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Time based audio latency, test, might fix some shits

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dec05eba
2024-04-11 14:40:27 +02:00
parent f8322c3c28
commit 52688dad72
3 changed files with 85 additions and 163 deletions
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5
include/sound.hpp
View File

@@ -24,6 +24,7 @@
typedef struct {
void *handle;
unsigned int frames;
double latency_seconds;
} SoundDevice;
struct AudioInput {
@@ -53,9 +54,9 @@ void sound_device_close(SoundDevice *device);
/*
Returns the next chunk of audio into @buffer.
Returns the number of frames read, or a negative value on failure.
Returns the number of bytes read, or a negative value on failure.
*/
int sound_device_read_next_chunk(SoundDevice *device, void **buffer);
int sound_device_read_next_chunk(SoundDevice *device, void **buffer, double timeout_sec);
std::vector<AudioInput> get_pulseaudio_inputs();

155
src/main.cpp
View File

@@ -315,7 +315,7 @@ static AVCodecContext* create_audio_codec_context(int fps, AudioCodec audio_code
#endif
codec_context->time_base.num = 1;
codec_context->time_base.den = codec_context->sample_rate;
codec_context->time_base.den = AV_TIME_BASE;
codec_context->framerate.num = fps;
codec_context->framerate.den = 1;
codec_context->thread_count = 1;
@@ -1699,10 +1699,10 @@ int main(int argc, char **argv) {
usage();
}
AudioCodec audio_codec = AudioCodec::OPUS;
AudioCodec audio_codec = AudioCodec::AAC;
const char *audio_codec_to_use = args["-ac"].value();
if(!audio_codec_to_use)
audio_codec_to_use = "opus";
audio_codec_to_use = "aac";
if(strcmp(audio_codec_to_use, "aac") == 0) {
audio_codec = AudioCodec::AAC;
@@ -1715,10 +1715,10 @@ 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;
if(audio_codec == AudioCodec::OPUS || audio_codec == AudioCodec::FLAC) {
fprintf(stderr, "Warning: opus and flac audio codecs has been temporary disabled, using aac audio codec instead\n");
audio_codec_to_use = "aac";
audio_codec = AudioCodec::AAC;
}
bool overclock = false;
@@ -2397,58 +2397,21 @@ int main(int argc, char **argv) {
swr_init(swr);
}
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);
// Remove this for now, it doesn't work well for everybody. The timing is different depending on system
#if 0
// 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.
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;
}
#endif
const int64_t no_input_sleep_ms = 500;
while(running) {
void *sound_buffer;
int sound_buffer_size = -1;
if(audio_device.sound_device.handle)
sound_buffer_size = sound_device_read_next_chunk(&audio_device.sound_device, &sound_buffer);
sound_buffer_size = sound_device_read_next_chunk(&audio_device.sound_device, &sound_buffer, 0.5);
const bool got_audio_data = sound_buffer_size >= 0;
const double this_audio_frame_time = clock_get_monotonic_seconds() - paused_time_offset;
if(paused) {
if(got_audio_data)
received_audio_time = this_audio_frame_time;
if(!audio_device.sound_device.handle)
usleep(timeout_ms * 1000);
usleep(no_input_sleep_ms * 1000);
continue;
}
@@ -2459,63 +2422,19 @@ int main(int argc, char **argv) {
break;
}
// TODO: Is this |received_audio_time| really correct?
int64_t num_missing_frames = std::round((this_audio_frame_time - received_audio_time) / target_audio_hz / (int64_t)audio_track.codec_context->frame_size);
if(got_audio_data)
num_missing_frames = std::max((int64_t)0, num_missing_frames - 1);
if(!audio_device.sound_device.handle)
num_missing_frames = std::max((int64_t)1, num_missing_frames);
if(got_audio_data)
received_audio_time = this_audio_frame_time;
// Fucking hell is there a better way to do this? I JUST WANT TO KEEP VIDEO AND AUDIO SYNCED HOLY FUCK I WANT TO KILL MYSELF NOW.
// THIS PIECE OF SHIT WANTS EMPTY FRAMES OTHERWISE VIDEO PLAYS TOO FAST TO KEEP UP WITH AUDIO OR THE AUDIO PLAYS TOO EARLY.
// BUT WE CANT USE DELAYS TO GIVE DUMMY DATA BECAUSE PULSEAUDIO MIGHT GIVE AUDIO A BIG DELAYED!!!
// This garbage is needed because we want to produce constant frame rate videos instead of variable frame rate
// videos because bad software such as video editing software and VLC do not support variable frame rate software,
// despite nvidia shadowplay and xbox game bar producing variable frame rate videos.
// So we have to make sure we produce frames at the same relative rate as the video.
if(num_missing_frames >= 5 || !audio_device.sound_device.handle) {
if(!got_audio_data) {
// TODO:
//audio_track.frame->data[0] = empty_audio;
received_audio_time = this_audio_frame_time;
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;
// TODO: Check if duplicate frame can be saved just by writing it with a different pts instead of sending it again
std::lock_guard<std::mutex> lock(audio_filter_mutex);
for(int i = 0; i < num_missing_frames; ++i) {
if(audio_track.graph) {
// 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 {
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;
}
}
if(!audio_device.sound_device.handle)
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?
if(needs_audio_conversion)
swr_convert(swr, &audio_device.frame->data[0], audio_track.codec_context->frame_size, (const uint8_t**)&sound_buffer, audio_track.codec_context->frame_size);
else
audio_device.frame->data[0] = (uint8_t*)sound_buffer;
const int64_t new_pts = (this_audio_frame_time - record_start_time) * AV_TIME_BASE;
if(new_pts == audio_device.frame->pts)
continue;
audio_device.frame->pts = new_pts;
//audio_device.frame->linesize[0] = sound_buffer_size / 2;
if(audio_track.graph) {
std::lock_guard<std::mutex> lock(audio_filter_mutex);
@@ -2532,8 +2451,39 @@ int main(int argc, char **argv) {
fprintf(stderr, "Failed to encode audio!\n");
}
}
}
audio_device.frame->pts += audio_track.codec_context->frame_size;
if(!audio_device.sound_device.handle)
usleep(no_input_sleep_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?
if(needs_audio_conversion)
swr_convert(swr, &audio_device.frame->data[0], audio_track.codec_context->frame_size, (const uint8_t**)&sound_buffer, audio_track.codec_context->frame_size);
else
audio_device.frame->data[0] = (uint8_t*)sound_buffer;
const int64_t new_pts = (this_audio_frame_time - record_start_time) * AV_TIME_BASE;
if(new_pts == audio_device.frame->pts)
continue;
audio_device.frame->pts = new_pts;
//audio_device.frame->linesize[0] = sound_buffer_size / 2;
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 {
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");
}
}
}
}
@@ -2571,7 +2521,11 @@ int main(int argc, char **argv) {
int err = 0;
while ((err = av_buffersink_get_frame(audio_track.sink, aframe)) >= 0) {
aframe->pts = audio_track.pts;
const int64_t new_pts = ((clock_get_monotonic_seconds() - paused_time_offset) - record_start_time) * AV_TIME_BASE;
if(new_pts == aframe->pts)
continue;
aframe->pts = new_pts;
//aframe->linesize[0] = sound_buffer_size / 2;
err = avcodec_send_frame(audio_track.codec_context, aframe);
if(err >= 0){
// TODO: Move to separate thread because this could write to network (for example when livestreaming)
@@ -2580,7 +2534,6 @@ int main(int argc, char **argv) {
fprintf(stderr, "Failed to encode audio!\n");
}
av_frame_unref(aframe);
audio_track.pts += audio_track.codec_context->frame_size;
}
}
}

88
src/sound.cpp
View File

@@ -41,6 +41,7 @@ struct pa_handle {
size_t output_index, output_length;
int operation_success;
double latency_seconds;
};
static void pa_sound_device_free(pa_handle *s) {
@@ -79,6 +80,7 @@ static pa_handle* pa_sound_device_new(const char *server,
p->read_data = NULL;
p->read_length = 0;
p->read_index = 0;
p->latency_seconds = 0;
const int buffer_size = attr->maxlength;
void *buffer = malloc(buffer_size);
@@ -153,78 +155,41 @@ fail:
return NULL;
}
// Returns a negative value on failure or if |p->output_length| data is not available within the time frame specified by the sample rate
static int pa_sound_device_read(pa_handle *p) {
static int pa_sound_device_read(pa_handle *p, double timeout_seconds) {
assert(p);
const int64_t timeout_ms = std::round((1000.0 / (double)pa_stream_get_sample_spec(p->stream)->rate) * 1000.0);
const double start_time = clock_get_monotonic_seconds();
bool success = false;
int r = 0;
//pa_usec_t latency = 0;
//int negative = 0;
int *rerror = &r;
CHECK_DEAD_GOTO(p, rerror, fail);
while (p->output_index < p->output_length) {
if((clock_get_monotonic_seconds() - start_time) * 1000 >= timeout_ms)
return -1;
while(clock_get_monotonic_seconds() - start_time < timeout_seconds) {
pa_mainloop_prepare(p->mainloop, 1 * 1000);
pa_mainloop_poll(p->mainloop);
pa_mainloop_dispatch(p->mainloop);
if(!p->read_data) {
pa_mainloop_prepare(p->mainloop, 1 * 1000); // 1 ms
pa_mainloop_poll(p->mainloop);
pa_mainloop_dispatch(p->mainloop);
if(pa_stream_peek(p->stream, &p->read_data, &p->read_length) < 0)
goto fail;
if(pa_stream_peek(p->stream, &p->read_data, &p->read_length) < 0)
goto fail;
if(!p->read_data && p->read_length == 0)
continue;
if(!p->read_data && p->read_length == 0)
continue;
// pa_operation_unref(pa_stream_update_timing_info(p->stream, NULL, NULL));
// if (pa_stream_get_latency(p->stream, &latency, &negative) >= 0) {
// fprintf(stderr, "latency: %lu ms, negative: %d, extra delay: %f ms\n", latency / 1000, negative, (clock_get_monotonic_seconds() - start_time) * 1000.0);
// }
if(!p->read_data && p->read_length > 0) {
// There is a hole in the stream :( drop it. Maybe we should generate silence instead? TODO
if(pa_stream_drop(p->stream) != 0)
goto fail;
continue;
}
if(p->read_length <= 0) {
p->read_data = NULL;
if(pa_stream_drop(p->stream) != 0)
goto fail;
CHECK_DEAD_GOTO(p, rerror, fail);
continue;
}
}
const size_t space_free_in_output_buffer = p->output_length - p->output_index;
if(space_free_in_output_buffer < p->read_length) {
memcpy(p->output_data + p->output_index, (const uint8_t*)p->read_data + p->read_index, space_free_in_output_buffer);
p->output_index = 0;
p->read_index += space_free_in_output_buffer;
p->read_length -= space_free_in_output_buffer;
break;
} else {
memcpy(p->output_data + p->output_index, (const uint8_t*)p->read_data + p->read_index, p->read_length);
p->output_index += p->read_length;
p->read_data = NULL;
p->read_length = 0;
p->read_index = 0;
if(pa_stream_drop(p->stream) != 0)
goto fail;
if(p->output_index == p->output_length) {
p->output_index = 0;
break;
}
}
memcpy(p->output_data, p->read_data, p->read_length);
pa_stream_drop(p->stream);
p->latency_seconds = clock_get_monotonic_seconds() - start_time;
return p->read_length;
}
success = true;
fail:
return success ? 0 : -1;
return -1;
}
static pa_sample_format_t audio_format_to_pulse_audio_format(AudioFormat audio_format) {
@@ -269,6 +234,7 @@ int sound_device_get_by_name(SoundDevice *device, const char *device_name, const
device->handle = handle;
device->frames = period_frame_size;
device->latency_seconds = 0.0;
return 0;
}
@@ -278,14 +244,16 @@ void sound_device_close(SoundDevice *device) {
device->handle = NULL;
}
int sound_device_read_next_chunk(SoundDevice *device, void **buffer) {
int sound_device_read_next_chunk(SoundDevice *device, void **buffer, double timeout_sec) {
pa_handle *pa = (pa_handle*)device->handle;
if(pa_sound_device_read(pa) < 0) {
int size = pa_sound_device_read(pa, timeout_sec);
if(size < 0) {
//fprintf(stderr, "pa_simple_read() failed: %s\n", pa_strerror(error));
return -1;
}
*buffer = pa->output_data;
return device->frames;
device->latency_seconds = pa->latency_seconds;
return size;
}
static void pa_state_cb(pa_context *c, void *userdata) {