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-rw-r--r--src/main.cpp50
-rw-r--r--src/sound.cpp81
2 files changed, 53 insertions, 78 deletions
diff --git a/src/main.cpp b/src/main.cpp
index 1c02984..1b88cd9 100644
--- a/src/main.cpp
+++ b/src/main.cpp
@@ -483,7 +483,7 @@ static AVCodecContext* create_audio_codec_context(AVFormatContext *av_format_con
#endif
codec_context->time_base.num = 1;
- codec_context->time_base.den = AV_TIME_BASE;
+ codec_context->time_base.den = codec_context->sample_rate;
codec_context->framerate.num = fps;
codec_context->framerate.den = 1;
@@ -1462,11 +1462,12 @@ 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 pts = 0;
+ const double target_audio_hz = 1.0 / (double)audio_track.codec_context->sample_rate;
+
while(running) {
void *sound_buffer;
int sound_buffer_size = sound_device_read_next_chunk(&audio_track.sound_device, &sound_buffer);
- if(sound_buffer_size < 0)
- sound_buffer = empty_audio;
int ret = av_frame_make_writable(audio_track.frame);
if (ret < 0) {
@@ -1474,15 +1475,42 @@ int main(int argc, char **argv) {
break;
}
- // 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);
- audio_track.frame->pts = (clock_get_monotonic_seconds() - start_time_pts) * AV_TIME_BASE;
+ const double this_audio_frame_time = clock_get_monotonic_seconds();
+ const int64_t expected_frames = std::round((this_audio_frame_time - start_time_pts) / target_audio_hz);
+ const int64_t num_missing_frames = std::max(0L, (expected_frames - pts) / audio_track.frame->nb_samples);
+ // Jesus 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!!!
+ if(num_missing_frames >= 5) {
+ // TODO:
+ //audio_track.frame->data[0] = empty_audio;
+ swr_convert(swr, &audio_track.frame->data[0], audio_track.frame->nb_samples, (const uint8_t**)&empty_audio, audio_track.sound_device.frames);
+ // TODO: Check if duplicate frame can be saved just by writing it with a different pts instead of sending it again
+ for(int i = 0; i < num_missing_frames; ++i) {
+ 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");
+ }
+ }
+ }
+
+ if(sound_buffer_size >= 0) {
+ // 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);
- 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");
+ 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");
+ }
}
}
diff --git a/src/sound.cpp b/src/sound.cpp
index ab0450e..928ee4a 100644
--- a/src/sound.cpp
+++ b/src/sound.cpp
@@ -20,7 +20,6 @@
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
-#include <time.h>
#include <cmath>
#include <pulse/pulseaudio.h>
@@ -28,29 +27,6 @@
#include <pulse/xmalloc.h>
#include <pulse/error.h>
-#define CHECK_DEAD_GOTO(p, rerror, label) \
- do { \
- if (!(p)->context || !PA_CONTEXT_IS_GOOD(pa_context_get_state((p)->context)) || \
- !(p)->stream || !PA_STREAM_IS_GOOD(pa_stream_get_state((p)->stream))) { \
- if (((p)->context && pa_context_get_state((p)->context) == PA_CONTEXT_FAILED) || \
- ((p)->stream && pa_stream_get_state((p)->stream) == PA_STREAM_FAILED)) { \
- if (rerror) \
- *(rerror) = pa_context_errno((p)->context); \
- } else \
- if (rerror) \
- *(rerror) = PA_ERR_BADSTATE; \
- goto label; \
- } \
- } while(false);
-
-static double clock_get_monotonic_seconds() {
- struct timespec ts;
- ts.tv_sec = 0;
- ts.tv_nsec = 0;
- clock_gettime(CLOCK_MONOTONIC, &ts);
- return (double)ts.tv_sec + (double)ts.tv_nsec * 0.000000001;
-}
-
static int sound_device_index = 0;
struct pa_handle {
@@ -154,59 +130,30 @@ fail:
return NULL;
}
-static void pa_sound_device_mainloop_timed(pa_handle *p, int64_t timeout_ms) {
- const double start_time = clock_get_monotonic_seconds();
- while((clock_get_monotonic_seconds() - start_time) * 1000.0 < timeout_ms) {
- pa_mainloop_prepare(p->mainloop, 1 * 1000);
- pa_mainloop_poll(p->mainloop);
- pa_mainloop_dispatch(p->mainloop);
- }
-}
-
-// Returns a negative value on failure. Always blocks a time specified matching the sampling rate of the audio.
+// Returns a negative value on failure or if no data is available at the moment
static int pa_sound_device_read(pa_handle *p, void *data, size_t length) {
assert(p);
- int r = 0;
- int *rerror = &r;
- bool retry = true;
-
- pa_mainloop_iterate(p->mainloop, 0, NULL);
const int64_t timeout_ms = std::round((1000.0 / (double)pa_stream_get_sample_spec(p->stream)->rate) * 1000.0);
+ pa_mainloop_prepare(p->mainloop, timeout_ms * 1000);
+ pa_mainloop_poll(p->mainloop);
+ pa_mainloop_dispatch(p->mainloop);
- CHECK_DEAD_GOTO(p, rerror, fail);
-
- while(true) {
- if(pa_stream_readable_size(p->stream) < length) {
- if(!retry)
- break;
-
- retry = false;
- pa_sound_device_mainloop_timed(p, timeout_ms);
- continue;
- }
-
- r = pa_stream_peek(p->stream, &p->read_data, &p->read_length);
- if(r != 0) {
- if(retry)
- pa_sound_device_mainloop_timed(p, timeout_ms);
- return -1;
- }
+ if(pa_stream_readable_size(p->stream) < length)
+ return -1;
- if(p->read_length < length || !p->read_data) {
- pa_stream_drop(p->stream);
- if(retry)
- pa_sound_device_mainloop_timed(p, timeout_ms);
- return -1;
- }
+ int r = pa_stream_peek(p->stream, &p->read_data, &p->read_length);
+ if(r != 0)
+ return -1;
- memcpy(data, p->read_data, length);
+ if(p->read_length < length || !p->read_data) {
pa_stream_drop(p->stream);
- return 0;
+ return -1;
}
- fail:
- return -1;
+ memcpy(data, p->read_data, length);
+ pa_stream_drop(p->stream);
+ return 0;
}
int sound_device_get_by_name(SoundDevice *device, const char *name, unsigned int num_channels, unsigned int period_frame_size) {