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Enable audio transmission

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This commit is contained in:
dijunkun
2023-11-29 22:04:53 -08:00
parent 733434f9b3
commit 4a65a59803
12 changed files with 1520 additions and 271 deletions
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196
test/audio_capture/sdl2_audio_capture.cpp
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@@ -2,6 +2,19 @@
#include <stdio.h>
#include <stdlib.h>
extern "C" {
#include <libavcodec/avcodec.h>
#include <libavdevice/avdevice.h>
#include <libavfilter/avfilter.h>
#include <libavformat/avformat.h>
#include <libavutil/channel_layout.h>
#include <libavutil/imgutils.h>
#include <libavutil/opt.h>
#include <libavutil/samplefmt.h>
#include <libswresample/swresample.h>
#include <libswscale/swscale.h>
};
static SDL_AudioDeviceID input_dev;
static SDL_AudioDeviceID output_dev;
@@ -9,54 +22,152 @@ static Uint8 *buffer = 0;
static int in_pos = 0;
static int out_pos = 0;
int64_t src_ch_layout = AV_CH_LAYOUT_MONO;
int src_rate = 48000;
enum AVSampleFormat src_sample_fmt = AV_SAMPLE_FMT_FLT;
int src_nb_channels = 0;
uint8_t **src_data = NULL; // <20><><EFBFBD><EFBFBD>ָ<EFBFBD><D6B8>
int src_linesize;
int src_nb_samples = 480;
// <20><><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD>
int64_t dst_ch_layout = AV_CH_LAYOUT_STEREO;
int dst_rate = 48000;
enum AVSampleFormat dst_sample_fmt = AV_SAMPLE_FMT_S16;
int dst_nb_channels = 0;
uint8_t **dst_data = NULL; // <20><><EFBFBD><EFBFBD>ָ<EFBFBD><D6B8>
int dst_linesize;
int dst_nb_samples;
int max_dst_nb_samples;
// <20><><EFBFBD><EFBFBD><EFBFBD>ļ<EFBFBD>
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>֤
FILE *dst_file;
int dst_bufsize;
const char *fmt;
// <20>ز<EFBFBD><D8B2><EFBFBD>ʵ<EFBFBD><CAB5>
struct SwrContext *swr_ctx;
double t;
int ret;
char *out = "audio_old.pcm";
FILE *outfile = fopen(out, "wb+");
void cb_in(void *userdata, Uint8 *stream, int len) {
// If len < 4, the printf below will probably segfault
{
fwrite(stream, 1, len, outfile);
fflush(outfile);
}
{
int64_t delay = swr_get_delay(swr_ctx, src_rate);
dst_nb_samples =
av_rescale_rnd(delay + src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
if (dst_nb_samples > max_dst_nb_samples) {
av_freep(&dst_data[0]);
ret = av_samples_alloc(dst_data, &dst_linesize, dst_nb_channels,
dst_nb_samples, dst_sample_fmt, 1);
if (ret < 0) return;
max_dst_nb_samples = dst_nb_samples;
}
// fwrite(stream, 1, len, outfile);
// fflush(outfile);
// SDL_memcpy(stream, buffer + in_pos, len);
// in_pos += len;
// printf("IN: %d\t%d %d %d %d\n", in_pos, stream[0], stream[1], stream[2],
// stream[3]);
ret = swr_convert(swr_ctx, dst_data, dst_nb_samples,
(const uint8_t **)&stream, src_nb_samples);
if (ret < 0) {
fprintf(stderr, "Error while converting\n");
return;
}
dst_bufsize = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels,
ret, dst_sample_fmt, 1);
if (dst_bufsize < 0) {
fprintf(stderr, "Could not get sample buffer size\n");
return;
}
printf("t:%f in:%d out:%d\n", t, src_nb_samples, ret);
fwrite(dst_data[0], 1, dst_bufsize, dst_file);
}
}
void cb_out(void *userdata, Uint8 *stream, int len) {
// If len < 4, the printf below will probably segfault
fwrite(stream, 1, len, outfile);
fflush(outfile);
// if (out_pos >= in_pos) {
// // Output is way ahead of input; fill with emptiness
// memset(buffer + out_pos, 0, len * sizeof(Uint8));
// printf("OUT: %d\t(Empty)\n", out_pos);
// } else if (out_pos + len > in_pos) {
// // Output is reaching input; read until reaching input, and leave the
// rest
// // empty
// memset(buffer + out_pos, 0, len * sizeof(Uint8));
// SDL_memcpy(buffer + out_pos, stream, in_pos - out_pos);
// out_pos = in_pos;
// printf("OUT: %d\t%d %d %d %d (Partial)\n", out_pos, stream[0], stream[1],
// stream[2], stream[3]);
// } else {
// // Input is way ahead of output; read as much as requested
// SDL_memcpy(buffer + out_pos, stream, len);
// out_pos += len;
// printf("OUT: %d\t%d %d %d %d\n", out_pos, stream[0], stream[1],
// stream[2],
// stream[3]);
// }
SDL_memcpy(buffer + out_pos, stream, len);
out_pos += len;
}
// This is to make sure the output device works
// for (int i = 0; i < len; i++)
// stream[i] = (Uint8) random();
int init() {
dst_filename = "res.pcm";
dst_file = fopen(dst_filename, "wb");
if (!dst_file) {
fprintf(stderr, "Could not open destination file %s\n", dst_filename);
exit(1);
}
// <20><><EFBFBD><EFBFBD><EFBFBD>ز<EFBFBD><D8B2><EFBFBD><EFBFBD><EFBFBD>
/* create resampler context */
swr_ctx = swr_alloc();
if (!swr_ctx) {
fprintf(stderr, "Could not allocate resampler context\n");
ret = AVERROR(ENOMEM);
return -1;
}
// <20><><EFBFBD><EFBFBD><EFBFBD>ز<EFBFBD><D8B2><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD>
/* set options */
// <20><><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD>
av_opt_set_int(swr_ctx, "in_channel_layout", src_ch_layout, 0);
av_opt_set_int(swr_ctx, "in_sample_rate", src_rate, 0);
av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", src_sample_fmt, 0);
// <20><><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD>
av_opt_set_int(swr_ctx, "out_channel_layout", dst_ch_layout, 0);
av_opt_set_int(swr_ctx, "out_sample_rate", dst_rate, 0);
av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", dst_sample_fmt, 0);
// <20><>ʼ<EFBFBD><CABC><EFBFBD>ز<EFBFBD><D8B2><EFBFBD>
/* initialize the resampling context */
if ((ret = swr_init(swr_ctx)) < 0) {
fprintf(stderr, "Failed to initialize the resampling context\n");
return -1;
}
/* allocate source and destination samples buffers */
// <20><><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD>Դ<EFBFBD><D4B4>ͨ<EFBFBD><CDA8><EFBFBD><EFBFBD><EFBFBD><EFBFBD>
src_nb_channels = av_get_channel_layout_nb_channels(src_ch_layout);
// <20><><EFBFBD><EFBFBD><EFBFBD><EFBFBD>Դ<EFBFBD><D4B4><EFBFBD><EFBFBD><EFBFBD>ڴ<EFBFBD><DAB4>ռ<EFBFBD>
ret = av_samples_alloc_array_and_samples(&src_data, &src_linesize,
src_nb_channels, src_nb_samples,
src_sample_fmt, 0);
if (ret < 0) {
fprintf(stderr, "Could not allocate source samples\n");
return -1;
}
/* compute the number of converted samples: buffering is avoided
* ensuring that the output buffer will contain at least all the
* converted input samples */
// <20><><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD>
max_dst_nb_samples = dst_nb_samples =
av_rescale_rnd(src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
/* buffer is going to be directly written to a rawaudio file, no alignment */
dst_nb_channels = av_get_channel_layout_nb_channels(dst_ch_layout);
// <20><><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD>ڴ<EFBFBD>
ret = av_samples_alloc_array_and_samples(&dst_data, &dst_linesize,
dst_nb_channels, dst_nb_samples,
dst_sample_fmt, 0);
if (ret < 0) {
fprintf(stderr, "Could not allocate destination samples\n");
return -1;
}
}
int main() {
init();
SDL_Init(SDL_INIT_AUDIO);
// 16Mb should be enough; the test lasts 5 seconds
@@ -64,29 +175,18 @@ int main() {
SDL_AudioSpec want_in, want_out, have_in, have_out;
SDL_zero(want_out);
want_out.freq = 48000;
want_out.format = AUDIO_U16LSB;
want_out.channels = 2;
want_out.samples = 960;
want_out.callback = cb_out;
output_dev = SDL_OpenAudioDevice(NULL, 0, &want_out, &have_out,
SDL_AUDIO_ALLOW_ANY_CHANGE);
if (output_dev == 0) {
SDL_Log("Failed to open output: %s", SDL_GetError());
return 1;
}
SDL_zero(want_in);
want_in.freq = 48000;
want_in.format = AUDIO_U16LSB;
want_in.format = AUDIO_F32LSB;
want_in.channels = 2;
want_in.samples = 960;
want_in.callback = cb_in;
input_dev = SDL_OpenAudioDevice(NULL, 1, &want_in, &have_in,
SDL_AUDIO_ALLOW_ANY_CHANGE);
printf("%d %d %d %d\n", have_in.freq, have_in.format, have_in.channels,
have_in.samples);
if (input_dev == 0) {
SDL_Log("Failed to open input: %s", SDL_GetError());
return 1;