mirror of
https://github.com/HappyTanuki/BumbleCee.git
synced 2025-10-25 17:35:58 +00:00
일단 ffmpeg파일 인코딩/디코딩 예제 코드 완성
This commit is contained in:
9
.vscode/settings.json
vendored
9
.vscode/settings.json
vendored
@@ -103,7 +103,14 @@
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"__locale": "cpp",
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"ios": "cpp",
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"locale": "cpp",
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"print": "cpp"
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"print": "cpp",
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"__bit_reference": "cpp",
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"__hash_table": "cpp",
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"__node_handle": "cpp",
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"__split_buffer": "cpp",
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"__threading_support": "cpp",
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"__verbose_abort": "cpp",
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"queue": "cpp"
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},
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"files.exclude": {
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"**/*.rpyc": true,
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@@ -16,6 +16,7 @@
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#include "boost/process.hpp"
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extern "C" {
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#include "libavcodec/avcodec.h"
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#include "libavformat/avformat.h"
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#include "libavutil/avutil.h"
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#include "libswresample/swresample.h"
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@@ -6,7 +6,7 @@ int main() {
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boost::asio::io_context ctx;
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boost::system::error_code ec;
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utils::CheckUpdate(ctx);
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// utils::CheckUpdate(ctx);
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char buf[8192];
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#ifdef WIN32
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@@ -1,193 +1,232 @@
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#include "precomp.h"
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/* check that a given sample format is supported by the encoder */
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static int check_sample_fmt(const AVCodec *codec,
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enum AVSampleFormat sample_fmt) {
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const enum AVSampleFormat *p = codec->sample_fmts;
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while (*p != AV_SAMPLE_FMT_NONE) {
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if (*p == sample_fmt) return 1;
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p++;
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#define OPUS_FRAME_SIZE 960 // 20ms @ 48kHz
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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.opus";
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AVFormatContext* fmt_ctx = NULL;
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AVCodecContext* dec_ctx = NULL;
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AVCodecContext* enc_ctx = NULL;
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const AVCodec* decoder = NULL;
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const AVCodec* encoder = NULL;
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AVPacket* packet = NULL;
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AVFrame* frame = NULL;
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AVFrame* enc_frame = NULL;
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SwrContext* swr_ctx = NULL;
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FILE* outfile = NULL;
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av_log_set_level(AV_LOG_ERROR);
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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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return 0;
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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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/* just pick the highest supported samplerate */
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static int select_sample_rate(const AVCodec *codec) {
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const int *p;
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int best_samplerate = 0;
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if (!codec->supported_samplerates) return 44100;
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p = codec->supported_samplerates;
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while (*p) {
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if (!best_samplerate || abs(44100 - *p) < abs(44100 - best_samplerate))
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best_samplerate = *p;
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p++;
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}
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return best_samplerate;
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}
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/* select layout with the highest channel count */
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static int select_channel_layout(const AVCodec *codec, AVChannelLayout *dst) {
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const AVChannelLayout *p, *best_ch_layout;
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int best_nb_channels = 0;
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if (!codec->ch_layouts)
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return av_channel_layout_copy(dst,
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&(AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO);
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p = codec->ch_layouts;
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while (p->nb_channels) {
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int nb_channels = p->nb_channels;
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if (nb_channels > best_nb_channels) {
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best_ch_layout = p;
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best_nb_channels = nb_channels;
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}
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p++;
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}
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return av_channel_layout_copy(dst, best_ch_layout);
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}
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static void encode(AVCodecContext *ctx, AVFrame *frame, AVPacket *pkt,
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FILE *output) {
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int ret;
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/* send the frame for encoding */
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ret = avcodec_send_frame(ctx, frame);
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if (ret < 0) {
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fprintf(stderr, "Error sending the frame to the encoder\n");
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exit(1);
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}
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/* read all the available output packets (in general there may be any
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* number of them */
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while (ret >= 0) {
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ret = avcodec_receive_packet(ctx, pkt);
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if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
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return;
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else if (ret < 0) {
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fprintf(stderr, "Error encoding audio frame\n");
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exit(1);
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}
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fwrite(pkt->data, 1, pkt->size, output);
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av_packet_unref(pkt);
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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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int main(int argc, char **argv) {
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const char *filename;
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const AVCodec *codec;
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AVCodecContext *c = NULL;
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AVFrame *frame;
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AVPacket *pkt;
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int i, j, k, ret;
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FILE *f;
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uint16_t *samples;
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float t, tincr;
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if (argc <= 1) {
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fprintf(stderr, "Usage: %s <output file>\n", argv[0]);
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return 0;
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}
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filename = argv[1];
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/* find the MP2 encoder */
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codec = avcodec_find_encoder(AV_CODEC_ID_MP2);
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if (!codec) {
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fprintf(stderr, "Codec not found\n");
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exit(1);
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if (stream_index == -1) {
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fprintf(stderr, "No audio stream found\n");
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return -1;
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}
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c = avcodec_alloc_context3(codec);
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if (!c) {
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fprintf(stderr, "Could not allocate audio codec context\n");
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exit(1);
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decoder =
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avcodec_find_decoder(fmt_ctx->streams[stream_index]->codecpar->codec_id);
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if (!decoder) {
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fprintf(stderr, "Decoder not found\n");
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return -1;
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}
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dec_ctx = avcodec_alloc_context3(decoder);
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avcodec_parameters_to_context(dec_ctx,
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fmt_ctx->streams[stream_index]->codecpar);
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avcodec_open2(dec_ctx, decoder, NULL);
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/* put sample parameters */
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c->bit_rate = 64000;
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/* check that the encoder supports s16 pcm input */
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c->sample_fmt = AV_SAMPLE_FMT_S16;
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if (!check_sample_fmt(codec, c->sample_fmt)) {
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fprintf(stderr, "Encoder does not support sample format %s",
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av_get_sample_fmt_name(c->sample_fmt));
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exit(1);
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encoder = avcodec_find_encoder(AV_CODEC_ID_OPUS);
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if (!encoder) {
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fprintf(stderr, "Opus encoder not found\n");
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return -1;
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}
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enc_ctx = avcodec_alloc_context3(encoder);
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/* select other audio parameters supported by the encoder */
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c->sample_rate = select_sample_rate(codec);
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ret = select_channel_layout(codec, &c->ch_layout);
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if (ret < 0) exit(1);
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AVChannelLayout enc_layout;
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av_channel_layout_default(&enc_layout, 2); // 스테레오
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av_channel_layout_copy(&enc_ctx->ch_layout, &enc_layout);
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/* open it */
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if (avcodec_open2(c, codec, NULL) < 0) {
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fprintf(stderr, "Could not open codec\n");
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exit(1);
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enc_ctx->sample_rate = 48000;
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enc_ctx->sample_fmt = AV_SAMPLE_FMT_FLT;
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enc_ctx->bit_rate = 128000;
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avcodec_open2(enc_ctx, encoder, NULL);
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swr_ctx = NULL;
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if (swr_alloc_set_opts2(&swr_ctx, &enc_ctx->ch_layout, enc_ctx->sample_fmt,
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enc_ctx->sample_rate, &dec_ctx->ch_layout,
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dec_ctx->sample_fmt, dec_ctx->sample_rate, 0,
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NULL) < 0) {
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fprintf(stderr, "Failed to allocate SwrContext\n");
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return -1;
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}
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swr_init(swr_ctx);
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f = fopen(filename, "wb");
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if (!f) {
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fprintf(stderr, "Could not open %s\n", filename);
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exit(1);
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}
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/* packet for holding encoded output */
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pkt = av_packet_alloc();
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if (!pkt) {
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fprintf(stderr, "could not allocate the packet\n");
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exit(1);
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}
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/* frame containing input raw audio */
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packet = av_packet_alloc();
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frame = av_frame_alloc();
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if (!frame) {
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fprintf(stderr, "Could not allocate audio frame\n");
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exit(1);
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enc_frame = av_frame_alloc();
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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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frame->nb_samples = c->frame_size;
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frame->format = c->sample_fmt;
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ret = av_channel_layout_copy(&frame->ch_layout, &c->ch_layout);
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if (ret < 0) exit(1);
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// 임시 PCM 버퍼 (float, 스테레오)
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float* pcm_buffer = (float*)malloc(sizeof(float) * 2 * OPUS_FRAME_SIZE *
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4); // 충분히 큰 버퍼
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int buffered_samples = 0;
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/* allocate the data buffers */
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ret = av_frame_get_buffer(frame, 0);
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if (ret < 0) {
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fprintf(stderr, "Could not allocate audio data buffers\n");
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exit(1);
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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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av_packet_unref(packet);
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continue;
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}
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/* encode a single tone sound */
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t = 0;
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tincr = 2 * M_PI * 440.0 / c->sample_rate;
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for (i = 0; i < 200; i++) {
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/* make sure the frame is writable -- makes a copy if the encoder
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* kept a reference internally */
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ret = av_frame_make_writable(frame);
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if (ret < 0) exit(1);
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samples = (uint16_t *)frame->data[0];
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avcodec_send_packet(dec_ctx, packet);
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while (avcodec_receive_frame(dec_ctx, frame) == 0) {
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int max_out = av_rescale_rnd(
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swr_get_delay(swr_ctx, dec_ctx->sample_rate) + frame->nb_samples,
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enc_ctx->sample_rate, dec_ctx->sample_rate, AV_ROUND_UP);
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for (j = 0; j < c->frame_size; j++) {
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samples[2 * j] = (int)(sin(t) * 10000);
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uint8_t** out_data = NULL;
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int out_linesize = 0;
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av_samples_alloc_array_and_samples(&out_data, &out_linesize, 2, max_out,
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enc_ctx->sample_fmt, 0);
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for (k = 1; k < c->ch_layout.nb_channels; k++)
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samples[2 * j + k] = samples[2 * j];
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t += tincr;
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int converted =
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swr_convert(swr_ctx, out_data, max_out, (const uint8_t**)frame->data,
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frame->nb_samples);
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// float PCM으로 임시 버퍼에 추가
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memcpy(pcm_buffer + buffered_samples * 2, out_data[0],
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converted * 2 * sizeof(float));
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buffered_samples += converted;
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av_freep(&out_data[0]);
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free(out_data);
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// OPUS_FRAME_SIZE 단위로 인코딩
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while (buffered_samples >= OPUS_FRAME_SIZE) {
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enc_frame->nb_samples = OPUS_FRAME_SIZE;
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enc_frame->format = enc_ctx->sample_fmt;
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enc_frame->sample_rate = enc_ctx->sample_rate;
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av_channel_layout_copy(&enc_frame->ch_layout, &enc_ctx->ch_layout);
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enc_frame->data[0] = (uint8_t*)pcm_buffer;
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AVPacket* out_pkt = av_packet_alloc();
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avcodec_send_frame(enc_ctx, enc_frame);
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while (avcodec_receive_packet(enc_ctx, out_pkt) == 0) {
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fwrite(out_pkt->data, 1, out_pkt->size, outfile);
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av_packet_unref(out_pkt);
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}
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encode(c, frame, pkt, f);
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av_packet_free(&out_pkt);
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// 버퍼 이동
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memmove(pcm_buffer, pcm_buffer + OPUS_FRAME_SIZE * 2,
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(buffered_samples - OPUS_FRAME_SIZE) * 2 * sizeof(float));
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buffered_samples -= OPUS_FRAME_SIZE;
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}
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}
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av_packet_unref(packet);
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}
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/* flush the encoder */
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encode(c, NULL, pkt, f);
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// 디코더 플러시
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avcodec_send_packet(dec_ctx, NULL);
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while (avcodec_receive_frame(dec_ctx, frame) == 0) {
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int max_out = av_rescale_rnd(
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swr_get_delay(swr_ctx, dec_ctx->sample_rate) + frame->nb_samples,
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enc_ctx->sample_rate, dec_ctx->sample_rate, AV_ROUND_UP);
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fclose(f);
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uint8_t** out_data = NULL;
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int out_linesize = 0;
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av_samples_alloc_array_and_samples(&out_data, &out_linesize, 2, max_out,
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enc_ctx->sample_fmt, 0);
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int converted =
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swr_convert(swr_ctx, out_data, max_out, (const uint8_t**)frame->data,
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frame->nb_samples);
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memcpy(pcm_buffer + buffered_samples * 2, out_data[0],
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converted * 2 * sizeof(float));
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buffered_samples += converted;
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av_freep(&out_data[0]);
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free(out_data);
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while (buffered_samples >= OPUS_FRAME_SIZE) {
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enc_frame->nb_samples = OPUS_FRAME_SIZE;
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enc_frame->format = enc_ctx->sample_fmt;
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enc_frame->sample_rate = enc_ctx->sample_rate;
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av_channel_layout_copy(&enc_frame->ch_layout, &enc_ctx->ch_layout);
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enc_frame->data[0] = (uint8_t*)pcm_buffer;
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AVPacket* out_pkt = av_packet_alloc();
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avcodec_send_frame(enc_ctx, enc_frame);
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while (avcodec_receive_packet(enc_ctx, out_pkt) == 0) {
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fwrite(out_pkt->data, 1, out_pkt->size, outfile);
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av_packet_unref(out_pkt);
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}
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av_packet_free(&out_pkt);
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memmove(pcm_buffer, pcm_buffer + OPUS_FRAME_SIZE * 2,
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(buffered_samples - OPUS_FRAME_SIZE) * 2 * sizeof(float));
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buffered_samples -= OPUS_FRAME_SIZE;
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}
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}
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// 마지막 남은 샘플 인코딩
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if (buffered_samples > 0) {
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enc_frame->nb_samples = buffered_samples;
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enc_frame->format = enc_ctx->sample_fmt;
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enc_frame->sample_rate = enc_ctx->sample_rate;
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av_channel_layout_copy(&enc_frame->ch_layout, &enc_ctx->ch_layout);
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enc_frame->data[0] = (uint8_t*)pcm_buffer;
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AVPacket* out_pkt = av_packet_alloc();
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avcodec_send_frame(enc_ctx, enc_frame);
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while (avcodec_receive_packet(enc_ctx, out_pkt) == 0) {
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fwrite(out_pkt->data, 1, out_pkt->size, outfile);
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av_packet_unref(out_pkt);
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}
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av_packet_free(&out_pkt);
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}
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// 인코더 플러시
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avcodec_send_frame(enc_ctx, NULL);
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AVPacket* out_pkt = av_packet_alloc();
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while (avcodec_receive_packet(enc_ctx, out_pkt) == 0) {
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fwrite(out_pkt->data, 1, out_pkt->size, outfile);
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av_packet_unref(out_pkt);
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}
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av_packet_free(&out_pkt);
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fclose(outfile);
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free(pcm_buffer);
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swr_free(&swr_ctx);
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av_frame_free(&frame);
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av_packet_free(&pkt);
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avcodec_free_context(&c);
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av_frame_free(&enc_frame);
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av_packet_free(&packet);
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avcodec_free_context(&dec_ctx);
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avcodec_free_context(&enc_ctx);
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avformat_close_input(&fmt_ctx);
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printf("Encoding finished: %s\n", output_filename);
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return 0;
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}
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155
tests/ffmpeg_decode_audio.cc
Normal file
155
tests/ffmpeg_decode_audio.cc
Normal file
@@ -0,0 +1,155 @@
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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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||||
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||||
int main() {
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const char* input_filename = "golden.webm";
|
||||
const char* output_filename = "output.pcm";
|
||||
|
||||
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;
|
||||
AVFrame* frame = NULL;
|
||||
SwrContext* swr_ctx = NULL;
|
||||
FILE* outfile = NULL;
|
||||
|
||||
if (avformat_open_input(&fmt_ctx, input_filename, NULL, NULL) < 0) {
|
||||
fprintf(stderr, "Could not open input file\n");
|
||||
return -1;
|
||||
}
|
||||
|
||||
if (avformat_find_stream_info(fmt_ctx, NULL) < 0) {
|
||||
fprintf(stderr, "Could not find stream info\n");
|
||||
return -1;
|
||||
}
|
||||
|
||||
int stream_index = -1;
|
||||
for (unsigned i = 0; i < fmt_ctx->nb_streams; i++) {
|
||||
if (fmt_ctx->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
|
||||
stream_index = i;
|
||||
break;
|
||||
}
|
||||
}
|
||||
if (stream_index == -1) {
|
||||
fprintf(stderr, "Could not find audio stream\n");
|
||||
return -1;
|
||||
}
|
||||
|
||||
codec =
|
||||
avcodec_find_decoder(fmt_ctx->streams[stream_index]->codecpar->codec_id);
|
||||
if (!codec) {
|
||||
fprintf(stderr, "Could not find decoder\n");
|
||||
return -1;
|
||||
}
|
||||
|
||||
codec_ctx = avcodec_alloc_context3(codec);
|
||||
if (!codec_ctx) {
|
||||
fprintf(stderr, "Could not allocate codec context\n");
|
||||
return -1;
|
||||
}
|
||||
|
||||
if (avcodec_parameters_to_context(
|
||||
codec_ctx, fmt_ctx->streams[stream_index]->codecpar) < 0) {
|
||||
fprintf(stderr, "Failed to copy codec parameters\n");
|
||||
return -1;
|
||||
}
|
||||
|
||||
if (avcodec_open2(codec_ctx, codec, NULL) < 0) {
|
||||
fprintf(stderr, "Could not open codec\n");
|
||||
return -1;
|
||||
}
|
||||
|
||||
packet = av_packet_alloc();
|
||||
frame = av_frame_alloc();
|
||||
if (!packet || !frame) {
|
||||
fprintf(stderr, "Could not allocate packet or frame\n");
|
||||
return -1;
|
||||
}
|
||||
|
||||
outfile = fopen(output_filename, "wb");
|
||||
if (!outfile) {
|
||||
fprintf(stderr, "Could not open output file\n");
|
||||
return -1;
|
||||
}
|
||||
|
||||
// AVChannelLayout 초기화
|
||||
AVChannelLayout in_layout, out_layout;
|
||||
if (av_channel_layout_copy(&in_layout, &codec_ctx->ch_layout) < 0) {
|
||||
fprintf(stderr, "Failed to copy channel layout\n");
|
||||
return -1;
|
||||
}
|
||||
av_channel_layout_default(&out_layout, 2); // 스테레오
|
||||
|
||||
swr_ctx = NULL; // 먼저 NULL로 선언
|
||||
if (swr_alloc_set_opts2(&swr_ctx, &out_layout, AV_SAMPLE_FMT_S16, 48000,
|
||||
&in_layout, codec_ctx->sample_fmt,
|
||||
codec_ctx->sample_rate, 0, NULL) < 0) {
|
||||
fprintf(stderr, "Failed to allocate and set SwrContext\n");
|
||||
return -1;
|
||||
}
|
||||
|
||||
if (swr_init(swr_ctx) < 0) {
|
||||
fprintf(stderr, "Failed to initialize SwrContext\n");
|
||||
return -1;
|
||||
}
|
||||
|
||||
while (av_read_frame(fmt_ctx, packet) >= 0) {
|
||||
if (packet->stream_index == stream_index) {
|
||||
if (avcodec_send_packet(codec_ctx, packet) == 0) {
|
||||
while (avcodec_receive_frame(codec_ctx, frame) == 0) {
|
||||
int out_samples =
|
||||
av_rescale_rnd(swr_get_delay(swr_ctx, codec_ctx->sample_rate) +
|
||||
frame->nb_samples,
|
||||
48000, codec_ctx->sample_rate, AV_ROUND_UP);
|
||||
|
||||
uint8_t** out_buf = NULL;
|
||||
int out_linesize = 0;
|
||||
av_samples_alloc_array_and_samples(&out_buf, &out_linesize, 2,
|
||||
out_samples, AV_SAMPLE_FMT_S16, 0);
|
||||
|
||||
int converted_samples =
|
||||
swr_convert(swr_ctx, out_buf, out_samples,
|
||||
(const uint8_t**)frame->data, frame->nb_samples);
|
||||
|
||||
fwrite(out_buf[0], 1, converted_samples * 2 * 2, outfile);
|
||||
av_freep(&out_buf[0]);
|
||||
free(out_buf);
|
||||
}
|
||||
}
|
||||
}
|
||||
av_packet_unref(packet);
|
||||
}
|
||||
|
||||
// 디코더 플러시
|
||||
avcodec_send_packet(codec_ctx, NULL);
|
||||
while (avcodec_receive_frame(codec_ctx, frame) == 0) {
|
||||
int out_samples = av_rescale_rnd(
|
||||
swr_get_delay(swr_ctx, codec_ctx->sample_rate) + frame->nb_samples,
|
||||
48000, codec_ctx->sample_rate, AV_ROUND_UP);
|
||||
|
||||
uint8_t** out_buf = NULL;
|
||||
int out_linesize = 0;
|
||||
av_samples_alloc_array_and_samples(&out_buf, &out_linesize, 2, out_samples,
|
||||
AV_SAMPLE_FMT_S16, 0);
|
||||
|
||||
int converted_samples =
|
||||
swr_convert(swr_ctx, out_buf, out_samples, (const uint8_t**)frame->data,
|
||||
frame->nb_samples);
|
||||
|
||||
fwrite(out_buf[0], 1, converted_samples * 2 * 2, outfile);
|
||||
av_freep(&out_buf[0]);
|
||||
free(out_buf);
|
||||
}
|
||||
|
||||
fclose(outfile);
|
||||
swr_free(&swr_ctx);
|
||||
av_frame_free(&frame);
|
||||
av_packet_free(&packet);
|
||||
avcodec_free_context(&codec_ctx);
|
||||
avformat_close_input(&fmt_ctx);
|
||||
|
||||
printf("Decoding finished, output saved to %s\n", output_filename);
|
||||
return 0;
|
||||
}
|
||||
Reference in New Issue
Block a user