#define WIDTH 1280
#define HEIGHT 720
+#define EXTRAHEIGHT 30
#undef Success
#include "mixer.h"
#include <assert.h>
-#include <effect.h>
-#include <effect_chain.h>
-#include <effect_util.h>
#include <epoxy/egl.h>
-#include <features.h>
-#include <image_format.h>
#include <init.h>
-#include <overlay_effect.h>
-#include <padding_effect.h>
-#include <resample_effect.h>
-#include <resource_pool.h>
-#include <saturation_effect.h>
+#include <movit/effect_chain.h>
+#include <movit/effect_util.h>
+#include <movit/flat_input.h>
+#include <movit/image_format.h>
+#include <movit/resource_pool.h>
#include <stdint.h>
#include <stdio.h>
#include <stdlib.h>
#include <sys/time.h>
#include <time.h>
#include <util.h>
-#include <white_balance_effect.h>
-#include <ycbcr.h>
-#include <ycbcr_input.h>
+#include <algorithm>
#include <cmath>
#include <condition_variable>
#include <cstddef>
#include <mutex>
#include <string>
#include <thread>
+#include <utility>
#include <vector>
#include "bmusb/bmusb.h"
#include "context.h"
+#include "defs.h"
#include "h264encode.h"
#include "pbo_frame_allocator.h"
#include "ref_counted_gl_sync.h"
} // namespace
-Mixer::Mixer(const QSurfaceFormat &format)
- : mixer_surface(create_surface(format)),
- h264_encoder_surface(create_surface(format))
+Mixer::Mixer(const QSurfaceFormat &format, unsigned num_cards)
+ : httpd("test.ts", WIDTH, HEIGHT),
+ num_cards(num_cards),
+ mixer_surface(create_surface(format)),
+ h264_encoder_surface(create_surface(format)),
+ level_compressor(OUTPUT_FREQUENCY),
+ limiter(OUTPUT_FREQUENCY),
+ compressor(OUTPUT_FREQUENCY)
{
+ httpd.start(9095);
+
CHECK(init_movit(MOVIT_SHADER_DIR, MOVIT_DEBUG_OFF));
check_error();
+ // Since we allow non-bouncing 4:2:2 YCbCrInputs, effective subpixel precision
+ // will be halved when sampling them, and we need to compensate here.
+ movit_texel_subpixel_precision /= 2.0;
+
resource_pool.reset(new ResourcePool);
- theme.reset(new Theme("theme.lua", resource_pool.get()));
- output_channel[OUTPUT_LIVE].parent = this;
- output_channel[OUTPUT_PREVIEW].parent = this;
- output_channel[OUTPUT_INPUT0].parent = this;
- output_channel[OUTPUT_INPUT1].parent = this;
+ theme.reset(new Theme("theme.lua", resource_pool.get(), num_cards));
+ for (unsigned i = 0; i < NUM_OUTPUTS; ++i) {
+ output_channel[i].parent = this;
+ }
ImageFormat inout_format;
inout_format.color_space = COLORSPACE_sRGB;
display_chain->set_dither_bits(0); // Don't bother.
display_chain->finalize();
- h264_encoder.reset(new H264Encoder(h264_encoder_surface, WIDTH, HEIGHT, "test.mp4"));
+ h264_encoder.reset(new H264Encoder(h264_encoder_surface, WIDTH, HEIGHT, &httpd));
- for (int card_index = 0; card_index < NUM_CARDS; ++card_index) {
+ 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(0x1edb, card_index == 0 ? 0xbd3b : 0xbd4f);
+ 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(WIDTH * (HEIGHT+30) * 2 + 44, WIDTH, HEIGHT));
+ card->frame_allocator.reset(new PBOFrameAllocator(WIDTH * (HEIGHT+EXTRAHEIGHT) * 2 + 44, WIDTH, HEIGHT));
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->context = create_context(card->surface);
if (!make_current(card->context, card->surface)) {
printf("failed to create bmusb context\n");
exit(1);
}
- printf("inited!\n");
},
[this]{
resource_pool->clean_context();
});
- card->resampler.reset(new Resampler(48000.0, 48000.0, 2));
+ card->resampling_queue.reset(new ResamplingQueue(OUTPUT_FREQUENCY, OUTPUT_FREQUENCY, 2));
card->usb->configure_card();
}
BMUSBCapture::start_bm_thread();
- for (int card_index = 0; card_index < NUM_CARDS; ++card_index) {
+ for (unsigned card_index = 0; card_index < num_cards; ++card_index) {
cards[card_index].usb->start_bm_capture();
}
" gl_FragColor = texture2D(cbcr_tex, tc0); \n"
"} \n";
cbcr_program_num = resource_pool->compile_glsl_program(cbcr_vert_shader, cbcr_frag_shader);
+
+ 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);
+
+ alsa.reset(new ALSAOutput(OUTPUT_FREQUENCY, /*num_channels=*/2));
}
Mixer::~Mixer()
resource_pool->release_glsl_program(cbcr_program_num);
BMUSBCapture::stop_bm_thread();
- for (int card_index = 0; card_index < NUM_CARDS; ++card_index) {
+ 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.
}
}
+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 = std::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(int card_index, uint16_t timecode,
+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)
{
return;
}
- // Convert the audio to stereo fp32 and add it.
- size_t num_samples = (audio_frame.len - audio_offset) / 8 / 3;
- vector<float> audio;
- audio.resize(num_samples * 2);
- convert_fixed24_to_fp32(&audio[0], 2, audio_frame.data + audio_offset, 8, num_samples);
-
int unwrapped_timecode = timecode;
int dropped_frames = 0;
if (card->last_timecode != -1) {
}
card->last_timecode = unwrapped_timecode;
+ // Convert the audio to stereo fp32 and add it.
+ size_t num_samples = (audio_frame.len >= audio_offset) ? (audio_frame.len - audio_offset) / 8 / 3 : 0;
+ 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);
int unwrapped_timecode = timecode;
- if (dropped_frames > 60 * 2) {
+ if (dropped_frames > FPS * 2) {
fprintf(stderr, "Card %d lost more than two seconds (or time code jumping around), resetting resampler\n",
card_index);
- card->resampler.reset(new Resampler(48000.0, 48000.0, 2));
+ card->resampling_queue.reset(new ResamplingQueue(OUTPUT_FREQUENCY, OUTPUT_FREQUENCY, 2));
} 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.resize((48000 / 60) * 2);
+ silence.resize((OUTPUT_FREQUENCY / FPS) * 2);
for (int i = 0; i < dropped_frames; ++i) {
- card->resampler->add_input_samples((unwrapped_timecode - dropped_frames + i) / 60.0, silence.data(), (48000 / 60));
+ card->resampling_queue->add_input_samples((unwrapped_timecode - dropped_frames + i) / double(FPS), silence.data(), (OUTPUT_FREQUENCY / FPS));
}
}
- card->resampler->add_input_samples(unwrapped_timecode / 60.0, audio.data(), num_samples);
+ card->resampling_queue->add_input_samples(unwrapped_timecode / double(FPS), audio.data(), num_samples);
}
// Done with the audio, so release it.
if (card->should_quit) return;
}
- if (video_frame.len - video_offset != WIDTH * (HEIGHT+30) * 2) {
+ if (video_frame.len - video_offset != WIDTH * (HEIGHT+EXTRAHEIGHT) * 2) {
if (video_frame.len != 0) {
printf("Card %d: Dropping video frame with wrong length (%ld)\n",
card_index, video_frame.len - video_offset);
// Upload the textures.
glBindTexture(GL_TEXTURE_2D, userdata->tex_y);
check_error();
- glTexSubImage2D(GL_TEXTURE_2D, 0, 0, 0, WIDTH, HEIGHT, GL_RED, GL_UNSIGNED_BYTE, BUFFER_OFFSET((WIDTH * (HEIGHT+30) * 2 + 44) / 2 + WIDTH * 25 + 22));
+ glTexSubImage2D(GL_TEXTURE_2D, 0, 0, 0, WIDTH, HEIGHT, GL_RED, GL_UNSIGNED_BYTE, BUFFER_OFFSET((WIDTH * (HEIGHT+EXTRAHEIGHT) * 2 + 44) / 2 + WIDTH * 25 + 22));
check_error();
glBindTexture(GL_TEXTURE_2D, userdata->tex_cbcr);
check_error();
void Mixer::thread_func()
{
eglBindAPI(EGL_OPENGL_API);
- QOpenGLContext *context = create_context();
+ QOpenGLContext *context = create_context(mixer_surface);
if (!make_current(context, mixer_surface)) {
printf("oops\n");
exit(1);
int dropped_frames = 0;
while (!should_quit) {
- CaptureCard card_copy[NUM_CARDS];
+ CaptureCard card_copy[MAX_CARDS];
{
unique_lock<mutex> lock(bmusb_mutex);
// TODO: Make configurable, and with a timeout.
cards[0].new_data_ready_changed.wait(lock, [this]{ return cards[0].new_data_ready; });
- for (int card_index = 0; card_index < NUM_CARDS; ++card_index) {
+ 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_data_ready_fence = card->new_data_ready_fence;
- card_copy[card_index].new_frame_audio = move(card->new_frame_audio);
card_copy[card_index].dropped_frames = card->dropped_frames;
card->new_data_ready = false;
card->new_data_ready_changed.notify_all();
}
// Resample the audio as needed, including from previously dropped frames.
- vector<float> samples_out;
- // TODO: Allow using audio from the other card(s) as well.
for (unsigned frame_num = 0; frame_num < card_copy[0].dropped_frames + 1; ++frame_num) {
- for (unsigned card_index = 0; card_index < NUM_CARDS; ++card_index) {
- samples_out.resize((48000 / 60) * 2);
- {
- unique_lock<mutex> lock(cards[card_index].audio_mutex);
- if (!cards[card_index].resampler->get_output_samples(pts(), &samples_out[0], 48000 / 60)) {
- printf("Card %d reported previous underrun.\n", card_index);
- }
- }
- if (card_index == 0) {
- h264_encoder->add_audio(pts_int, move(samples_out));
- }
- }
+ process_audio_one_frame();
if (frame_num != card_copy[0].dropped_frames) {
// For dropped frames, increase the pts.
++dropped_frames;
- pts_int += TIMEBASE / 60;
+ pts_int += TIMEBASE / FPS;
+ }
+ }
+
+ if (audio_level_callback != nullptr) {
+ 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,
+ last_gain_staging_db);
+ }
+
+ 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;
+ }
+ if (card_copy[card_index].dropped_frames > 0) {
+ printf("Card %u dropped %d frames before this\n",
+ card_index, int(card_copy[card_index].dropped_frames));
}
}
// just increase the pts (skipping over this frame) and don't try to compute anything new.
if (card_copy[0].new_frame->len == 0) {
++dropped_frames;
- pts_int += TIMEBASE / 60;
+ pts_int += TIMEBASE / FPS;
continue;
}
- for (int card_index = 0; card_index < NUM_CARDS; ++card_index) {
+ 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)
continue;
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);
// input frames needed, so that they are not released back
// until the rendering is done.
vector<RefCountedFrame> input_frames;
- for (int card_index = 0; card_index < NUM_CARDS; ++card_index) {
+ for (unsigned card_index = 0; card_index < num_cards; ++card_index) {
input_frames.push_back(bmusb_current_rendering_frame[card_index]);
}
- const int64_t av_delay = TIMEBASE / 10; // Corresponds to the fixed delay in resampler.h. TODO: Make less hard-coded.
+ 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, input_frames);
++frame;
- pts_int += TIMEBASE / 60;
+ pts_int += TIMEBASE / FPS;
// The live frame just shows the RGBA texture we just rendered.
// It owns rgba_tex now.
display_frame.chain = chain.first;
display_frame.setup_chain = chain.second;
display_frame.ready_fence = fence;
- display_frame.input_frames = { bmusb_current_rendering_frame[0], bmusb_current_rendering_frame[1] }; // FIXME: possible to do better?
+
+ // FIXME: possible to do better?
+ for (unsigned card_index = 0; card_index < num_cards; ++card_index) {
+ display_frame.input_frames.push_back(bmusb_current_rendering_frame[card_index]);
+ }
display_frame.temp_textures = {};
output_channel[i].output_frame(display_frame);
}
resource_pool->clean_context();
}
+void Mixer::process_audio_one_frame()
+{
+ vector<float> samples_card;
+ vector<float> samples_out;
+ for (unsigned card_index = 0; card_index < num_cards; ++card_index) {
+ samples_card.resize((OUTPUT_FREQUENCY / FPS) * 2);
+ {
+ unique_lock<mutex> lock(cards[card_index].audio_mutex);
+ if (!cards[card_index].resampling_queue->get_output_samples(pts(), &samples_card[0], OUTPUT_FREQUENCY / FPS)) {
+ 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);
+ }
+ }
+
+ // Cut away everything under 150 Hz; 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.)
+ 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.
+ float ref_level_dbfs = -14.0f;
+ {
+ 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);
+ last_gain_staging_db = 20.0 * log10(level_compressor.get_attenuation() * makeup_gain);
+ }
+
+#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));
+ }
+
+ // Find R128 levels.
+ vector<float> left, right;
+ deinterleave_samples(samples_out, &left, &right);
+ float *ptrs[] = { left.data(), right.data() };
+ r128.process(left.size(), ptrs);
+
+ // Send the samples to the sound card.
+ if (alsa) {
+ alsa->write(samples_out);
+ }
+
+ // And finally add them to the output.
+ h264_encoder->add_audio(pts_int, move(samples_out));
+}
+
void Mixer::subsample_chroma(GLuint src_tex, GLuint dst_tex)
{
GLuint vao;
theme->channel_clicked(preview_num);
}
+void Mixer::reset_meters()
+{
+ peak_resampler.reset();
+ peak = 0.0f;
+ r128.reset();
+ r128.integr_start();
+}
+
Mixer::OutputChannel::~OutputChannel()
{
if (has_current_frame) {
projects / nageru / blobdiff