ffmpeg예제 코드 작성 중..

.vscode/settings.json

@@ -4,6 +4,8 @@
             "column": 80
         }
     ],
+    "editor.renderWhitespace": "boundary",
+    "editor.formatOnSave": true,
     "cmake.generator": "Ninja",
     "files.associations": {
         "cctype": "cpp",

include/core/bumblebee.h

@@ -0,0 +1,19 @@
+#ifndef BUMBLEBEE_INCLUDE_CORE_BUMBLEBEE_H_
+#define BUMBLEBEE_INCLUDE_CORE_BUMBLEBEE_H_
+
+#include "precomp.h"
+
+namespace bumblebee {
+
+class BumbleBee {
+ public:
+  BumbleBee();
+  ~BumbleBee();
+
+ private:
+  dpp::cluster cluster;
+};
+
+}  // namespace bumblebee
+
+#endif

include/precomp.h

@@ -4,6 +4,8 @@
 #include <winsock2.h>
 #endif
 
+#include <dpp/dpp.h>
+
 #include <iostream>
 #include <queue>
 

tests/ffmpeg_any_to_opus.cc

@@ -0,0 +1,193 @@
+#include "precomp.h"
+/* check that a given sample format is supported by the encoder */
+static int check_sample_fmt(const AVCodec *codec,
+                            enum AVSampleFormat sample_fmt) {
+  const enum AVSampleFormat *p = codec->sample_fmts;
+
+  while (*p != AV_SAMPLE_FMT_NONE) {
+    if (*p == sample_fmt) return 1;
+    p++;
+  }
+  return 0;
+}
+
+/* just pick the highest supported samplerate */
+static int select_sample_rate(const AVCodec *codec) {
+  const int *p;
+  int best_samplerate = 0;
+
+  if (!codec->supported_samplerates) return 44100;
+
+  p = codec->supported_samplerates;
+  while (*p) {
+    if (!best_samplerate || abs(44100 - *p) < abs(44100 - best_samplerate))
+      best_samplerate = *p;
+    p++;
+  }
+  return best_samplerate;
+}
+
+/* select layout with the highest channel count */
+static int select_channel_layout(const AVCodec *codec, AVChannelLayout *dst) {
+  const AVChannelLayout *p, *best_ch_layout;
+  int best_nb_channels = 0;
+
+  if (!codec->ch_layouts)
+    return av_channel_layout_copy(dst,
+                                  &(AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO);
+
+  p = codec->ch_layouts;
+  while (p->nb_channels) {
+    int nb_channels = p->nb_channels;
+
+    if (nb_channels > best_nb_channels) {
+      best_ch_layout = p;
+      best_nb_channels = nb_channels;
+    }
+    p++;
+  }
+  return av_channel_layout_copy(dst, best_ch_layout);
+}
+
+static void encode(AVCodecContext *ctx, AVFrame *frame, AVPacket *pkt,
+                   FILE *output) {
+  int ret;
+
+  /* send the frame for encoding */
+  ret = avcodec_send_frame(ctx, frame);
+  if (ret < 0) {
+    fprintf(stderr, "Error sending the frame to the encoder\n");
+    exit(1);
+  }
+
+  /* read all the available output packets (in general there may be any
+   * number of them */
+  while (ret >= 0) {
+    ret = avcodec_receive_packet(ctx, pkt);
+    if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
+      return;
+    else if (ret < 0) {
+      fprintf(stderr, "Error encoding audio frame\n");
+      exit(1);
+    }
+
+    fwrite(pkt->data, 1, pkt->size, output);
+    av_packet_unref(pkt);
+  }
+}
+
+int main(int argc, char **argv) {
+  const char *filename;
+  const AVCodec *codec;
+  AVCodecContext *c = NULL;
+  AVFrame *frame;
+  AVPacket *pkt;
+  int i, j, k, ret;
+  FILE *f;
+  uint16_t *samples;
+  float t, tincr;
+
+  if (argc <= 1) {
+    fprintf(stderr, "Usage: %s <output file>\n", argv[0]);
+    return 0;
+  }
+  filename = argv[1];
+
+  /* find the MP2 encoder */
+  codec = avcodec_find_encoder(AV_CODEC_ID_MP2);
+  if (!codec) {
+    fprintf(stderr, "Codec not found\n");
+    exit(1);
+  }
+
+  c = avcodec_alloc_context3(codec);
+  if (!c) {
+    fprintf(stderr, "Could not allocate audio codec context\n");
+    exit(1);
+  }
+
+  /* put sample parameters */
+  c->bit_rate = 64000;
+
+  /* check that the encoder supports s16 pcm input */
+  c->sample_fmt = AV_SAMPLE_FMT_S16;
+  if (!check_sample_fmt(codec, c->sample_fmt)) {
+    fprintf(stderr, "Encoder does not support sample format %s",
+            av_get_sample_fmt_name(c->sample_fmt));
+    exit(1);
+  }
+
+  /* select other audio parameters supported by the encoder */
+  c->sample_rate = select_sample_rate(codec);
+  ret = select_channel_layout(codec, &c->ch_layout);
+  if (ret < 0) exit(1);
+
+  /* open it */
+  if (avcodec_open2(c, codec, NULL) < 0) {
+    fprintf(stderr, "Could not open codec\n");
+    exit(1);
+  }
+
+  f = fopen(filename, "wb");
+  if (!f) {
+    fprintf(stderr, "Could not open %s\n", filename);
+    exit(1);
+  }
+
+  /* packet for holding encoded output */
+  pkt = av_packet_alloc();
+  if (!pkt) {
+    fprintf(stderr, "could not allocate the packet\n");
+    exit(1);
+  }
+
+  /* frame containing input raw audio */
+  frame = av_frame_alloc();
+  if (!frame) {
+    fprintf(stderr, "Could not allocate audio frame\n");
+    exit(1);
+  }
+
+  frame->nb_samples = c->frame_size;
+  frame->format = c->sample_fmt;
+  ret = av_channel_layout_copy(&frame->ch_layout, &c->ch_layout);
+  if (ret < 0) exit(1);
+
+  /* allocate the data buffers */
+  ret = av_frame_get_buffer(frame, 0);
+  if (ret < 0) {
+    fprintf(stderr, "Could not allocate audio data buffers\n");
+    exit(1);
+  }
+
+  /* encode a single tone sound */
+  t = 0;
+  tincr = 2 * M_PI * 440.0 / c->sample_rate;
+  for (i = 0; i < 200; i++) {
+    /* make sure the frame is writable -- makes a copy if the encoder
+     * kept a reference internally */
+    ret = av_frame_make_writable(frame);
+    if (ret < 0) exit(1);
+    samples = (uint16_t *)frame->data[0];
+
+    for (j = 0; j < c->frame_size; j++) {
+      samples[2 * j] = (int)(sin(t) * 10000);
+
+      for (k = 1; k < c->ch_layout.nb_channels; k++)
+        samples[2 * j + k] = samples[2 * j];
+      t += tincr;
+    }
+    encode(c, frame, pkt, f);
+  }
+
+  /* flush the encoder */
+  encode(c, NULL, pkt, f);
+
+  fclose(f);
+
+  av_frame_free(&frame);
+  av_packet_free(&pkt);
+  avcodec_free_context(&c);
+
+  return 0;
+}