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>

7 files changed, 246 insertions, 107 deletions

diff --git a/libavresample/audio_mix.c b/libavresample/audio_mix.c
index 93192221cd..2c2a356844 100644
--- a/libavresample/audio_mix.c
+++ b/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;
diff --git a/libavresample/avresample.h b/libavresample/avresample.h
index 002bec21fb..b93aba5d73 100644
--- a/libavresample/avresample.h
+++ b/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.
  */
diff --git a/libavresample/internal.h b/libavresample/internal.h
index fa9499a8ef..7b7648f0be 100644
--- a/libavresample/internal.h
+++ b/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                  */
diff --git a/libavresample/options.c b/libavresample/options.c
index a1a0b0ca21..02e1f86308 100644
--- a/libavresample/options.c
+++ b/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 },
 };
 
diff --git a/libavresample/resample.c b/libavresample/resample.c
index 5529fafe8d..1c3d13ae0a 100644
--- a/libavresample/resample.c
+++ b/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
- *                          1->blackman nuttall windowed sinc
- *                          2..16->kaiser windowed sinc beta=2..16
+ * @param      filter_type  filter type
+ * @param      kaiser_beta  kaiser window beta
  * @return                  0 on success, negative AVERROR code on failure
  */
-static int build_filter(FELEM *filter, double factor, int tap_count,
-                        int phase_count, int scale, int type)
+static int build_filter(ResampleContext *c)
 {
     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 = 1.0;
+    factor = FFMIN(c->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) {
-            case 0: {
+            switch (c->filter_type) {
+            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++) {
-#ifdef CONFIG_RESAMPLE_FLT
-            filter[ph * tap_count + i] = tab[i] / norm;
-#else
-            filter[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm),
-                                                 FELEM_MIN, FELEM_MAX);
-#endif
-        }
+        for (i = 0; i < tap_count; i++)
+            tab[i] = tab[i] / norm;
+
+        c->set_filter(c->filter_bank, tab, ph, tap_count);
     }
 
     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;
+
+    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;
+    }
 
-    c->filter_bank = av_mallocz(c->filter_length * (phase_count + 1) * sizeof(FELEM));
+    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,
-                     1 << FILTER_SHIFT, WINDOW_TYPE) < 0)
+    if (build_filter(c) < 0)
         goto error;
 
-    memcpy(&c->filter_bank[c->filter_length * phase_count + 1],
-           c->filter_bank, (c->filter_length - 1) * sizeof(FELEM));
-    c->filter_bank[c->filter_length * phase_count] = c->filter_bank[c->filter_length - 1];
+    memcpy(&c->filter_bank[(c->filter_length * phase_count + 1) * felem_size],
+           c->filter_bank, (c->filter_length - 1) * felem_size);
+    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 ||
-                         -sample_index >= src_size))
+            if (sample_index + c->filter_length > src_size ||
+                -sample_index >= src_size)
                 break;
 
-            if (dst) {
-                FELEM2 val = 0;
-
-                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
-            }
+            if (dst)
+                c->resample_one(c, 0, dst, dst_index, src, src_size, index, frac);
 
             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],
-                               (const int16_t *)c->buffer->data[ch], consumed,
+        out_samples = resample(c, (void *)dst->data[ch],
+                               (const void *)c->buffer->data[ch], consumed,
                                c->buffer->nb_samples, dst->allocated_samples,
                                ch + 1 == c->buffer->channels);
     }
diff --git a/libavresample/resample_template.c b/libavresample/resample_template.c
new file mode 100644
index 0000000000..06da90fe9f
--- /dev/null
+++ b/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
diff --git a/libavresample/utils.c b/libavresample/utils.c
index 21bb9309b3..caf9081e5d 100644
--- a/libavresample/utils.c
+++ b/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 */
-    /* override user-requested internal format to avoid unexpected failures
-       TODO: support more internal formats */
-    if (avr->resample_needed && avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P) {
-        av_log(avr, AV_LOG_WARNING, "Using s16p as internal sample format\n");
-        avr->internal_sample_fmt = AV_SAMPLE_FMT_S16P;
-    } else if (avr->mixing_needed &&
-               avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P &&
-               avr->internal_sample_fmt != AV_SAMPLE_FMT_FLTP) {
-        av_log(avr, AV_LOG_WARNING, "Using fltp as internal sample format\n");
-        avr->internal_sample_fmt = AV_SAMPLE_FMT_FLTP;
+    /* select internal sample format if not specified by the user */
+    if (avr->internal_sample_fmt == AV_SAMPLE_FMT_NONE &&
+        (avr->mixing_needed || avr->resample_needed)) {
+        enum AVSampleFormat  in_fmt = av_get_planar_sample_fmt(avr->in_sample_fmt);
+        enum AVSampleFormat out_fmt = av_get_planar_sample_fmt(avr->out_sample_fmt);
+        int max_bps = FFMAX(av_get_bytes_per_sample(in_fmt),
+                            av_get_bytes_per_sample(out_fmt));
+        if (max_bps <= 2) {
+            avr->internal_sample_fmt = AV_SAMPLE_FMT_S16P;
+        } else if (avr->mixing_needed) {
+            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)