Add libavresample

This is a new library for audio sample format, channel layout, and sample rate
conversion.

1 files changed, 480 insertions, 0 deletions

diff --git a/libavresample/resample.c b/libavresample/resample.c
new file mode 100644
index 0000000000..5529fafe8d
--- /dev/null
+++ b/libavresample/resample.c
@@ -0,0 +1,480 @@
+/*
+ * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
+ * 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
+ */
+
+#include "libavutil/libm.h"
+#include "libavutil/log.h"
+#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;
+    int filter_length;
+    int ideal_dst_incr;
+    int dst_incr;
+    int index;
+    int frac;
+    int src_incr;
+    int compensation_distance;
+    int phase_shift;
+    int phase_mask;
+    int linear;
+    double factor;
+};
+
+/**
+ * 0th order modified bessel function of the first kind.
+ */
+static double bessel(double x)
+{
+    double v     = 1;
+    double lastv = 0;
+    double t     = 1;
+    int i;
+
+    x = x * x / 4;
+    for (i = 1; v != lastv; i++) {
+        lastv = v;
+        t    *= x / (i * i);
+        v    += t;
+    }
+    return v;
+}
+
+/**
+ * Build a polyphase filterbank.
+ *
+ * @param[out] filter       filter coefficients
+ * @param      factor       resampling factor
+ * @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
+ * @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)
+{
+    int ph, i;
+    double x, y, w;
+    double *tab;
+    const int center = (tap_count - 1) / 2;
+
+    tab = av_malloc(tap_count * sizeof(*tab));
+    if (!tab)
+        return AVERROR(ENOMEM);
+
+    /* if upsampling, only need to interpolate, no filter */
+    if (factor > 1.0)
+        factor = 1.0;
+
+    for (ph = 0; ph < phase_count; ph++) {
+        double norm = 0;
+        for (i = 0; i < tap_count; i++) {
+            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: {
+                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:
+                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:
+                w  = 2.0 * x / (factor * tap_count * M_PI);
+                y *= bessel(type * 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
+        }
+    }
+
+    av_free(tab);
+    return 0;
+}
+
+ResampleContext *ff_audio_resample_init(AVAudioResampleContext *avr)
+{
+    ResampleContext *c;
+    int out_rate    = avr->out_sample_rate;
+    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;
+
+    /* TODO: add support for s32 and float internal formats */
+    if (avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P) {
+        av_log(avr, AV_LOG_ERROR, "Unsupported internal format for "
+               "resampling: %s\n",
+               av_get_sample_fmt_name(avr->internal_sample_fmt));
+        return NULL;
+    }
+    c = av_mallocz(sizeof(*c));
+    if (!c)
+        return NULL;
+
+    c->avr           = avr;
+    c->phase_shift   = avr->phase_shift;
+    c->phase_mask    = phase_count - 1;
+    c->linear        = avr->linear_interp;
+    c->factor        = factor;
+    c->filter_length = FFMAX((int)ceil(avr->filter_size / factor), 1);
+
+    c->filter_bank = av_mallocz(c->filter_length * (phase_count + 1) * sizeof(FELEM));
+    if (!c->filter_bank)
+        goto error;
+
+    if (build_filter(c->filter_bank, factor, c->filter_length, phase_count,
+                     1 << FILTER_SHIFT, WINDOW_TYPE) < 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];
+
+    c->compensation_distance = 0;
+    if (!av_reduce(&c->src_incr, &c->dst_incr, out_rate,
+                   in_rate * (int64_t)phase_count, INT32_MAX / 2))
+        goto error;
+    c->ideal_dst_incr = c->dst_incr;
+
+    c->index = -phase_count * ((c->filter_length - 1) / 2);
+    c->frac  = 0;
+
+    /* allocate internal buffer */
+    c->buffer = ff_audio_data_alloc(avr->resample_channels, 0,
+                                    avr->internal_sample_fmt,
+                                    "resample buffer");
+    if (!c->buffer)
+        goto error;
+
+    av_log(avr, AV_LOG_DEBUG, "resample: %s from %d Hz to %d Hz\n",
+           av_get_sample_fmt_name(avr->internal_sample_fmt),
+           avr->in_sample_rate, avr->out_sample_rate);
+
+    return c;
+
+error:
+    ff_audio_data_free(&c->buffer);
+    av_free(c->filter_bank);
+    av_free(c);
+    return NULL;
+}
+
+void ff_audio_resample_free(ResampleContext **c)
+{
+    if (!*c)
+        return;
+    ff_audio_data_free(&(*c)->buffer);
+    av_free((*c)->filter_bank);
+    av_freep(c);
+}
+
+int avresample_set_compensation(AVAudioResampleContext *avr, int sample_delta,
+                                int compensation_distance)
+{
+    ResampleContext *c;
+    AudioData *fifo_buf = NULL;
+    int ret = 0;
+
+    if (compensation_distance < 0)
+        return AVERROR(EINVAL);
+    if (!compensation_distance && sample_delta)
+        return AVERROR(EINVAL);
+
+    /* if resampling was not enabled previously, re-initialize the
+       AVAudioResampleContext and force resampling */
+    if (!avr->resample_needed) {
+        int fifo_samples;
+        double matrix[AVRESAMPLE_MAX_CHANNELS * AVRESAMPLE_MAX_CHANNELS] = { 0 };
+
+        /* buffer any remaining samples in the output FIFO before closing */
+        fifo_samples = av_audio_fifo_size(avr->out_fifo);
+        if (fifo_samples > 0) {
+            fifo_buf = ff_audio_data_alloc(avr->out_channels, fifo_samples,
+                                           avr->out_sample_fmt, NULL);
+            if (!fifo_buf)
+                return AVERROR(EINVAL);
+            ret = ff_audio_data_read_from_fifo(avr->out_fifo, fifo_buf,
+                                               fifo_samples);
+            if (ret < 0)
+                goto reinit_fail;
+        }
+        /* save the channel mixing matrix */
+        ret = avresample_get_matrix(avr, matrix, AVRESAMPLE_MAX_CHANNELS);
+        if (ret < 0)
+            goto reinit_fail;
+
+        /* close the AVAudioResampleContext */
+        avresample_close(avr);
+
+        avr->force_resampling = 1;
+
+        /* restore the channel mixing matrix */
+        ret = avresample_set_matrix(avr, matrix, AVRESAMPLE_MAX_CHANNELS);
+        if (ret < 0)
+            goto reinit_fail;
+
+        /* re-open the AVAudioResampleContext */
+        ret = avresample_open(avr);
+        if (ret < 0)
+            goto reinit_fail;
+
+        /* restore buffered samples to the output FIFO */
+        if (fifo_samples > 0) {
+            ret = ff_audio_data_add_to_fifo(avr->out_fifo, fifo_buf, 0,
+                                            fifo_samples);
+            if (ret < 0)
+                goto reinit_fail;
+            ff_audio_data_free(&fifo_buf);
+        }
+    }
+    c = avr->resample;
+    c->compensation_distance = compensation_distance;
+    if (compensation_distance) {
+        c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr *
+                      (int64_t)sample_delta / compensation_distance;
+    } else {
+        c->dst_incr = c->ideal_dst_incr;
+    }
+    return 0;
+
+reinit_fail:
+    ff_audio_data_free(&fifo_buf);
+    return ret;
+}
+
+static int resample(ResampleContext *c, int16_t *dst, const int16_t *src,
+                    int *consumed, int src_size, int dst_size, int update_ctx)
+{
+    int dst_index, i;
+    int index         = c->index;
+    int frac          = c->frac;
+    int dst_incr_frac = c->dst_incr % c->src_incr;
+    int dst_incr      = c->dst_incr / c->src_incr;
+    int compensation_distance = c->compensation_distance;
+
+    if (!dst != !src)
+        return AVERROR(EINVAL);
+
+    if (compensation_distance == 0 && c->filter_length == 1 &&
+        c->phase_shift == 0) {
+        int64_t index2 = ((int64_t)index) << 32;
+        int64_t incr   = (1LL << 32) * c->dst_incr / c->src_incr;
+        dst_size       = FFMIN(dst_size,
+                               (src_size-1-index) * (int64_t)c->src_incr /
+                               c->dst_incr);
+
+        if (dst) {
+            for(dst_index = 0; dst_index < dst_size; dst_index++) {
+                dst[dst_index] = src[index2 >> 32];
+                index2 += incr;
+            }
+        } else {
+            dst_index = dst_size;
+        }
+        index += dst_index * dst_incr;
+        index += (frac + dst_index * (int64_t)dst_incr_frac) / c->src_incr;
+        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))
+                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
+            }
+
+            frac  += dst_incr_frac;
+            index += dst_incr;
+            if (frac >= c->src_incr) {
+                frac -= c->src_incr;
+                index++;
+            }
+            if (dst_index + 1 == compensation_distance) {
+                compensation_distance = 0;
+                dst_incr_frac = c->ideal_dst_incr % c->src_incr;
+                dst_incr      = c->ideal_dst_incr / c->src_incr;
+            }
+        }
+    }
+    if (consumed)
+        *consumed = FFMAX(index, 0) >> c->phase_shift;
+
+    if (update_ctx) {
+        if (index >= 0)
+            index &= c->phase_mask;
+
+        if (compensation_distance) {
+            compensation_distance -= dst_index;
+            if (compensation_distance <= 0)
+                return AVERROR_BUG;
+        }
+        c->frac     = frac;
+        c->index    = index;
+        c->dst_incr = dst_incr_frac + c->src_incr*dst_incr;
+        c->compensation_distance = compensation_distance;
+    }
+
+    return dst_index;
+}
+
+int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src,
+                      int *consumed)
+{
+    int ch, in_samples, in_leftover, out_samples = 0;
+    int ret = AVERROR(EINVAL);
+
+    in_samples  = src ? src->nb_samples : 0;
+    in_leftover = c->buffer->nb_samples;
+
+    /* add input samples to the internal buffer */
+    if (src) {
+        ret = ff_audio_data_combine(c->buffer, in_leftover, src, 0, in_samples);
+        if (ret < 0)
+            return ret;
+    } else if (!in_leftover) {
+        /* no remaining samples to flush */
+        return 0;
+    } else {
+        /* TODO: pad buffer to flush completely */
+    }
+
+    /* calculate output size and reallocate output buffer if needed */
+    /* TODO: try to calculate this without the dummy resample() run */
+    if (!dst->read_only && dst->allow_realloc) {
+        out_samples = resample(c, NULL, NULL, NULL, c->buffer->nb_samples,
+                               INT_MAX, 0);
+        ret = ff_audio_data_realloc(dst, out_samples);
+        if (ret < 0) {
+            av_log(c->avr, AV_LOG_ERROR, "error reallocating output\n");
+            return ret;
+        }
+    }
+
+    /* 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,
+                               c->buffer->nb_samples, dst->allocated_samples,
+                               ch + 1 == c->buffer->channels);
+    }
+    if (out_samples < 0) {
+        av_log(c->avr, AV_LOG_ERROR, "error during resampling\n");
+        return out_samples;
+    }
+
+    /* drain consumed samples from the internal buffer */
+    ff_audio_data_drain(c->buffer, *consumed);
+
+    av_dlog(c->avr, "resampled %d in + %d leftover to %d out + %d leftover\n",
+            in_samples, in_leftover, out_samples, c->buffer->nb_samples);
+
+    dst->nb_samples = out_samples;
+    return 0;
+}
+
+int avresample_get_delay(AVAudioResampleContext *avr)
+{
+    if (!avr->resample_needed || !avr->resample)
+        return 0;
+
+    return avr->resample->buffer->nb_samples;
+}