From 356714a6eb92f7d458d3419e8bb2333141bd4638 Mon Sep 17 00:00:00 2001
From: dec05eba <dec05eba@protonmail.com>
Date: Wed, 10 Apr 2024 00:11:09 +0200
Subject: More audio bullshit, repeat write audio frames for audio delay
---
src/main.cpp | 36 ++++++++++++++++++++++++++++++++++--
1 file changed, 34 insertions(+), 2 deletions(-)
(limited to 'src')
diff --git a/src/main.cpp b/src/main.cpp
index bb54ca2..be0bba2 100644
--- a/src/main.cpp
+++ b/src/main.cpp
@@ -1713,6 +1713,12 @@ int main(int argc, char **argv) {
usage();
}
+ if(audio_codec == AudioCodec::FLAC) {
+ fprintf(stderr, "Warning: flac audio codec has been temporary disabled, using opus audio codec instead\n");
+ audio_codec_to_use = "opus";
+ audio_codec = AudioCodec::OPUS;
+ }
+
bool overclock = false;
const char *overclock_str = args["-oc"].value();
if(!overclock_str)
@@ -2388,9 +2394,35 @@ int main(int argc, char **argv) {
double received_audio_time = clock_get_monotonic_seconds();
const int64_t timeout_ms = std::round((1000.0 / (double)audio_track.codec_context->sample_rate) * 1000.0);
- // Move audio back by around 252 ms. This is just a shitty way to handle audio latency but pulseaudio latency calculation
+ // Move audio forward by around 252 ms (for opus/aac), or 42ms for flac. This is just a shitty way to handle audio latency but pulseaudio latency calculation
// returns much lower value which isn't helpful.
- audio_device.frame->pts = audio_track.codec_context->frame_size * 12;
+ if(needs_audio_conversion)
+ swr_convert(swr, &audio_device.frame->data[0], audio_track.codec_context->frame_size, (const uint8_t**)&empty_audio, audio_track.codec_context->frame_size);
+ else
+ audio_device.frame->data[0] = empty_audio;
+
+ int num_frames_to_delay = 12;
+ if(audio_codec == AudioCodec::FLAC)
+ num_frames_to_delay = 2;
+
+ for(int i = 0; i < num_frames_to_delay; ++i) {
+ if(audio_track.graph) {
+ std::lock_guard<std::mutex> lock(audio_filter_mutex);
+ // TODO: av_buffersrc_add_frame
+ if(av_buffersrc_write_frame(audio_device.src_filter_ctx, audio_device.frame) < 0) {
+ fprintf(stderr, "Error: failed to add audio frame to filter\n");
+ }
+ } else {
+ int ret = avcodec_send_frame(audio_track.codec_context, audio_device.frame);
+ if(ret >= 0) {
+ // TODO: Move to separate thread because this could write to network (for example when livestreaming)
+ receive_frames(audio_track.codec_context, audio_track.stream_index, audio_track.stream, audio_device.frame->pts, av_format_context, record_start_time, frame_data_queue, replay_buffer_size_secs, frames_erased, write_output_mutex, paused_time_offset);
+ } else {
+ fprintf(stderr, "Failed to encode audio!\n");
+ }
+ }
+ audio_device.frame->pts += audio_track.codec_context->frame_size;
+ }
while(running) {
void *sound_buffer;
--
cgit v1.2.3-70-g09d2