#include <stdint.h>
#include <stdio.h>
#include <stdlib.h>
-#include <sys/time.h>
-#include <time.h>
+#include <sys/resource.h>
#include <algorithm>
-#include <cmath>
+#include <chrono>
#include <condition_variable>
#include <cstddef>
+#include <cstdint>
#include <memory>
#include <mutex>
+#include <ratio>
#include <string>
#include <thread>
#include <utility>
#include <vector>
-#include <arpa/inet.h>
+#include "DeckLinkAPI.h"
+#include "LinuxCOM.h"
+#include "alsa_output.h"
#include "bmusb/bmusb.h"
+#include "bmusb/fake_capture.h"
#include "context.h"
#include "decklink_capture.h"
#include "defs.h"
+#include "disk_space_estimator.h"
#include "flags.h"
-#include "h264encode.h"
+#include "input_mapping.h"
#include "pbo_frame_allocator.h"
#include "ref_counted_gl_sync.h"
+#include "resampling_queue.h"
#include "timebase.h"
+#include "video_encoder.h"
+class IDeckLink;
class QOpenGLContext;
using namespace movit;
using namespace std;
+using namespace std::chrono;
using namespace std::placeholders;
+using namespace bmusb;
Mixer *global_mixer = nullptr;
+bool uses_mlock = false;
namespace {
-void convert_fixed24_to_fp32(float *dst, size_t out_channels, const uint8_t *src, size_t in_channels, size_t num_samples)
-{
- assert(in_channels >= out_channels);
- for (size_t i = 0; i < num_samples; ++i) {
- for (size_t j = 0; j < out_channels; ++j) {
- uint32_t s1 = *src++;
- uint32_t s2 = *src++;
- uint32_t s3 = *src++;
- uint32_t s = s1 | (s1 << 8) | (s2 << 16) | (s3 << 24);
- dst[i * out_channels + j] = int(s) * (1.0f / 4294967296.0f);
- }
- src += 3 * (in_channels - out_channels);
- }
-}
-
-void convert_fixed32_to_fp32(float *dst, size_t out_channels, const uint8_t *src, size_t in_channels, size_t num_samples)
-{
- assert(in_channels >= out_channels);
- for (size_t i = 0; i < num_samples; ++i) {
- for (size_t j = 0; j < out_channels; ++j) {
- // Note: Assumes little-endian.
- int32_t s = *(int32_t *)src;
- dst[i * out_channels + j] = s * (1.0f / 4294967296.0f);
- src += 4;
- }
- src += 4 * (in_channels - out_channels);
- }
-}
-
void insert_new_frame(RefCountedFrame frame, unsigned field_num, bool interlaced, unsigned card_index, InputState *input_state)
{
if (interlaced) {
}
}
-string generate_local_dump_filename(int frame)
-{
- time_t now = time(NULL);
- tm now_tm;
- localtime_r(&now, &now_tm);
-
- char timestamp[256];
- strftime(timestamp, sizeof(timestamp), "%F-%T%z", &now_tm);
-
- // Use the frame number to disambiguate between two cuts starting
- // on the same second.
- char filename[256];
- snprintf(filename, sizeof(filename), "%s%s-f%02d%s",
- LOCAL_DUMP_PREFIX, timestamp, frame % 100, LOCAL_DUMP_SUFFIX);
- return filename;
-}
-
} // namespace
void QueueLengthPolicy::update_policy(int queue_length)
if (queue_length >= int(safe_queue_length)) {
been_at_safe_point_since_last_starvation = true;
}
- if (++frames_with_at_least_one >= 50 && safe_queue_length > 0) {
+ if (++frames_with_at_least_one >= 1000 && safe_queue_length > 0) {
--safe_queue_length;
- fprintf(stderr, "Card %u: Spare frames for more than 50 frames, reducing safe limit to %u frames\n",
+ fprintf(stderr, "Card %u: Spare frames for more than 1000 frames, reducing safe limit to %u frames\n",
card_index, safe_queue_length);
frames_with_at_least_one = 0;
}
}
Mixer::Mixer(const QSurfaceFormat &format, unsigned num_cards)
- : httpd(WIDTH, HEIGHT),
+ : httpd(),
num_cards(num_cards),
mixer_surface(create_surface(format)),
h264_encoder_surface(create_surface(format)),
- correlation(OUTPUT_FREQUENCY),
- level_compressor(OUTPUT_FREQUENCY),
- limiter(OUTPUT_FREQUENCY),
- compressor(OUTPUT_FREQUENCY)
+ audio_mixer(num_cards)
{
- httpd.open_output_file(generate_local_dump_filename(/*frame=*/0).c_str());
- httpd.start(9095);
-
CHECK(init_movit(MOVIT_SHADER_DIR, MOVIT_DEBUG_OFF));
check_error();
movit_texel_subpixel_precision /= 2.0;
resource_pool.reset(new ResourcePool);
- theme.reset(new Theme(global_flags.theme_filename.c_str(), resource_pool.get(), num_cards));
+ theme.reset(new Theme(global_flags.theme_filename, global_flags.theme_dirs, resource_pool.get(), num_cards));
for (unsigned i = 0; i < NUM_OUTPUTS; ++i) {
output_channel[i].parent = this;
+ output_channel[i].channel = i;
}
ImageFormat inout_format;
display_chain->set_dither_bits(0); // Don't bother.
display_chain->finalize();
- h264_encoder.reset(new H264Encoder(h264_encoder_surface, global_flags.va_display, WIDTH, HEIGHT, &httpd));
+ video_encoder.reset(new VideoEncoder(resource_pool.get(), h264_encoder_surface, global_flags.va_display, WIDTH, HEIGHT, &httpd, global_disk_space_estimator));
- // First try initializing the PCI devices, then USB, until we have the desired number of cards.
- unsigned num_pci_devices = 0, num_usb_devices = 0;
+ // Start listening for clients only once VideoEncoder has written its header, if any.
+ httpd.start(9095);
+
+ // First try initializing the then PCI devices, then USB, then
+ // fill up with fake cards until we have the desired number of cards.
+ unsigned num_pci_devices = 0;
unsigned card_index = 0;
- IDeckLinkIterator *decklink_iterator = CreateDeckLinkIteratorInstance();
- if (decklink_iterator != nullptr) {
- for ( ; card_index < num_cards; ++card_index) {
- IDeckLink *decklink;
- if (decklink_iterator->Next(&decklink) != S_OK) {
- break;
+ {
+ IDeckLinkIterator *decklink_iterator = CreateDeckLinkIteratorInstance();
+ if (decklink_iterator != nullptr) {
+ for ( ; card_index < num_cards; ++card_index) {
+ IDeckLink *decklink;
+ if (decklink_iterator->Next(&decklink) != S_OK) {
+ break;
+ }
+
+ configure_card(card_index, new DeckLinkCapture(decklink, card_index), /*is_fake_capture=*/false);
+ ++num_pci_devices;
}
-
- configure_card(card_index, format, new DeckLinkCapture(decklink, card_index));
- ++num_pci_devices;
+ decklink_iterator->Release();
+ fprintf(stderr, "Found %u DeckLink PCI card(s).\n", num_pci_devices);
+ } else {
+ fprintf(stderr, "DeckLink drivers not found. Probing for USB cards only.\n");
}
- decklink_iterator->Release();
- fprintf(stderr, "Found %d DeckLink PCI card(s).\n", num_pci_devices);
- } else {
- fprintf(stderr, "DeckLink drivers not found. Probing for USB cards only.\n");
}
- for ( ; card_index < num_cards; ++card_index) {
- configure_card(card_index, format, new BMUSBCapture(card_index - num_pci_devices));
- ++num_usb_devices;
+ unsigned num_usb_devices = BMUSBCapture::num_cards();
+ for (unsigned usb_card_index = 0; usb_card_index < num_usb_devices && card_index < num_cards; ++usb_card_index, ++card_index) {
+ BMUSBCapture *capture = new BMUSBCapture(usb_card_index);
+ capture->set_card_disconnected_callback(bind(&Mixer::bm_hotplug_remove, this, card_index));
+ configure_card(card_index, capture, /*is_fake_capture=*/false);
+ }
+ fprintf(stderr, "Found %u USB card(s).\n", num_usb_devices);
+
+ unsigned num_fake_cards = 0;
+ for ( ; card_index < num_cards; ++card_index, ++num_fake_cards) {
+ FakeCapture *capture = new FakeCapture(WIDTH, HEIGHT, FAKE_FPS, OUTPUT_FREQUENCY, card_index, global_flags.fake_cards_audio);
+ configure_card(card_index, capture, /*is_fake_capture=*/true);
}
- if (num_usb_devices > 0) {
- BMUSBCapture::start_bm_thread();
+ if (num_fake_cards > 0) {
+ fprintf(stderr, "Initialized %u fake cards.\n", num_fake_cards);
}
+ BMUSBCapture::set_card_connected_callback(bind(&Mixer::bm_hotplug_add, this, _1));
+ BMUSBCapture::start_bm_thread();
+
for (card_index = 0; card_index < num_cards; ++card_index) {
cards[card_index].queue_length_policy.reset(card_index);
cards[card_index].capture->start_bm_capture();
cbcr_position_attribute_index = glGetAttribLocation(cbcr_program_num, "position");
cbcr_texcoord_attribute_index = glGetAttribLocation(cbcr_program_num, "texcoord");
- r128.init(2, OUTPUT_FREQUENCY);
- r128.integr_start();
-
- locut.init(FILTER_HPF, 2);
-
- // If --flat-audio is given, turn off everything that messes with the sound,
- // except the final makeup gain.
- if (global_flags.flat_audio) {
- set_locut_enabled(false);
- set_limiter_enabled(false);
- set_compressor_enabled(false);
+ if (global_flags.enable_alsa_output) {
+ alsa.reset(new ALSAOutput(OUTPUT_FREQUENCY, /*num_channels=*/2));
}
-
- // hlen=16 is pretty low quality, but we use quite a bit of CPU otherwise,
- // and there's a limit to how important the peak meter is.
- peak_resampler.setup(OUTPUT_FREQUENCY, OUTPUT_FREQUENCY * 4, /*num_channels=*/2, /*hlen=*/16, /*frel=*/1.0);
-
- alsa.reset(new ALSAOutput(OUTPUT_FREQUENCY, /*num_channels=*/2));
}
Mixer::~Mixer()
cards[card_index].capture->stop_dequeue_thread();
}
- h264_encoder.reset(nullptr);
+ video_encoder.reset(nullptr);
}
-void Mixer::configure_card(unsigned card_index, const QSurfaceFormat &format, CaptureInterface *capture)
+void Mixer::configure_card(unsigned card_index, CaptureInterface *capture, bool is_fake_capture)
{
printf("Configuring card %d...\n", card_index);
CaptureCard *card = &cards[card_index];
+ if (card->capture != nullptr) {
+ card->capture->stop_dequeue_thread();
+ delete card->capture;
+ }
card->capture = capture;
+ card->is_fake_capture = is_fake_capture;
card->capture->set_frame_callback(bind(&Mixer::bm_frame, this, card_index, _1, _2, _3, _4, _5, _6, _7));
- card->frame_allocator.reset(new PBOFrameAllocator(8 << 20, WIDTH, HEIGHT)); // 8 MB.
+ if (card->frame_allocator == nullptr) {
+ card->frame_allocator.reset(new PBOFrameAllocator(8 << 20, WIDTH, HEIGHT)); // 8 MB.
+ }
card->capture->set_video_frame_allocator(card->frame_allocator.get());
- card->surface = create_surface(format);
- card->capture->set_dequeue_thread_callbacks(
- [card]{
- eglBindAPI(EGL_OPENGL_API);
- card->context = create_context(card->surface);
- if (!make_current(card->context, card->surface)) {
- printf("failed to create bmusb context\n");
- exit(1);
- }
- },
- [this]{
- resource_pool->clean_context();
- });
- card->resampling_queue.reset(new ResamplingQueue(OUTPUT_FREQUENCY, OUTPUT_FREQUENCY, 2));
+ if (card->surface == nullptr) {
+ card->surface = create_surface_with_same_format(mixer_surface);
+ }
+ while (!card->new_frames.empty()) card->new_frames.pop();
+ card->fractional_samples = 0;
+ card->last_timecode = -1;
card->capture->configure_card();
+
+ DeviceSpec device{InputSourceType::CAPTURE_CARD, card_index};
+ audio_mixer.reset_resampler(device);
+ audio_mixer.set_display_name(device, card->capture->get_description());
+ audio_mixer.trigger_state_changed_callback();
}
}
}
-float find_peak(const float *samples, size_t num_samples)
-{
- float m = fabs(samples[0]);
- for (size_t i = 1; i < num_samples; ++i) {
- m = max(m, fabs(samples[i]));
- }
- return m;
-}
-
-void deinterleave_samples(const vector<float> &in, vector<float> *out_l, vector<float> *out_r)
-{
- size_t num_samples = in.size() / 2;
- out_l->resize(num_samples);
- out_r->resize(num_samples);
-
- const float *inptr = in.data();
- float *lptr = &(*out_l)[0];
- float *rptr = &(*out_r)[0];
- for (size_t i = 0; i < num_samples; ++i) {
- *lptr++ = *inptr++;
- *rptr++ = *inptr++;
- }
-}
-
} // namespace
void Mixer::bm_frame(unsigned card_index, uint16_t timecode,
FrameAllocator::Frame video_frame, size_t video_offset, VideoFormat video_format,
FrameAllocator::Frame audio_frame, size_t audio_offset, AudioFormat audio_format)
{
+ DeviceSpec device{InputSourceType::CAPTURE_CARD, card_index};
CaptureCard *card = &cards[card_index];
if (is_mode_scanning[card_index]) {
is_mode_scanning[card_index] = false;
} else {
static constexpr double switch_time_s = 0.5; // Should be enough time for the signal to stabilize.
- timespec now;
- clock_gettime(CLOCK_MONOTONIC, &now);
- double sec_since_last_switch = (now.tv_sec - last_mode_scan_change[card_index].tv_sec) +
- 1e-9 * (now.tv_nsec - last_mode_scan_change[card_index].tv_nsec);
+ steady_clock::time_point now = steady_clock::now();
+ double sec_since_last_switch = duration<double>(steady_clock::now() - last_mode_scan_change[card_index]).count();
if (sec_since_last_switch > switch_time_s) {
// It isn't this mode; try the next one.
mode_scanlist_index[card_index]++;
}
}
- int64_t frame_length = int64_t(TIMEBASE * video_format.frame_rate_den) / video_format.frame_rate_nom;
+ int64_t frame_length = int64_t(TIMEBASE) * video_format.frame_rate_den / video_format.frame_rate_nom;
+ assert(frame_length > 0);
size_t num_samples = (audio_frame.len > audio_offset) ? (audio_frame.len - audio_offset) / audio_format.num_channels / (audio_format.bits_per_sample / 8) : 0;
if (num_samples > OUTPUT_FREQUENCY / 10) {
return;
}
- int64_t local_pts = card->next_local_pts;
int dropped_frames = 0;
if (card->last_timecode != -1) {
dropped_frames = unwrap_timecode(timecode, card->last_timecode) - card->last_timecode - 1;
}
- // Convert the audio to stereo fp32 and add it.
- vector<float> audio;
- audio.resize(num_samples * 2);
- switch (audio_format.bits_per_sample) {
- case 0:
- assert(num_samples == 0);
- break;
- case 24:
- convert_fixed24_to_fp32(&audio[0], 2, audio_frame.data + audio_offset, audio_format.num_channels, num_samples);
- break;
- case 32:
- convert_fixed32_to_fp32(&audio[0], 2, audio_frame.data + audio_offset, audio_format.num_channels, num_samples);
- break;
- default:
- fprintf(stderr, "Cannot handle audio with %u bits per sample\n", audio_format.bits_per_sample);
- assert(false);
- }
-
- // Add the audio.
- {
- unique_lock<mutex> lock(card->audio_mutex);
-
- // Number of samples per frame if we need to insert silence.
- // (Could be nonintegral, but resampling will save us then.)
- int silence_samples = OUTPUT_FREQUENCY * video_format.frame_rate_den / video_format.frame_rate_nom;
-
- if (dropped_frames > MAX_FPS * 2) {
- fprintf(stderr, "Card %d lost more than two seconds (or time code jumping around; from 0x%04x to 0x%04x), resetting resampler\n",
- card_index, card->last_timecode, timecode);
- card->resampling_queue.reset(new ResamplingQueue(OUTPUT_FREQUENCY, OUTPUT_FREQUENCY, 2));
- dropped_frames = 0;
- } else if (dropped_frames > 0) {
- // Insert silence as needed.
- fprintf(stderr, "Card %d dropped %d frame(s) (before timecode 0x%04x), inserting silence.\n",
- card_index, dropped_frames, timecode);
- vector<float> silence(silence_samples * 2, 0.0f);
- for (int i = 0; i < dropped_frames; ++i) {
- card->resampling_queue->add_input_samples(local_pts / double(TIMEBASE), silence.data(), silence_samples);
- // Note that if the format changed in the meantime, we have
- // no way of detecting that; we just have to assume the frame length
- // is always the same.
- local_pts += frame_length;
- }
- }
- if (num_samples == 0) {
- audio.resize(silence_samples * 2);
- num_samples = silence_samples;
- }
- card->resampling_queue->add_input_samples(local_pts / double(TIMEBASE), audio.data(), num_samples);
- card->next_local_pts = local_pts + frame_length;
+ // Number of samples per frame if we need to insert silence.
+ // (Could be nonintegral, but resampling will save us then.)
+ const int silence_samples = OUTPUT_FREQUENCY * video_format.frame_rate_den / video_format.frame_rate_nom;
+
+ if (dropped_frames > MAX_FPS * 2) {
+ fprintf(stderr, "Card %d lost more than two seconds (or time code jumping around; from 0x%04x to 0x%04x), resetting resampler\n",
+ card_index, card->last_timecode, timecode);
+ audio_mixer.reset_resampler(device);
+ dropped_frames = 0;
+ } else if (dropped_frames > 0) {
+ // Insert silence as needed.
+ fprintf(stderr, "Card %d dropped %d frame(s) (before timecode 0x%04x), inserting silence.\n",
+ card_index, dropped_frames, timecode);
+
+ bool success;
+ do {
+ success = audio_mixer.add_silence(device, silence_samples, dropped_frames, frame_length);
+ } while (!success);
}
- card->last_timecode = timecode;
+ audio_mixer.add_audio(device, audio_frame.data + audio_offset, num_samples, audio_format, frame_length);
// Done with the audio, so release it.
if (audio_frame.owner) {
audio_frame.owner->release_frame(audio_frame);
}
+ card->last_timecode = timecode;
+
size_t expected_length = video_format.width * (video_format.height + video_format.extra_lines_top + video_format.extra_lines_bottom) * 2;
if (video_frame.len - video_offset == 0 ||
video_frame.len - video_offset != expected_length) {
PBOFrameAllocator::Userdata *userdata = (PBOFrameAllocator::Userdata *)video_frame.userdata;
unsigned num_fields = video_format.interlaced ? 2 : 1;
- timespec frame_upload_start;
+ steady_clock::time_point frame_upload_start;
if (video_format.interlaced) {
// Send the two fields along as separate frames; the other side will need to add
// a deinterlacer to actually get this right.
assert(frame_length % 2 == 0);
frame_length /= 2;
num_fields = 2;
- clock_gettime(CLOCK_MONOTONIC, &frame_upload_start);
+ frame_upload_start = steady_clock::now();
}
userdata->last_interlaced = video_format.interlaced;
userdata->last_has_signal = video_format.has_signal;
+ userdata->last_is_connected = video_format.is_connected;
userdata->last_frame_rate_nom = video_format.frame_rate_nom;
userdata->last_frame_rate_den = video_format.frame_rate_den;
RefCountedFrame frame(video_frame);
size_t y_offset = video_frame.size / 2 + video_offset / 2;
for (unsigned field = 0; field < num_fields; ++field) {
- unsigned field_start_line = (field == 1) ? video_format.second_field_start : video_format.extra_lines_top + field * (video_format.height + 22);
-
- if (userdata->tex_y[field] == 0 ||
- userdata->tex_cbcr[field] == 0 ||
- video_format.width != userdata->last_width[field] ||
- video_format.height != userdata->last_height[field]) {
- // We changed resolution since last use of this texture, so we need to create
- // a new object. Note that this each card has its own PBOFrameAllocator,
- // we don't need to worry about these flip-flopping between resolutions.
+ // Put the actual texture upload in a lambda that is executed in the main thread.
+ // It is entirely possible to do this in the same thread (and it might even be
+ // faster, depending on the GPU and driver), but it appears to be trickling
+ // driver bugs very easily.
+ //
+ // Note that this means we must hold on to the actual frame data in <userdata>
+ // until the upload command is run, but we hold on to <frame> much longer than that
+ // (in fact, all the way until we no longer use the texture in rendering).
+ auto upload_func = [field, video_format, y_offset, cbcr_offset, cbcr_width, userdata]() {
+ unsigned field_start_line = (field == 1) ? video_format.second_field_start : video_format.extra_lines_top + field * (video_format.height + 22);
+
+ if (userdata->tex_y[field] == 0 ||
+ userdata->tex_cbcr[field] == 0 ||
+ video_format.width != userdata->last_width[field] ||
+ video_format.height != userdata->last_height[field]) {
+ // We changed resolution since last use of this texture, so we need to create
+ // a new object. Note that this each card has its own PBOFrameAllocator,
+ // we don't need to worry about these flip-flopping between resolutions.
+ glBindTexture(GL_TEXTURE_2D, userdata->tex_cbcr[field]);
+ check_error();
+ glTexImage2D(GL_TEXTURE_2D, 0, GL_RG8, cbcr_width, video_format.height, 0, GL_RG, GL_UNSIGNED_BYTE, nullptr);
+ check_error();
+ glBindTexture(GL_TEXTURE_2D, userdata->tex_y[field]);
+ check_error();
+ glTexImage2D(GL_TEXTURE_2D, 0, GL_R8, video_format.width, video_format.height, 0, GL_RED, GL_UNSIGNED_BYTE, nullptr);
+ check_error();
+ userdata->last_width[field] = video_format.width;
+ userdata->last_height[field] = video_format.height;
+ }
+
+ GLuint pbo = userdata->pbo;
+ check_error();
+ glBindBuffer(GL_PIXEL_UNPACK_BUFFER, pbo);
+ check_error();
+
+ size_t field_y_start = y_offset + video_format.width * field_start_line;
+ size_t field_cbcr_start = cbcr_offset + cbcr_width * field_start_line * sizeof(uint16_t);
+
+ if (global_flags.flush_pbos) {
+ glFlushMappedBufferRange(GL_PIXEL_UNPACK_BUFFER, field_y_start, video_format.width * video_format.height);
+ check_error();
+ glFlushMappedBufferRange(GL_PIXEL_UNPACK_BUFFER, field_cbcr_start, cbcr_width * video_format.height * sizeof(uint16_t));
+ check_error();
+ }
+
glBindTexture(GL_TEXTURE_2D, userdata->tex_cbcr[field]);
check_error();
- glTexImage2D(GL_TEXTURE_2D, 0, GL_RG8, cbcr_width, video_format.height, 0, GL_RG, GL_UNSIGNED_BYTE, nullptr);
+ glTexSubImage2D(GL_TEXTURE_2D, 0, 0, 0, cbcr_width, video_format.height, GL_RG, GL_UNSIGNED_BYTE, BUFFER_OFFSET(field_cbcr_start));
check_error();
glBindTexture(GL_TEXTURE_2D, userdata->tex_y[field]);
check_error();
- glTexImage2D(GL_TEXTURE_2D, 0, GL_R8, video_format.width, video_format.height, 0, GL_RED, GL_UNSIGNED_BYTE, nullptr);
+ glTexSubImage2D(GL_TEXTURE_2D, 0, 0, 0, video_format.width, video_format.height, GL_RED, GL_UNSIGNED_BYTE, BUFFER_OFFSET(field_y_start));
check_error();
- userdata->last_width[field] = video_format.width;
- userdata->last_height[field] = video_format.height;
- }
-
- GLuint pbo = userdata->pbo;
- check_error();
- glBindBuffer(GL_PIXEL_UNPACK_BUFFER, pbo);
- check_error();
- glFlushMappedBufferRange(GL_PIXEL_UNPACK_BUFFER, 0, video_frame.size);
- check_error();
-
- glBindTexture(GL_TEXTURE_2D, userdata->tex_cbcr[field]);
- check_error();
- glTexSubImage2D(GL_TEXTURE_2D, 0, 0, 0, cbcr_width, video_format.height, GL_RG, GL_UNSIGNED_BYTE, BUFFER_OFFSET(cbcr_offset + cbcr_width * field_start_line * sizeof(uint16_t)));
- check_error();
- glBindTexture(GL_TEXTURE_2D, userdata->tex_y[field]);
- check_error();
- glTexSubImage2D(GL_TEXTURE_2D, 0, 0, 0, video_format.width, video_format.height, GL_RED, GL_UNSIGNED_BYTE, BUFFER_OFFSET(y_offset + video_format.width * field_start_line));
- check_error();
- glBindTexture(GL_TEXTURE_2D, 0);
- check_error();
- glBindBuffer(GL_PIXEL_UNPACK_BUFFER, 0);
- check_error();
- RefCountedGLsync fence(GL_SYNC_GPU_COMMANDS_COMPLETE, /*flags=*/0);
- check_error();
- assert(fence.get() != nullptr);
+ glBindTexture(GL_TEXTURE_2D, 0);
+ check_error();
+ glBindBuffer(GL_PIXEL_UNPACK_BUFFER, 0);
+ check_error();
+ };
if (field == 1) {
// Don't upload the second field as fast as we can; wait until
// against the video display, although the latter is not as critical.)
// This requires our system clock to be reasonably close to the
// video clock, but that's not an unreasonable assumption.
- timespec second_field_start;
- second_field_start.tv_nsec = frame_upload_start.tv_nsec +
- frame_length * 1000000000 / TIMEBASE;
- second_field_start.tv_sec = frame_upload_start.tv_sec +
- second_field_start.tv_nsec / 1000000000;
- second_field_start.tv_nsec %= 1000000000;
-
- while (clock_nanosleep(CLOCK_MONOTONIC, TIMER_ABSTIME,
- &second_field_start, nullptr) == -1 &&
- errno == EINTR) ;
+ steady_clock::time_point second_field_start = frame_upload_start +
+ nanoseconds(frame_length * 1000000000 / TIMEBASE);
+ this_thread::sleep_until(second_field_start);
}
{
new_frame.length = frame_length;
new_frame.field = field;
new_frame.interlaced = video_format.interlaced;
- new_frame.ready_fence = fence;
+ new_frame.upload_func = upload_func;
new_frame.dropped_frames = dropped_frames;
card->new_frames.push(move(new_frame));
card->new_frames_changed.notify_all();
}
}
+void Mixer::bm_hotplug_add(libusb_device *dev)
+{
+ lock_guard<mutex> lock(hotplug_mutex);
+ hotplugged_cards.push_back(dev);
+}
+
+void Mixer::bm_hotplug_remove(unsigned card_index)
+{
+ cards[card_index].new_frames_changed.notify_all();
+}
+
void Mixer::thread_func()
{
eglBindAPI(EGL_OPENGL_API);
exit(1);
}
- struct timespec start, now;
- clock_gettime(CLOCK_MONOTONIC, &start);
+ steady_clock::time_point start, now;
+ start = steady_clock::now();
int frame = 0;
int stats_dropped_frames = 0;
while (!should_quit) {
- CaptureCard::NewFrame new_frames[MAX_CARDS];
- bool has_new_frame[MAX_CARDS] = { false };
- int num_samples[MAX_CARDS] = { 0 };
+ CaptureCard::NewFrame new_frames[MAX_VIDEO_CARDS];
+ bool has_new_frame[MAX_VIDEO_CARDS] = { false };
+ int num_samples[MAX_VIDEO_CARDS] = { 0 };
- // TODO: Add a timeout.
unsigned master_card_index = theme->map_signal(master_clock_channel);
assert(master_card_index < num_cards);
get_one_frame_from_each_card(master_card_index, new_frames, has_new_frame, num_samples);
schedule_audio_resampling_tasks(new_frames[master_card_index].dropped_frames, num_samples[master_card_index], new_frames[master_card_index].length);
stats_dropped_frames += new_frames[master_card_index].dropped_frames;
- send_audio_level_callback();
+
+ handle_hotplugged_cards();
for (unsigned card_index = 0; card_index < num_cards; ++card_index) {
if (card_index == master_card_index || !has_new_frame[card_index]) {
insert_new_frame(new_frame->frame, new_frame->field, new_frame->interlaced, card_index, &input_state);
check_error();
- // The new texture might still be uploaded,
- // tell the GPU to wait until it's there.
- if (new_frame->ready_fence) {
- glWaitSync(new_frame->ready_fence.get(), /*flags=*/0, GL_TIMEOUT_IGNORED);
- check_error();
- new_frame->ready_fence.reset();
- check_error();
+ // The new texture might need uploading before use.
+ if (new_frame->upload_func) {
+ new_frame->upload_func();
+ new_frame->upload_func = nullptr;
}
}
- render_one_frame();
+ int64_t frame_duration = new_frames[master_card_index].length;
+ render_one_frame(frame_duration);
++frame;
- pts_int += new_frames[master_card_index].length;
+ pts_int += frame_duration;
- clock_gettime(CLOCK_MONOTONIC, &now);
- double elapsed = now.tv_sec - start.tv_sec +
- 1e-9 * (now.tv_nsec - start.tv_nsec);
+ now = steady_clock::now();
+ double elapsed = duration<double>(now - start).count();
if (frame % 100 == 0) {
- printf("%d frames (%d dropped) in %.3f seconds = %.1f fps (%.1f ms/frame)\n",
+ printf("%d frames (%d dropped) in %.3f seconds = %.1f fps (%.1f ms/frame)",
frame, stats_dropped_frames, elapsed, frame / elapsed,
1e3 * elapsed / frame);
// chain->print_phase_timing();
+
+ // Check our memory usage, to see if we are close to our mlockall()
+ // limit (if at all set).
+ rusage used;
+ if (getrusage(RUSAGE_SELF, &used) == -1) {
+ perror("getrusage(RUSAGE_SELF)");
+ assert(false);
+ }
+
+ if (uses_mlock) {
+ rlimit limit;
+ if (getrlimit(RLIMIT_MEMLOCK, &limit) == -1) {
+ perror("getrlimit(RLIMIT_MEMLOCK)");
+ assert(false);
+ }
+
+ printf(", using %ld / %ld MB lockable memory (%.1f%%)",
+ long(used.ru_maxrss / 1024),
+ long(limit.rlim_cur / 1048576),
+ float(100.0 * (used.ru_maxrss * 1024.0) / limit.rlim_cur));
+ } else {
+ printf(", using %ld MB memory (not locked)",
+ long(used.ru_maxrss / 1024));
+ }
+
+ printf("\n");
}
+
if (should_cut.exchange(false)) { // Test and clear.
- string filename = generate_local_dump_filename(frame);
- printf("Starting new recording: %s\n", filename.c_str());
- h264_encoder->shutdown();
- httpd.close_output_file();
- httpd.open_output_file(filename.c_str());
- h264_encoder.reset(new H264Encoder(h264_encoder_surface, global_flags.va_display, WIDTH, HEIGHT, &httpd));
+ video_encoder->do_cut(frame);
}
#if 0
resource_pool->clean_context();
}
-void Mixer::get_one_frame_from_each_card(unsigned master_card_index, CaptureCard::NewFrame new_frames[MAX_CARDS], bool has_new_frame[MAX_CARDS], int num_samples[MAX_CARDS])
+void Mixer::get_one_frame_from_each_card(unsigned master_card_index, CaptureCard::NewFrame new_frames[MAX_VIDEO_CARDS], bool has_new_frame[MAX_VIDEO_CARDS], int num_samples[MAX_VIDEO_CARDS])
{
+start:
// The first card is the master timer, so wait for it to have a new frame.
+ // TODO: Add a timeout.
unique_lock<mutex> lock(bmusb_mutex);
- cards[master_card_index].new_frames_changed.wait(lock, [this, master_card_index]{ return !cards[master_card_index].new_frames.empty(); });
+ cards[master_card_index].new_frames_changed.wait(lock, [this, master_card_index]{ return !cards[master_card_index].new_frames.empty() || cards[master_card_index].capture->get_disconnected(); });
+
+ if (cards[master_card_index].new_frames.empty()) {
+ // We were woken up, but not due to a new frame. Deal with it
+ // and then restart.
+ assert(cards[master_card_index].capture->get_disconnected());
+ handle_hotplugged_cards();
+ goto start;
+ }
for (unsigned card_index = 0; card_index < num_cards; ++card_index) {
CaptureCard *card = &cards[card_index];
}
}
+void Mixer::handle_hotplugged_cards()
+{
+ // Check for cards that have been disconnected since last frame.
+ for (unsigned card_index = 0; card_index < num_cards; ++card_index) {
+ CaptureCard *card = &cards[card_index];
+ if (card->capture->get_disconnected()) {
+ fprintf(stderr, "Card %u went away, replacing with a fake card.\n", card_index);
+ FakeCapture *capture = new FakeCapture(WIDTH, HEIGHT, FAKE_FPS, OUTPUT_FREQUENCY, card_index, global_flags.fake_cards_audio);
+ configure_card(card_index, capture, /*is_fake_capture=*/true);
+ card->queue_length_policy.reset(card_index);
+ card->capture->start_bm_capture();
+ }
+ }
+
+ // Check for cards that have been connected since last frame.
+ vector<libusb_device *> hotplugged_cards_copy;
+ {
+ lock_guard<mutex> lock(hotplug_mutex);
+ swap(hotplugged_cards, hotplugged_cards_copy);
+ }
+ for (libusb_device *new_dev : hotplugged_cards_copy) {
+ // Look for a fake capture card where we can stick this in.
+ int free_card_index = -1;
+ for (unsigned card_index = 0; card_index < num_cards; ++card_index) {
+ if (cards[card_index].is_fake_capture) {
+ free_card_index = int(card_index);
+ break;
+ }
+ }
+
+ if (free_card_index == -1) {
+ fprintf(stderr, "New card plugged in, but no free slots -- ignoring.\n");
+ libusb_unref_device(new_dev);
+ } else {
+ // BMUSBCapture takes ownership.
+ fprintf(stderr, "New card plugged in, choosing slot %d.\n", free_card_index);
+ CaptureCard *card = &cards[free_card_index];
+ BMUSBCapture *capture = new BMUSBCapture(free_card_index, new_dev);
+ configure_card(free_card_index, capture, /*is_fake_capture=*/false);
+ card->queue_length_policy.reset(free_card_index);
+ capture->set_card_disconnected_callback(bind(&Mixer::bm_hotplug_remove, this, free_card_index));
+ capture->start_bm_capture();
+ }
+ }
+}
+
+
void Mixer::schedule_audio_resampling_tasks(unsigned dropped_frames, int num_samples_per_frame, int length_per_frame)
{
// Resample the audio as needed, including from previously dropped frames.
assert(num_cards > 0);
for (unsigned frame_num = 0; frame_num < dropped_frames + 1; ++frame_num) {
+ const bool dropped_frame = (frame_num != dropped_frames);
{
// Signal to the audio thread to process this frame.
+ // Note that if the frame is a dropped frame, we signal that
+ // we don't want to use this frame as base for adjusting
+ // the resampler rate. The reason for this is that the timing
+ // of these frames is often way too late; they typically don't
+ // “arrive” before we synthesize them. Thus, we could end up
+ // in a situation where we have inserted e.g. five audio frames
+ // into the queue before we then start pulling five of them
+ // back out. This makes ResamplingQueue overestimate the delay,
+ // causing undue resampler changes. (We _do_ use the last,
+ // non-dropped frame; perhaps we should just discard that as well,
+ // since dropped frames are expected to be rare, and it might be
+ // better to just wait until we have a slightly more normal situation).
unique_lock<mutex> lock(audio_mutex);
- audio_task_queue.push(AudioTask{pts_int, num_samples_per_frame});
+ bool adjust_rate = !dropped_frame;
+ audio_task_queue.push(AudioTask{pts_int, num_samples_per_frame, adjust_rate});
audio_task_queue_changed.notify_one();
}
- if (frame_num != dropped_frames) {
+ if (dropped_frame) {
// For dropped frames, increase the pts. Note that if the format changed
// in the meantime, we have no way of detecting that; we just have to
// assume the frame length is always the same.
}
}
-void Mixer::render_one_frame()
+void Mixer::render_one_frame(int64_t duration)
{
// Get the main chain from the theme, and set its state immediately.
Theme::Chain theme_main_chain = theme->get_chain(0, pts(), WIDTH, HEIGHT, input_state);
//theme_main_chain.chain->enable_phase_timing(true);
GLuint y_tex, cbcr_tex;
- bool got_frame = h264_encoder->begin_frame(&y_tex, &cbcr_tex);
+ bool got_frame = video_encoder->begin_frame(&y_tex, &cbcr_tex);
assert(got_frame);
// Render main chain.
glTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_WRAP_S, GL_CLAMP_TO_EDGE);
glTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_WRAP_T, GL_CLAMP_TO_EDGE);
- RefCountedGLsync fence(GL_SYNC_GPU_COMMANDS_COMPLETE, /*flags=*/0);
- check_error();
-
const int64_t av_delay = TIMEBASE / 10; // Corresponds to the fixed delay in resampling_queue.h. TODO: Make less hard-coded.
- h264_encoder->end_frame(fence, pts_int + av_delay, theme_main_chain.input_frames);
+ RefCountedGLsync fence = video_encoder->end_frame(pts_int + av_delay, duration, theme_main_chain.input_frames);
// The live frame just shows the RGBA texture we just rendered.
// It owns rgba_tex now.
}
}
-void Mixer::send_audio_level_callback()
-{
- if (audio_level_callback == nullptr) {
- return;
- }
-
- unique_lock<mutex> lock(compressor_mutex);
- double loudness_s = r128.loudness_S();
- double loudness_i = r128.integrated();
- double loudness_range_low = r128.range_min();
- double loudness_range_high = r128.range_max();
-
- audio_level_callback(loudness_s, 20.0 * log10(peak),
- loudness_i, loudness_range_low, loudness_range_high,
- gain_staging_db, 20.0 * log10(final_makeup_gain),
- correlation.get_correlation());
-}
-
void Mixer::audio_thread_func()
{
while (!should_quit) {
{
unique_lock<mutex> lock(audio_mutex);
- audio_task_queue_changed.wait(lock, [this]{ return !audio_task_queue.empty(); });
+ audio_task_queue_changed.wait(lock, [this]{ return should_quit || !audio_task_queue.empty(); });
+ if (should_quit) {
+ return;
+ }
task = audio_task_queue.front();
audio_task_queue.pop();
}
- process_audio_one_frame(task.pts_int, task.num_samples);
- }
-}
-
-void Mixer::process_audio_one_frame(int64_t frame_pts_int, int num_samples)
-{
- vector<float> samples_card;
- vector<float> samples_out;
-
- // TODO: Allow mixing audio from several sources.
- unsigned selected_audio_card = theme->map_signal(audio_source_channel);
- assert(selected_audio_card < num_cards);
-
- for (unsigned card_index = 0; card_index < num_cards; ++card_index) {
- samples_card.resize(num_samples * 2);
- {
- unique_lock<mutex> lock(cards[card_index].audio_mutex);
- if (!cards[card_index].resampling_queue->get_output_samples(double(frame_pts_int) / TIMEBASE, &samples_card[0], num_samples)) {
- printf("Card %d reported previous underrun.\n", card_index);
- }
- }
- if (card_index == selected_audio_card) {
- samples_out = move(samples_card);
- }
- }
-
- // Cut away everything under 120 Hz (or whatever the cutoff is);
- // we don't need it for voice, and it will reduce headroom
- // and confuse the compressor. (In particular, any hums at 50 or 60 Hz
- // should be dampened.)
- if (locut_enabled) {
- locut.render(samples_out.data(), samples_out.size() / 2, locut_cutoff_hz * 2.0 * M_PI / OUTPUT_FREQUENCY, 0.5f);
- }
-
- // Apply a level compressor to get the general level right.
- // Basically, if it's over about -40 dBFS, we squeeze it down to that level
- // (or more precisely, near it, since we don't use infinite ratio),
- // then apply a makeup gain to get it to -14 dBFS. -14 dBFS is, of course,
- // entirely arbitrary, but from practical tests with speech, it seems to
- // put ut around -23 LUFS, so it's a reasonable starting point for later use.
- {
- unique_lock<mutex> lock(compressor_mutex);
- if (level_compressor_enabled) {
- float threshold = 0.01f; // -40 dBFS.
- float ratio = 20.0f;
- float attack_time = 0.5f;
- float release_time = 20.0f;
- float makeup_gain = pow(10.0f, (ref_level_dbfs - (-40.0f)) / 20.0f); // +26 dB.
- level_compressor.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
- gain_staging_db = 20.0 * log10(level_compressor.get_attenuation() * makeup_gain);
- } else {
- // Just apply the gain we already had.
- float g = pow(10.0f, gain_staging_db / 20.0f);
- for (size_t i = 0; i < samples_out.size(); ++i) {
- samples_out[i] *= g;
- }
- }
- }
-
-#if 0
- printf("level=%f (%+5.2f dBFS) attenuation=%f (%+5.2f dB) end_result=%+5.2f dB\n",
- level_compressor.get_level(), 20.0 * log10(level_compressor.get_level()),
- level_compressor.get_attenuation(), 20.0 * log10(level_compressor.get_attenuation()),
- 20.0 * log10(level_compressor.get_level() * level_compressor.get_attenuation() * makeup_gain));
-#endif
-
-// float limiter_att, compressor_att;
-
- // The real compressor.
- if (compressor_enabled) {
- float threshold = pow(10.0f, compressor_threshold_dbfs / 20.0f);
- float ratio = 20.0f;
- float attack_time = 0.005f;
- float release_time = 0.040f;
- float makeup_gain = 2.0f; // +6 dB.
- compressor.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
-// compressor_att = compressor.get_attenuation();
- }
-
- // Finally a limiter at -4 dB (so, -10 dBFS) to take out the worst peaks only.
- // Note that since ratio is not infinite, we could go slightly higher than this.
- if (limiter_enabled) {
- float threshold = pow(10.0f, limiter_threshold_dbfs / 20.0f);
- float ratio = 30.0f;
- float attack_time = 0.0f; // Instant.
- float release_time = 0.020f;
- float makeup_gain = 1.0f; // 0 dB.
- limiter.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
-// limiter_att = limiter.get_attenuation();
- }
-
-// printf("limiter=%+5.1f compressor=%+5.1f\n", 20.0*log10(limiter_att), 20.0*log10(compressor_att));
-
- // Upsample 4x to find interpolated peak.
- peak_resampler.inp_data = samples_out.data();
- peak_resampler.inp_count = samples_out.size() / 2;
-
- vector<float> interpolated_samples_out;
- interpolated_samples_out.resize(samples_out.size());
- while (peak_resampler.inp_count > 0) { // About four iterations.
- peak_resampler.out_data = &interpolated_samples_out[0];
- peak_resampler.out_count = interpolated_samples_out.size() / 2;
- peak_resampler.process();
- size_t out_stereo_samples = interpolated_samples_out.size() / 2 - peak_resampler.out_count;
- peak = max<float>(peak, find_peak(interpolated_samples_out.data(), out_stereo_samples * 2));
- peak_resampler.out_data = nullptr;
- }
-
- // At this point, we are most likely close to +0 LU, but all of our
- // measurements have been on raw sample values, not R128 values.
- // So we have a final makeup gain to get us to +0 LU; the gain
- // adjustments required should be relatively small, and also, the
- // offset shouldn't change much (only if the type of audio changes
- // significantly). Thus, we shoot for updating this value basically
- // “whenever we process buffers”, since the R128 calculation isn't exactly
- // something we get out per-sample.
- //
- // Note that there's a feedback loop here, so we choose a very slow filter
- // (half-time of 100 seconds).
- double target_loudness_factor, alpha;
- {
- unique_lock<mutex> lock(compressor_mutex);
- double loudness_lu = r128.loudness_M() - ref_level_lufs;
- double current_makeup_lu = 20.0f * log10(final_makeup_gain);
- target_loudness_factor = pow(10.0f, -loudness_lu / 20.0f);
-
- // If we're outside +/- 5 LU uncorrected, we don't count it as
- // a normal signal (probably silence) and don't change the
- // correction factor; just apply what we already have.
- if (fabs(loudness_lu - current_makeup_lu) >= 5.0 || !final_makeup_gain_auto) {
- alpha = 0.0;
- } else {
- // Formula adapted from
- // https://en.wikipedia.org/wiki/Low-pass_filter#Simple_infinite_impulse_response_filter.
- const double half_time_s = 100.0;
- const double fc_mul_2pi_delta_t = 1.0 / (half_time_s * OUTPUT_FREQUENCY);
- alpha = fc_mul_2pi_delta_t / (fc_mul_2pi_delta_t + 1.0);
- }
+ ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy =
+ task.adjust_rate ? ResamplingQueue::ADJUST_RATE : ResamplingQueue::DO_NOT_ADJUST_RATE;
+ vector<float> samples_out = audio_mixer.get_output(
+ double(task.pts_int) / TIMEBASE,
+ task.num_samples,
+ rate_adjustment_policy);
- double m = final_makeup_gain;
- for (size_t i = 0; i < samples_out.size(); i += 2) {
- samples_out[i + 0] *= m;
- samples_out[i + 1] *= m;
- m += (target_loudness_factor - m) * alpha;
+ // Send the samples to the sound card, then add them to the output.
+ if (alsa) {
+ alsa->write(samples_out);
}
- final_makeup_gain = m;
+ video_encoder->add_audio(task.pts_int, move(samples_out));
}
-
- // Find R128 levels and L/R correlation.
- vector<float> left, right;
- deinterleave_samples(samples_out, &left, &right);
- float *ptrs[] = { left.data(), right.data() };
- {
- unique_lock<mutex> lock(compressor_mutex);
- r128.process(left.size(), ptrs);
- correlation.process_samples(samples_out);
- }
-
- // Send the samples to the sound card.
- if (alsa) {
- alsa->write(samples_out);
- }
-
- // And finally add them to the output.
- h264_encoder->add_audio(frame_pts_int, move(samples_out));
}
void Mixer::subsample_chroma(GLuint src_tex, GLuint dst_tex)
void Mixer::quit()
{
should_quit = true;
+ audio_task_queue_changed.notify_one();
mixer_thread.join();
audio_thread.join();
}
theme->channel_clicked(preview_num);
}
-void Mixer::reset_meters()
-{
- peak_resampler.reset();
- peak = 0.0f;
- r128.reset();
- r128.integr_start();
- correlation.reset();
-}
-
void Mixer::start_mode_scanning(unsigned card_index)
{
assert(card_index < num_cards);
assert(!mode_scanlist[card_index].empty());
mode_scanlist_index[card_index] = 0;
cards[card_index].capture->set_video_mode(mode_scanlist[card_index][0]);
- clock_gettime(CLOCK_MONOTONIC, &last_mode_scan_change[card_index]);
+ last_mode_scan_change[card_index] = steady_clock::now();
}
Mixer::OutputChannel::~OutputChannel()
has_ready_frame = true;
}
- if (has_new_frame_ready_callback) {
+ if (new_frame_ready_callback) {
new_frame_ready_callback();
}
+
+ // Reduce the number of callbacks by filtering duplicates. The reason
+ // why we bother doing this is that Qt seemingly can get into a state
+ // where its builds up an essentially unbounded queue of signals,
+ // consuming more and more memory, and there's no good way of collapsing
+ // user-defined signals or limiting the length of the queue.
+ if (transition_names_updated_callback) {
+ vector<string> transition_names = global_mixer->get_transition_names();
+ bool changed = false;
+ if (transition_names.size() != last_transition_names.size()) {
+ changed = true;
+ } else {
+ for (unsigned i = 0; i < transition_names.size(); ++i) {
+ if (transition_names[i] != last_transition_names[i]) {
+ changed = true;
+ break;
+ }
+ }
+ }
+ if (changed) {
+ transition_names_updated_callback(transition_names);
+ last_transition_names = transition_names;
+ }
+ }
+ if (name_updated_callback) {
+ string name = global_mixer->get_channel_name(channel);
+ if (name != last_name) {
+ name_updated_callback(name);
+ last_name = name;
+ }
+ }
+ if (color_updated_callback) {
+ string color = global_mixer->get_channel_color(channel);
+ if (color != last_color) {
+ color_updated_callback(color);
+ last_color = color;
+ }
+ }
}
bool Mixer::OutputChannel::get_display_frame(DisplayFrame *frame)
void Mixer::OutputChannel::set_frame_ready_callback(Mixer::new_frame_ready_callback_t callback)
{
new_frame_ready_callback = callback;
- has_new_frame_ready_callback = true;
}
+
+void Mixer::OutputChannel::set_transition_names_updated_callback(Mixer::transition_names_updated_callback_t callback)
+{
+ transition_names_updated_callback = callback;
+}
+
+void Mixer::OutputChannel::set_name_updated_callback(Mixer::name_updated_callback_t callback)
+{
+ name_updated_callback = callback;
+}
+
+void Mixer::OutputChannel::set_color_updated_callback(Mixer::color_updated_callback_t callback)
+{
+ color_updated_callback = callback;
+}
+
+mutex RefCountedGLsync::fence_lock;
projects / nageru / blobdiff