From 0059724fdc4a8eba89fdfa4d65ad72e3f8c75ff4 Mon Sep 17 00:00:00 2001
From: dec05eba <dec05eba@protonmail.com>
Date: Thu, 22 Sep 2022 01:44:45 +0200
Subject: Read audio data from pulseaudio as it's available and buffer it.
Fixes audio recording on pulseaudio (and some pipewire configs)
---
include/sound.hpp | 2 -
src/main.cpp | 15 ++++--
src/sound.cpp | 153 +++++++++++++++++++++++++++++++++++++++++++-----------
3 files changed, 132 insertions(+), 38 deletions(-)
diff --git a/include/sound.hpp b/include/sound.hpp
index 666d009..39d9248 100644
--- a/include/sound.hpp
+++ b/include/sound.hpp
@@ -20,8 +20,6 @@
typedef struct {
void *handle;
- void *buffer;
- int buffer_size;
unsigned int frames;
} SoundDevice;
diff --git a/src/main.cpp b/src/main.cpp
index 67552f4..7a17c84 100644
--- a/src/main.cpp
+++ b/src/main.cpp
@@ -1470,10 +1470,16 @@ 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();
while(running) {
void *sound_buffer;
int 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();
+ if(got_audio_data)
+ received_audio_time = this_audio_frame_time;
int ret = av_frame_make_writable(audio_track.frame);
if (ret < 0) {
@@ -1481,15 +1487,14 @@ int main(int argc, char **argv) {
break;
}
- 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);
+ const int64_t num_missing_frames = std::round((this_audio_frame_time - received_audio_time) / target_audio_hz / (int64_t)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) {
+ if(num_missing_frames >= 5 || (num_missing_frames > 0 && got_audio_data)) {
// 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.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) {
@@ -1504,7 +1509,7 @@ int main(int argc, char **argv) {
}
}
- if(sound_buffer_size >= 0) {
+ 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);
diff --git a/src/sound.cpp b/src/sound.cpp
index 928ee4a..9978e6f 100644
--- a/src/sound.cpp
+++ b/src/sound.cpp
@@ -21,14 +21,38 @@
#include <stdio.h>
#include <string.h>
#include <cmath>
+#include <time.h>
#include <pulse/pulseaudio.h>
#include <pulse/mainloop.h>
#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 int sound_device_index = 0;
+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;
+}
+
struct pa_handle {
pa_context *context;
pa_stream *stream;
@@ -37,6 +61,9 @@ struct pa_handle {
const void *read_data;
size_t read_index, read_length;
+ uint8_t *output_data;
+ size_t output_index, output_length;
+
int operation_success;
};
@@ -54,6 +81,11 @@ static void pa_sound_device_free(pa_handle *s) {
if (s->mainloop)
pa_mainloop_free(s->mainloop);
+ if (s->output_data) {
+ free(s->output_data);
+ s->output_data = NULL;
+ }
+
pa_xfree(s);
}
@@ -68,6 +100,21 @@ static pa_handle* pa_sound_device_new(const char *server,
int error = PA_ERR_INTERNAL, r;
p = pa_xnew0(pa_handle, 1);
+ p->read_data = NULL;
+ p->read_length = 0;
+ p->read_index = 0;
+
+ const int buffer_size = attr->maxlength;
+ void *buffer = malloc(buffer_size);
+ if(!buffer) {
+ fprintf(stderr, "failed to allocate buffer for audio\n");
+ *rerror = -1;
+ return NULL;
+ }
+
+ p->output_data = (uint8_t*)buffer;
+ p->output_length = buffer_size;
+ p->output_index = 0;
if (!(p->mainloop = pa_mainloop_new()))
goto fail;
@@ -130,30 +177,86 @@ fail:
return NULL;
}
-// 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) {
+// 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) {
assert(p);
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);
+ const double start_time = clock_get_monotonic_seconds();
+
+ bool success = false;
+ int r = 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;
+
+ if(p->read_data) {
+ assert(p->output_index == 0);
+ memcpy(p->output_data, (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(pa_stream_readable_size(p->stream) < length)
- return -1;
+ pa_mainloop_prepare(p->mainloop, 1 * 1000); // 1 ms
+ pa_mainloop_poll(p->mainloop);
+ pa_mainloop_dispatch(p->mainloop);
- int r = pa_stream_peek(p->stream, &p->read_data, &p->read_length);
- if(r != 0)
- return -1;
+ if(pa_stream_peek(p->stream, &p->read_data, &p->read_length) < 0)
+ goto fail;
- if(p->read_length < length || !p->read_data) {
- pa_stream_drop(p->stream);
- return -1;
+ if(!p->read_data && p->read_length == 0)
+ continue;
+
+ 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) {
+ 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) {
+ assert(p->read_index == 0);
+ memcpy(p->output_data + p->output_index, p->read_data, 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 {
+ assert(p->read_index == 0);
+ memcpy(p->output_data + p->output_index, p->read_data, 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(data, p->read_data, length);
- pa_stream_drop(p->stream);
- return 0;
+ success = true;
+
+ fail:
+ return success ? 0 : -1;
}
int sound_device_get_by_name(SoundDevice *device, const char *name, unsigned int num_channels, unsigned int period_frame_size) {
@@ -181,33 +284,21 @@ int sound_device_get_by_name(SoundDevice *device, const char *name, unsigned int
return -1;
}
- int buffer_size = buffer_attr.maxlength;
- void *buffer = malloc(buffer_size);
- if(!buffer) {
- fprintf(stderr, "failed to allocate buffer for audio\n");
- pa_sound_device_free(handle);
- return -1;
- }
-
- fprintf(stderr, "Using pulseaudio\n");
-
device->handle = handle;
- device->buffer = buffer;
- device->buffer_size = buffer_size;
device->frames = period_frame_size;
return 0;
}
void sound_device_close(SoundDevice *device) {
pa_sound_device_free((pa_handle*)device->handle);
- free(device->buffer);
}
int sound_device_read_next_chunk(SoundDevice *device, void **buffer) {
- if(pa_sound_device_read((pa_handle*)device->handle, device->buffer, device->buffer_size) < 0) {
+ pa_handle *pa = (pa_handle*)device->handle;
+ if(pa_sound_device_read(pa) < 0) {
//fprintf(stderr, "pa_simple_read() failed: %s\n", pa_strerror(error));
return -1;
}
- *buffer = device->buffer;
+ *buffer = pa->output_data;
return device->frames;
}
\ No newline at end of file
--
cgit v1.2.3-70-g09d2