#include "mixer.h"
#include <assert.h>
+#include <endian.h>
#include <epoxy/egl.h>
-#include <init.h>
#include <movit/effect_chain.h>
#include <movit/effect_util.h>
#include <movit/flat_input.h>
#include <movit/image_format.h>
+#include <movit/init.h>
#include <movit/resource_pool.h>
#include <movit/util.h>
#include <stdint.h>
#include <stdio.h>
#include <stdlib.h>
+#include <sys/resource.h>
#include <sys/time.h>
#include <time.h>
#include <algorithm>
+#include <chrono>
#include <cmath>
#include <condition_variable>
#include <cstddef>
#include <vector>
#include "bmusb/bmusb.h"
+#include "bmusb/fake_capture.h"
#include "context.h"
+#include "db.h"
+#include "decklink_capture.h"
#include "defs.h"
-#include "h264encode.h"
+#include "disk_space_estimator.h"
+#include "flags.h"
#include "pbo_frame_allocator.h"
#include "ref_counted_gl_sync.h"
#include "timebase.h"
+#include "video_encoder.h"
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)
-{
- 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 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);
+} // namespace
- // 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;
+void QueueLengthPolicy::update_policy(int queue_length)
+{
+ if (queue_length < 0) { // Starvation.
+ if (been_at_safe_point_since_last_starvation && safe_queue_length < 5) {
+ ++safe_queue_length;
+ fprintf(stderr, "Card %u: Starvation, increasing safe limit to %u frames\n",
+ card_index, safe_queue_length);
+ }
+ frames_with_at_least_one = 0;
+ been_at_safe_point_since_last_starvation = false;
+ return;
+ }
+ if (queue_length > 0) {
+ if (queue_length >= int(safe_queue_length)) {
+ been_at_safe_point_since_last_starvation = true;
+ }
+ if (++frames_with_at_least_one >= 1000 && safe_queue_length > 0) {
+ --safe_queue_length;
+ 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;
+ }
+ } else {
+ frames_with_at_least_one = 0;
+ }
}
-} // namespace
-
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),
+ correlation(OUTPUT_FREQUENCY)
{
- 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("theme.lua", 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, 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));
- for (unsigned card_index = 0; card_index < num_cards; ++card_index) {
- printf("Configuring card %d...\n", card_index);
- CaptureCard *card = &cards[card_index];
- card->usb = new BMUSBCapture(card_index);
- card->usb->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.
- card->usb->set_video_frame_allocator(card->frame_allocator.get());
- card->surface = create_surface(format);
- card->usb->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);
+ // 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;
}
- },
- [this]{
- resource_pool->clean_context();
- });
- card->resampling_queue.reset(new ResamplingQueue(OUTPUT_FREQUENCY, OUTPUT_FREQUENCY, 2));
- card->usb->configure_card();
+
+ configure_card(card_index, new DeckLinkCapture(decklink, card_index), /*is_fake_capture=*/false);
+ ++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");
+ }
+ }
+ 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_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 (unsigned card_index = 0; card_index < num_cards; ++card_index) {
- cards[card_index].usb->start_bm_capture();
+ 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();
}
// Set up stuff for NV12 conversion.
// Cb/Cr shader.
- string cbcr_vert_shader = read_file("vs-cbcr.130.vert");
+ string cbcr_vert_shader =
+ "#version 130 \n"
+ " \n"
+ "in vec2 position; \n"
+ "in vec2 texcoord; \n"
+ "out vec2 tc0; \n"
+ "uniform vec2 foo_chroma_offset_0; \n"
+ " \n"
+ "void main() \n"
+ "{ \n"
+ " // The result of glOrtho(0.0, 1.0, 0.0, 1.0, 0.0, 1.0) is: \n"
+ " // \n"
+ " // 2.000 0.000 0.000 -1.000 \n"
+ " // 0.000 2.000 0.000 -1.000 \n"
+ " // 0.000 0.000 -2.000 -1.000 \n"
+ " // 0.000 0.000 0.000 1.000 \n"
+ " gl_Position = vec4(2.0 * position.x - 1.0, 2.0 * position.y - 1.0, -1.0, 1.0); \n"
+ " vec2 flipped_tc = texcoord; \n"
+ " tc0 = flipped_tc + foo_chroma_offset_0; \n"
+ "} \n";
string cbcr_frag_shader =
"#version 130 \n"
"in vec2 tc0; \n"
"uniform sampler2D cbcr_tex; \n"
+ "out vec4 FragColor; \n"
"void main() { \n"
- " gl_FragColor = texture2D(cbcr_tex, tc0); \n"
+ " FragColor = texture(cbcr_tex, tc0); \n"
"} \n";
vector<string> frag_shader_outputs;
cbcr_program_num = resource_pool->compile_glsl_program(cbcr_vert_shader, cbcr_frag_shader, frag_shader_outputs);
+ float vertices[] = {
+ 0.0f, 2.0f,
+ 0.0f, 0.0f,
+ 2.0f, 0.0f
+ };
+ cbcr_vbo = generate_vbo(2, GL_FLOAT, sizeof(vertices), vertices);
+ 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);
-
// 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));
+ if (global_flags.enable_alsa_output) {
+ alsa.reset(new ALSAOutput(OUTPUT_FREQUENCY, /*num_channels=*/2));
+ }
}
Mixer::~Mixer()
{
resource_pool->release_glsl_program(cbcr_program_num);
+ glDeleteBuffers(1, &cbcr_vbo);
BMUSBCapture::stop_bm_thread();
for (unsigned card_index = 0; card_index < num_cards; ++card_index) {
{
unique_lock<mutex> lock(bmusb_mutex);
cards[card_index].should_quit = true; // Unblock thread.
- cards[card_index].new_data_ready_changed.notify_all();
+ cards[card_index].new_frames_changed.notify_all();
}
- cards[card_index].usb->stop_dequeue_thread();
+ cards[card_index].capture->stop_dequeue_thread();
}
- h264_encoder.reset(nullptr);
+ video_encoder.reset(nullptr);
}
+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));
+ 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());
+ 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_device(device);
+ audio_mixer.set_name(device, card->capture->get_description());
+}
+
+
namespace {
int unwrap_timecode(uint16_t current_wrapped, int last)
} // namespace
void Mixer::bm_frame(unsigned card_index, uint16_t timecode,
- FrameAllocator::Frame video_frame, size_t video_offset, uint16_t video_format,
- FrameAllocator::Frame audio_frame, size_t audio_offset, uint16_t audio_format)
+ 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];
- unsigned width, height, second_field_start, frame_rate_nom, frame_rate_den, extra_lines_top, extra_lines_bottom;
- bool interlaced;
+ if (is_mode_scanning[card_index]) {
+ if (video_format.has_signal) {
+ // Found a stable signal, so stop scanning.
+ is_mode_scanning[card_index] = false;
+ } else {
+ static constexpr double switch_time_s = 0.5; // Should be enough time for the signal to stabilize.
+ 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]++;
+ mode_scanlist_index[card_index] %= mode_scanlist[card_index].size();
+ cards[card_index].capture->set_video_mode(mode_scanlist[card_index][mode_scanlist_index[card_index]]);
+ last_mode_scan_change[card_index] = now;
+ }
+ }
+ }
- decode_video_format(video_format, &width, &height, &second_field_start, &extra_lines_top, &extra_lines_bottom,
- &frame_rate_nom, &frame_rate_den, &interlaced); // Ignore return value for now.
- int64_t frame_length = TIMEBASE * frame_rate_den / 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) / 8 / 3 : 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) {
printf("Card %d: Dropping frame with implausible audio length (len=%d, offset=%d) [timecode=0x%04x video_len=%d video_offset=%d video_format=%x)\n",
card_index, int(audio_frame.len), int(audio_offset),
- timecode, int(video_frame.len), int(video_offset), video_format);
+ timecode, int(video_frame.len), int(video_offset), video_format.id);
if (video_frame.owner) {
video_frame.owner->release_frame(video_frame);
}
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);
- convert_fixed24_to_fp32(&audio[0], 2, audio_frame.data + audio_offset, 8, num_samples);
-
- // 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 * frame_rate_den / 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_device(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);
+
+ audio_mixer.add_silence(device, silence_samples, dropped_frames, frame_length);
}
- 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);
}
- {
- // Wait until the previous frame was consumed.
- unique_lock<mutex> lock(bmusb_mutex);
- card->new_data_ready_changed.wait(lock, [card]{ return !card->new_data_ready || card->should_quit; });
- if (card->should_quit) return;
- }
+ card->last_timecode = timecode;
- size_t expected_length = width * (height + extra_lines_top + extra_lines_bottom) * 2;
+ 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) {
if (video_frame.len != 0) {
// so that pts can go up accordingly.
{
unique_lock<mutex> lock(bmusb_mutex);
- card->new_data_ready = true;
- card->new_frame = RefCountedFrame(FrameAllocator::Frame());
- card->new_frame_length = frame_length;
- card->new_frame_interlaced = false;
- card->new_data_ready_fence = nullptr;
- card->dropped_frames = dropped_frames;
- card->new_data_ready_changed.notify_all();
+ CaptureCard::NewFrame new_frame;
+ new_frame.frame = RefCountedFrame(FrameAllocator::Frame());
+ new_frame.length = frame_length;
+ new_frame.interlaced = false;
+ new_frame.dropped_frames = dropped_frames;
+ card->new_frames.push(move(new_frame));
+ card->new_frames_changed.notify_all();
}
return;
}
PBOFrameAllocator::Userdata *userdata = (PBOFrameAllocator::Userdata *)video_frame.userdata;
- unsigned num_fields = interlaced ? 2 : 1;
- timespec frame_upload_start;
- if (interlaced) {
+ unsigned num_fields = video_format.interlaced ? 2 : 1;
+ 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(height % 2 == 0);
- height /= 2;
+ assert(video_format.height % 2 == 0);
+ video_format.height /= 2;
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 = interlaced;
- userdata->last_frame_rate_nom = frame_rate_nom;
- userdata->last_frame_rate_den = frame_rate_den;
- RefCountedFrame new_frame(video_frame);
+ 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);
// Upload the textures.
- size_t cbcr_width = width / 2;
+ size_t cbcr_width = video_format.width / 2;
size_t cbcr_offset = video_offset / 2;
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) ? second_field_start : extra_lines_top + field * (height + 22);
-
- if (userdata->tex_y[field] == 0 ||
- userdata->tex_cbcr[field] == 0 ||
- width != userdata->last_width[field] ||
- 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, 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, width, 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] = width;
- userdata->last_height[field] = height;
- }
-
- GLuint pbo = userdata->pbo;
- check_error();
- glBindBuffer(GL_PIXEL_UNPACK_BUFFER_ARB, pbo);
- check_error();
- glMemoryBarrier(GL_CLIENT_MAPPED_BUFFER_BARRIER_BIT);
- check_error();
-
- glBindTexture(GL_TEXTURE_2D, userdata->tex_cbcr[field]);
- check_error();
- glTexSubImage2D(GL_TEXTURE_2D, 0, 0, 0, cbcr_width, 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, width, height, GL_RED, GL_UNSIGNED_BYTE, BUFFER_OFFSET(y_offset + width * field_start_line));
- check_error();
- glBindTexture(GL_TEXTURE_2D, 0);
- check_error();
- GLsync fence = glFenceSync(GL_SYNC_GPU_COMMANDS_COMPLETE, /*flags=*/0);
- check_error();
- assert(fence != 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);
}
{
unique_lock<mutex> lock(bmusb_mutex);
- card->new_data_ready = true;
- card->new_frame = new_frame;
- card->new_frame_length = frame_length;
- card->new_frame_field = field;
- card->new_frame_interlaced = interlaced;
- card->new_data_ready_fence = fence;
- card->dropped_frames = dropped_frames;
- card->new_data_ready_changed.notify_all();
-
- if (field != num_fields - 1) {
- // Wait until the previous frame was consumed.
- card->new_data_ready_changed.wait(lock, [card]{ return !card->new_data_ready || card->should_quit; });
- if (card->should_quit) return;
- }
+ CaptureCard::NewFrame new_frame;
+ new_frame.frame = frame;
+ new_frame.length = frame_length;
+ new_frame.field = field;
+ new_frame.interlaced = video_format.interlaced;
+ 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 card_copy[MAX_CARDS];
- int num_samples[MAX_CARDS];
+ CaptureCard::NewFrame new_frames[MAX_CARDS];
+ bool has_new_frame[MAX_CARDS] = { false };
+ int num_samples[MAX_CARDS] = { 0 };
- {
- unique_lock<mutex> lock(bmusb_mutex);
+ unsigned master_card_index = theme->map_signal(master_clock_channel);
+ assert(master_card_index < num_cards);
- // The first card is the master timer, so wait for it to have a new frame.
- // TODO: Make configurable, and with a timeout.
- cards[0].new_data_ready_changed.wait(lock, [this]{ return cards[0].new_data_ready; });
-
- for (unsigned card_index = 0; card_index < num_cards; ++card_index) {
- CaptureCard *card = &cards[card_index];
- card_copy[card_index].usb = card->usb;
- card_copy[card_index].new_data_ready = card->new_data_ready;
- card_copy[card_index].new_frame = card->new_frame;
- card_copy[card_index].new_frame_length = card->new_frame_length;
- card_copy[card_index].new_frame_field = card->new_frame_field;
- card_copy[card_index].new_frame_interlaced = card->new_frame_interlaced;
- card_copy[card_index].new_data_ready_fence = card->new_data_ready_fence;
- card_copy[card_index].dropped_frames = card->dropped_frames;
- card->new_data_ready = false;
- card->new_data_ready_changed.notify_all();
-
- int num_samples_times_timebase = OUTPUT_FREQUENCY * card->new_frame_length + card->fractional_samples;
- num_samples[card_index] = num_samples_times_timebase / TIMEBASE;
- card->fractional_samples = num_samples_times_timebase % TIMEBASE;
- assert(num_samples[card_index] >= 0);
- }
- }
-
- // Resample the audio as needed, including from previously dropped frames.
- for (unsigned frame_num = 0; frame_num < card_copy[0].dropped_frames + 1; ++frame_num) {
- {
- // Signal to the audio thread to process this frame.
- unique_lock<mutex> lock(audio_mutex);
- audio_task_queue.push(AudioTask{pts_int, num_samples[0]});
- audio_task_queue_changed.notify_one();
- }
- if (frame_num != card_copy[0].dropped_frames) {
- // 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.
- ++stats_dropped_frames;
- pts_int += card_copy[0].new_frame_length;
- }
- }
-
- if (audio_level_callback != nullptr) {
- 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();
+ 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();
- 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());
- }
+ handle_hotplugged_cards();
- for (unsigned card_index = 1; card_index < num_cards; ++card_index) {
- if (card_copy[card_index].new_data_ready && card_copy[card_index].new_frame->len == 0) {
- ++card_copy[card_index].dropped_frames;
+ for (unsigned card_index = 0; card_index < num_cards; ++card_index) {
+ if (card_index == master_card_index || !has_new_frame[card_index]) {
+ continue;
+ }
+ if (new_frames[card_index].frame->len == 0) {
+ ++new_frames[card_index].dropped_frames;
}
- if (card_copy[card_index].dropped_frames > 0) {
+ if (new_frames[card_index].dropped_frames > 0) {
printf("Card %u dropped %d frames before this\n",
- card_index, int(card_copy[card_index].dropped_frames));
+ card_index, int(new_frames[card_index].dropped_frames));
}
}
// If the first card is reporting a corrupted or otherwise dropped frame,
// just increase the pts (skipping over this frame) and don't try to compute anything new.
- if (card_copy[0].new_frame->len == 0) {
+ if (new_frames[master_card_index].frame->len == 0) {
++stats_dropped_frames;
- pts_int += card_copy[0].new_frame_length;
+ pts_int += new_frames[master_card_index].length;
continue;
}
for (unsigned card_index = 0; card_index < num_cards; ++card_index) {
- CaptureCard *card = &card_copy[card_index];
- if (!card->new_data_ready || card->new_frame->len == 0)
+ if (!has_new_frame[card_index] || new_frames[card_index].frame->len == 0)
continue;
- assert(card->new_frame != nullptr);
- insert_new_frame(card->new_frame, card->new_frame_field, card->new_frame_interlaced, card_index, &input_state);
+ CaptureCard::NewFrame *new_frame = &new_frames[card_index];
+ assert(new_frame->frame != nullptr);
+ 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 (card->new_data_ready_fence) {
- glWaitSync(card->new_data_ready_fence, /*flags=*/0, GL_TIMEOUT_IGNORED);
- check_error();
- glDeleteSync(card->new_data_ready_fence);
- check_error();
+ // The new texture might need uploading before use.
+ if (new_frame->upload_func) {
+ new_frame->upload_func();
+ new_frame->upload_func = nullptr;
}
}
- // 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);
- EffectChain *chain = theme_main_chain.chain;
- theme_main_chain.setup_chain();
- //theme_main_chain.chain->enable_phase_timing(true);
-
- GLuint y_tex, cbcr_tex;
- bool got_frame = h264_encoder->begin_frame(&y_tex, &cbcr_tex);
- assert(got_frame);
-
- // Render main chain.
- GLuint cbcr_full_tex = resource_pool->create_2d_texture(GL_RG8, WIDTH, HEIGHT);
- GLuint rgba_tex = resource_pool->create_2d_texture(GL_RGB565, WIDTH, HEIGHT); // Saves texture bandwidth, although dithering gets messed up.
- GLuint fbo = resource_pool->create_fbo(y_tex, cbcr_full_tex, rgba_tex);
- check_error();
- chain->render_to_fbo(fbo, WIDTH, HEIGHT);
- resource_pool->release_fbo(fbo);
-
- subsample_chroma(cbcr_full_tex, cbcr_tex);
- resource_pool->release_2d_texture(cbcr_full_tex);
-
- // Set the right state for rgba_tex.
- glBindFramebuffer(GL_FRAMEBUFFER, 0);
- glBindTexture(GL_TEXTURE_2D, rgba_tex);
- glTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_MIN_FILTER, GL_LINEAR);
- 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);
+ int64_t frame_duration = new_frames[master_card_index].length;
+ render_one_frame(frame_duration);
++frame;
- pts_int += card_copy[0].new_frame_length;
-
- // The live frame just shows the RGBA texture we just rendered.
- // It owns rgba_tex now.
- DisplayFrame live_frame;
- live_frame.chain = display_chain.get();
- live_frame.setup_chain = [this, rgba_tex]{
- display_input->set_texture_num(rgba_tex);
- };
- live_frame.ready_fence = fence;
- live_frame.input_frames = {};
- live_frame.temp_textures = { rgba_tex };
- output_channel[OUTPUT_LIVE].output_frame(live_frame);
-
- // Set up preview and any additional channels.
- for (int i = 1; i < theme->get_num_channels() + 2; ++i) {
- DisplayFrame display_frame;
- Theme::Chain chain = theme->get_chain(i, pts(), WIDTH, HEIGHT, input_state); // FIXME: dimensions
- display_frame.chain = chain.chain;
- display_frame.setup_chain = chain.setup_chain;
- display_frame.ready_fence = fence;
- display_frame.input_frames = chain.input_frames;
- display_frame.temp_textures = {};
- output_channel[i].output_frame(display_frame);
- }
-
- clock_gettime(CLOCK_MONOTONIC, &now);
- double elapsed = now.tv_sec - start.tv_sec +
- 1e-9 * (now.tv_nsec - start.tv_nsec);
+ pts_int += frame_duration;
+
+ 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, WIDTH, HEIGHT, &httpd));
+ video_encoder->do_cut(frame);
}
#if 0
resource_pool->clean_context();
}
-void Mixer::audio_thread_func()
+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])
{
- while (!should_quit) {
- AudioTask task;
+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].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;
+ }
- {
- unique_lock<mutex> lock(audio_mutex);
- audio_task_queue_changed.wait(lock, [this]{ return !audio_task_queue.empty(); });
- task = audio_task_queue.front();
- audio_task_queue.pop();
+ for (unsigned card_index = 0; card_index < num_cards; ++card_index) {
+ CaptureCard *card = &cards[card_index];
+ if (card->new_frames.empty()) {
+ assert(card_index != master_card_index);
+ card->queue_length_policy.update_policy(-1);
+ continue;
+ }
+ new_frames[card_index] = move(card->new_frames.front());
+ has_new_frame[card_index] = true;
+ card->new_frames.pop();
+ card->new_frames_changed.notify_all();
+
+ int num_samples_times_timebase = OUTPUT_FREQUENCY * new_frames[card_index].length + card->fractional_samples;
+ num_samples[card_index] = num_samples_times_timebase / TIMEBASE;
+ card->fractional_samples = num_samples_times_timebase % TIMEBASE;
+ assert(num_samples[card_index] >= 0);
+
+ if (card_index == master_card_index) {
+ // We don't use the queue length policy for the master card,
+ // but we will if it stops being the master. Thus, clear out
+ // the policy in case we switch in the future.
+ card->queue_length_policy.reset(card_index);
+ } else {
+ // If we have excess frames compared to the policy for this card,
+ // drop frames from the head.
+ card->queue_length_policy.update_policy(card->new_frames.size());
+ while (card->new_frames.size() > card->queue_length_policy.get_safe_queue_length()) {
+ card->new_frames.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)
+void Mixer::handle_hotplugged_cards()
{
- vector<float> samples_card;
- vector<float> samples_out;
+ // Check for cards that have been disconnected since last frame.
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);
- }
- }
- // TODO: Allow using audio from the other card(s) as well.
- if (card_index == 0) {
- samples_out = move(samples_card);
+ 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();
}
}
- // 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);
+ // 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;
+ }
+ }
- // 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);
+ if (free_card_index == -1) {
+ fprintf(stderr, "New card plugged in, but no free slots -- ignoring.\n");
+ libusb_unref_device(new_dev);
} 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;
- }
+ // 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();
}
}
+}
-#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;
+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);
+ 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 (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.
+ pts_int += length_per_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);
+ EffectChain *chain = theme_main_chain.chain;
+ theme_main_chain.setup_chain();
+ //theme_main_chain.chain->enable_phase_timing(true);
+
+ GLuint y_tex, cbcr_tex;
+ bool got_frame = video_encoder->begin_frame(&y_tex, &cbcr_tex);
+ assert(got_frame);
+
+ // Render main chain.
+ GLuint cbcr_full_tex = resource_pool->create_2d_texture(GL_RG8, WIDTH, HEIGHT);
+ GLuint rgba_tex = resource_pool->create_2d_texture(GL_RGB565, WIDTH, HEIGHT); // Saves texture bandwidth, although dithering gets messed up.
+ GLuint fbo = resource_pool->create_fbo(y_tex, cbcr_full_tex, rgba_tex);
+ check_error();
+ chain->render_to_fbo(fbo, WIDTH, HEIGHT);
+ resource_pool->release_fbo(fbo);
+
+ subsample_chroma(cbcr_full_tex, cbcr_tex);
+ resource_pool->release_2d_texture(cbcr_full_tex);
+
+ // Set the right state for rgba_tex.
+ glBindFramebuffer(GL_FRAMEBUFFER, 0);
+ glBindTexture(GL_TEXTURE_2D, rgba_tex);
+ glTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_MIN_FILTER, GL_LINEAR);
+ glTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_WRAP_S, GL_CLAMP_TO_EDGE);
+ glTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_WRAP_T, GL_CLAMP_TO_EDGE);
+
+ const int64_t av_delay = TIMEBASE / 10; // Corresponds to the fixed delay in resampling_queue.h. TODO: Make less hard-coded.
+ 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.
+ DisplayFrame live_frame;
+ live_frame.chain = display_chain.get();
+ live_frame.setup_chain = [this, rgba_tex]{
+ display_input->set_texture_num(rgba_tex);
+ };
+ live_frame.ready_fence = fence;
+ live_frame.input_frames = {};
+ live_frame.temp_textures = { rgba_tex };
+ output_channel[OUTPUT_LIVE].output_frame(live_frame);
+
+ // Set up preview and any additional channels.
+ for (int i = 1; i < theme->get_num_channels() + 2; ++i) {
+ DisplayFrame display_frame;
+ Theme::Chain chain = theme->get_chain(i, pts(), WIDTH, HEIGHT, input_state); // FIXME: dimensions
+ display_frame.chain = chain.chain;
+ display_frame.setup_chain = chain.setup_chain;
+ display_frame.ready_fence = fence;
+ display_frame.input_frames = chain.input_frames;
+ display_frame.temp_textures = {};
+ output_channel[i].output_frame(display_frame);
+ }
+}
- // 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();
+void Mixer::send_audio_level_callback()
+{
+ if (audio_level_callback == nullptr) {
+ return;
}
- // 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();
+ unique_lock<mutex> lock(audio_measure_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, to_db(peak),
+ loudness_i, loudness_range_low, loudness_range_high,
+ audio_mixer.get_gain_staging_db(),
+ audio_mixer.get_final_makeup_gain_db(),
+ correlation.get_correlation());
+}
+
+void Mixer::audio_thread_func()
+{
+ while (!should_quit) {
+ AudioTask task;
+
+ {
+ unique_lock<mutex> lock(audio_mutex);
+ 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();
+ }
+
+ ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy =
+ task.adjust_rate ? ResamplingQueue::ADJUST_RATE : ResamplingQueue::DO_NOT_ADJUST_RATE;
+ process_audio_one_frame(task.pts_int, task.num_samples, rate_adjustment_policy);
}
+}
-// printf("limiter=%+5.1f compressor=%+5.1f\n", 20.0*log10(limiter_att), 20.0*log10(compressor_att));
+void Mixer::process_audio_one_frame(int64_t frame_pts_int, int num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy)
+{
+ vector<float> samples_out = audio_mixer.get_output(double(frame_pts_int) / TIMEBASE, num_samples, rate_adjustment_policy);
// Upsample 4x to find interpolated peak.
peak_resampler.inp_data = samples_out.data();
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));
- }
-
- // 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);
+ unique_lock<mutex> lock(audio_measure_mutex);
+
+ 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;
}
-
- 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;
- }
- final_makeup_gain = m;
}
// Find R128 levels and L/R correlation.
deinterleave_samples(samples_out, &left, &right);
float *ptrs[] = { left.data(), right.data() };
{
- unique_lock<mutex> lock(compressor_mutex);
+ unique_lock<mutex> lock(audio_measure_mutex);
r128.process(left.size(), ptrs);
+ audio_mixer.set_current_loudness(r128.loudness_M());
correlation.process_samples(samples_out);
}
}
// And finally add them to the output.
- h264_encoder->add_audio(frame_pts_int, move(samples_out));
+ video_encoder->add_audio(frame_pts_int, move(samples_out));
}
void Mixer::subsample_chroma(GLuint src_tex, GLuint dst_tex)
glGenVertexArrays(1, &vao);
check_error();
- float vertices[] = {
- 0.0f, 2.0f,
- 0.0f, 0.0f,
- 2.0f, 0.0f
- };
-
glBindVertexArray(vao);
check_error();
float chroma_offset_0[] = { -0.5f / WIDTH, 0.0f };
set_uniform_vec2(cbcr_program_num, "foo", "chroma_offset_0", chroma_offset_0);
- GLuint position_vbo = fill_vertex_attribute(cbcr_program_num, "position", 2, GL_FLOAT, sizeof(vertices), vertices);
- GLuint texcoord_vbo = fill_vertex_attribute(cbcr_program_num, "texcoord", 2, GL_FLOAT, sizeof(vertices), vertices); // Same as vertices.
+ glBindBuffer(GL_ARRAY_BUFFER, cbcr_vbo);
+ check_error();
+
+ for (GLint attr_index : { cbcr_position_attribute_index, cbcr_texcoord_attribute_index }) {
+ glEnableVertexAttribArray(attr_index);
+ check_error();
+ glVertexAttribPointer(attr_index, 2, GL_FLOAT, GL_FALSE, 0, BUFFER_OFFSET(0));
+ check_error();
+ }
glDrawArrays(GL_TRIANGLES, 0, 3);
check_error();
- cleanup_vertex_attribute(cbcr_program_num, "position", position_vbo);
- cleanup_vertex_attribute(cbcr_program_num, "texcoord", texcoord_vbo);
+ for (GLint attr_index : { cbcr_position_attribute_index, cbcr_texcoord_attribute_index }) {
+ glDisableVertexAttribArray(attr_index);
+ check_error();
+ }
glUseProgram(0);
check_error();
+ glBindFramebuffer(GL_FRAMEBUFFER, 0);
+ check_error();
resource_pool->release_fbo(fbo);
glDeleteVertexArrays(1, &vao);
void Mixer::quit()
{
should_quit = true;
+ audio_task_queue_changed.notify_one();
mixer_thread.join();
audio_thread.join();
}
void Mixer::reset_meters()
{
+ unique_lock<mutex> lock(audio_measure_mutex);
peak_resampler.reset();
peak = 0.0f;
r128.reset();
correlation.reset();
}
+void Mixer::start_mode_scanning(unsigned card_index)
+{
+ assert(card_index < num_cards);
+ if (is_mode_scanning[card_index]) {
+ return;
+ }
+ is_mode_scanning[card_index] = true;
+ mode_scanlist[card_index].clear();
+ for (const auto &mode : cards[card_index].capture->get_available_video_modes()) {
+ mode_scanlist[card_index].push_back(mode.first);
+ }
+ assert(!mode_scanlist[card_index].empty());
+ mode_scanlist_index[card_index] = 0;
+ cards[card_index].capture->set_video_mode(mode_scanlist[card_index][0]);
+ last_mode_scan_change[card_index] = steady_clock::now();
+}
+
Mixer::OutputChannel::~OutputChannel()
{
if (has_current_frame) {
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;