Add libavresample

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

1 files changed, 340 insertions, 0 deletions

diff --git a/libavresample/avresample-test.c b/libavresample/avresample-test.c
new file mode 100644
index 0000000000..ad2f16d6f6
--- /dev/null
+++ b/libavresample/avresample-test.c
@@ -0,0 +1,340 @@
+/*
+ * Copyright (c) 2002 Fabrice Bellard
+ * 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 <stdint.h>
+#include <stdio.h>
+
+#include "libavutil/avstring.h"
+#include "libavutil/lfg.h"
+#include "libavutil/libm.h"
+#include "libavutil/log.h"
+#include "libavutil/mem.h"
+#include "libavutil/opt.h"
+#include "libavutil/samplefmt.h"
+#include "avresample.h"
+
+static double dbl_rand(AVLFG *lfg)
+{
+    return 2.0 * (av_lfg_get(lfg) / (double)UINT_MAX) - 1.0;
+}
+
+#define PUT_FUNC(name, fmt, type, expr)                                     \
+static void put_sample_ ## name(void **data, enum AVSampleFormat sample_fmt,\
+                                int channels, int sample, int ch,           \
+                                double v_dbl)                               \
+{                                                                           \
+    type v = expr;                                                          \
+    type **out = (type **)data;                                             \
+    if (av_sample_fmt_is_planar(sample_fmt))                                \
+        out[ch][sample] = v;                                                \
+    else                                                                    \
+        out[0][sample * channels + ch] = v;                                 \
+}
+
+PUT_FUNC(u8,  AV_SAMPLE_FMT_U8,  uint8_t, av_clip_uint8 ( lrint(v_dbl * (1  <<  7)) + 128))
+PUT_FUNC(s16, AV_SAMPLE_FMT_S16, int16_t, av_clip_int16 ( lrint(v_dbl * (1  << 15))))
+PUT_FUNC(s32, AV_SAMPLE_FMT_S32, int32_t, av_clipl_int32(llrint(v_dbl * (1U << 31))))
+PUT_FUNC(flt, AV_SAMPLE_FMT_FLT, float,   v_dbl)
+PUT_FUNC(dbl, AV_SAMPLE_FMT_DBL, double,  v_dbl)
+
+static void put_sample(void **data, enum AVSampleFormat sample_fmt,
+                       int channels, int sample, int ch, double v_dbl)
+{
+    switch (av_get_packed_sample_fmt(sample_fmt)) {
+    case AV_SAMPLE_FMT_U8:
+        put_sample_u8(data, sample_fmt, channels, sample, ch, v_dbl);
+        break;
+    case AV_SAMPLE_FMT_S16:
+        put_sample_s16(data, sample_fmt, channels, sample, ch, v_dbl);
+        break;
+    case AV_SAMPLE_FMT_S32:
+        put_sample_s32(data, sample_fmt, channels, sample, ch, v_dbl);
+        break;
+    case AV_SAMPLE_FMT_FLT:
+        put_sample_flt(data, sample_fmt, channels, sample, ch, v_dbl);
+        break;
+    case AV_SAMPLE_FMT_DBL:
+        put_sample_dbl(data, sample_fmt, channels, sample, ch, v_dbl);
+        break;
+    }
+}
+
+static void audiogen(AVLFG *rnd, void **data, enum AVSampleFormat sample_fmt,
+                     int channels, int sample_rate, int nb_samples)
+{
+    int i, ch, k;
+    double v, f, a, ampa;
+    double tabf1[AVRESAMPLE_MAX_CHANNELS];
+    double tabf2[AVRESAMPLE_MAX_CHANNELS];
+    double taba[AVRESAMPLE_MAX_CHANNELS];
+
+#define PUT_SAMPLE put_sample(data, sample_fmt, channels, k, ch, v);
+
+    k = 0;
+
+    /* 1 second of single freq sinus at 1000 Hz */
+    a = 0;
+    for (i = 0; i < 1 * sample_rate && k < nb_samples; i++, k++) {
+        v = sin(a) * 0.30;
+        for (ch = 0; ch < channels; ch++)
+            PUT_SAMPLE
+        a += M_PI * 1000.0 * 2.0 / sample_rate;
+    }
+
+    /* 1 second of varing frequency between 100 and 10000 Hz */
+    a = 0;
+    for (i = 0; i < 1 * sample_rate && k < nb_samples; i++, k++) {
+        v = sin(a) * 0.30;
+        for (ch = 0; ch < channels; ch++)
+            PUT_SAMPLE
+        f  = 100.0 + (((10000.0 - 100.0) * i) / sample_rate);
+        a += M_PI * f * 2.0 / sample_rate;
+    }
+
+    /* 0.5 second of low amplitude white noise */
+    for (i = 0; i < sample_rate / 2 && k < nb_samples; i++, k++) {
+        v = dbl_rand(rnd) * 0.30;
+        for (ch = 0; ch < channels; ch++)
+            PUT_SAMPLE
+    }
+
+    /* 0.5 second of high amplitude white noise */
+    for (i = 0; i < sample_rate / 2 && k < nb_samples; i++, k++) {
+        v = dbl_rand(rnd);
+        for (ch = 0; ch < channels; ch++)
+            PUT_SAMPLE
+    }
+
+    /* 1 second of unrelated ramps for each channel */
+    for (ch = 0; ch < channels; ch++) {
+        taba[ch]  = 0;
+        tabf1[ch] = 100 + av_lfg_get(rnd) % 5000;
+        tabf2[ch] = 100 + av_lfg_get(rnd) % 5000;
+    }
+    for (i = 0; i < 1 * sample_rate && k < nb_samples; i++, k++) {
+        for (ch = 0; ch < channels; ch++) {
+            v = sin(taba[ch]) * 0.30;
+            PUT_SAMPLE
+            f = tabf1[ch] + (((tabf2[ch] - tabf1[ch]) * i) / sample_rate);
+            taba[ch] += M_PI * f * 2.0 / sample_rate;
+        }
+    }
+
+    /* 2 seconds of 500 Hz with varying volume */
+    a    = 0;
+    ampa = 0;
+    for (i = 0; i < 2 * sample_rate && k < nb_samples; i++, k++) {
+        for (ch = 0; ch < channels; ch++) {
+            double amp = (1.0 + sin(ampa)) * 0.15;
+            if (ch & 1)
+                amp = 0.30 - amp;
+            v = sin(a) * amp;
+            PUT_SAMPLE
+            a    += M_PI * 500.0 * 2.0 / sample_rate;
+            ampa += M_PI *  2.0 / sample_rate;
+        }
+    }
+}
+
+/* formats, rates, and layouts are ordered for priority in testing.
+   e.g. 'avresample-test 4 2 2' will test all input/output combinations of
+   S16/FLTP/S16P/FLT, 48000/44100, and stereo/mono */
+
+static const enum AVSampleFormat formats[] = {
+    AV_SAMPLE_FMT_S16,
+    AV_SAMPLE_FMT_FLTP,
+    AV_SAMPLE_FMT_S16P,
+    AV_SAMPLE_FMT_FLT,
+    AV_SAMPLE_FMT_S32P,
+    AV_SAMPLE_FMT_S32,
+    AV_SAMPLE_FMT_U8P,
+    AV_SAMPLE_FMT_U8,
+    AV_SAMPLE_FMT_DBLP,
+    AV_SAMPLE_FMT_DBL,
+};
+
+static const int rates[] = {
+    48000,
+    44100,
+    16000
+};
+
+static const uint64_t layouts[] = {
+    AV_CH_LAYOUT_STEREO,
+    AV_CH_LAYOUT_MONO,
+    AV_CH_LAYOUT_5POINT1,
+    AV_CH_LAYOUT_7POINT1,
+};
+
+int main(int argc, char **argv)
+{
+    AVAudioResampleContext *s;
+    AVLFG rnd;
+    int ret = 0;
+    uint8_t *in_buf = NULL;
+    uint8_t *out_buf = NULL;
+    unsigned int in_buf_size;
+    unsigned int out_buf_size;
+    uint8_t  *in_data[AVRESAMPLE_MAX_CHANNELS] = { 0 };
+    uint8_t *out_data[AVRESAMPLE_MAX_CHANNELS] = { 0 };
+    int in_linesize;
+    int out_linesize;
+    uint64_t in_ch_layout;
+    int in_channels;
+    enum AVSampleFormat in_fmt;
+    int in_rate;
+    uint64_t out_ch_layout;
+    int out_channels;
+    enum AVSampleFormat out_fmt;
+    int out_rate;
+    int num_formats, num_rates, num_layouts;
+    int i, j, k, l, m, n;
+
+    num_formats = 2;
+    num_rates   = 2;
+    num_layouts = 2;
+    if (argc > 1) {
+        if (!av_strncasecmp(argv[1], "-h", 3)) {
+            av_log(NULL, AV_LOG_INFO, "Usage: avresample-test [<num formats> "
+                   "[<num sample rates> [<num channel layouts>]]]\n"
+                   "Default is 2 2 2\n");
+            return 0;
+        }
+        num_formats = strtol(argv[1], NULL, 0);
+        num_formats = av_clip(num_formats, 1, FF_ARRAY_ELEMS(formats));
+    }
+    if (argc > 2) {
+        num_rates = strtol(argv[2], NULL, 0);
+        num_rates = av_clip(num_rates, 1, FF_ARRAY_ELEMS(rates));
+    }
+    if (argc > 3) {
+        num_layouts = strtol(argv[3], NULL, 0);
+        num_layouts = av_clip(num_layouts, 1, FF_ARRAY_ELEMS(layouts));
+    }
+
+    av_log_set_level(AV_LOG_DEBUG);
+
+    av_lfg_init(&rnd, 0xC0FFEE);
+
+    in_buf_size = av_samples_get_buffer_size(&in_linesize, 8, 48000 * 6,
+                                             AV_SAMPLE_FMT_DBLP, 0);
+    out_buf_size = in_buf_size;
+
+    in_buf = av_malloc(in_buf_size);
+    if (!in_buf)
+        goto end;
+    out_buf = av_malloc(out_buf_size);
+    if (!out_buf)
+        goto end;
+
+    s = avresample_alloc_context();
+    if (!s) {
+        av_log(NULL, AV_LOG_ERROR, "Error allocating AVAudioResampleContext\n");
+        ret = 1;
+        goto end;
+    }
+
+    for (i = 0; i < num_formats; i++) {
+        in_fmt = formats[i];
+        for (k = 0; k < num_layouts; k++) {
+            in_ch_layout = layouts[k];
+            in_channels  = av_get_channel_layout_nb_channels(in_ch_layout);
+            for (m = 0; m < num_rates; m++) {
+                in_rate = rates[m];
+
+                ret = av_samples_fill_arrays(in_data, &in_linesize, in_buf,
+                                             in_channels, in_rate * 6,
+                                             in_fmt, 0);
+                if (ret < 0) {
+                    av_log(s, AV_LOG_ERROR, "failed in_data fill arrays\n");
+                    goto end;
+                }
+                audiogen(&rnd, (void **)in_data, in_fmt, in_channels, in_rate, in_rate * 6);
+
+                for (j = 0; j < num_formats; j++) {
+                    out_fmt = formats[j];
+                    for (l = 0; l < num_layouts; l++) {
+                        out_ch_layout = layouts[l];
+                        out_channels  = av_get_channel_layout_nb_channels(out_ch_layout);
+                        for (n = 0; n < num_rates; n++) {
+                            out_rate = rates[n];
+
+                            av_log(NULL, AV_LOG_INFO, "%s to %s, %d to %d channels, %d Hz to %d Hz\n",
+                                   av_get_sample_fmt_name(in_fmt), av_get_sample_fmt_name(out_fmt),
+                                   in_channels, out_channels, in_rate, out_rate);
+
+                            ret = av_samples_fill_arrays(out_data, &out_linesize,
+                                                         out_buf, out_channels,
+                                                         out_rate * 6, out_fmt, 0);
+                            if (ret < 0) {
+                                av_log(s, AV_LOG_ERROR, "failed out_data fill arrays\n");
+                                goto end;
+                            }
+
+                            av_opt_set_int(s, "in_channel_layout",  in_ch_layout,  0);
+                            av_opt_set_int(s, "in_sample_fmt",      in_fmt,        0);
+                            av_opt_set_int(s, "in_sample_rate",     in_rate,       0);
+                            av_opt_set_int(s, "out_channel_layout", out_ch_layout, 0);
+                            av_opt_set_int(s, "out_sample_fmt",     out_fmt,       0);
+                            av_opt_set_int(s, "out_sample_rate",    out_rate,      0);
+
+                            av_opt_set_int(s, "internal_sample_fmt", AV_SAMPLE_FMT_FLTP, 0);
+
+                            ret = avresample_open(s);
+                            if (ret < 0) {
+                                av_log(s, AV_LOG_ERROR, "Error opening context\n");
+                                goto end;
+                            }
+
+                            ret = avresample_convert(s, (void **)out_data, out_linesize, out_rate * 6,
+                                                        (void **) in_data,  in_linesize,  in_rate * 6);
+                            if (ret < 0) {
+                                char errbuf[256];
+                                av_strerror(ret, errbuf, sizeof(errbuf));
+                                av_log(NULL, AV_LOG_ERROR, "%s\n", errbuf);
+                                goto end;
+                            }
+                            av_log(NULL, AV_LOG_INFO, "Converted %d samples to %d samples\n",
+                                   in_rate * 6, ret);
+                            if (avresample_get_delay(s) > 0)
+                                av_log(NULL, AV_LOG_INFO, "%d delay samples not converted\n",
+                                       avresample_get_delay(s));
+                            if (avresample_available(s) > 0)
+                                av_log(NULL, AV_LOG_INFO, "%d samples available for output\n",
+                                       avresample_available(s));
+                            av_log(NULL, AV_LOG_INFO, "\n");
+
+                            avresample_close(s);
+                        }
+                    }
+                }
+            }
+        }
+    }
+
+    ret = 0;
+
+end:
+    av_freep(&in_buf);
+    av_freep(&out_buf);
+    avresample_free(&s);
+    return ret;
+}