examples: add resampling_audio.c file

doc/examples/Makefile

@@ -17,6 +17,7 @@ EXAMPLES=       decoding_encoding                  \
                 filtering_audio                    \
                 metadata                           \
                 muxing                             \
+                resampling_audio                   \
                 scaling_video                      \
 
 OBJS=$(addsuffix .o,$(EXAMPLES))

doc/examples/resampling_audio.c

+/*
+ * Copyright (c) 2012 Stefano Sabatini
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+/**
+ * @example
+ * libswresample API use example.
+ */
+
+#include <libavutil/opt.h>
+#include <libavutil/channel_layout.h>
+#include <libavutil/samplefmt.h>
+#include <libswresample/swresample.h>
+
+static int get_format_from_sample_fmt(const char **fmt,
+                                      enum AVSampleFormat sample_fmt)
+{
+    int i;
+    struct sample_fmt_entry {
+        enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
+    } sample_fmt_entries[] = {
+        { AV_SAMPLE_FMT_U8,  "u8",    "u8"    },
+        { AV_SAMPLE_FMT_S16, "s16be", "s16le" },
+        { AV_SAMPLE_FMT_S32, "s32be", "s32le" },
+        { AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
+        { AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
+    };
+    *fmt = NULL;
+
+    for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
+        struct sample_fmt_entry *entry = &sample_fmt_entries[i];
+        if (sample_fmt == entry->sample_fmt) {
+            *fmt = AV_NE(entry->fmt_be, entry->fmt_le);
+            return 0;
+        }
+    }
+
+    fprintf(stderr,
+            "Sample format %s not supported as output format\n",
+            av_get_sample_fmt_name(sample_fmt));
+    return AVERROR(EINVAL);
+}
+
+/**
+ * Fill dst buffer with nb_samples, generated starting from t.
+ */
+void fill_samples(double *dst, int nb_samples, int nb_channels, int sample_rate, double *t)
+{
+    int i, j;
+    double tincr = 1.0 / sample_rate, *dstp = dst;
+    const double c = 2 * M_PI * 440.0;
+
+    /* generate sin tone with 440Hz frequency and duplicated channels */
+    for (i = 0; i < nb_samples; i++) {
+        *dstp = sin(c * *t);
+        for (j = 1; j < nb_channels; j++)
+            dstp[j] = dstp[0];
+        dstp += nb_channels;
+        *t += tincr;
+    }
+}
+
+int alloc_samples_array_and_data(uint8_t ***data, int *linesize, int nb_channels,
+                                    int nb_samples, enum AVSampleFormat sample_fmt, int align)
+{
+    int nb_planes = av_sample_fmt_is_planar(sample_fmt) ? nb_channels : 1;
+
+    *data = av_malloc(sizeof(*data) * nb_planes);
+    if (!*data)
+        return AVERROR(ENOMEM);
+    return av_samples_alloc(*data, linesize, nb_channels,
+                            nb_samples, sample_fmt, align);
+}
+
+int main(int argc, char **argv)
+{
+    int64_t src_ch_layout = AV_CH_LAYOUT_STEREO, dst_ch_layout = AV_CH_LAYOUT_SURROUND;
+    int src_rate = 48000, dst_rate = 44100;
+    uint8_t **src_data = NULL, **dst_data = NULL;
+    int src_nb_channels = 0, dst_nb_channels = 0;
+    int src_linesize, dst_linesize;
+    int src_nb_samples = 1024, dst_nb_samples, max_dst_nb_samples;
+    enum AVSampleFormat src_sample_fmt = AV_SAMPLE_FMT_DBL, dst_sample_fmt = AV_SAMPLE_FMT_S16;
+    const char *dst_filename = NULL;
+    FILE *dst_file;
+    int dst_bufsize;
+    const char *fmt;
+    struct SwrContext *swr_ctx;
+    double t;
+    int ret;
+
+    if (argc != 2) {
+        fprintf(stderr, "Usage: %s output_file\n"
+                "API example program to show how to resample an audio stream with libswresample.\n"
+                "This program generates a series of audio frames, resamples them to a specified "
+                "output format and rate and saves them to an output file named output_file.\n",
+            argv[0]);
+        exit(1);
+    }
+    dst_filename = argv[1];
+
+    dst_file = fopen(dst_filename, "wb");
+    if (!dst_file) {
+        fprintf(stderr, "Could not open destination file %s\n", dst_filename);
+        exit(1);
+    }
+
+    /* create resampler context */
+    swr_ctx = swr_alloc();
+    if (!swr_ctx) {
+        fprintf(stderr, "Could not allocate resampler context\n");
+        ret = AVERROR(ENOMEM);
+        goto end;
+    }
+
+    /* set options */
+    av_opt_set_int(swr_ctx, "in_channel_layout",    src_ch_layout, 0);
+    av_opt_set_int(swr_ctx, "in_sample_rate",       src_rate, 0);
+    av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", src_sample_fmt, 0);
+
+    av_opt_set_int(swr_ctx, "out_channel_layout",    dst_ch_layout, 0);
+    av_opt_set_int(swr_ctx, "out_sample_rate",       dst_rate, 0);
+    av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", dst_sample_fmt, 0);
+
+    /* initialize the resampling context */
+    if ((ret = swr_init(swr_ctx)) < 0) {
+        fprintf(stderr, "Failed to initialize the resampling context\n");
+        goto end;
+    }
+
+    /* allocate source and destination samples buffers */
+
+    src_nb_channels = av_get_channel_layout_nb_channels(src_ch_layout);
+    ret = alloc_samples_array_and_data(&src_data, &src_linesize, src_nb_channels,
+                                       src_nb_samples, src_sample_fmt, 0);
+    if (ret < 0) {
+        fprintf(stderr, "Could not allocate source samples\n");
+        goto end;
+    }
+
+    /* compute the number of converted samples: buffering is avoided
+     * ensuring that the output buffer will contain at least all the
+     * converted input samples */
+    max_dst_nb_samples = dst_nb_samples =
+        av_rescale_rnd(src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
+
+    /* buffer is going to be directly written to a rawaudio file, no alignment */
+    dst_nb_channels = av_get_channel_layout_nb_channels(dst_ch_layout);
+    ret = alloc_samples_array_and_data(&dst_data, &dst_linesize, dst_nb_channels,
+                                       dst_nb_samples, dst_sample_fmt, 0);
+    if (ret < 0) {
+        fprintf(stderr, "Could not allocate destination samples\n");
+        goto end;
+    }
+
+    t = 0;
+    do {
+        /* generate synthetic audio */
+        fill_samples((double *)src_data[0], src_nb_samples, src_nb_channels, src_rate, &t);
+
+        /* compute destination number of samples */
+        dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, src_rate) +
+                                        src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
+        if (dst_nb_samples > max_dst_nb_samples) {
+            av_free(dst_data[0]);
+            ret = av_samples_alloc(dst_data, &dst_linesize, dst_nb_channels,
+                                   dst_nb_samples, dst_sample_fmt, 1);
+            if (ret < 0)
+                break;
+            max_dst_nb_samples = dst_nb_samples;
+        }
+
+        /* convert to destination format */
+        ret = swr_convert(swr_ctx, dst_data, dst_nb_samples, (const uint8_t **)src_data, src_nb_samples);
+        if (ret < 0) {
+            fprintf(stderr, "Error while converting\n");
+            goto end;
+        }
+        dst_bufsize = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels,
+                                                 ret, dst_sample_fmt, 1);
+        printf("t:%f in:%d out:%d\n", t, src_nb_samples, ret);
+        fwrite(dst_data[0], 1, dst_bufsize, dst_file);
+    } while (t < 10);
+
+    if ((ret = get_format_from_sample_fmt(&fmt, dst_sample_fmt) < 0))
+        goto end;
+    fprintf(stderr, "Resampling succeeded. Play the output file with the command:\n"
+            "ffplay -f %s -channel_layout %"PRId64" -channels %d -ar %d %s\n",
+            fmt, dst_ch_layout, dst_nb_channels, dst_rate, dst_filename);
+
+end:
+    if (dst_file)
+        fclose(dst_file);
+
+    if (src_data)
+        av_freep(&src_data[0]);
+    av_freep(&src_data);
+
+    if (dst_data)
+        av_freep(&dst_data[0]);
+    av_freep(&dst_data);
+
+    swr_free(&swr_ctx);
+    return ret < 0;
+}