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https://github.com/kunkundi/crossdesk.git
synced 2026-03-25 18:07:34 +08:00
Opus codec module test pass
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dijunkun
@@ -1,3 +1,20 @@
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#include <SDL2/SDL.h>
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#include <stdio.h>
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#include <stdlib.h>
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extern "C" {
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#include <libavcodec/avcodec.h>
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#include <libavdevice/avdevice.h>
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#include <libavfilter/avfilter.h>
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#include <libavformat/avformat.h>
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#include <libavutil/channel_layout.h>
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#include <libavutil/imgutils.h>
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#include <libavutil/opt.h>
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#include <libavutil/samplefmt.h>
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#include <libswresample/swresample.h>
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#include <libswscale/swscale.h>
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};
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#include <fstream>
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#include <iostream>
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#include <vector>
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@@ -5,27 +22,197 @@
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#include "OpusEncoderImpl.h"
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#include "opus/opus.h"
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int main() {
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OpusEncoderImpl* opusEncoder = new OpusEncoderImpl(48000, 2);
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static SDL_AudioDeviceID input_dev;
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static SDL_AudioDeviceID output_dev;
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std::ifstream inputFile("ls.pcm", std::ios::binary);
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if (!inputFile) {
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std::cerr << "Failed to open input file." << std::endl;
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static Uint8 *buffer = 0;
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static int in_pos = 0;
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static int out_pos = 0;
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char *out = "audio_old.pcm";
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FILE *outfile = fopen(out, "wb+");
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static OpusEncoderImpl *opusEncoder = nullptr;
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int64_t src_ch_layout = AV_CH_LAYOUT_MONO;
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int src_rate = 48000;
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enum AVSampleFormat src_sample_fmt = AV_SAMPLE_FMT_FLT;
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int src_nb_channels = 0;
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uint8_t **src_data = NULL; // 二级指针
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int src_linesize;
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int src_nb_samples = 480;
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// 输出参数
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int64_t dst_ch_layout = AV_CH_LAYOUT_STEREO;
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int dst_rate = 48000;
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enum AVSampleFormat dst_sample_fmt = AV_SAMPLE_FMT_S16;
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int dst_nb_channels = 0;
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uint8_t **dst_data = NULL; // 二级指针
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int dst_linesize;
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int dst_nb_samples;
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int max_dst_nb_samples;
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// 输出文件
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const char *dst_filename = NULL; // 保存输出的pcm到本地，然后播放验证
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FILE *dst_file;
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int dst_bufsize;
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const char *fmt;
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// 重采样实例
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struct SwrContext *swr_ctx;
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double t;
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int ret;
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void cb_in(void *userdata, Uint8 *stream, int len) {
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// If len < 4, the printf below will probably segfault
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{
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fwrite(stream, 1, len, outfile);
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fflush(outfile);
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}
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{
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int64_t delay = swr_get_delay(swr_ctx, src_rate);
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dst_nb_samples =
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av_rescale_rnd(delay + src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
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if (dst_nb_samples > max_dst_nb_samples) {
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av_freep(&dst_data[0]);
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ret = av_samples_alloc(dst_data, &dst_linesize, dst_nb_channels,
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dst_nb_samples, dst_sample_fmt, 1);
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if (ret < 0) return;
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max_dst_nb_samples = dst_nb_samples;
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}
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ret = swr_convert(swr_ctx, dst_data, dst_nb_samples,
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(const uint8_t **)&stream, src_nb_samples);
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if (ret < 0) {
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fprintf(stderr, "Error while converting\n");
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return;
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}
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dst_bufsize = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels,
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ret, dst_sample_fmt, 1);
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if (dst_bufsize < 0) {
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fprintf(stderr, "Could not get sample buffer size\n");
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return;
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}
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printf("t:%f in:%d out:%d\n", t, src_nb_samples, ret);
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fwrite(dst_data[0], 1, dst_bufsize, dst_file);
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opusEncoder->Feed(dst_data[0], dst_bufsize);
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}
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}
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void cb_out(void *userdata, Uint8 *stream, int len) {
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// If len < 4, the printf below will probably segfault
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SDL_memcpy(buffer + out_pos, stream, len);
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out_pos += len;
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}
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int init() {
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dst_filename = "res.pcm";
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dst_file = fopen(dst_filename, "wb");
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if (!dst_file) {
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fprintf(stderr, "Could not open destination file %s\n", dst_filename);
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exit(1);
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}
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// 创建重采样器
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/* create resampler context */
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swr_ctx = swr_alloc();
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if (!swr_ctx) {
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fprintf(stderr, "Could not allocate resampler context\n");
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ret = AVERROR(ENOMEM);
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return -1;
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}
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char sample[960];
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while (inputFile.read(sample, 960)) {
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opusEncoder->Feed((unsigned char*)sample, 960);
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// 设置重采样参数
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/* set options */
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// 输入参数
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av_opt_set_int(swr_ctx, "in_channel_layout", src_ch_layout, 0);
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av_opt_set_int(swr_ctx, "in_sample_rate", src_rate, 0);
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av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", src_sample_fmt, 0);
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// 输出参数
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av_opt_set_int(swr_ctx, "out_channel_layout", dst_ch_layout, 0);
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av_opt_set_int(swr_ctx, "out_sample_rate", dst_rate, 0);
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av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", dst_sample_fmt, 0);
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// 初始化重采样
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/* initialize the resampling context */
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if ((ret = swr_init(swr_ctx)) < 0) {
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fprintf(stderr, "Failed to initialize the resampling context\n");
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return -1;
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}
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// // 读取编码后的opus，一般放在单独线程，这里只是为了方便
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// StreamInfo info;
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// while (opusEncoder.PopFrame(info)) {
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// .....
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// }
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/* allocate source and destination samples buffers */
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// 计算出输入源的通道数量
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src_nb_channels = av_get_channel_layout_nb_channels(src_ch_layout);
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// 给输入源分配内存空间
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ret = av_samples_alloc_array_and_samples(&src_data, &src_linesize,
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src_nb_channels, src_nb_samples,
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src_sample_fmt, 0);
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if (ret < 0) {
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fprintf(stderr, "Could not allocate source samples\n");
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return -1;
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}
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/* compute the number of converted samples: buffering is avoided
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* ensuring that the output buffer will contain at least all the
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* converted input samples */
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// 计算输出采样数量
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max_dst_nb_samples = dst_nb_samples =
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av_rescale_rnd(src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
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/* buffer is going to be directly written to a rawaudio file, no alignment */
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dst_nb_channels = av_get_channel_layout_nb_channels(dst_ch_layout);
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// 分配输出缓存内存
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ret = av_samples_alloc_array_and_samples(&dst_data, &dst_linesize,
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dst_nb_channels, dst_nb_samples,
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dst_sample_fmt, 0);
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if (ret < 0) {
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fprintf(stderr, "Could not allocate destination samples\n");
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return -1;
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}
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}
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int main() {
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init();
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SDL_Init(SDL_INIT_AUDIO);
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// 16Mb should be enough; the test lasts 5 seconds
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buffer = (Uint8 *)malloc(16777215);
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SDL_AudioSpec want_in, want_out, have_in, have_out;
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SDL_zero(want_in);
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want_in.freq = 48000;
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want_in.format = AUDIO_F32LSB;
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want_in.channels = 2;
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want_in.samples = 960;
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want_in.callback = cb_in;
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input_dev = SDL_OpenAudioDevice(NULL, 1, &want_in, &have_in,
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SDL_AUDIO_ALLOW_ANY_CHANGE);
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printf("%d %d %d %d\n", have_in.freq, have_in.format, have_in.channels,
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have_in.samples);
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if (input_dev == 0) {
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SDL_Log("Failed to open input: %s", SDL_GetError());
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return 1;
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}
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SDL_PauseAudioDevice(input_dev, 0);
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SDL_PauseAudioDevice(output_dev, 0);
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opusEncoder = new OpusEncoderImpl(have_in.freq, have_in.channels);
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SDL_Delay(5000);
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opusEncoder->Stop();
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SDL_CloseAudioDevice(output_dev);
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SDL_CloseAudioDevice(input_dev);
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free(buffer);
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return 0;
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}
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fclose(outfile);
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}
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