Read audio data from pulseaudio as it's available and buffer it. Fixes audio recording on pulseaudio (and some pipewire configs)
This commit is contained in:
dec05eba
parent
5ba4c05953
commit
0059724fdc
@ -20,8 +20,6 @@
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typedef struct {
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void *handle;
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void *buffer;
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int buffer_size;
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unsigned int frames;
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} SoundDevice;
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15
src/main.cpp
15
src/main.cpp
@ -1470,10 +1470,16 @@ int main(int argc, char **argv) {
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int64_t pts = 0;
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const double target_audio_hz = 1.0 / (double)audio_track.codec_context->sample_rate;
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double received_audio_time = clock_get_monotonic_seconds();
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while(running) {
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void *sound_buffer;
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int sound_buffer_size = sound_device_read_next_chunk(&audio_track.sound_device, &sound_buffer);
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const bool got_audio_data = sound_buffer_size >= 0;
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const double this_audio_frame_time = clock_get_monotonic_seconds();
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if(got_audio_data)
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received_audio_time = this_audio_frame_time;
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int ret = av_frame_make_writable(audio_track.frame);
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if (ret < 0) {
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@ -1481,15 +1487,14 @@ int main(int argc, char **argv) {
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break;
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}
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const double this_audio_frame_time = clock_get_monotonic_seconds();
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const int64_t expected_frames = std::round((this_audio_frame_time - start_time_pts) / target_audio_hz);
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const int64_t num_missing_frames = std::max(0L, (expected_frames - pts) / audio_track.frame->nb_samples);
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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);
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// 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.
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// THIS PIECE OF SHIT WANTS EMPTY FRAMES OTHERWISE VIDEO PLAYS TOO FAST TO KEEP UP WITH AUDIO OR THE AUDIO PLAYS TOO EARLY.
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// BUT WE CANT USE DELAYS TO GIVE DUMMY DATA BECAUSE PULSEAUDIO MIGHT GIVE AUDIO A BIG DELAYED!!!
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if(num_missing_frames >= 5) {
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if(num_missing_frames >= 5 || (num_missing_frames > 0 && got_audio_data)) {
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// TODO:
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//audio_track.frame->data[0] = empty_audio;
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received_audio_time = this_audio_frame_time;
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swr_convert(swr, &audio_track.frame->data[0], audio_track.frame->nb_samples, (const uint8_t**)&empty_audio, audio_track.sound_device.frames);
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// TODO: Check if duplicate frame can be saved just by writing it with a different pts instead of sending it again
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for(int i = 0; i < num_missing_frames; ++i) {
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@ -1504,7 +1509,7 @@ int main(int argc, char **argv) {
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}
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}
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if(sound_buffer_size >= 0) {
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if(got_audio_data) {
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// TODO: Instead of converting audio, get float audio from alsa. Or does alsa do conversion internally to get this format?
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swr_convert(swr, &audio_track.frame->data[0], audio_track.frame->nb_samples, (const uint8_t**)&sound_buffer, audio_track.sound_device.frames);
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149
src/sound.cpp
149
src/sound.cpp
@ -21,14 +21,38 @@
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#include <stdio.h>
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#include <string.h>
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#include <cmath>
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#include <time.h>
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#include <pulse/pulseaudio.h>
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#include <pulse/mainloop.h>
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#include <pulse/xmalloc.h>
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#include <pulse/error.h>
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#define CHECK_DEAD_GOTO(p, rerror, label) \
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do { \
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if (!(p)->context || !PA_CONTEXT_IS_GOOD(pa_context_get_state((p)->context)) || \
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!(p)->stream || !PA_STREAM_IS_GOOD(pa_stream_get_state((p)->stream))) { \
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if (((p)->context && pa_context_get_state((p)->context) == PA_CONTEXT_FAILED) || \
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((p)->stream && pa_stream_get_state((p)->stream) == PA_STREAM_FAILED)) { \
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if (rerror) \
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*(rerror) = pa_context_errno((p)->context); \
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} else \
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if (rerror) \
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*(rerror) = PA_ERR_BADSTATE; \
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goto label; \
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} \
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} while(false);
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static int sound_device_index = 0;
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static double clock_get_monotonic_seconds() {
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struct timespec ts;
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ts.tv_sec = 0;
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ts.tv_nsec = 0;
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clock_gettime(CLOCK_MONOTONIC, &ts);
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return (double)ts.tv_sec + (double)ts.tv_nsec * 0.000000001;
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}
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struct pa_handle {
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pa_context *context;
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pa_stream *stream;
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@ -37,6 +61,9 @@ struct pa_handle {
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const void *read_data;
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size_t read_index, read_length;
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uint8_t *output_data;
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size_t output_index, output_length;
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int operation_success;
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};
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@ -54,6 +81,11 @@ static void pa_sound_device_free(pa_handle *s) {
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if (s->mainloop)
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pa_mainloop_free(s->mainloop);
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if (s->output_data) {
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free(s->output_data);
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s->output_data = NULL;
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}
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pa_xfree(s);
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}
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@ -68,6 +100,21 @@ static pa_handle* pa_sound_device_new(const char *server,
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int error = PA_ERR_INTERNAL, r;
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p = pa_xnew0(pa_handle, 1);
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p->read_data = NULL;
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p->read_length = 0;
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p->read_index = 0;
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const int buffer_size = attr->maxlength;
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void *buffer = malloc(buffer_size);
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if(!buffer) {
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fprintf(stderr, "failed to allocate buffer for audio\n");
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*rerror = -1;
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return NULL;
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}
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p->output_data = (uint8_t*)buffer;
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p->output_length = buffer_size;
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p->output_index = 0;
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if (!(p->mainloop = pa_mainloop_new()))
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goto fail;
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@ -130,30 +177,86 @@ fail:
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return NULL;
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}
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// Returns a negative value on failure or if no data is available at the moment
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static int pa_sound_device_read(pa_handle *p, void *data, size_t length) {
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// Returns a negative value on failure or if |p->output_length| data is not available within the time frame specified by the sample rate
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static int pa_sound_device_read(pa_handle *p) {
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assert(p);
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const int64_t timeout_ms = std::round((1000.0 / (double)pa_stream_get_sample_spec(p->stream)->rate) * 1000.0);
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pa_mainloop_prepare(p->mainloop, timeout_ms * 1000);
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const double start_time = clock_get_monotonic_seconds();
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bool success = false;
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int r = 0;
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int *rerror = &r;
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CHECK_DEAD_GOTO(p, rerror, fail);
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while (p->output_index < p->output_length) {
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if((clock_get_monotonic_seconds() - start_time) * 1000 >= timeout_ms)
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return -1;
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if(p->read_data) {
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assert(p->output_index == 0);
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memcpy(p->output_data, (const uint8_t*)p->read_data + p->read_index, p->read_length);
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p->output_index += p->read_length;
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p->read_data = NULL;
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p->read_length = 0;
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p->read_index = 0;
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if(pa_stream_drop(p->stream) != 0)
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goto fail;
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}
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pa_mainloop_prepare(p->mainloop, 1 * 1000); // 1 ms
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pa_mainloop_poll(p->mainloop);
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pa_mainloop_dispatch(p->mainloop);
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if(pa_stream_readable_size(p->stream) < length)
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return -1;
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if(pa_stream_peek(p->stream, &p->read_data, &p->read_length) < 0)
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goto fail;
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int r = pa_stream_peek(p->stream, &p->read_data, &p->read_length);
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if(r != 0)
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return -1;
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if(!p->read_data && p->read_length == 0)
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continue;
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if(p->read_length < length || !p->read_data) {
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pa_stream_drop(p->stream);
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return -1;
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if(!p->read_data && p->read_length > 0) {
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// There is a hole in the stream :( drop it. Maybe we should generate silence instead? TODO
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if(pa_stream_drop(p->stream) != 0)
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goto fail;
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continue;
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}
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memcpy(data, p->read_data, length);
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pa_stream_drop(p->stream);
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return 0;
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if(p->read_length <= 0) {
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CHECK_DEAD_GOTO(p, rerror, fail);
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continue;
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}
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const size_t space_free_in_output_buffer = p->output_length - p->output_index;
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if(space_free_in_output_buffer < p->read_length) {
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assert(p->read_index == 0);
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memcpy(p->output_data + p->output_index, p->read_data, space_free_in_output_buffer);
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p->output_index = 0;
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p->read_index += space_free_in_output_buffer;
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p->read_length -= space_free_in_output_buffer;
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break;
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} else {
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assert(p->read_index == 0);
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memcpy(p->output_data + p->output_index, p->read_data, p->read_length);
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p->output_index += p->read_length;
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p->read_data = NULL;
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p->read_length = 0;
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p->read_index = 0;
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if(pa_stream_drop(p->stream) != 0)
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goto fail;
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if(p->output_index == p->output_length) {
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p->output_index = 0;
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break;
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}
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}
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}
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success = true;
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fail:
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return success ? 0 : -1;
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}
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int sound_device_get_by_name(SoundDevice *device, const char *name, unsigned int num_channels, unsigned int period_frame_size) {
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@ -181,33 +284,21 @@ int sound_device_get_by_name(SoundDevice *device, const char *name, unsigned int
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return -1;
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}
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int buffer_size = buffer_attr.maxlength;
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void *buffer = malloc(buffer_size);
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if(!buffer) {
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fprintf(stderr, "failed to allocate buffer for audio\n");
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pa_sound_device_free(handle);
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return -1;
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}
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fprintf(stderr, "Using pulseaudio\n");
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device->handle = handle;
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device->buffer = buffer;
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device->buffer_size = buffer_size;
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device->frames = period_frame_size;
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return 0;
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}
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void sound_device_close(SoundDevice *device) {
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pa_sound_device_free((pa_handle*)device->handle);
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free(device->buffer);
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}
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int sound_device_read_next_chunk(SoundDevice *device, void **buffer) {
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if(pa_sound_device_read((pa_handle*)device->handle, device->buffer, device->buffer_size) < 0) {
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pa_handle *pa = (pa_handle*)device->handle;
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if(pa_sound_device_read(pa) < 0) {
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//fprintf(stderr, "pa_simple_read() failed: %s\n", pa_strerror(error));
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return -1;
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}
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*buffer = device->buffer;
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*buffer = pa->output_data;
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return device->frames;
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}
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