#include "mixer.h"
#include <assert.h>
-#include <endian.h>
#include <epoxy/egl.h>
#include <movit/effect_chain.h>
#include <movit/effect_util.h>
#include <stdint.h>
#include <stdio.h>
#include <stdlib.h>
-#include <sys/time.h>
-#include <time.h>
+#include <sys/resource.h>
#include <algorithm>
#include <chrono>
-#include <cmath>
#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 <sys/time.h>
-#include <sys/resource.h>
+#include "DeckLinkAPI.h"
+#include "LinuxCOM.h"
+#include "alsa_output.h"
#include "bmusb/bmusb.h"
#include "bmusb/fake_capture.h"
#include "context.h"
#include "defs.h"
#include "disk_space_estimator.h"
#include "flags.h"
-#include "video_encoder.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;
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 / 2147483648.0f);
- }
- 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) {
- int32_t s = le32toh(*(int32_t *)src);
- dst[i * out_channels + j] = s * (1.0f / 2147483648.0f);
- 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) {
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)
{
CHECK(init_movit(MOVIT_SHADER_DIR, MOVIT_DEBUG_OFF));
check_error();
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);
-
- set_locut_enabled(global_flags.locut_enabled);
- set_gain_staging_db(global_flags.initial_gain_staging_db);
- set_gain_staging_auto(global_flags.gain_staging_auto);
- set_compressor_enabled(global_flags.compressor_enabled);
- set_limiter_enabled(global_flags.limiter_enabled);
- set_final_makeup_gain_auto(global_flags.final_makeup_gain_auto);
-
- // 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);
-
if (global_flags.enable_alsa_output) {
alsa.reset(new ALSAOutput(OUTPUT_FREQUENCY, /*num_channels=*/2));
}
if (card->surface == nullptr) {
card->surface = create_surface_with_same_format(mixer_surface);
}
- {
- unique_lock<mutex> lock(cards[card_index].audio_mutex);
- card->resampling_queue.reset(new ResamplingQueue(card_index, OUTPUT_FREQUENCY, OUTPUT_FREQUENCY, 2));
- }
while (!card->new_frames.empty()) card->new_frames.pop();
card->fractional_samples = 0;
card->last_timecode = -1;
- card->next_local_pts = 0;
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]) {
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);
- }
+ // 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;
- // 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(card_index, 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;
+ 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) {
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 };
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();
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.
}
}
-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) {
audio_task_queue.pop();
}
- process_audio_one_frame(task.pts_int, task.num_samples, task.adjust_rate);
- }
-}
-
-void Mixer::process_audio_one_frame(int64_t frame_pts_int, int num_samples, bool adjust_rate)
-{
- 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);
- ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy =
- adjust_rate ? ResamplingQueue::ADJUST_RATE : ResamplingQueue::DO_NOT_ADJUST_RATE;
- cards[card_index].resampling_queue->get_output_samples(
- double(frame_pts_int) / TIMEBASE,
- &samples_card[0],
- num_samples,
- rate_adjustment_policy);
- }
- 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);
- }
+ 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);
- // 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;
- }
+ // Send the samples to the sound card, then add them to the output.
+ if (alsa) {
+ alsa->write(samples_out);
}
+ video_encoder->add_audio(task.pts_int, move(samples_out));
}
-
-#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));
-
- // 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);
- }
-
- 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;
- }
-
- // 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;
- }
-
- // 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.
- video_encoder->add_audio(frame_pts_int, move(samples_out));
}
void Mixer::subsample_chroma(GLuint src_tex, GLuint dst_tex)
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);
projects / nageru / blobdiff