From efc15622a9edf3eccaec0e31b28cc8cbef18c7d2 Mon Sep 17 00:00:00 2001 From: dec05eba Date: Wed, 23 Mar 2022 19:19:00 +0100 Subject: Attempt to fix audio timestamp being totally wrong on fedora --- src/main.cpp | 60 +++++++++++++++++++++++++++++++++++++++++++----------------- 1 file changed, 43 insertions(+), 17 deletions(-) (limited to 'src/main.cpp') diff --git a/src/main.cpp b/src/main.cpp index e624797..11bc57d 100644 --- a/src/main.cpp +++ b/src/main.cpp @@ -24,6 +24,7 @@ #include #include #include +#include #include #include @@ -45,6 +46,8 @@ extern "C" { #include #include #include +#include +#include } #include @@ -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 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; -- cgit v1.2.3