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https://github.com/kunkundi/crossdesk.git
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Enable audio transmission
This commit is contained in:
dijunkun
232
test/audio_capture/ffmpeg_audio.cpp
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232
test/audio_capture/ffmpeg_audio.cpp
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@@ -0,0 +1,232 @@
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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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static int get_format_from_sample_fmt(const char **fmt,
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enum AVSampleFormat sample_fmt) {
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int i;
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struct sample_fmt_entry {
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enum AVSampleFormat sample_fmt;
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const char *fmt_be, *fmt_le;
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} sample_fmt_entries[] = {
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{AV_SAMPLE_FMT_U8, "u8", "u8"},
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{AV_SAMPLE_FMT_S16, "s16be", "s16le"},
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{AV_SAMPLE_FMT_S32, "s32be", "s32le"},
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{AV_SAMPLE_FMT_FLT, "f32be", "f32le"},
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{AV_SAMPLE_FMT_DBL, "f64be", "f64le"},
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};
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*fmt = NULL;
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for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
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struct sample_fmt_entry *entry = &sample_fmt_entries[i];
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if (sample_fmt == entry->sample_fmt) {
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*fmt = AV_NE(entry->fmt_be, entry->fmt_le);
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return 0;
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}
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}
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fprintf(stderr, "Sample format %s not supported as output format\n",
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av_get_sample_fmt_name(sample_fmt));
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return AVERROR(EINVAL);
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}
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/**
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* Fill dst buffer with nb_samples, generated starting from t. <20><><EFBFBD><EFBFBD>ģʽ<C4A3><CABD>
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*/
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static void fill_samples(double *dst, int nb_samples, int nb_channels,
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int sample_rate, double *t) {
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int i, j;
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double tincr = 1.0 / sample_rate, *dstp = dst;
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const double c = 2 * M_PI * 440.0;
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/* generate sin tone with 440Hz frequency and duplicated channels */
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for (i = 0; i < nb_samples; i++) {
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*dstp = sin(c * *t);
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for (j = 1; j < nb_channels; j++) dstp[j] = dstp[0];
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dstp += nb_channels;
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*t += tincr;
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}
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}
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int main(int argc, char **argv) {
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// <20><><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD>
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int64_t src_ch_layout = AV_CH_LAYOUT_MONO;
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int src_rate = 44100;
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enum AVSampleFormat src_sample_fmt = AV_SAMPLE_FMT_DBL;
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int src_nb_channels = 0;
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uint8_t **src_data = NULL; // <20><><EFBFBD><EFBFBD>ָ<EFBFBD><D6B8>
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int src_linesize;
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int src_nb_samples = 1024;
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// <20><><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD>
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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; // <20><><EFBFBD><EFBFBD>ָ<EFBFBD><D6B8>
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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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// <20><><EFBFBD><EFBFBD><EFBFBD>ļ<EFBFBD>
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const char *dst_filename = NULL; // <20><><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD>pcm<63><6D><EFBFBD><EFBFBD><EFBFBD>أ<EFBFBD>Ȼ<EFBFBD><EFBFBD><F3B2A5B7><EFBFBD>֤
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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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// <20>ز<EFBFBD><D8B2><EFBFBD>ʵ<EFBFBD><CAB5>
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struct SwrContext *swr_ctx;
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double t;
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int ret;
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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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// <20><><EFBFBD><EFBFBD><EFBFBD>ز<EFBFBD><D8B2><EFBFBD><EFBFBD><EFBFBD>
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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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goto end;
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}
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// <20><><EFBFBD><EFBFBD><EFBFBD>ز<EFBFBD><D8B2><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD>
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/* set options */
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// <20><><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD>
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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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// <20><><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD>
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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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// <20><>ʼ<EFBFBD><CABC><EFBFBD>ز<EFBFBD><D8B2><EFBFBD>
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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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goto end;
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}
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/* allocate source and destination samples buffers */
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// <20><><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD>Դ<EFBFBD><D4B4>ͨ<EFBFBD><CDA8><EFBFBD><EFBFBD><EFBFBD><EFBFBD>
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src_nb_channels = av_get_channel_layout_nb_channels(src_ch_layout);
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// <20><><EFBFBD><EFBFBD><EFBFBD><EFBFBD>Դ<EFBFBD><D4B4><EFBFBD><EFBFBD><EFBFBD>ڴ<EFBFBD><DAB4>ռ<EFBFBD>
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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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goto end;
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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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// <20><><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD>
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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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*/
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dst_nb_channels = av_get_channel_layout_nb_channels(dst_ch_layout);
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// <20><><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD>ڴ<EFBFBD>
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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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goto end;
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}
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t = 0;
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do {
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/* generate synthetic audio */
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// <20><><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD>Դ
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fill_samples((double *)src_data[0], src_nb_samples, src_nb_channels,
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src_rate, &t);
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/* compute destination number of samples */
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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) break;
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max_dst_nb_samples = dst_nb_samples;
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}
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// int fifo_size = swr_get_out_samples(swr_ctx,src_nb_samples);
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// printf("fifo_size:%d\n", fifo_size);
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// if(fifo_size < 1024)
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// continue;
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/* convert to destination format */
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// ret = swr_convert(swr_ctx, dst_data, dst_nb_samples, (const
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// uint8_t **)src_data, src_nb_samples);
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ret = swr_convert(swr_ctx, dst_data, dst_nb_samples,
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(const uint8_t **)src_data, src_nb_samples);
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if (ret < 0) {
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fprintf(stderr, "Error while converting\n");
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goto end;
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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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goto end;
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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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} while (t < 10);
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ret = swr_convert(swr_ctx, dst_data, dst_nb_samples, NULL, 0);
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if (ret < 0) {
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fprintf(stderr, "Error while converting\n");
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goto end;
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}
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dst_bufsize = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels, ret,
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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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goto end;
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}
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printf("flush in:%d out:%d\n", 0, ret);
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fwrite(dst_data[0], 1, dst_bufsize, dst_file);
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if ((ret = get_format_from_sample_fmt(&fmt, dst_sample_fmt)) < 0) goto end;
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fprintf(stderr,
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"Resampling succeeded. Play the output file with the command:\n"
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"ffplay -f %s -channel_layout %" PRId64 " -channels %d -ar %d %s\n",
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fmt, dst_ch_layout, dst_nb_channels, dst_rate, dst_filename);
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end:
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fclose(dst_file);
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if (src_data) av_freep(&src_data[0]);
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av_freep(&src_data);
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if (dst_data) av_freep(&dst_data[0]);
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av_freep(&dst_data);
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swr_free(&swr_ctx);
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return ret < 0;
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}
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