Attempt to fix audio timestamp being totally wrong on fedora

This commit is contained in:
dec05eba 2022-03-23 19:19:00 +01:00
parent c595c7ffb5
commit efc15622a9
2 changed files with 44 additions and 17 deletions

1
.gitignore vendored
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@ -5,6 +5,7 @@ tests/sibs-build/
tests/compile_commands.json
.clangd/
.cache/
main.o
sound.o

View File

@ -24,6 +24,7 @@
#include <thread>
#include <mutex>
#include <map>
#include <atomic>
#include <signal.h>
#include <unistd.h>
@ -45,6 +46,8 @@ extern "C" {
#include <libavutil/hwcontext_cuda.h>
#include <libavutil/opt.h>
#include <libswresample/swresample.h>
#include <libavutil/avutil.h>
#include <libavutil/time.h>
}
#include <cudaGL.h>
@ -325,10 +328,14 @@ static void receive_frames(AVCodecContext *av_codec_context, AVStream *stream,
av_packet.size = 0;
int res = avcodec_receive_packet(av_codec_context, &av_packet);
if (res == 0) { // we have a packet, send the packet to the muxer
av_packet_rescale_ts(&av_packet, av_codec_context->time_base,
stream->time_base);
//av_packet_rescale_ts(&av_packet, av_codec_context->time_base,
// stream->time_base);
if(av_packet.pts != AV_NOPTS_VALUE)
av_packet.pts = av_rescale_q(av_packet.pts, av_codec_context->time_base, stream->time_base);
if(av_packet.dts != AV_NOPTS_VALUE)
av_packet.dts = av_rescale_q(av_packet.dts, av_codec_context->time_base, stream->time_base);
av_packet.stream_index = stream->index;
av_packet.dts = AV_NOPTS_VALUE;
//av_packet.dts = AV_NOPTS_VALUE;
std::lock_guard<std::mutex> lock(write_output_mutex);
if(replay_buffer_size_secs != -1) {
double time_now = glfwGetTime();
@ -887,7 +894,7 @@ int main(int argc, char **argv) {
if(!record_area) {
record_width = window_pixmap.texture_width;
record_height = window_pixmap.texture_height;
fprintf(stderr, "Record size: %dx%x\n", record_width, record_height);
fprintf(stderr, "Record size: %dx%d\n", record_width, record_height);
}
} else {
window_pixmap.texture_id = 0;
@ -913,11 +920,18 @@ int main(int argc, char **argv) {
return 1;
}
av_format_context->flags |= AVFMT_FLAG_GENPTS;
const AVOutputFormat *output_format = av_format_context->oformat;
bool use_hevc = strcmp(window_str, "screen") == 0;
if(use_hevc && strcmp(container_format, "flv") == 0) {
use_hevc = false;
fprintf(stderr, "Warning: hevc is not compatible with flv, falling back to h264 instead.\n");
}
AVCodecContext *video_codec_context;
AVStream *video_stream =
add_video_stream(av_format_context, &video_codec_context, quality, record_width, record_height, fps, strcmp(window_str, "screen") == 0);
add_video_stream(av_format_context, &video_codec_context, quality, record_width, record_height, fps, use_hevc);
if (!video_stream) {
fprintf(stderr, "Error: Failed to create video stream\n");
return 1;
@ -992,7 +1006,7 @@ int main(int argc, char **argv) {
XDamageSubtract(dpy, damage,None,None);
*/
int frame_count = 0;
std::atomic_int frame_count(0);
CUcontext old_ctx;
CUarray mapped_array;
@ -1062,7 +1076,7 @@ int main(int argc, char **argv) {
uint8_t *audio_frame_buf = (uint8_t *)av_malloc(audio_buffer_size);
avcodec_fill_audio_frame(audio_frame, audio_codec_context->channels, audio_codec_context->sample_fmt, (const uint8_t*)audio_frame_buf, audio_buffer_size, 1);
audio_thread = std::thread([record_start_time, replay_buffer_size_secs, &frame_data_queue, &frames_erased, audio_codec_context](AVFormatContext *av_format_context, AVStream *audio_stream, uint8_t *audio_frame_buf, SoundDevice *sound_device, AVFrame *audio_frame, std::mutex *write_output_mutex) mutable {
audio_thread = std::thread([record_start_time, replay_buffer_size_secs, &frame_data_queue, &frames_erased, audio_codec_context, &frame_count](AVFormatContext *av_format_context, AVStream *audio_stream, uint8_t *audio_frame_buf, SoundDevice *sound_device, AVFrame *audio_frame, std::mutex *write_output_mutex) mutable {
SwrContext *swr = swr_alloc();
if(!swr) {
@ -1077,6 +1091,8 @@ int main(int argc, char **argv) {
av_opt_set_sample_fmt(swr, "out_sample_fmt", AV_SAMPLE_FMT_FLTP, 0);
swr_init(swr);
int64_t prev_frame_count = 0;
while(running) {
void *sound_buffer;
int sound_buffer_size = sound_device_read_next_chunk(sound_device, &sound_buffer);
@ -1084,12 +1100,15 @@ int main(int argc, char **argv) {
// TODO: Instead of converting audio, get float audio from alsa. Or does alsa do conversion internally to get this format?
swr_convert(swr, &audio_frame_buf, audio_frame->nb_samples, (const uint8_t**)&sound_buffer, sound_buffer_size);
audio_frame->extended_data = &audio_frame_buf;
// TODO: Fix this. Warning from ffmpeg:
// Timestamps are unset in a packet for stream 1. This is deprecated and will stop working in the future. Fix your code to set the timestamps properly
//audio_frame->pts=audio_frame_index*100;
//++audio_frame_index;
//audio_frame->pts = frame_count;
const int64_t pts = frame_count;
if(pts == prev_frame_count) {
prev_frame_count = pts;
continue;
}
prev_frame_count = pts;
audio_frame->pts = pts;
int ret = avcodec_send_frame(audio_codec_context, audio_frame);
if(ret < 0){
printf("Failed to encode!\n");
@ -1178,8 +1197,6 @@ int main(int argc, char **argv) {
break;
}
frame->pts = frame_count;
if(window_pixmap.texture_width < record_width)
frame->width = window_pixmap.texture_width & ~1;
else
@ -1311,15 +1328,24 @@ int main(int argc, char **argv) {
//fprintf(stderr, "Frame start index: %zu\n", start_index);
int64_t pos = 0;
int64_t pts_offset = 0;
if(frames_erased)
int64_t dts_offset = 0;
if(frames_erased) {
pos = frame_data_queue[start_index]->pos;
pts_offset = frame_data_queue[start_index]->pts;
dts_offset = frame_data_queue[start_index]->dts;
}
for(size_t i = start_index; i < frame_data_queue.size(); ++i) {
AVPacket *av_packet = frame_data_queue[i];
if(av_packet->stream_index == video_stream->index) {
av_packet->pos = -1;
if(av_packet->stream_index == audio_stream->index) {
av_packet->pts = AV_NOPTS_VALUE;
av_packet->dts = AV_NOPTS_VALUE;
} else {
av_packet->pts -= pts_offset;
//av_packet->pos -= pos;
av_packet->dts = AV_NOPTS_VALUE;
}
av_packet->pos = -1;