Attempt to fix audio timestamp being totally wrong on fedora
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dec05eba
parent
c595c7ffb5
commit
efc15622a9
1
.gitignore
vendored
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.gitignore
vendored
@ -5,6 +5,7 @@ tests/sibs-build/
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tests/compile_commands.json
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.clangd/
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.cache/
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main.o
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sound.o
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60
src/main.cpp
60
src/main.cpp
@ -24,6 +24,7 @@
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#include <thread>
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#include <mutex>
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#include <map>
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#include <atomic>
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#include <signal.h>
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#include <unistd.h>
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@ -45,6 +46,8 @@ extern "C" {
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#include <libavutil/hwcontext_cuda.h>
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#include <libavutil/opt.h>
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#include <libswresample/swresample.h>
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#include <libavutil/avutil.h>
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#include <libavutil/time.h>
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}
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#include <cudaGL.h>
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@ -325,10 +328,14 @@ static void receive_frames(AVCodecContext *av_codec_context, AVStream *stream,
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av_packet.size = 0;
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int res = avcodec_receive_packet(av_codec_context, &av_packet);
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if (res == 0) { // we have a packet, send the packet to the muxer
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av_packet_rescale_ts(&av_packet, av_codec_context->time_base,
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stream->time_base);
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//av_packet_rescale_ts(&av_packet, av_codec_context->time_base,
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// stream->time_base);
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if(av_packet.pts != AV_NOPTS_VALUE)
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av_packet.pts = av_rescale_q(av_packet.pts, av_codec_context->time_base, stream->time_base);
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if(av_packet.dts != AV_NOPTS_VALUE)
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av_packet.dts = av_rescale_q(av_packet.dts, av_codec_context->time_base, stream->time_base);
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av_packet.stream_index = stream->index;
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av_packet.dts = AV_NOPTS_VALUE;
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//av_packet.dts = AV_NOPTS_VALUE;
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std::lock_guard<std::mutex> lock(write_output_mutex);
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if(replay_buffer_size_secs != -1) {
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double time_now = glfwGetTime();
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@ -887,7 +894,7 @@ int main(int argc, char **argv) {
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if(!record_area) {
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record_width = window_pixmap.texture_width;
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record_height = window_pixmap.texture_height;
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fprintf(stderr, "Record size: %dx%x\n", record_width, record_height);
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fprintf(stderr, "Record size: %dx%d\n", record_width, record_height);
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}
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} else {
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window_pixmap.texture_id = 0;
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@ -913,11 +920,18 @@ int main(int argc, char **argv) {
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return 1;
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}
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av_format_context->flags |= AVFMT_FLAG_GENPTS;
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const AVOutputFormat *output_format = av_format_context->oformat;
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bool use_hevc = strcmp(window_str, "screen") == 0;
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if(use_hevc && strcmp(container_format, "flv") == 0) {
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use_hevc = false;
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fprintf(stderr, "Warning: hevc is not compatible with flv, falling back to h264 instead.\n");
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}
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AVCodecContext *video_codec_context;
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AVStream *video_stream =
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add_video_stream(av_format_context, &video_codec_context, quality, record_width, record_height, fps, strcmp(window_str, "screen") == 0);
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add_video_stream(av_format_context, &video_codec_context, quality, record_width, record_height, fps, use_hevc);
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if (!video_stream) {
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fprintf(stderr, "Error: Failed to create video stream\n");
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return 1;
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@ -992,7 +1006,7 @@ int main(int argc, char **argv) {
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XDamageSubtract(dpy, damage,None,None);
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*/
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int frame_count = 0;
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std::atomic_int frame_count(0);
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CUcontext old_ctx;
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CUarray mapped_array;
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@ -1062,7 +1076,7 @@ int main(int argc, char **argv) {
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uint8_t *audio_frame_buf = (uint8_t *)av_malloc(audio_buffer_size);
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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);
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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 {
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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 {
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SwrContext *swr = swr_alloc();
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if(!swr) {
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@ -1077,6 +1091,8 @@ int main(int argc, char **argv) {
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av_opt_set_sample_fmt(swr, "out_sample_fmt", AV_SAMPLE_FMT_FLTP, 0);
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swr_init(swr);
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int64_t prev_frame_count = 0;
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while(running) {
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void *sound_buffer;
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int sound_buffer_size = sound_device_read_next_chunk(sound_device, &sound_buffer);
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@ -1084,12 +1100,15 @@ int main(int argc, char **argv) {
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// TODO: Instead of converting audio, get float audio from alsa. Or does alsa do conversion internally to get this format?
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swr_convert(swr, &audio_frame_buf, audio_frame->nb_samples, (const uint8_t**)&sound_buffer, sound_buffer_size);
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audio_frame->extended_data = &audio_frame_buf;
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// TODO: Fix this. Warning from ffmpeg:
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// 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
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//audio_frame->pts=audio_frame_index*100;
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//++audio_frame_index;
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//audio_frame->pts = frame_count;
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const int64_t pts = frame_count;
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if(pts == prev_frame_count) {
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prev_frame_count = pts;
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continue;
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}
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prev_frame_count = pts;
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audio_frame->pts = pts;
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int ret = avcodec_send_frame(audio_codec_context, audio_frame);
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if(ret < 0){
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printf("Failed to encode!\n");
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@ -1178,8 +1197,6 @@ int main(int argc, char **argv) {
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break;
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}
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frame->pts = frame_count;
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if(window_pixmap.texture_width < record_width)
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frame->width = window_pixmap.texture_width & ~1;
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else
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@ -1311,15 +1328,24 @@ int main(int argc, char **argv) {
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//fprintf(stderr, "Frame start index: %zu\n", start_index);
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int64_t pos = 0;
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int64_t pts_offset = 0;
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if(frames_erased)
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int64_t dts_offset = 0;
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if(frames_erased) {
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pos = frame_data_queue[start_index]->pos;
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pts_offset = frame_data_queue[start_index]->pts;
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dts_offset = frame_data_queue[start_index]->dts;
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}
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for(size_t i = start_index; i < frame_data_queue.size(); ++i) {
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AVPacket *av_packet = frame_data_queue[i];
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if(av_packet->stream_index == video_stream->index) {
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av_packet->pos = -1;
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if(av_packet->stream_index == audio_stream->index) {
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av_packet->pts = AV_NOPTS_VALUE;
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av_packet->dts = AV_NOPTS_VALUE;
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} else {
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av_packet->pts -= pts_offset;
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//av_packet->pos -= pos;
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av_packet->dts = AV_NOPTS_VALUE;
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}
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av_packet->pos = -1;
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