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Remove resampling process during SDL2 capture audio stream

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dijunkun
2024-05-21 17:01:08 +08:00
parent 9962829885
commit 669b944cfd
1 changed files with 13 additions and 111 deletions
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124
src/gui/main.cpp
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@@ -55,23 +55,6 @@ int fps = 0;
static std::atomic<bool> audio_buffer_fresh{false}; static std::atomic<bool> audio_buffer_fresh{false};
static uint32_t last_ts = 0; static uint32_t last_ts = 0;
// int64_t src_ch_layout = AV_CH_LAYOUT_MONO;
// int src_rate = 48000;
// enum AVSampleFormat src_sample_fmt = AV_SAMPLE_FMT_S16;
// int src_nb_channels = 0;
// uint8_t **src_data = NULL;
// int src_linesize;
// int src_nb_samples = 480;
// int64_t dst_ch_layout = AV_CH_LAYOUT_MONO;
// int dst_rate = 48000;
// enum AVSampleFormat dst_sample_fmt = AV_SAMPLE_FMT_S16;
// int dst_nb_channels = 0;
// uint8_t **dst_data = NULL;
// int dst_linesize;
// int dst_nb_samples;
// int max_dst_nb_samples;
int dst_bufsize; int dst_bufsize;
struct SwrContext *swr_ctx; struct SwrContext *swr_ctx;
@@ -187,47 +170,20 @@ inline int ProcessMouseKeyEven(SDL_Event &ev) {
} }
void SdlCaptureAudioIn(void *userdata, Uint8 *stream, int len) { void SdlCaptureAudioIn(void *userdata, Uint8 *stream, int len) {
// int64_t delay = swr_get_delay(swr_ctx, src_rate); if (1) {
// dst_nb_samples = (int)av_rescale_rnd(delay + src_nb_samples, dst_rate, if ("ClientConnected" == client_connection_status_str) {
// src_rate, AV_ROUND_UP); SendData(peer_client, DATA_TYPE::AUDIO, (const char *)stream, len);
// 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;
// }
// ret = swr_convert(swr_ctx, dst_data, dst_nb_samples, if ("ServerConnected" == server_connection_status_str) {
// (const uint8_t **)&stream, src_nb_samples); SendData(peer_server, DATA_TYPE::AUDIO, (const char *)stream, len);
// if (ret < 0) { }
// fprintf(stderr, "Error while converting\n"); } else {
// return; memcpy(audio_buffer, stream, len);
// } audio_len = len;
// dst_bufsize = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels, SDL_Delay(10);
// ret, audio_buffer_fresh = true;
// dst_sample_fmt, 1); }
// if (dst_bufsize < 0) {
// fprintf(stderr, "Could not get sample buffer size\n");
// return;
// }
// if (1) {
// if ("ClientConnected" == client_connection_status_str) {
// SendData(peer_client, DATA_TYPE::AUDIO, (const char *)dst_data[0],
// dst_bufsize);
// }
// if ("ServerConnected" == server_connection_status_str) {
// SendData(peer_server, DATA_TYPE::AUDIO, (const char *)dst_data[0],
// dst_bufsize);
// }
// } else {
// memcpy(audio_buffer, dst_data[0], dst_bufsize);
// audio_len = dst_bufsize;
// SDL_Delay(10);
// audio_buffer_fresh = true;
// }
} }
void SdlCaptureAudioOut(void *userdata, Uint8 *stream, int len) { void SdlCaptureAudioOut(void *userdata, Uint8 *stream, int len) {
@@ -386,58 +342,6 @@ void ClientConnectionStatus(ConnectionStatus status) {
} }
} }
int initResampler() {
/* create resampler context */
// swr_ctx = swr_alloc();
// if (!swr_ctx) {
// fprintf(stderr, "Could not allocate resampler context\n");
// ret = AVERROR(ENOMEM);
// return -1;
// }
// /* set options */
// 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);
// 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);
// /* 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 */
// src_nb_channels = av_get_channel_layout_nb_channels(src_ch_layout);
// 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;
// }
// 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);
// 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;
// }
return 0;
}
int main(int argc, char *argv[]) { int main(int argc, char *argv[]) {
LOG_INFO("Remote desk"); LOG_INFO("Remote desk");
@@ -446,8 +350,6 @@ int main(int argc, char *argv[]) {
std::chrono::high_resolution_clock::now().time_since_epoch()) std::chrono::high_resolution_clock::now().time_since_epoch())
.count()); .count());
initResampler();
cd_cache_file = fopen("cache.cd", "r+"); cd_cache_file = fopen("cache.cd", "r+");
if (cd_cache_file) { if (cd_cache_file) {
fseek(cd_cache_file, 0, SEEK_SET); fseek(cd_cache_file, 0, SEEK_SET);