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-rw-r--r--src/main.cpp131
-rw-r--r--src/sound.cpp88
2 files changed, 70 insertions, 149 deletions
diff --git a/src/main.cpp b/src/main.cpp
index faab93b..b3b206c 100644
--- a/src/main.cpp
+++ b/src/main.cpp
@@ -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;
}
@@ -2459,56 +2422,39 @@ 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");
- }
+ 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 {
- 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");
- }
+ 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);
+ 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?
@@ -2517,6 +2463,12 @@ int main(int argc, char **argv) {
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
@@ -2532,8 +2484,6 @@ int main(int argc, char **argv) {
fprintf(stderr, "Failed to encode audio!\n");
}
}
-
- audio_device.frame->pts += audio_track.codec_context->frame_size;
}
}
@@ -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;
}
}
}
diff --git a/src/sound.cpp b/src/sound.cpp
index c3aa4d4..99342f2 100644
--- a/src/sound.cpp
+++ b/src/sound.cpp
@@ -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(!p->read_data && p->read_length == 0)
- continue;
+ if(pa_stream_peek(p->stream, &p->read_data, &p->read_length) < 0)
+ goto fail;
- 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_data && p->read_length == 0)
+ continue;
- if(p->read_length <= 0) {
- p->read_data = NULL;
- if(pa_stream_drop(p->stream) != 0)
- goto fail;
+ // 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);
+ // }
- 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) {