lavr: add option for dithering during sample format conversion to s16

libavresample/Makefile

@@ -8,6 +8,7 @@ OBJS = audio_convert.o                                                  \
        audio_data.o                                                     \
        audio_mix.o                                                      \
        audio_mix_matrix.o                                               \
+       dither.o                                                         \
        options.o                                                        \
        resample.o                                                       \
        utils.o                                                          \

libavresample/audio_convert.c

@@ -29,6 +29,8 @@
 #include "libavutil/samplefmt.h"
 #include "audio_convert.h"
 #include "audio_data.h"
+#include "dither.h"
+#include "internal.h"
 
 enum ConvFuncType {
     CONV_FUNC_TYPE_FLAT,
@@ -46,6 +48,7 @@ typedef void (conv_func_deinterleave)(uint8_t **out, const uint8_t *in, int len,
 
 struct AudioConvert {
     AVAudioResampleContext *avr;
+    DitherContext *dc;
     enum AVSampleFormat in_fmt;
     enum AVSampleFormat out_fmt;
     int channels;
@@ -246,10 +249,18 @@ static void set_generic_function(AudioConvert *ac)
     SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBL)
 }
 
+void ff_audio_convert_free(AudioConvert **ac)
+{
+    if (!*ac)
+        return;
+    ff_dither_free(&(*ac)->dc);
+    av_freep(ac);
+}
+
 AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr,
                                      enum AVSampleFormat out_fmt,
                                      enum AVSampleFormat in_fmt,
-                                     int channels)
+                                     int channels, int sample_rate)
 {
     AudioConvert *ac;
     int in_planar, out_planar;
@@ -263,6 +274,17 @@ AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr,
     ac->in_fmt   = in_fmt;
     ac->channels = channels;
 
+    if (avr->dither_method != AV_RESAMPLE_DITHER_NONE          &&
+        av_get_packed_sample_fmt(out_fmt) == AV_SAMPLE_FMT_S16 &&
+        av_get_bytes_per_sample(in_fmt) > 2) {
+        ac->dc = ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate);
+        if (!ac->dc) {
+            av_free(ac);
+            return NULL;
+        }
+        return ac;
+    }
+
     in_planar  = av_sample_fmt_is_planar(in_fmt);
     out_planar = av_sample_fmt_is_planar(out_fmt);
 
@@ -289,6 +311,15 @@ int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in)
     int use_generic = 1;
     int len         = in->nb_samples;
 
+    if (ac->dc) {
+        /* dithered conversion */
+        av_dlog(ac->avr, "%d samples - audio_convert: %s to %s (dithered)\n",
+                len, av_get_sample_fmt_name(ac->in_fmt),
+                av_get_sample_fmt_name(ac->out_fmt));
+
+        return ff_convert_dither(ac->dc, out, in);
+    }
+
     /* determine whether to use the optimized function based on pointer and
        samples alignment in both the input and output */
     if (ac->has_optimized_func) {

libavresample/audio_convert.h

@@ -54,16 +54,26 @@ void ff_audio_convert_set_func(AudioConvert *ac, enum AVSampleFormat out_fmt,
 /**
  * Allocate and initialize AudioConvert context for sample format conversion.
  *
- * @param avr      AVAudioResampleContext
- * @param out_fmt  output sample format
- * @param in_fmt   input sample format
- * @param channels number of channels
- * @return         newly-allocated AudioConvert context
+ * @param avr         AVAudioResampleContext
+ * @param out_fmt     output sample format
+ * @param in_fmt      input sample format
+ * @param channels    number of channels
+ * @param sample_rate sample rate (used for dithering)
+ * @return            newly-allocated AudioConvert context
  */
 AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr,
                                      enum AVSampleFormat out_fmt,
                                      enum AVSampleFormat in_fmt,
-                                     int channels);
+                                     int channels, int sample_rate);
+
+/**
+ * Free AudioConvert.
+ *
+ * The AudioConvert must have been previously allocated with ff_audio_convert_alloc().
+ *
+ * @param ac  AudioConvert struct
+ */
+void ff_audio_convert_free(AudioConvert **ac);
 
 /**
  * Convert audio data from one sample format to another.

libavresample/avresample.h

@@ -119,6 +119,15 @@ enum AVResampleFilterType {
     AV_RESAMPLE_FILTER_TYPE_KAISER,             /**< Kaiser Windowed Sinc */
 };
 
+enum AVResampleDitherMethod {
+    AV_RESAMPLE_DITHER_NONE,            /**< Do not use dithering */
+    AV_RESAMPLE_DITHER_RECTANGULAR,     /**< Rectangular Dither */
+    AV_RESAMPLE_DITHER_TRIANGULAR,      /**< Triangular Dither*/
+    AV_RESAMPLE_DITHER_TRIANGULAR_HP,   /**< Triangular Dither with High Pass */
+    AV_RESAMPLE_DITHER_TRIANGULAR_NS,   /**< Triangular Dither with Noise Shaping */
+    AV_RESAMPLE_DITHER_NB,              /**< Number of dither types. Not part of ABI. */
+};
+
 /**
  * Return the LIBAVRESAMPLE_VERSION_INT constant.
  */

libavresample/dither.c

+/*
+ * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
+ *
+ * Triangular with Noise Shaping is based on opusfile.
+ * Copyright (c) 1994-2012 by the Xiph.Org Foundation and contributors
+ *
+ * This file is part of Libav.
+ *
+ * Libav 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.
+ *
+ * Libav 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 Libav; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * Dithered Audio Sample Quantization
+ *
+ * Converts from dbl, flt, or s32 to s16 using dithering.
+ */
+
+#include <math.h>
+#include <stdint.h>
+
+#include "libavutil/common.h"
+#include "libavutil/lfg.h"
+#include "libavutil/mem.h"
+#include "libavutil/samplefmt.h"
+#include "audio_convert.h"
+#include "dither.h"
+#include "internal.h"
+
+typedef struct DitherState {
+    int mute;
+    unsigned int seed;
+    AVLFG lfg;
+    float *noise_buf;
+    int noise_buf_size;
+    int noise_buf_ptr;
+    float dither_a[4];
+    float dither_b[4];
+} DitherState;
+
+struct DitherContext {
+    DitherDSPContext  ddsp;
+    enum AVResampleDitherMethod method;
+
+    int mute_dither_threshold;  // threshold for disabling dither
+    int mute_reset_threshold;   // threshold for resetting noise shaping
+    const float *ns_coef_b;     // noise shaping coeffs
+    const float *ns_coef_a;     // noise shaping coeffs
+
+    int channels;
+    DitherState *state;         // dither states for each channel
+
+    AudioData *flt_data;        // input data in fltp
+    AudioData *s16_data;        // dithered output in s16p
+    AudioConvert *ac_in;        // converter for input to fltp
+    AudioConvert *ac_out;       // converter for s16p to s16 (if needed)
+
+    void (*quantize)(int16_t *dst, const float *src, float *dither, int len);
+    int samples_align;
+};
+
+/* mute threshold, in seconds */
+#define MUTE_THRESHOLD_SEC 0.000333
+
+/* scale factor for 16-bit output.
+   The signal is attenuated slightly to avoid clipping */
+#define S16_SCALE 32753.0f
+
+/* scale to convert lfg from INT_MIN/INT_MAX to -0.5/0.5 */
+#define LFG_SCALE (1.0f / (2.0f * INT32_MAX))
+
+/* noise shaping coefficients */
+
+static const float ns_48_coef_b[4] = {
+    2.2374f, -0.7339f, -0.1251f, -0.6033f
+};
+
+static const float ns_48_coef_a[4] = {
+    0.9030f, 0.0116f, -0.5853f, -0.2571f
+};
+
+static const float ns_44_coef_b[4] = {
+    2.2061f, -0.4707f, -0.2534f, -0.6213f
+};
+
+static const float ns_44_coef_a[4] = {
+    1.0587f, 0.0676f, -0.6054f, -0.2738f
+};
+
+static void dither_int_to_float_rectangular_c(float *dst, int *src, int len)
+{
+    int i;
+    for (i = 0; i < len; i++)
+        dst[i] = src[i] * LFG_SCALE;
+}
+
+static void dither_int_to_float_triangular_c(float *dst, int *src0, int len)
+{
+    int i;
+    int *src1  = src0 + len;
+
+    for (i = 0; i < len; i++) {
+        float r = src0[i] * LFG_SCALE;
+        r      += src1[i] * LFG_SCALE;
+        dst[i]  = r;
+    }
+}
+
+static void quantize_c(int16_t *dst, const float *src, float *dither, int len)
+{
+    int i;
+    for (i = 0; i < len; i++)
+        dst[i] = av_clip_int16(lrintf(src[i] * S16_SCALE + dither[i]));
+}
+
+#define SQRT_1_6 0.40824829046386301723f
+
+static void dither_highpass_filter(float *src, int len)
+{
+    int i;
+
+    /* filter is from libswresample in FFmpeg */
+    for (i = 0; i < len - 2; i++)
+        src[i] = (-src[i] + 2 * src[i + 1] - src[i + 2]) * SQRT_1_6;
+}
+
+static int generate_dither_noise(DitherContext *c, DitherState *state,
+                                 int min_samples)
+{
+    int i;
+    int nb_samples  = FFALIGN(min_samples, 16) + 16;
+    int buf_samples = nb_samples *
+                      (c->method == AV_RESAMPLE_DITHER_RECTANGULAR ? 1 : 2);
+    unsigned int *noise_buf_ui;
+
+    av_freep(&state->noise_buf);
+    state->noise_buf_size = state->noise_buf_ptr = 0;
+
+    state->noise_buf = av_malloc(buf_samples * sizeof(*state->noise_buf));
+    if (!state->noise_buf)
+        return AVERROR(ENOMEM);
+    state->noise_buf_size = FFALIGN(min_samples, 16);
+    noise_buf_ui          = (unsigned int *)state->noise_buf;
+
+    av_lfg_init(&state->lfg, state->seed);
+    for (i = 0; i < buf_samples; i++)
+        noise_buf_ui[i] = av_lfg_get(&state->lfg);
+
+    c->ddsp.dither_int_to_float(state->noise_buf, noise_buf_ui, nb_samples);
+
+    if (c->method == AV_RESAMPLE_DITHER_TRIANGULAR_HP)
+        dither_highpass_filter(state->noise_buf, nb_samples);
+
+    return 0;
+}
+
+static void quantize_triangular_ns(DitherContext *c, DitherState *state,
+                                   int16_t *dst, const float *src,
+                                   int nb_samples)
+{
+    int i, j;
+    float *dither = &state->noise_buf[state->noise_buf_ptr];
+
+    if (state->mute > c->mute_reset_threshold)
+        memset(state->dither_a, 0, sizeof(state->dither_a));
+
+    for (i = 0; i < nb_samples; i++) {
+        float err = 0;
+        float sample = src[i] * S16_SCALE;
+
+        for (j = 0; j < 4; j++) {
+            err += c->ns_coef_b[j] * state->dither_b[j] -
+                   c->ns_coef_a[j] * state->dither_a[j];
+        }
+        for (j = 3; j > 0; j--) {
+            state->dither_a[j] = state->dither_a[j - 1];
+            state->dither_b[j] = state->dither_b[j - 1];
+        }
+        state->dither_a[0] = err;
+        sample -= err;
+
+        if (state->mute > c->mute_dither_threshold) {
+            dst[i]             = av_clip_int16(lrintf(sample));
+            state->dither_b[0] = 0;
+        } else {
+            dst[i]             = av_clip_int16(lrintf(sample + dither[i]));
+            state->dither_b[0] = av_clipf(dst[i] - sample, -1.5f, 1.5f);
+        }
+
+        state->mute++;
+        if (src[i])
+            state->mute = 0;
+    }
+}
+
+static int convert_samples(DitherContext *c, int16_t **dst, float * const *src,
+                           int channels, int nb_samples)
+{
+    int ch, ret;
+    int aligned_samples = FFALIGN(nb_samples, 16);
+
+    for (ch = 0; ch < channels; ch++) {
+        DitherState *state = &c->state[ch];
+
+        if (state->noise_buf_size < aligned_samples) {
+            ret = generate_dither_noise(c, state, nb_samples);
+            if (ret < 0)
+                return ret;
+        } else if (state->noise_buf_size - state->noise_buf_ptr < aligned_samples) {
+            state->noise_buf_ptr = 0;
+        }
+
+        if (c->method == AV_RESAMPLE_DITHER_TRIANGULAR_NS) {
+            quantize_triangular_ns(c, state, dst[ch], src[ch], nb_samples);
+        } else {
+            c->quantize(dst[ch], src[ch],
+                        &state->noise_buf[state->noise_buf_ptr],
+                        FFALIGN(nb_samples, c->samples_align));
+        }
+
+        state->noise_buf_ptr += aligned_samples;
+    }
+
+    return 0;
+}
+
+int ff_convert_dither(DitherContext *c, AudioData *dst, AudioData *src)
+{
+    int ret;
+    AudioData *flt_data;
+
+    /* output directly to dst if it is planar */
+    if (dst->sample_fmt == AV_SAMPLE_FMT_S16P)
+        c->s16_data = dst;
+    else {
+        /* make sure s16_data is large enough for the output */
+        ret = ff_audio_data_realloc(c->s16_data, src->nb_samples);
+        if (ret < 0)
+            return ret;
+    }
+
+    if (src->sample_fmt != AV_SAMPLE_FMT_FLTP) {
+        /* make sure flt_data is large enough for the input */
+        ret = ff_audio_data_realloc(c->flt_data, src->nb_samples);
+        if (ret < 0)
+            return ret;
+        flt_data = c->flt_data;
+
+        /* convert input samples to fltp and scale to s16 range */
+        ret = ff_audio_convert(c->ac_in, flt_data, src);
+        if (ret < 0)
+            return ret;
+    } else {
+        flt_data = src;
+    }
+
+    /* check alignment and padding constraints */
+    if (c->method != AV_RESAMPLE_DITHER_TRIANGULAR_NS) {
+        int ptr_align     = FFMIN(flt_data->ptr_align,     c->s16_data->ptr_align);
+        int samples_align = FFMIN(flt_data->samples_align, c->s16_data->samples_align);
+        int aligned_len   = FFALIGN(src->nb_samples, c->ddsp.samples_align);
+
+        if (!(ptr_align % c->ddsp.ptr_align) && samples_align >= aligned_len) {
+            c->quantize      = c->ddsp.quantize;
+            c->samples_align = c->ddsp.samples_align;
+        } else {
+            c->quantize      = quantize_c;
+            c->samples_align = 1;
+        }
+    }
+
+    ret = convert_samples(c, (int16_t **)c->s16_data->data,
+                          (float * const *)flt_data->data, src->channels,
+                          src->nb_samples);
+    if (ret < 0)
+        return ret;
+
+    c->s16_data->nb_samples = src->nb_samples;
+
+    /* interleave output to dst if needed */
+    if (dst->sample_fmt == AV_SAMPLE_FMT_S16) {
+        ret = ff_audio_convert(c->ac_out, dst, c->s16_data);
+        if (ret < 0)
+            return ret;
+    } else
+        c->s16_data = NULL;
+
+    return 0;
+}
+
+void ff_dither_free(DitherContext **cp)
+{
+    DitherContext *c = *cp;
+    int ch;
+
+    if (!c)
+        return;
+    ff_audio_data_free(&c->flt_data);
+    ff_audio_data_free(&c->s16_data);
+    ff_audio_convert_free(&c->ac_in);
+    ff_audio_convert_free(&c->ac_out);
+    for (ch = 0; ch < c->channels; ch++)
+        av_free(c->state[ch].noise_buf);
+    av_free(c->state);
+    av_freep(cp);
+}
+
+static void dither_init(DitherDSPContext *ddsp,
+                        enum AVResampleDitherMethod method)
+{
+    ddsp->quantize      = quantize_c;
+    ddsp->ptr_align     = 1;
+    ddsp->samples_align = 1;
+
+    if (method == AV_RESAMPLE_DITHER_RECTANGULAR)
+        ddsp->dither_int_to_float = dither_int_to_float_rectangular_c;
+    else
+        ddsp->dither_int_to_float = dither_int_to_float_triangular_c;
+}
+
+DitherContext *ff_dither_alloc(AVAudioResampleContext *avr,
+                               enum AVSampleFormat out_fmt,
+                               enum AVSampleFormat in_fmt,
+                               int channels, int sample_rate)
+{
+    AVLFG seed_gen;
+    DitherContext *c;
+    int ch;
+
+    if (av_get_packed_sample_fmt(out_fmt) != AV_SAMPLE_FMT_S16 ||
+        av_get_bytes_per_sample(in_fmt) <= 2) {
+        av_log(avr, AV_LOG_ERROR, "dithering %s to %s is not supported\n",
+               av_get_sample_fmt_name(in_fmt), av_get_sample_fmt_name(out_fmt));
+        return NULL;
+    }
+
+    c = av_mallocz(sizeof(*c));
+    if (!c)
+        return NULL;
+
+    if (avr->dither_method == AV_RESAMPLE_DITHER_TRIANGULAR_NS &&
+        sample_rate != 48000 && sample_rate != 44100) {
+        av_log(avr, AV_LOG_WARNING, "sample rate must be 48000 or 44100 Hz "
+               "for triangular_ns dither. using triangular_hp instead.\n");
+        avr->dither_method = AV_RESAMPLE_DITHER_TRIANGULAR_HP;
+    }
+    c->method = avr->dither_method;
+    dither_init(&c->ddsp, c->method);
+
+    if (c->method == AV_RESAMPLE_DITHER_TRIANGULAR_NS) {
+        if (sample_rate == 48000) {
+            c->ns_coef_b = ns_48_coef_b;
+            c->ns_coef_a = ns_48_coef_a;
+        } else {
+            c->ns_coef_b = ns_44_coef_b;
+            c->ns_coef_a = ns_44_coef_a;
+        }
+    }
+
+    /* Either s16 or s16p output format is allowed, but s16p is used
+       internally, so we need to use a temp buffer and interleave if the output
+       format is s16 */
+    if (out_fmt != AV_SAMPLE_FMT_S16P) {
+        c->s16_data = ff_audio_data_alloc(channels, 1024, AV_SAMPLE_FMT_S16P,
+                                          "dither s16 buffer");
+        if (!c->s16_data)
+            goto fail;
+
+        c->ac_out = ff_audio_convert_alloc(avr, out_fmt, AV_SAMPLE_FMT_S16P,
+                                           channels, sample_rate);
+        if (!c->ac_out)
+            goto fail;
+    }
+
+    if (in_fmt != AV_SAMPLE_FMT_FLTP) {
+        c->flt_data = ff_audio_data_alloc(channels, 1024, AV_SAMPLE_FMT_FLTP,
+                                          "dither flt buffer");
+        if (!c->flt_data)
+            goto fail;
+
+        c->ac_in = ff_audio_convert_alloc(avr, AV_SAMPLE_FMT_FLTP, in_fmt,
+                                          channels, sample_rate);
+        if (!c->ac_in)
+            goto fail;
+    }
+
+    c->state = av_mallocz(channels * sizeof(*c->state));
+    if (!c->state)
+        goto fail;
+    c->channels = channels;
+
+    /* calculate thresholds for turning off dithering during periods of
+       silence to avoid replacing digital silence with quiet dither noise */
+    c->mute_dither_threshold = lrintf(sample_rate * MUTE_THRESHOLD_SEC);
+    c->mute_reset_threshold  = c->mute_dither_threshold * 4;
+
+    /* initialize dither states */
+    av_lfg_init(&seed_gen, 0xC0FFEE);
+    for (ch = 0; ch < channels; ch++) {
+        DitherState *state = &c->state[ch];
+        state->mute = c->mute_reset_threshold + 1;
+        state->seed = av_lfg_get(&seed_gen);
+        generate_dither_noise(c, state, FFMAX(32768, sample_rate / 2));
+    }
+
+    return c;
+
+fail:
+    ff_dither_free(&c);
+    return NULL;
+}

libavresample/dither.h

+/*
+ * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
+ *
+ * This file is part of Libav.
+ *
+ * Libav 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.
+ *
+ * Libav 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 Libav; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVRESAMPLE_DITHER_H
+#define AVRESAMPLE_DITHER_H
+
+#include "avresample.h"
+#include "audio_data.h"
+
+typedef struct DitherContext DitherContext;
+
+typedef struct DitherDSPContext {
+    /**
+     * Convert samples from flt to s16 with added dither noise.
+     *
+     * @param dst    destination float array, range -0.5 to 0.5
+     * @param src    source int array, range INT_MIN to INT_MAX.
+     * @param dither float dither noise array
+     * @param len    number of samples
+     */
+    void (*quantize)(int16_t *dst, const float *src, float *dither, int len);
+
+    int ptr_align;      ///< src and dst constraits for quantize()
+    int samples_align;  ///< len constraits for quantize()
+
+    /**
+     * Convert dither noise from int to float with triangular distribution.
+     *
+     * @param dst  destination float array, range -0.5 to 0.5
+     *             constraints: 32-byte aligned
+     * @param src0 source int array, range INT_MIN to INT_MAX.
+     *             the array size is len * 2
+     *             constraints: 32-byte aligned
+     * @param len  number of output noise samples
+     *             constraints: multiple of 16
+     */
+    void (*dither_int_to_float)(float *dst, int *src0, int len);
+} DitherDSPContext;
+
+/**
+ * Allocate and initialize a DitherContext.
+ *
+ * The parameters in the AVAudioResampleContext are used to initialize the
+ * DitherContext.
+ *
+ * @param avr  AVAudioResampleContext
+ * @return     newly-allocated DitherContext
+ */
+DitherContext *ff_dither_alloc(AVAudioResampleContext *avr,
+                               enum AVSampleFormat out_fmt,
+                               enum AVSampleFormat in_fmt,
+                               int channels, int sample_rate);
+
+/**
+ * Free a DitherContext.
+ *
+ * @param c  DitherContext
+ */
+void ff_dither_free(DitherContext **c);
+
+/**
+ * Convert audio sample format with dithering.
+ *
+ * @param c    DitherContext
+ * @param dst  destination audio data
+ * @param src  source audio data
+ * @return     0 if ok, negative AVERROR code on failure
+ */
+int ff_convert_dither(DitherContext *c, AudioData *dst, AudioData *src);
+
+#endif /* AVRESAMPLE_DITHER_H */

libavresample/internal.h

@@ -53,6 +53,7 @@ struct AVAudioResampleContext {
     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) */
+    enum AVResampleDitherMethod dither_method;  /**< dither method          */
 
     int in_channels;        /**< number of input channels                   */
     int out_channels;       /**< number of output channels                  */

libavresample/options.c

@@ -63,6 +63,12 @@ static const AVOption options[] = {
         { "blackman_nuttall", "Blackman Nuttall Windowed Sinc", 0, AV_OPT_TYPE_CONST, { .i64 = AV_RESAMPLE_FILTER_TYPE_BLACKMAN_NUTTALL }, INT_MIN, INT_MAX, PARAM, "filter_type" },
         { "kaiser",           "Kaiser Windowed Sinc",           0, AV_OPT_TYPE_CONST, { .i64 = AV_RESAMPLE_FILTER_TYPE_KAISER           }, INT_MIN, INT_MAX, PARAM, "filter_type" },
     { "kaiser_beta",            "Kaiser Window Beta",       OFFSET(kaiser_beta),            AV_OPT_TYPE_INT,    { .i64 = 9              }, 2,                    16,                     PARAM },
+    { "dither_method",          "Dither Method",            OFFSET(dither_method),          AV_OPT_TYPE_INT,    { .i64 = AV_RESAMPLE_DITHER_NONE }, 0, AV_RESAMPLE_DITHER_NB-1, PARAM, "dither_method"},
+        {"none",          "No Dithering",                         0, AV_OPT_TYPE_CONST, { .i64 = AV_RESAMPLE_DITHER_NONE          }, INT_MIN, INT_MAX, PARAM, "dither_method"},
+        {"rectangular",   "Rectangular Dither",                   0, AV_OPT_TYPE_CONST, { .i64 = AV_RESAMPLE_DITHER_RECTANGULAR   }, INT_MIN, INT_MAX, PARAM, "dither_method"},
+        {"triangular",    "Triangular Dither",                    0, AV_OPT_TYPE_CONST, { .i64 = AV_RESAMPLE_DITHER_TRIANGULAR    }, INT_MIN, INT_MAX, PARAM, "dither_method"},
+        {"triangular_hp", "Triangular Dither With High Pass",     0, AV_OPT_TYPE_CONST, { .i64 = AV_RESAMPLE_DITHER_TRIANGULAR_HP }, INT_MIN, INT_MAX, PARAM, "dither_method"},
+        {"triangular_ns", "Triangular Dither With Noise Shaping", 0, AV_OPT_TYPE_CONST, { .i64 = AV_RESAMPLE_DITHER_TRIANGULAR_NS }, INT_MIN, INT_MAX, PARAM, "dither_method"},
     { NULL },
 };
 

libavresample/utils.c

@@ -142,7 +142,8 @@ int avresample_open(AVAudioResampleContext *avr)
     /* setup contexts */
     if (avr->in_convert_needed) {
         avr->ac_in = ff_audio_convert_alloc(avr, avr->internal_sample_fmt,
-                                            avr->in_sample_fmt, avr->in_channels);
+                                            avr->in_sample_fmt, avr->in_channels,
+                                            avr->in_sample_rate);
         if (!avr->ac_in) {
             ret = AVERROR(ENOMEM);
             goto error;
@@ -155,7 +156,8 @@ int avresample_open(AVAudioResampleContext *avr)
         else
             src_fmt = avr->in_sample_fmt;
         avr->ac_out = ff_audio_convert_alloc(avr, avr->out_sample_fmt, src_fmt,
-                                             avr->out_channels);
+                                             avr->out_channels,
+                                             avr->out_sample_rate);
         if (!avr->ac_out) {
             ret = AVERROR(ENOMEM);
             goto error;
@@ -190,8 +192,8 @@ void avresample_close(AVAudioResampleContext *avr)
     ff_audio_data_free(&avr->out_buffer);
     av_audio_fifo_free(avr->out_fifo);
     avr->out_fifo = NULL;
-    av_freep(&avr->ac_in);
-    av_freep(&avr->ac_out);
+    ff_audio_convert_free(&avr->ac_in);
+    ff_audio_convert_free(&avr->ac_out);
     ff_audio_resample_free(&avr->resample);
     ff_audio_mix_free(&avr->am);
     av_freep(&avr->mix_matrix);

libavresample/version.h

@@ -21,7 +21,7 @@
 
 #define LIBAVRESAMPLE_VERSION_MAJOR  1
 #define LIBAVRESAMPLE_VERSION_MINOR  0
-#define LIBAVRESAMPLE_VERSION_MICRO  0
+#define LIBAVRESAMPLE_VERSION_MICRO  1
 
 #define LIBAVRESAMPLE_VERSION_INT  AV_VERSION_INT(LIBAVRESAMPLE_VERSION_MAJOR, \
                                                   LIBAVRESAMPLE_VERSION_MINOR, \