *samplesref)
+static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *samplesref)
 {
     AVFilterContext *ctx = inlink->dst;
     ShowInfoContext *showinfo = ctx->priv;
@@ -83,7 +83,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *samplesref)
     av_log(ctx, AV_LOG_INFO, "]\n");
 
     showinfo->frame++;
-    ff_filter_samples(inlink->dst->outputs[0], samplesref);
+    return ff_filter_samples(inlink->dst->outputs[0], samplesref);
 }
 
 AVFilter avfilter_af_ashowinfo = {

libavfilter/af_astreamsync.c

@@ -107,11 +107,12 @@ static int config_output(AVFilterLink *outlink)
     return 0;
 }
 
-static void send_out(AVFilterContext *ctx, int out_id)
+static int send_out(AVFilterContext *ctx, int out_id)
 {
     AStreamSyncContext *as = ctx->priv;
     struct buf_queue *queue = &as->queue[out_id];
     AVFilterBufferRef *buf = queue->buf[queue->tail];
+    int ret;
 
     queue->buf[queue->tail] = NULL;
     as->var_values[VAR_B1 + out_id]++;
@@ -121,11 +122,12 @@ static void send_out(AVFilterContext *ctx, int out_id)
             av_q2d(ctx->outputs[out_id]->time_base) * buf->pts;
     as->var_values[VAR_T1 + out_id] += buf->audio->nb_samples /
                                    (double)ctx->inputs[out_id]->sample_rate;
-    ff_filter_samples(ctx->outputs[out_id], buf);
+    ret = ff_filter_samples(ctx->outputs[out_id], buf);
     queue->nb--;
     queue->tail = (queue->tail + 1) % QUEUE_SIZE;
     if (as->req[out_id])
         as->req[out_id]--;
+    return ret;
 }
 
 static void send_next(AVFilterContext *ctx)
@@ -165,7 +167,7 @@ static int request_frame(AVFilterLink *outlink)
     return 0;
 }
 
-static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
+static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
 {
     AVFilterContext *ctx = inlink->dst;
     AStreamSyncContext *as = ctx->priv;
@@ -175,6 +177,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
         insamples;
     as->eof &= ~(1 << id);
     send_next(ctx);
+    return 0;
 }
 
 AVFilter avfilter_af_astreamsync = {

libavfilter/af_asyncts.c

@@ -37,6 +37,9 @@ typedef struct ASyncContext {
     int resample;
     float min_delta_sec;
     int max_comp;
+
+    /* set by filter_samples() to signal an output frame to request_frame() */
+    int got_output;
 } ASyncContext;
 
 #define OFFSET(x) offsetof(ASyncContext, x)
@@ -112,9 +115,13 @@ static int request_frame(AVFilterLink *link)
 {
     AVFilterContext *ctx = link->src;
     ASyncContext      *s = ctx->priv;
-    int ret = ff_request_frame(ctx->inputs[0]);
+    int ret = 0;
     int nb_samples;
 
+    s->got_output = 0;
+    while (ret >= 0 && !s->got_output)
+        ret = ff_request_frame(ctx->inputs[0]);
+
     /* flush the fifo */
     if (ret == AVERROR_EOF && (nb_samples = avresample_get_delay(s->avr))) {
         AVFilterBufferRef *buf = ff_get_audio_buffer(link, AV_PERM_WRITE,
@@ -124,18 +131,18 @@ static int request_frame(AVFilterLink *link)
         avresample_convert(s->avr, (void**)buf->extended_data, buf->linesize[0],
                            nb_samples, NULL, 0, 0);
         buf->pts = s->pts;
-        ff_filter_samples(link, buf);
+        return ff_filter_samples(link, buf);
-        return 0;
     }
 
     return ret;
 }
 
-static void write_to_fifo(ASyncContext *s, AVFilterBufferRef *buf)
+static int write_to_fifo(ASyncContext *s, AVFilterBufferRef *buf)
 {
-    avresample_convert(s->avr, NULL, 0, 0, (void**)buf->extended_data,
+    int ret = avresample_convert(s->avr, NULL, 0, 0, (void**)buf->extended_data,
-                       buf->linesize[0], buf->audio->nb_samples);
+                                 buf->linesize[0], buf->audio->nb_samples);
     avfilter_unref_buffer(buf);
+    return ret;
 }
 
 /* get amount of data currently buffered, in samples */
@@ -144,7 +151,7 @@ static int64_t get_delay(ASyncContext *s)
     return avresample_available(s->avr) + avresample_get_delay(s->avr);
 }
 
-static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
+static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
 {
     AVFilterContext  *ctx = inlink->dst;
     ASyncContext       *s = ctx->priv;
@@ -152,7 +159,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
     int nb_channels = av_get_channel_layout_nb_channels(buf->audio->channel_layout);
     int64_t pts = (buf->pts == AV_NOPTS_VALUE) ? buf->pts :
                   av_rescale_q(buf->pts, inlink->time_base, outlink->time_base);
-    int out_size;
+    int out_size, ret;
     int64_t delta;
 
     /* buffer data until we get the first timestamp */
@@ -160,14 +167,12 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
         if (pts != AV_NOPTS_VALUE) {
             s->pts = pts - get_delay(s);
         }
-        write_to_fifo(s, buf);
+        return write_to_fifo(s, buf);
-        return;
     }
 
     /* now wait for the next timestamp */
     if (pts == AV_NOPTS_VALUE) {
-        write_to_fifo(s, buf);
+        return write_to_fifo(s, buf);
-        return;
     }
 
     /* when we have two timestamps, compute how many samples would we have
@@ -190,8 +195,10 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
     if (out_size > 0) {
         AVFilterBufferRef *buf_out = ff_get_audio_buffer(outlink, AV_PERM_WRITE,
                                                          out_size);
-        if (!buf_out)
+        if (!buf_out) {
-            return;
+            ret = AVERROR(ENOMEM);
+            goto fail;
+        }
 
         avresample_read(s->avr, (void**)buf_out->extended_data, out_size);
         buf_out->pts = s->pts;
@@ -200,7 +207,10 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
             av_samples_set_silence(buf_out->extended_data, out_size - delta,
                                    delta, nb_channels, buf->format);
         }
-        ff_filter_samples(outlink, buf_out);
+        ret = ff_filter_samples(outlink, buf_out);
+        if (ret < 0)
+            goto fail;
+        s->got_output = 1;
     } else {
         av_log(ctx, AV_LOG_WARNING, "Non-monotonous timestamps, dropping "
                "whole buffer.\n");
@@ -210,9 +220,13 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
     avresample_read(s->avr, NULL, avresample_available(s->avr));
 
     s->pts = pts - avresample_get_delay(s->avr);
-    avresample_convert(s->avr, NULL, 0, 0, (void**)buf->extended_data,
+    ret = avresample_convert(s->avr, NULL, 0, 0, (void**)buf->extended_data,
-                       buf->linesize[0], buf->audio->nb_samples);
+                             buf->linesize[0], buf->audio->nb_samples);
+
+fail:
     avfilter_unref_buffer(buf);
+
+    return ret;
 }
 
 AVFilter avfilter_af_asyncts = {

libavfilter/af_atempo.c

@@ -1040,7 +1040,7 @@ static void push_samples(ATempoContext *atempo,
     atempo->nsamples_out += n_out;
 }
 
-static void filter_samples(AVFilterLink *inlink,
+static int filter_samples(AVFilterLink *inlink,
                            AVFilterBufferRef *src_buffer)
 {
     AVFilterContext  *ctx = inlink->dst;
@@ -1074,6 +1074,7 @@ static void filter_samples(AVFilterLink *inlink,
 
     atempo->nsamples_in += n_in;
     avfilter_unref_bufferp(&src_buffer);
+    return 0;
 }
 
 static int request_frame(AVFilterLink *outlink)

libavfilter/af_channelmap.c

@@ -313,7 +313,7 @@ static int channelmap_query_formats(AVFilterContext *ctx)
     return 0;
 }
 
-static void channelmap_filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
+static int channelmap_filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
 {
     AVFilterContext  *ctx = inlink->dst;
     AVFilterLink *outlink = ctx->outputs[0];
@@ -330,8 +330,10 @@ static void channelmap_filter_samples(AVFilterLink *inlink, AVFilterBufferRef *b
         if (nch_out > FF_ARRAY_ELEMS(buf->data)) {
             uint8_t **new_extended_data =
                 av_mallocz(nch_out * sizeof(*buf->extended_data));
-            if (!new_extended_data)
+            if (!new_extended_data) {
-                return;
+                avfilter_unref_buffer(buf);
+                return AVERROR(ENOMEM);
+            }
             if (buf->extended_data == buf->data) {
                 buf->extended_data = new_extended_data;
             } else {
@@ -353,7 +355,7 @@ static void channelmap_filter_samples(AVFilterLink *inlink, AVFilterBufferRef *b
         memcpy(buf->data, buf->extended_data,
            FFMIN(FF_ARRAY_ELEMS(buf->data), nch_out) * sizeof(buf->data[0]));
 
-    ff_filter_samples(outlink, buf);
+    return ff_filter_samples(outlink, buf);
 }
 
 static int channelmap_config_input(AVFilterLink *inlink)

libavfilter/af_channelsplit.c

@@ -105,24 +105,29 @@ static int query_formats(AVFilterContext *ctx)
     return 0;
 }
 
-static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
+static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
 {
     AVFilterContext *ctx = inlink->dst;
-    int i;
+    int i, ret = 0;
 
     for (i = 0; i < ctx->nb_outputs; i++) {
         AVFilterBufferRef *buf_out = avfilter_ref_buffer(buf, ~AV_PERM_WRITE);
 
-        if (!buf_out)
+        if (!buf_out) {
-            return;
+            ret = AVERROR(ENOMEM);
+            break;
+        }
 
         buf_out->data[0] = buf_out->extended_data[0] = buf_out->extended_data[i];
         buf_out->audio->channel_layout =
             av_channel_layout_extract_channel(buf->audio->channel_layout, i);
 
-        ff_filter_samples(ctx->outputs[i], buf_out);
+        ret = ff_filter_samples(ctx->outputs[i], buf_out);
+        if (ret < 0)
+            break;
     }
     avfilter_unref_buffer(buf);
+    return ret;
 }
 
 AVFilter avfilter_af_channelsplit = {

libavfilter/af_earwax.c

@@ -120,13 +120,15 @@ static inline int16_t *scalarproduct(const int16_t *in, const int16_t *endin, in
     return out;
 }
 
-static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
+static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
 {
     AVFilterLink *outlink = inlink->dst->outputs[0];
     int16_t *taps, *endin, *in, *out;
     AVFilterBufferRef *outsamples =
         ff_get_audio_buffer(inlink, AV_PERM_WRITE,
                                   insamples->audio->nb_samples);
+    int ret;
+
     avfilter_copy_buffer_ref_props(outsamples, insamples);
 
     taps  = ((EarwaxContext *)inlink->dst->priv)->taps;
@@ -144,8 +146,9 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
     // save part of input for next round
     memcpy(taps, endin, NUMTAPS * sizeof(*taps));
 
-    ff_filter_samples(outlink, outsamples);
+    ret = ff_filter_samples(outlink, outsamples);
     avfilter_unref_buffer(insamples);
+    return ret;
 }
 
 AVFilter avfilter_af_earwax = {

libavfilter/af_join.c

@@ -92,7 +92,7 @@ static const AVClass join_class = {
     .version    = LIBAVUTIL_VERSION_INT,
 };
 
-static void filter_samples(AVFilterLink *link, AVFilterBufferRef *buf)
+static int filter_samples(AVFilterLink *link, AVFilterBufferRef *buf)
 {
     AVFilterContext *ctx = link->dst;
     JoinContext       *s = ctx->priv;
@@ -104,6 +104,8 @@ static void filter_samples(AVFilterLink *link, AVFilterBufferRef *buf)
     av_assert0(i < ctx->nb_inputs);
     av_assert0(!s->input_frames[i]);
     s->input_frames[i] = buf;
+
+    return 0;
 }
 
 static int parse_maps(AVFilterContext *ctx)
@@ -468,11 +470,11 @@ static int join_request_frame(AVFilterLink *outlink)
     priv->nb_in_buffers = ctx->nb_inputs;
     buf->buf->priv      = priv;
 
-    ff_filter_samples(outlink, buf);
+    ret = ff_filter_samples(outlink, buf);
 
     memset(s->input_frames, 0, sizeof(*s->input_frames) * ctx->nb_inputs);
 
-    return 0;
+    return ret;
 
 fail:
     avfilter_unref_buffer(buf);

libavfilter/af_pan.c

@@ -343,8 +343,9 @@ static int config_props(AVFilterLink *link)
     return 0;
 }
 
-static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
+static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
 {
+    int ret;
     int n = insamples->audio->nb_samples;
     AVFilterLink *const outlink = inlink->dst->outputs[0];
     AVFilterBufferRef *outsamples = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n);
@@ -354,8 +355,9 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
     avfilter_copy_buffer_ref_props(outsamples, insamples);
     outsamples->audio->channel_layout = outlink->channel_layout;
 
-    ff_filter_samples(outlink, outsamples);
+    ret = ff_filter_samples(outlink, outsamples);
     avfilter_unref_buffer(insamples);
+    return ret;
 }
 
 static av_cold void uninit(AVFilterContext *ctx)

libavfilter/af_resample.c

@@ -38,6 +38,9 @@ typedef struct ResampleContext {
     AVAudioResampleContext *avr;
 
     int64_t next_pts;
+
+    /* set by filter_samples() to signal an output frame to request_frame() */
+    int got_output;
 } ResampleContext;
 
 static av_cold void uninit(AVFilterContext *ctx)
@@ -102,12 +105,6 @@ static int config_output(AVFilterLink *outlink)
     av_opt_set_int(s->avr,  "in_sample_rate",    inlink ->sample_rate,    0);
     av_opt_set_int(s->avr, "out_sample_rate",    outlink->sample_rate,    0);
 
-    /* if both the input and output formats are s16 or u8, use s16 as
-       the internal sample format */
-    if (av_get_bytes_per_sample(inlink->format)  <= 2 &&
-        av_get_bytes_per_sample(outlink->format) <= 2)
-        av_opt_set_int(s->avr, "internal_sample_fmt", AV_SAMPLE_FMT_S16P, 0);
-
     if ((ret = avresample_open(s->avr)) < 0)
         return ret;
 
@@ -130,7 +127,11 @@ static int request_frame(AVFilterLink *outlink)
 {
     AVFilterContext *ctx = outlink->src;
     ResampleContext   *s = ctx->priv;
-    int ret = ff_request_frame(ctx->inputs[0]);
+    int ret = 0;
+
+    s->got_output = 0;
+    while (ret >= 0 && !s->got_output)
+        ret = ff_request_frame(ctx->inputs[0]);
 
     /* flush the lavr delay buffer */
     if (ret == AVERROR_EOF && s->avr) {
@@ -156,21 +157,21 @@ static int request_frame(AVFilterLink *outlink)
         }
 
         buf->pts = s->next_pts;
-        ff_filter_samples(outlink, buf);
+        return ff_filter_samples(outlink, buf);
-        return 0;
     }
     return ret;
 }
 
-static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
+static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
 {
     AVFilterContext  *ctx = inlink->dst;
     ResampleContext    *s = ctx->priv;
     AVFilterLink *outlink = ctx->outputs[0];
+    int ret;
 
     if (s->avr) {
         AVFilterBufferRef *buf_out;
-        int delay, nb_samples, ret;
+        int delay, nb_samples;
 
         /* maximum possible samples lavr can output */
         delay      = avresample_get_delay(s->avr);
@@ -179,10 +180,19 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
                                     AV_ROUND_UP);
 
         buf_out = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
+        if (!buf_out) {
+            ret = AVERROR(ENOMEM);
+            goto fail;
+        }
+
         ret     = avresample_convert(s->avr, (void**)buf_out->extended_data,
                                      buf_out->linesize[0], nb_samples,
                                      (void**)buf->extended_data, buf->linesize[0],
                                      buf->audio->nb_samples);
+        if (ret < 0) {
+            avfilter_unref_buffer(buf_out);
+            goto fail;
+        }
 
         av_assert0(!avresample_available(s->avr));
 
@@ -208,11 +218,18 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
 
             s->next_pts = buf_out->pts + buf_out->audio->nb_samples;
 
-            ff_filter_samples(outlink, buf_out);
+            ret = ff_filter_samples(outlink, buf_out);
+            s->got_output = 1;
         }
+
+fail:
         avfilter_unref_buffer(buf);
-    } else
+    } else {
-        ff_filter_samples(outlink, buf);
+        ret = ff_filter_samples(outlink, buf);
+        s->got_output = 1;
+    }
+
+    return ret;
 }
 
 AVFilter avfilter_af_resample = {

libavfilter/af_silencedetect.c

@@ -78,7 +78,7 @@ static av_cold int init(AVFilterContext *ctx, const char *args)
     return 0;
 }
 
-static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
+static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
 {
     int i;
     SilenceDetectContext *silence = inlink->dst->priv;
@@ -118,7 +118,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
         }
     }
 
-    ff_filter_samples(inlink->dst->outputs[0], insamples);
+    return ff_filter_samples(inlink->dst->outputs[0], insamples);
 }
 
 static int query_formats(AVFilterContext *ctx)

libavfilter/af_volume.c

@@ -110,7 +110,7 @@ static int query_formats(AVFilterContext *ctx)
     return 0;
 }
 
-static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
+static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
 {
     VolumeContext *vol = inlink->dst->priv;
     AVFilterLink *outlink = inlink->dst->outputs[0];
@@ -169,7 +169,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
         }
         }
     }
-    ff_filter_samples(outlink, insamples);
+    return ff_filter_samples(outlink, insamples);
 }
 
 AVFilter avfilter_af_volume = {

libavfilter/asink_anullsink.c

@@ -21,7 +21,10 @@
 #include "avfilter.h"
 #include "internal.h"
 
-static void null_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref) { }
+static int null_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
+{
+    return 0;
+}
 
 AVFilter avfilter_asink_anullsink = {
     .name        = "anullsink",

libavfilter/audio.c

@@ -150,19 +150,19 @@ fail:
     return NULL;
 }
 
-static void default_filter_samples(AVFilterLink *link,
+static int default_filter_samples(AVFilterLink *link,
-                                   AVFilterBufferRef *samplesref)
+                                  AVFilterBufferRef *samplesref)
 {
-    ff_filter_samples(link->dst->outputs[0], samplesref);
+    return ff_filter_samples(link->dst->outputs[0], samplesref);
 }
 
-void ff_filter_samples_framed(AVFilterLink *link,
+int ff_filter_samples_framed(AVFilterLink *link, AVFilterBufferRef *samplesref)
-                              AVFilterBufferRef *samplesref)
 {
-    void (*filter_samples)(AVFilterLink *, AVFilterBufferRef *);
+    int (*filter_samples)(AVFilterLink *, AVFilterBufferRef *);
     AVFilterPad *dst = link->dstpad;
     int64_t pts;
     AVFilterBufferRef *buf_out;
+    int ret;
 
     FF_TPRINTF_START(NULL, filter_samples); ff_tlog_link(NULL, link, 1);
 
@@ -193,21 +193,22 @@ void ff_filter_samples_framed(AVFilterLink *link,
 
     link->cur_buf = buf_out;
     pts = buf_out->pts;
-    filter_samples(link, buf_out);
+    ret = filter_samples(link, buf_out);
     ff_update_link_current_pts(link, pts);
+    return ret;
 }
 
-void ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
+int ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
 {
     int insamples = samplesref->audio->nb_samples, inpos = 0, nb_samples;
     AVFilterBufferRef *pbuf = link->partial_buf;
     int nb_channels = av_get_channel_layout_nb_channels(link->channel_layout);
+    int ret = 0;
 
     if (!link->min_samples ||
         (!pbuf &&
          insamples >= link->min_samples && insamples <= link->max_samples)) {
-        ff_filter_samples_framed(link, samplesref);
+        return ff_filter_samples_framed(link, samplesref);
-        return;
     }
     /* Handle framing (min_samples, max_samples) */
     while (insamples) {
@@ -218,7 +219,7 @@ void ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
             if (!pbuf) {
                 av_log(link->dst, AV_LOG_WARNING,
                        "Samples dropped due to memory allocation failure.\n");
-                return;
+                return 0;
             }
             avfilter_copy_buffer_ref_props(pbuf, samplesref);
             pbuf->pts = samplesref->pts +
@@ -234,10 +235,11 @@ void ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
         insamples               -= nb_samples;
         pbuf->audio->nb_samples += nb_samples;
         if (pbuf->audio->nb_samples >= link->min_samples) {
-            ff_filter_samples_framed(link, pbuf);
+            ret = ff_filter_samples_framed(link, pbuf);
             pbuf = NULL;
         }
     }
     avfilter_unref_buffer(samplesref);
     link->partial_buf = pbuf;
+    return ret;
 }

libavfilter/audio.h

@@ -70,14 +70,17 @@ AVFilterBufferRef *ff_get_audio_buffer(AVFilterLink *link, int perms,
  * @param samplesref a reference to the buffer of audio samples being sent. The
  *                   receiving filter will free this reference when it no longer
  *                   needs it or pass it on to the next filter.
+ *
+ * @return >= 0 on success, a negative AVERROR on error. The receiving filter
+ * is responsible for unreferencing samplesref in case of error.
  */
-void ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref);
+int ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref);
 
 /**
  * Send a buffer of audio samples to the next link, without checking
  * min_samples.
  */
-void ff_filter_samples_framed(AVFilterLink *link,
+int ff_filter_samples_framed(AVFilterLink *link,
                               AVFilterBufferRef *samplesref);
 
 #endif /* AVFILTER_AUDIO_H */

libavfilter/avf_showwaves.c

@@ -180,7 +180,7 @@ static int request_frame(AVFilterLink *outlink)
 
 #define MAX_INT16 ((1<<15) -1)
 
-static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
+static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
 {
     AVFilterContext *ctx = inlink->dst;
     AVFilterLink *outlink = ctx->outputs[0];
@@ -225,6 +225,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
     }
 
     avfilter_unref_buffer(insamples);
+    return 0;
 }
 
 AVFilter avfilter_avf_showwaves = {

libavfilter/avfilter.h

@@ -301,8 +301,12 @@ struct AVFilterPad {
      * and should do its processing.
      *
      * Input audio pads only.
+     *
+     * @return >= 0 on success, a negative AVERROR on error. This function
+     * must ensure that samplesref is properly unreferenced on error if it
+     * hasn't been passed on to another filter.
      */
-    void (*filter_samples)(AVFilterLink *link, AVFilterBufferRef *samplesref);
+    int (*filter_samples)(AVFilterLink *link, AVFilterBufferRef *samplesref);
 
     /**
      * Frame poll callback. This returns the number of immediately available

libavfilter/buffersink.c

@@ -56,6 +56,12 @@ static void start_frame(AVFilterLink *link, AVFilterBufferRef *buf)
     link->cur_buf = NULL;
 };
 
+static int filter_samples(AVFilterLink *link, AVFilterBufferRef *buf)
+{
+    start_frame(link, buf);
+    return 0;
+}
+
 int av_buffersink_read(AVFilterContext *ctx, AVFilterBufferRef **buf)
 {
     BufferSinkContext *s    = ctx->priv;
@@ -160,7 +166,7 @@ AVFilter avfilter_asink_abuffer = {
 
     .inputs    = (AVFilterPad[]) {{ .name           = "default",
                                     .type           = AVMEDIA_TYPE_AUDIO,
-                                    .filter_samples = start_frame,
+                                    .filter_samples = filter_samples,
                                     .min_perms      = AV_PERM_READ,
                                     .needs_fifo     = 1 },
                                   { .name = NULL }},

libavfilter/buffersrc.c

@@ -408,6 +408,7 @@ static int request_frame(AVFilterLink *link)
 {
     BufferSourceContext *c = link->src->priv;
     AVFilterBufferRef *buf;
+    int ret = 0;
 
     if (!av_fifo_size(c->fifo)) {
         if (c->eof)
@@ -424,7 +425,7 @@ static int request_frame(AVFilterLink *link)
         ff_end_frame(link);
         break;
     case AVMEDIA_TYPE_AUDIO:
-        ff_filter_samples(link, avfilter_ref_buffer(buf, ~0));
+        ret = ff_filter_samples(link, avfilter_ref_buffer(buf, ~0));
         break;
     default:
         return AVERROR(EINVAL);
@@ -432,7 +433,7 @@ static int request_frame(AVFilterLink *link)
 
     avfilter_unref_buffer(buf);
 
-    return 0;
+    return ret;
 }
 
 static int poll_frame(AVFilterLink *link)

libavfilter/f_settb.c

@@ -117,7 +117,7 @@ static void start_frame(AVFilterLink *inlink, AVFilterBufferRef *picref)
     ff_start_frame(outlink, picref2);
 }
 
-static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
+static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
 {
     AVFilterContext *ctx = inlink->dst;
     AVFilterLink *outlink = ctx->outputs[0];
@@ -132,7 +132,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
         avfilter_unref_buffer(insamples);
     }
 
-    ff_filter_samples(outlink, outsamples);
+    return ff_filter_samples(outlink, outsamples);
 }
 
 #if CONFIG_SETTB_FILTER

libavfilter/fifo.c

@@ -72,13 +72,25 @@ static av_cold void uninit(AVFilterContext *ctx)
     avfilter_unref_buffer(fifo->buf_out);
 }
 
-static void add_to_queue(AVFilterLink *inlink, AVFilterBufferRef *buf)
+static int add_to_queue(AVFilterLink *inlink, AVFilterBufferRef *buf)
 {
     FifoContext *fifo = inlink->dst->priv;
 
     fifo->last->next = av_mallocz(sizeof(Buf));
+    if (!fifo->last->next) {
+        avfilter_unref_buffer(buf);
+        return AVERROR(ENOMEM);
+    }
+
     fifo->last = fifo->last->next;
     fifo->last->buf = buf;
+
+    return 0;
+}
+
+static void start_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)
+{
+    add_to_queue(inlink, buf);
 }
 
 static void queue_pop(FifoContext *s)
@@ -210,15 +222,13 @@ static int return_audio_frame(AVFilterContext *ctx)
         buf_out = s->buf_out;
         s->buf_out = NULL;
     }
-    ff_filter_samples(link, buf_out);
+    return ff_filter_samples(link, buf_out);
-
-    return 0;
 }
 
 static int request_frame(AVFilterLink *outlink)
 {
     FifoContext *fifo = outlink->src->priv;
-    int ret;
+    int ret = 0;
 
     if (!fifo->root.next) {
         if ((ret = ff_request_frame(outlink->src->inputs[0])) < 0)
@@ -238,7 +248,7 @@ static int request_frame(AVFilterLink *outlink)
         if (outlink->request_samples) {
             return return_audio_frame(outlink->src);
         } else {
-            ff_filter_samples(outlink, fifo->root.next->buf);
+            ret = ff_filter_samples(outlink, fifo->root.next->buf);
             queue_pop(fifo);
         }
         break;
@@ -246,7 +256,7 @@ static int request_frame(AVFilterLink *outlink)
         return AVERROR(EINVAL);
     }
 
-    return 0;
+    return ret;
 }
 
 AVFilter avfilter_vf_fifo = {
@@ -261,7 +271,7 @@ AVFilter avfilter_vf_fifo = {
     .inputs    = (const AVFilterPad[]) {{ .name      = "default",
                                     .type            = AVMEDIA_TYPE_VIDEO,
                                     .get_video_buffer= ff_null_get_video_buffer,
-                                    .start_frame     = add_to_queue,
+                                    .start_frame     = start_frame,
                                     .draw_slice      = draw_slice,
                                     .end_frame       = end_frame,
                                     .rej_perms       = AV_PERM_REUSE2, },

libavfilter/internal.h

@@ -135,8 +135,12 @@ struct AVFilterPad {
      * and should do its processing.
      *
      * Input audio pads only.
+     *
+     * @return >= 0 on success, a negative AVERROR on error. This function
+     * must ensure that samplesref is properly unreferenced on error if it
+     * hasn't been passed on to another filter.
      */
-    void (*filter_samples)(AVFilterLink *link, AVFilterBufferRef *samplesref);
+    int (*filter_samples)(AVFilterLink *link, AVFilterBufferRef *samplesref);
 
     /**
      * Frame poll callback. This returns the number of immediately available

libavfilter/sink_buffer.c

@@ -244,9 +244,10 @@ AVFilter avfilter_vsink_buffersink = {
 
 #if CONFIG_ABUFFERSINK_FILTER
 
-static void filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
+static int filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
 {
     end_frame(link);
+    return 0;
 }
 
 static av_cold int asink_init(AVFilterContext *ctx, const char *args)

libavfilter/split.c

@@ -110,15 +110,19 @@ AVFilter avfilter_vf_split = {
     .outputs   = (AVFilterPad[]) {{ .name = NULL}},
 };
 
-static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *samplesref)
+static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *samplesref)
 {
     AVFilterContext *ctx = inlink->dst;
-    int i;
+    int i, ret = 0;
 
-    for (i = 0; i < ctx->nb_outputs; i++)
+    for (i = 0; i < ctx->nb_outputs; i++) {
-        ff_filter_samples(inlink->dst->outputs[i],
+        ret = ff_filter_samples(inlink->dst->outputs[i],
-                          avfilter_ref_buffer(samplesref, ~AV_PERM_WRITE));
+                                avfilter_ref_buffer(samplesref, ~AV_PERM_WRITE));
+        if (ret < 0)
+            break;
+    }
     avfilter_unref_buffer(samplesref);
+    return ret;
 }
 
 AVFilter avfilter_af_asplit = {

libavformat/matroskadec.c

@@ -842,8 +842,11 @@ static int ebml_parse_id(MatroskaDemuxContext *matroska, EbmlSyntax *syntax,
         matroska->num_levels > 0 &&
         matroska->levels[matroska->num_levels-1].length == 0xffffffffffffff)
         return 0;  // we reached the end of an unknown size cluster
-    if (!syntax[i].id && id != EBML_ID_VOID && id != EBML_ID_CRC32)
+    if (!syntax[i].id && id != EBML_ID_VOID && id != EBML_ID_CRC32) {
         av_log(matroska->ctx, AV_LOG_INFO, "Unknown entry 0x%X\n", id);
+        if (matroska->ctx->error_recognition & AV_EF_EXPLODE)
+            return AVERROR_INVALIDDATA;
+    }
     return ebml_parse_elem(matroska, &syntax[i], data);
 }
 

libavformat/tcp.c

@@ -43,7 +43,7 @@ static int tcp_open(URLContext *h, const char *uri, int flags)
     char buf[256];
     int ret;
     socklen_t optlen;
-    int timeout = 50;
+    int timeout = 50, listen_timeout = -1;
     char hostname[1024],proto[1024],path[1024];
     char portstr[10];
 
@@ -59,6 +59,9 @@ static int tcp_open(URLContext *h, const char *uri, int flags)
         if (av_find_info_tag(buf, sizeof(buf), "timeout", p)) {
             timeout = strtol(buf, NULL, 10);
         }
+        if (av_find_info_tag(buf, sizeof(buf), "listen_timeout", p)) {
+            listen_timeout = strtol(buf, NULL, 10);
+        }
     }
     hints.ai_family = AF_UNSPEC;
     hints.ai_socktype = SOCK_STREAM;
@@ -87,6 +90,7 @@ static int tcp_open(URLContext *h, const char *uri, int flags)
     if (listen_socket) {
         int fd1;
         int reuse = 1;
+        struct pollfd lp = { fd, POLLIN, 0 };
         setsockopt(fd, SOL_SOCKET, SO_REUSEADDR, &reuse, sizeof(reuse));
         ret = bind(fd, cur_ai->ai_addr, cur_ai->ai_addrlen);
         if (ret) {
@@ -98,6 +102,11 @@ static int tcp_open(URLContext *h, const char *uri, int flags)
             ret = ff_neterrno();
             goto fail1;
         }
+        ret = poll(&lp, 1, listen_timeout >= 0 ? listen_timeout : -1);
+        if (ret <= 0) {
+            ret = AVERROR(ETIMEDOUT);
+            goto fail1;
+        }
         fd1 = accept(fd, NULL, NULL);
         if (fd1 < 0) {
             ret = ff_neterrno();

libavresample/audio_mix.c

@@ -305,6 +305,14 @@ int ff_audio_mix_init(AVAudioResampleContext *avr)
 {
     int ret;
 
+    if (avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P &&
+        avr->internal_sample_fmt != AV_SAMPLE_FMT_FLTP) {
+        av_log(avr, AV_LOG_ERROR, "Unsupported internal format for "
+               "mixing: %s\n",
+               av_get_sample_fmt_name(avr->internal_sample_fmt));
+        return AVERROR(EINVAL);
+    }
+
     /* build matrix if the user did not already set one */
     if (!avr->am->matrix) {
         int i, j;

libavresample/avresample.h

@@ -45,6 +45,13 @@ enum AVMixCoeffType {
     AV_MIX_COEFF_TYPE_NB,   /** Number of coeff types. Not part of ABI      */
 };
 
+/** Resampling Filter Types */
+enum AVResampleFilterType {
+    AV_RESAMPLE_FILTER_TYPE_CUBIC,              /**< Cubic */
+    AV_RESAMPLE_FILTER_TYPE_BLACKMAN_NUTTALL,   /**< Blackman Nuttall Windowed Sinc */
+    AV_RESAMPLE_FILTER_TYPE_KAISER,             /**< Kaiser Windowed Sinc */
+};
+
 /**
  * Return the LIBAVRESAMPLE_VERSION_INT constant.
  */

libavresample/internal.h

@@ -50,6 +50,8 @@ struct AVAudioResampleContext {
     int phase_shift;                            /**< log2 of the number of entries in the resampling polyphase filterbank */
     int linear_interp;                          /**< if 1 then the resampling FIR filter will be linearly interpolated */
     double cutoff;                              /**< resampling cutoff frequency. 1.0 corresponds to half the output sample rate */
+    enum AVResampleFilterType filter_type;      /**< resampling filter type */
+    int kaiser_beta;                            /**< beta value for Kaiser window (only applicable if filter_type == AV_FILTER_TYPE_KAISER) */
 
     int in_channels;        /**< number of input channels                   */
     int out_channels;       /**< number of output channels                  */

libavresample/options.c

@@ -39,7 +39,7 @@ static const AVOption options[] = {
     { "out_channel_layout",     "Output Channel Layout",    OFFSET(out_channel_layout),     AV_OPT_TYPE_INT64,  { 0                     }, INT64_MIN,            INT64_MAX,              PARAM },
     { "out_sample_fmt",         "Output Sample Format",     OFFSET(out_sample_fmt),         AV_OPT_TYPE_INT,    { AV_SAMPLE_FMT_S16     }, AV_SAMPLE_FMT_U8,     AV_SAMPLE_FMT_NB-1,     PARAM },
     { "out_sample_rate",        "Output Sample Rate",       OFFSET(out_sample_rate),        AV_OPT_TYPE_INT,    { 48000                 }, 1,                    INT_MAX,                PARAM },
-    { "internal_sample_fmt",    "Internal Sample Format",   OFFSET(internal_sample_fmt),    AV_OPT_TYPE_INT,    { AV_SAMPLE_FMT_FLTP    }, AV_SAMPLE_FMT_NONE,   AV_SAMPLE_FMT_NB-1,     PARAM },
+    { "internal_sample_fmt",    "Internal Sample Format",   OFFSET(internal_sample_fmt),    AV_OPT_TYPE_INT,    { AV_SAMPLE_FMT_NONE    }, AV_SAMPLE_FMT_NONE,   AV_SAMPLE_FMT_NB-1,     PARAM },
     { "mix_coeff_type",         "Mixing Coefficient Type",  OFFSET(mix_coeff_type),         AV_OPT_TYPE_INT,    { AV_MIX_COEFF_TYPE_FLT }, AV_MIX_COEFF_TYPE_Q8, AV_MIX_COEFF_TYPE_NB-1, PARAM, "mix_coeff_type" },
         { "q8",  "16-bit 8.8 Fixed-Point",   0, AV_OPT_TYPE_CONST, { AV_MIX_COEFF_TYPE_Q8  }, INT_MIN, INT_MAX, PARAM, "mix_coeff_type" },
         { "q15", "32-bit 17.15 Fixed-Point", 0, AV_OPT_TYPE_CONST, { AV_MIX_COEFF_TYPE_Q15 }, INT_MIN, INT_MAX, PARAM, "mix_coeff_type" },
@@ -56,6 +56,11 @@ static const AVOption options[] = {
         { "none",  "None",               0, AV_OPT_TYPE_CONST, { AV_MATRIX_ENCODING_NONE  }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
         { "dolby", "Dolby",              0, AV_OPT_TYPE_CONST, { AV_MATRIX_ENCODING_DOLBY }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
         { "dplii", "Dolby Pro Logic II", 0, AV_OPT_TYPE_CONST, { AV_MATRIX_ENCODING_DPLII }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
+    { "filter_type",            "Filter Type",              OFFSET(filter_type),            AV_OPT_TYPE_INT,    { AV_RESAMPLE_FILTER_TYPE_KAISER }, AV_RESAMPLE_FILTER_TYPE_CUBIC, AV_RESAMPLE_FILTER_TYPE_KAISER,  PARAM, "filter_type" },
+        { "cubic",            "Cubic",                          0, AV_OPT_TYPE_CONST, { AV_RESAMPLE_FILTER_TYPE_CUBIC            }, INT_MIN, INT_MAX, PARAM, "filter_type" },
+        { "blackman_nuttall", "Blackman Nuttall Windowed Sinc", 0, AV_OPT_TYPE_CONST, { AV_RESAMPLE_FILTER_TYPE_BLACKMAN_NUTTALL }, INT_MIN, INT_MAX, PARAM, "filter_type" },
+        { "kaiser",           "Kaiser Windowed Sinc",           0, AV_OPT_TYPE_CONST, { AV_RESAMPLE_FILTER_TYPE_KAISER           }, INT_MIN, INT_MAX, PARAM, "filter_type" },
+    { "kaiser_beta",            "Kaiser Window Beta",       OFFSET(kaiser_beta),            AV_OPT_TYPE_INT,    { 9                     }, 2,                    16,                     PARAM },
     { NULL },
 };
 

libavresample/resample.c

@@ -24,37 +24,10 @@
 #include "internal.h"
 #include "audio_data.h"
 
-#ifdef CONFIG_RESAMPLE_FLT
-/* float template */
-#define FILTER_SHIFT  0
-#define FELEM         float
-#define FELEM2        float
-#define FELEML        float
-#define WINDOW_TYPE   24
-#elifdef CONFIG_RESAMPLE_S32
-/* s32 template */
-#define FILTER_SHIFT  30
-#define FELEM         int32_t
-#define FELEM2        int64_t
-#define FELEML        int64_t
-#define FELEM_MAX     INT32_MAX
-#define FELEM_MIN     INT32_MIN
-#define WINDOW_TYPE   12
-#else
-/* s16 template */
-#define FILTER_SHIFT  15
-#define FELEM         int16_t
-#define FELEM2        int32_t
-#define FELEML        int64_t
-#define FELEM_MAX     INT16_MAX
-#define FELEM_MIN     INT16_MIN
-#define WINDOW_TYPE   9
-#endif
-
 struct ResampleContext {
     AVAudioResampleContext *avr;
     AudioData *buffer;
-    FELEM *filter_bank;
+    uint8_t *filter_bank;
     int filter_length;
     int ideal_dst_incr;
     int dst_incr;
@@ -65,9 +38,35 @@ struct ResampleContext {
     int phase_shift;
     int phase_mask;
     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, int no_filter, void *dst0,
+                         int dst_index, const void *src0, int src_size,
+                         int index, int frac);
 };
 
+
+/* double template */
+#define CONFIG_RESAMPLE_DBL
+#include "resample_template.c"
+#undef CONFIG_RESAMPLE_DBL
+
+/* float template */
+#define CONFIG_RESAMPLE_FLT
+#include "resample_template.c"
+#undef CONFIG_RESAMPLE_FLT
+
+/* s32 template */
+#define CONFIG_RESAMPLE_S32
+#include "resample_template.c"
+#undef CONFIG_RESAMPLE_S32
+
+/* s16 template */
+#include "resample_template.c"
+
+
 /**
  * 0th order modified bessel function of the first kind.
  */
@@ -95,17 +94,17 @@ static double bessel(double x)
  * @param      tap_count    tap count
  * @param      phase_count  phase count
  * @param      scale        wanted sum of coefficients for each filter
- * @param      type         0->cubic
+ * @param      filter_type  filter type
- *                          1->blackman nuttall windowed sinc
+ * @param      kaiser_beta  kaiser window beta
- *                          2..16->kaiser windowed sinc beta=2..16
  * @return                  0 on success, negative AVERROR code on failure
  */
-static int build_filter(FELEM *filter, double factor, int tap_count,
+static int build_filter(ResampleContext *c)
-                        int phase_count, int scale, int type)
 {
     int ph, i;
-    double x, y, w;
+    double x, y, w, factor;
     double *tab;
+    int tap_count    = c->filter_length;
+    int phase_count  = 1 << c->phase_shift;
     const int center = (tap_count - 1) / 2;
 
     tab = av_malloc(tap_count * sizeof(*tab));
@@ -113,8 +112,7 @@ static int build_filter(FELEM *filter, double factor, int tap_count,
         return AVERROR(ENOMEM);
 
     /* if upsampling, only need to interpolate, no filter */
-    if (factor > 1.0)
+    factor = FFMIN(c->factor, 1.0);
-        factor = 1.0;
 
     for (ph = 0; ph < phase_count; ph++) {
         double norm = 0;
@@ -122,39 +120,34 @@ static int build_filter(FELEM *filter, double factor, int tap_count,
             x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
             if (x == 0) y = 1.0;
             else        y = sin(x) / x;
-            switch (type) {
+            switch (c->filter_type) {
-            case 0: {
+            case AV_RESAMPLE_FILTER_TYPE_CUBIC: {
                 const float d = -0.5; //first order derivative = -0.5
                 x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
                 if (x < 1.0) y = 1 - 3 * x*x + 2 * x*x*x + d * (                -x*x + x*x*x);
                 else         y =                           d * (-4 + 8 * x - 5 * x*x + x*x*x);
                 break;
             }
-            case 1:
+            case AV_RESAMPLE_FILTER_TYPE_BLACKMAN_NUTTALL:
                 w  = 2.0 * x / (factor * tap_count) + M_PI;
                 y *= 0.3635819 - 0.4891775 * cos(    w) +
                                  0.1365995 * cos(2 * w) -
                                  0.0106411 * cos(3 * w);
                 break;
-            default:
+            case AV_RESAMPLE_FILTER_TYPE_KAISER:
                 w  = 2.0 * x / (factor * tap_count * M_PI);
-                y *= bessel(type * sqrt(FFMAX(1 - w * w, 0)));
+                y *= bessel(c->kaiser_beta * sqrt(FFMAX(1 - w * w, 0)));
                 break;
             }
 
             tab[i] = y;
             norm  += y;
         }
-
         /* normalize so that an uniform color remains the same */
-        for (i = 0; i < tap_count; i++) {
+        for (i = 0; i < tap_count; i++)
-#ifdef CONFIG_RESAMPLE_FLT
+            tab[i] = tab[i] / norm;
-            filter[ph * tap_count + i] = tab[i] / norm;
+
-#else
+        c->set_filter(c->filter_bank, tab, ph, tap_count);
-            filter[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm),
-                                                 FELEM_MIN, FELEM_MAX);
-#endif
-        }
     }
 
     av_free(tab);
@@ -168,9 +161,12 @@ ResampleContext *ff_audio_resample_init(AVAudioResampleContext *avr)
     int in_rate     = avr->in_sample_rate;
     double factor   = FFMIN(out_rate * avr->cutoff / in_rate, 1.0);
     int phase_count = 1 << avr->phase_shift;
+    int felem_size;
 
-    /* TODO: add support for s32 and float internal formats */
+    if (avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P &&
-    if (avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P) {
+        avr->internal_sample_fmt != AV_SAMPLE_FMT_S32P &&
+        avr->internal_sample_fmt != AV_SAMPLE_FMT_FLTP &&
+        avr->internal_sample_fmt != AV_SAMPLE_FMT_DBLP) {
         av_log(avr, AV_LOG_ERROR, "Unsupported internal format for "
                "resampling: %s\n",
                av_get_sample_fmt_name(avr->internal_sample_fmt));
@@ -186,18 +182,40 @@ ResampleContext *ff_audio_resample_init(AVAudioResampleContext *avr)
     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;
 
-    c->filter_bank = av_mallocz(c->filter_length * (phase_count + 1) * sizeof(FELEM));
+    switch (avr->internal_sample_fmt) {
+    case AV_SAMPLE_FMT_DBLP:
+        c->resample_one  = resample_one_dbl;
+        c->set_filter    = set_filter_dbl;
+        break;
+    case AV_SAMPLE_FMT_FLTP:
+        c->resample_one  = resample_one_flt;
+        c->set_filter    = set_filter_flt;
+        break;
+    case AV_SAMPLE_FMT_S32P:
+        c->resample_one  = resample_one_s32;
+        c->set_filter    = set_filter_s32;
+        break;
+    case AV_SAMPLE_FMT_S16P:
+        c->resample_one  = resample_one_s16;
+        c->set_filter    = set_filter_s16;
+        break;
+    }
+
+    felem_size = av_get_bytes_per_sample(avr->internal_sample_fmt);
+    c->filter_bank = av_mallocz(c->filter_length * (phase_count + 1) * felem_size);
     if (!c->filter_bank)
         goto error;
 
-    if (build_filter(c->filter_bank, factor, c->filter_length, phase_count,
+    if (build_filter(c) < 0)
-                     1 << FILTER_SHIFT, WINDOW_TYPE) < 0)
         goto error;
 
-    memcpy(&c->filter_bank[c->filter_length * phase_count + 1],
+    memcpy(&c->filter_bank[(c->filter_length * phase_count + 1) * felem_size],
-           c->filter_bank, (c->filter_length - 1) * sizeof(FELEM));
+           c->filter_bank, (c->filter_length - 1) * felem_size);
-    c->filter_bank[c->filter_length * phase_count] = c->filter_bank[c->filter_length - 1];
+    memcpy(&c->filter_bank[c->filter_length * phase_count * felem_size],
+           &c->filter_bank[(c->filter_length - 1) * felem_size], felem_size);
 
     c->compensation_distance = 0;
     if (!av_reduce(&c->src_incr, &c->dst_incr, out_rate,
@@ -311,10 +329,10 @@ reinit_fail:
     return ret;
 }
 
-static int resample(ResampleContext *c, int16_t *dst, const int16_t *src,
+static int resample(ResampleContext *c, void *dst, const void *src,
                     int *consumed, int src_size, int dst_size, int update_ctx)
 {
-    int dst_index, i;
+    int dst_index;
     int index         = c->index;
     int frac          = c->frac;
     int dst_incr_frac = c->dst_incr % c->src_incr;
@@ -334,7 +352,7 @@ static int resample(ResampleContext *c, int16_t *dst, const int16_t *src,
 
         if (dst) {
             for(dst_index = 0; dst_index < dst_size; dst_index++) {
-                dst[dst_index] = src[index2 >> 32];
+                c->resample_one(c, 1, dst, dst_index, src, 0, index2 >> 32, 0);
                 index2 += incr;
             }
         } else {
@@ -345,42 +363,14 @@ static int resample(ResampleContext *c, int16_t *dst, const int16_t *src,
         frac   = (frac + dst_index * (int64_t)dst_incr_frac) % c->src_incr;
     } else {
         for (dst_index = 0; dst_index < dst_size; dst_index++) {
-            FELEM *filter = c->filter_bank +
-                            c->filter_length * (index & c->phase_mask);
             int sample_index = index >> c->phase_shift;
 
-            if (!dst && (sample_index + c->filter_length > src_size ||
+            if (sample_index + c->filter_length > src_size ||
-                         -sample_index >= src_size))
+                -sample_index >= src_size)
                 break;
 
-            if (dst) {
+            if (dst)
-                FELEM2 val = 0;
+                c->resample_one(c, 0, dst, dst_index, src, src_size, index, frac);
-
-                if (sample_index < 0) {
-                    for (i = 0; i < c->filter_length; i++)
-                        val += src[FFABS(sample_index + i) % src_size] *
-                               (FELEM2)filter[i];
-                } else if (sample_index + c->filter_length > src_size) {
-                    break;
-                } else if (c->linear) {
-                    FELEM2 v2 = 0;
-                    for (i = 0; i < c->filter_length; i++) {
-                        val += src[abs(sample_index + i)] * (FELEM2)filter[i];
-                        v2  += src[abs(sample_index + i)] * (FELEM2)filter[i + c->filter_length];
-                    }
-                    val += (v2 - val) * (FELEML)frac / c->src_incr;
-                } else {
-                    for (i = 0; i < c->filter_length; i++)
-                        val += src[sample_index + i] * (FELEM2)filter[i];
-                }
-
-#ifdef CONFIG_RESAMPLE_FLT
-                dst[dst_index] = av_clip_int16(lrintf(val));
-#else
-                val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;
-                dst[dst_index] = av_clip_int16(val);
-#endif
-            }
 
             frac  += dst_incr_frac;
             index += dst_incr;
@@ -451,8 +441,8 @@ int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src,
 
     /* resample each channel plane */
     for (ch = 0; ch < c->buffer->channels; ch++) {
-        out_samples = resample(c, (int16_t *)dst->data[ch],
+        out_samples = resample(c, (void *)dst->data[ch],
-                               (const int16_t *)c->buffer->data[ch], consumed,
+                               (const void *)c->buffer->data[ch], consumed,
                                c->buffer->nb_samples, dst->allocated_samples,
                                ch + 1 == c->buffer->channels);
     }

libavresample/resample_template.c

@@ -0,0 +1,102 @@
+/*
+ * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#if defined(CONFIG_RESAMPLE_DBL)
+#define SET_TYPE(func)  func ## _dbl
+#define FELEM         double
+#define FELEM2        double
+#define FELEML        double
+#define OUT(d, v) d = v
+#define DBL_TO_FELEM(d, v) d = v
+#elif defined(CONFIG_RESAMPLE_FLT)
+#define SET_TYPE(func)  func ## _flt
+#define FELEM         float
+#define FELEM2        float
+#define FELEML        float
+#define OUT(d, v) d = v
+#define DBL_TO_FELEM(d, v) d = v
+#elif defined(CONFIG_RESAMPLE_S32)
+#define SET_TYPE(func)  func ## _s32
+#define FELEM         int32_t
+#define FELEM2        int64_t
+#define FELEML        int64_t
+#define OUT(d, v) d = av_clipl_int32((v + (1 << 29)) >> 30)
+#define DBL_TO_FELEM(d, v) d = av_clipl_int32(llrint(v * (1 << 30)));
+#else
+#define SET_TYPE(func)  func ## _s16
+#define FELEM         int16_t
+#define FELEM2        int32_t
+#define FELEML        int64_t
+#define OUT(d, v) d = av_clip_int16((v + (1 << 14)) >> 15)
+#define DBL_TO_FELEM(d, v) d = av_clip_int16(lrint(v * (1 << 15)))
+#endif
+
+static void SET_TYPE(resample_one)(ResampleContext *c, int no_filter,
+                                   void *dst0, int dst_index, const void *src0,
+                                   int src_size, int index, int frac)
+{
+    FELEM *dst = dst0;
+    const FELEM *src = src0;
+
+    if (no_filter) {
+        dst[dst_index] = src[index];
+    } else {
+        int i;
+        int sample_index = index >> c->phase_shift;
+        FELEM2 val = 0;
+        FELEM *filter = ((FELEM *)c->filter_bank) +
+                        c->filter_length * (index & c->phase_mask);
+
+        if (sample_index < 0) {
+            for (i = 0; i < c->filter_length; i++)
+                val += src[FFABS(sample_index + i) % src_size] *
+                       (FELEM2)filter[i];
+        } else if (c->linear) {
+            FELEM2 v2 = 0;
+            for (i = 0; i < c->filter_length; i++) {
+                val += src[abs(sample_index + i)] * (FELEM2)filter[i];
+                v2  += src[abs(sample_index + i)] * (FELEM2)filter[i + c->filter_length];
+            }
+            val += (v2 - val) * (FELEML)frac / c->src_incr;
+        } else {
+            for (i = 0; i < c->filter_length; i++)
+                val += src[sample_index + i] * (FELEM2)filter[i];
+        }
+
+        OUT(dst[dst_index], val);
+    }
+}
+
+static void SET_TYPE(set_filter)(void *filter0, double *tab, int phase,
+                                 int tap_count)
+{
+    int i;
+    FELEM *filter = ((FELEM *)filter0) + phase * tap_count;
+    for (i = 0; i < tap_count; i++) {
+        DBL_TO_FELEM(filter[i], tab[i]);
+    }
+}
+
+#undef SET_TYPE
+#undef FELEM
+#undef FELEM2
+#undef FELEML
+#undef OUT
+#undef DBL_TO_FELEM

libavresample/utils.c

@@ -57,18 +57,43 @@ int avresample_open(AVAudioResampleContext *avr)
     avr->resample_needed   = avr->in_sample_rate != avr->out_sample_rate ||
                              avr->force_resampling;
 
-    /* set sample format conversion parameters */
+    /* select internal sample format if not specified by the user */
-    /* override user-requested internal format to avoid unexpected failures
+    if (avr->internal_sample_fmt == AV_SAMPLE_FMT_NONE &&
-       TODO: support more internal formats */
+        (avr->mixing_needed || avr->resample_needed)) {
-    if (avr->resample_needed && avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P) {
+        enum AVSampleFormat  in_fmt = av_get_planar_sample_fmt(avr->in_sample_fmt);
-        av_log(avr, AV_LOG_WARNING, "Using s16p as internal sample format\n");
+        enum AVSampleFormat out_fmt = av_get_planar_sample_fmt(avr->out_sample_fmt);
-        avr->internal_sample_fmt = AV_SAMPLE_FMT_S16P;
+        int max_bps = FFMAX(av_get_bytes_per_sample(in_fmt),
-    } else if (avr->mixing_needed &&
+                            av_get_bytes_per_sample(out_fmt));
-               avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P &&
+        if (max_bps <= 2) {
-               avr->internal_sample_fmt != AV_SAMPLE_FMT_FLTP) {
+            avr->internal_sample_fmt = AV_SAMPLE_FMT_S16P;
-        av_log(avr, AV_LOG_WARNING, "Using fltp as internal sample format\n");
+        } else if (avr->mixing_needed) {
-        avr->internal_sample_fmt = AV_SAMPLE_FMT_FLTP;
+            avr->internal_sample_fmt = AV_SAMPLE_FMT_FLTP;
+        } else {
+            if (max_bps <= 4) {
+                if (in_fmt  == AV_SAMPLE_FMT_S32P ||
+                    out_fmt == AV_SAMPLE_FMT_S32P) {
+                    if (in_fmt  == AV_SAMPLE_FMT_FLTP ||
+                        out_fmt == AV_SAMPLE_FMT_FLTP) {
+                        /* if one is s32 and the other is flt, use dbl */
+                        avr->internal_sample_fmt = AV_SAMPLE_FMT_DBLP;
+                    } else {
+                        /* if one is s32 and the other is s32, s16, or u8, use s32 */
+                        avr->internal_sample_fmt = AV_SAMPLE_FMT_S32P;
+                    }
+                } else {
+                    /* if one is flt and the other is flt, s16 or u8, use flt */
+                    avr->internal_sample_fmt = AV_SAMPLE_FMT_FLTP;
+                }
+            } else {
+                /* if either is dbl, use dbl */
+                avr->internal_sample_fmt = AV_SAMPLE_FMT_DBLP;
+            }
+        }
+        av_log(avr, AV_LOG_DEBUG, "Using %s as internal sample format\n",
+               av_get_sample_fmt_name(avr->internal_sample_fmt));
     }
+
+    /* set sample format conversion parameters */
     if (avr->in_channels == 1)
         avr->in_sample_fmt = av_get_planar_sample_fmt(avr->in_sample_fmt);
     if (avr->out_channels == 1)

tests/fate-run.sh

@@ -75,7 +75,13 @@ probefmt(){
 }
 
 ffmpeg(){
-    run ffmpeg -nostats -threads $threads -thread_type $thread_type -cpuflags $cpuflags "$@"
+    dec_opts="-threads $threads -thread_type $thread_type"
+    ffmpeg_args="-nostats -cpuflags $cpuflags"
+    for arg in $@; do
+        [ ${arg} = -i ] && ffmpeg_args="${ffmpeg_args} ${dec_opts}"
+        ffmpeg_args="${ffmpeg_args} ${arg}"
+    done
+    run ffmpeg ${ffmpeg_args}
 }
 
 framecrc(){