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-rw-r--r--src/main.cpp45
1 files changed, 38 insertions, 7 deletions
diff --git a/src/main.cpp b/src/main.cpp
index 682b896..79fee50 100644
--- a/src/main.cpp
+++ b/src/main.cpp
@@ -645,7 +645,7 @@ static AVCodecContext *create_video_codec_context(AVFormatContext *av_format_con
codec_context->codec_tag = MKTAG('h', 'v', 'c', '1');
switch(video_quality) {
case VideoQuality::MEDIUM:
- codec_context->bit_rate = 5000000 + (codec_context->width * codec_context->height) / 2;
+ codec_context->bit_rate = 6000000 + (codec_context->width * codec_context->height) / 2;
/*
if(use_hevc) {
codec_context->qmin = 20;
@@ -796,7 +796,7 @@ static void open_video(AVCodecContext *codec_context,
AVDictionary *options = nullptr;
switch(video_quality) {
case VideoQuality::MEDIUM:
- av_dict_set_int(&options, "qp", 36, 0);
+ av_dict_set_int(&options, "qp", 35, 0);
//av_dict_set(&options, "preset", "hq", 0);
break;
case VideoQuality::HIGH:
@@ -815,7 +815,7 @@ static void open_video(AVCodecContext *codec_context,
if(is_livestream) {
av_dict_set_int(&options, "zerolatency", 1, 0);
- av_dict_set(&options, "preset", "llhq", 0);
+ //av_dict_set(&options, "preset", "llhq", 0);
}
av_opt_set(&options, "rc", "vbr", 0);
@@ -1432,6 +1432,12 @@ int main(int argc, char **argv) {
const AVOutputFormat *output_format = av_format_context->oformat;
const bool is_livestream = is_livestream_path(filename);
+ // (Some?) livestreaming services require at least one audio track to work.
+ // If not audio is provided then create one silent audio track.
+ if(is_livestream && requested_audio_inputs.empty()) {
+ fprintf(stderr, "Info: live streaming but no audio track was added. Adding a silent audio track\n");
+ requested_audio_inputs.push_back({ "", "gsr-silent" });
+ }
//bool use_hevc = strcmp(window_str, "screen") == 0 || strcmp(window_str, "screen-direct") == 0;
if(video_codec != VideoCodec::H264 && strcmp(container_format, "flv") == 0) {
@@ -1472,9 +1478,14 @@ int main(int argc, char **argv) {
const int num_channels = audio_codec_context->ch_layout.nb_channels;
#endif
- if(sound_device_get_by_name(&audio_tracks.back().sound_device, audio_input.name.c_str(), audio_input.description.c_str(), num_channels, audio_codec_context->frame_size) != 0) {
- fprintf(stderr, "failed to get 'pulse' sound device\n");
- exit(1);
+ if(audio_input.name.empty()) {
+ audio_tracks.back().sound_device.handle = NULL;
+ audio_tracks.back().sound_device.frames = 0;
+ } else {
+ if(sound_device_get_by_name(&audio_tracks.back().sound_device, audio_input.name.c_str(), audio_input.description.c_str(), num_channels, audio_codec_context->frame_size) != 0) {
+ fprintf(stderr, "failed to get 'pulse' sound device\n");
+ exit(1);
+ }
}
++audio_stream_index;
@@ -1609,10 +1620,13 @@ int main(int argc, char **argv) {
int64_t pts = 0;
const double target_audio_hz = 1.0 / (double)audio_track.codec_context->sample_rate;
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);
while(running) {
void *sound_buffer;
- int sound_buffer_size = sound_device_read_next_chunk(&audio_track.sound_device, &sound_buffer);
+ int sound_buffer_size = -1;
+ if(audio_track.sound_device.handle)
+ sound_buffer_size = sound_device_read_next_chunk(&audio_track.sound_device, &sound_buffer);
const bool got_audio_data = sound_buffer_size >= 0;
const double this_audio_frame_time = clock_get_monotonic_seconds();
@@ -1647,6 +1661,23 @@ int main(int argc, char **argv) {
}
}
+ if(!audio_track.sound_device.handle) {
+ // TODO:
+ //audio_track.frame->data[0] = empty_audio;
+ received_audio_time = this_audio_frame_time;
+ swr_convert(swr, &audio_track.frame->data[0], audio_track.frame->nb_samples, (const uint8_t**)&empty_audio, audio_track.codec_context->frame_size);
+ audio_track.frame->pts = pts;
+ pts += audio_track.frame->nb_samples;
+ ret = avcodec_send_frame(audio_track.codec_context, audio_track.frame);
+ if(ret >= 0){
+ receive_frames(audio_track.codec_context, audio_track.stream_index, audio_track.stream, audio_track.frame, av_format_context, record_start_time, frame_data_queue, replay_buffer_size_secs, frames_erased, *write_output_mutex);
+ } else {
+ fprintf(stderr, "Failed to encode audio!\n");
+ }
+
+ usleep(timeout_ms * 1000);
+ }
+
if(got_audio_data) {
// TODO: Instead of converting audio, get float audio from alsa. Or does alsa do conversion internally to get this format?
swr_convert(swr, &audio_track.frame->data[0], audio_track.frame->nb_samples, (const uint8_t**)&sound_buffer, audio_track.sound_device.frames);