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https://github.com/HappyTanuki/BumbleCee.git
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일단 ffmpeg파일 인코딩/디코딩 예제 코드 완성
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155
tests/ffmpeg_decode_audio.cc
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155
tests/ffmpeg_decode_audio.cc
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#include <fstream>
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#include "ffmpeg/libavcodec.h"
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#include "precomp.h"
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int main() {
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const char* input_filename = "golden.webm";
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const char* output_filename = "output.pcm";
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AVFormatContext* fmt_ctx = NULL;
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AVCodecContext* codec_ctx = NULL;
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const AVCodec* codec = NULL;
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AVPacket* packet = NULL;
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AVFrame* frame = NULL;
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SwrContext* swr_ctx = NULL;
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FILE* outfile = NULL;
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if (avformat_open_input(&fmt_ctx, input_filename, NULL, NULL) < 0) {
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fprintf(stderr, "Could not open input file\n");
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return -1;
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}
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if (avformat_find_stream_info(fmt_ctx, NULL) < 0) {
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fprintf(stderr, "Could not find stream info\n");
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return -1;
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}
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int stream_index = -1;
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for (unsigned i = 0; i < fmt_ctx->nb_streams; i++) {
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if (fmt_ctx->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
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stream_index = i;
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break;
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}
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}
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if (stream_index == -1) {
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fprintf(stderr, "Could not find audio stream\n");
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return -1;
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}
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codec =
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avcodec_find_decoder(fmt_ctx->streams[stream_index]->codecpar->codec_id);
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if (!codec) {
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fprintf(stderr, "Could not find decoder\n");
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return -1;
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}
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codec_ctx = avcodec_alloc_context3(codec);
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if (!codec_ctx) {
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fprintf(stderr, "Could not allocate codec context\n");
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return -1;
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}
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if (avcodec_parameters_to_context(
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codec_ctx, fmt_ctx->streams[stream_index]->codecpar) < 0) {
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fprintf(stderr, "Failed to copy codec parameters\n");
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return -1;
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}
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if (avcodec_open2(codec_ctx, codec, NULL) < 0) {
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fprintf(stderr, "Could not open codec\n");
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return -1;
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}
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packet = av_packet_alloc();
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frame = av_frame_alloc();
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if (!packet || !frame) {
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fprintf(stderr, "Could not allocate packet or frame\n");
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return -1;
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}
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outfile = fopen(output_filename, "wb");
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if (!outfile) {
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fprintf(stderr, "Could not open output file\n");
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return -1;
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}
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// AVChannelLayout 초기화
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AVChannelLayout in_layout, out_layout;
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if (av_channel_layout_copy(&in_layout, &codec_ctx->ch_layout) < 0) {
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fprintf(stderr, "Failed to copy channel layout\n");
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return -1;
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}
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av_channel_layout_default(&out_layout, 2); // 스테레오
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swr_ctx = NULL; // 먼저 NULL로 선언
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if (swr_alloc_set_opts2(&swr_ctx, &out_layout, AV_SAMPLE_FMT_S16, 48000,
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&in_layout, codec_ctx->sample_fmt,
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codec_ctx->sample_rate, 0, NULL) < 0) {
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fprintf(stderr, "Failed to allocate and set SwrContext\n");
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return -1;
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}
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if (swr_init(swr_ctx) < 0) {
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fprintf(stderr, "Failed to initialize SwrContext\n");
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return -1;
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}
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while (av_read_frame(fmt_ctx, packet) >= 0) {
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if (packet->stream_index == stream_index) {
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if (avcodec_send_packet(codec_ctx, packet) == 0) {
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while (avcodec_receive_frame(codec_ctx, frame) == 0) {
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int out_samples =
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av_rescale_rnd(swr_get_delay(swr_ctx, codec_ctx->sample_rate) +
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frame->nb_samples,
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48000, codec_ctx->sample_rate, AV_ROUND_UP);
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uint8_t** out_buf = NULL;
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int out_linesize = 0;
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av_samples_alloc_array_and_samples(&out_buf, &out_linesize, 2,
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out_samples, AV_SAMPLE_FMT_S16, 0);
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int converted_samples =
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swr_convert(swr_ctx, out_buf, out_samples,
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(const uint8_t**)frame->data, frame->nb_samples);
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fwrite(out_buf[0], 1, converted_samples * 2 * 2, outfile);
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av_freep(&out_buf[0]);
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free(out_buf);
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}
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}
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}
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av_packet_unref(packet);
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}
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// 디코더 플러시
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avcodec_send_packet(codec_ctx, NULL);
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while (avcodec_receive_frame(codec_ctx, frame) == 0) {
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int out_samples = av_rescale_rnd(
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swr_get_delay(swr_ctx, codec_ctx->sample_rate) + frame->nb_samples,
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48000, codec_ctx->sample_rate, AV_ROUND_UP);
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uint8_t** out_buf = NULL;
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int out_linesize = 0;
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av_samples_alloc_array_and_samples(&out_buf, &out_linesize, 2, out_samples,
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AV_SAMPLE_FMT_S16, 0);
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int converted_samples =
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swr_convert(swr_ctx, out_buf, out_samples, (const uint8_t**)frame->data,
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frame->nb_samples);
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fwrite(out_buf[0], 1, converted_samples * 2 * 2, outfile);
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av_freep(&out_buf[0]);
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free(out_buf);
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}
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fclose(outfile);
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swr_free(&swr_ctx);
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av_frame_free(&frame);
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av_packet_free(&packet);
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avcodec_free_context(&codec_ctx);
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avformat_close_input(&fmt_ctx);
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printf("Decoding finished, output saved to %s\n", output_filename);
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return 0;
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}
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