From 4a6d9312cc1c9d62d66c4def71246d9faae29edb Mon Sep 17 00:00:00 2001
From: zhangmeng <775834166@qq.com>
Date: 星期三, 03 三月 2021 14:26:16 +0800
Subject: [PATCH] environment settings
---
module/VideoPlayer/lib/linux/ffmpeg/share/ffmpeg/examples/resampling_audio.c | 214 +++++++++++++++++++++++++++++++++++++++++++++++++++++
1 files changed, 214 insertions(+), 0 deletions(-)
diff --git a/module/VideoPlayer/lib/linux/ffmpeg/share/ffmpeg/examples/resampling_audio.c b/module/VideoPlayer/lib/linux/ffmpeg/share/ffmpeg/examples/resampling_audio.c
new file mode 100644
index 0000000..f35e7e1
--- /dev/null
+++ b/module/VideoPlayer/lib/linux/ffmpeg/share/ffmpeg/examples/resampling_audio.c
@@ -0,0 +1,214 @@
+/*
+ * 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 resampling_audio.c
+ * 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.
+ */
+static 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 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 = av_samples_alloc_array_and_samples(&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 = av_samples_alloc_array_and_samples(&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_freep(&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);
+ if (dst_bufsize < 0) {
+ fprintf(stderr, "Could not get sample buffer size\n");
+ goto end;
+ }
+ 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:
+ 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;
+}
--
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