-
- // Cut away everything under 120 Hz (or whatever the cutoff is);
- // we don't need it for voice, and it will reduce headroom
- // and confuse the compressor. (In particular, any hums at 50 or 60 Hz
- // should be dampened.)
- if (locut_enabled) {
- locut.render(samples_out.data(), samples_out.size() / 2, locut_cutoff_hz * 2.0 * M_PI / OUTPUT_FREQUENCY, 0.5f);
- }
-
- // Apply a level compressor to get the general level right.
- // Basically, if it's over about -40 dBFS, we squeeze it down to that level
- // (or more precisely, near it, since we don't use infinite ratio),
- // then apply a makeup gain to get it to -14 dBFS. -14 dBFS is, of course,
- // entirely arbitrary, but from practical tests with speech, it seems to
- // put ut around -23 LUFS, so it's a reasonable starting point for later use.
- {
- unique_lock<mutex> lock(compressor_mutex);
- if (level_compressor_enabled) {
- float threshold = 0.01f; // -40 dBFS.
- float ratio = 20.0f;
- float attack_time = 0.5f;
- float release_time = 20.0f;
- float makeup_gain = pow(10.0f, (ref_level_dbfs - (-40.0f)) / 20.0f); // +26 dB.
- level_compressor.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
- gain_staging_db = 20.0 * log10(level_compressor.get_attenuation() * makeup_gain);
- } else {
- // Just apply the gain we already had.
- float g = pow(10.0f, gain_staging_db / 20.0f);
- for (size_t i = 0; i < samples_out.size(); ++i) {
- samples_out[i] *= g;
- }
- }
- }
-
-#if 0
- printf("level=%f (%+5.2f dBFS) attenuation=%f (%+5.2f dB) end_result=%+5.2f dB\n",
- level_compressor.get_level(), 20.0 * log10(level_compressor.get_level()),
- level_compressor.get_attenuation(), 20.0 * log10(level_compressor.get_attenuation()),
- 20.0 * log10(level_compressor.get_level() * level_compressor.get_attenuation() * makeup_gain));
-#endif
-
-// float limiter_att, compressor_att;
-
- // The real compressor.
- if (compressor_enabled) {
- float threshold = pow(10.0f, compressor_threshold_dbfs / 20.0f);
- float ratio = 20.0f;
- float attack_time = 0.005f;
- float release_time = 0.040f;
- float makeup_gain = 2.0f; // +6 dB.
- compressor.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
-// compressor_att = compressor.get_attenuation();
- }
-
- // Finally a limiter at -4 dB (so, -10 dBFS) to take out the worst peaks only.
- // Note that since ratio is not infinite, we could go slightly higher than this.
- if (limiter_enabled) {
- float threshold = pow(10.0f, limiter_threshold_dbfs / 20.0f);
- float ratio = 30.0f;
- float attack_time = 0.0f; // Instant.
- float release_time = 0.020f;
- float makeup_gain = 1.0f; // 0 dB.
- limiter.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
-// limiter_att = limiter.get_attenuation();
- }
-
-// printf("limiter=%+5.1f compressor=%+5.1f\n", 20.0*log10(limiter_att), 20.0*log10(compressor_att));
-
- // Upsample 4x to find interpolated peak.
- peak_resampler.inp_data = samples_out.data();
- peak_resampler.inp_count = samples_out.size() / 2;
-
- vector<float> interpolated_samples_out;
- interpolated_samples_out.resize(samples_out.size());
- while (peak_resampler.inp_count > 0) { // About four iterations.
- peak_resampler.out_data = &interpolated_samples_out[0];
- peak_resampler.out_count = interpolated_samples_out.size() / 2;
- peak_resampler.process();
- size_t out_stereo_samples = interpolated_samples_out.size() / 2 - peak_resampler.out_count;
- peak = max<float>(peak, find_peak(interpolated_samples_out.data(), out_stereo_samples * 2));
- peak_resampler.out_data = nullptr;
- }
-
- // At this point, we are most likely close to +0 LU, but all of our
- // measurements have been on raw sample values, not R128 values.
- // So we have a final makeup gain to get us to +0 LU; the gain
- // adjustments required should be relatively small, and also, the
- // offset shouldn't change much (only if the type of audio changes
- // significantly). Thus, we shoot for updating this value basically
- // “whenever we process buffers”, since the R128 calculation isn't exactly
- // something we get out per-sample.
- //
- // Note that there's a feedback loop here, so we choose a very slow filter
- // (half-time of 100 seconds).
- double target_loudness_factor, alpha;
- {
- unique_lock<mutex> lock(compressor_mutex);
- double loudness_lu = r128.loudness_M() - ref_level_lufs;
- double current_makeup_lu = 20.0f * log10(final_makeup_gain);
- target_loudness_factor = pow(10.0f, -loudness_lu / 20.0f);
-
- // If we're outside +/- 5 LU uncorrected, we don't count it as
- // a normal signal (probably silence) and don't change the
- // correction factor; just apply what we already have.
- if (fabs(loudness_lu - current_makeup_lu) >= 5.0 || !final_makeup_gain_auto) {
- alpha = 0.0;
- } else {
- // Formula adapted from
- // https://en.wikipedia.org/wiki/Low-pass_filter#Simple_infinite_impulse_response_filter.
- const double half_time_s = 100.0;
- const double fc_mul_2pi_delta_t = 1.0 / (half_time_s * OUTPUT_FREQUENCY);
- alpha = fc_mul_2pi_delta_t / (fc_mul_2pi_delta_t + 1.0);
- }
-
- 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.
- vector<float> left, right;
- deinterleave_samples(samples_out, &left, &right);
- float *ptrs[] = { left.data(), right.data() };
- {
- unique_lock<mutex> lock(compressor_mutex);
- r128.process(left.size(), ptrs);
- correlation.process_samples(samples_out);
- }
-
- // Send the samples to the sound card.
- if (alsa) {
- alsa->write(samples_out);
- }
-
- // And finally add them to the output.
- h264_encoder->add_audio(frame_pts_int, move(samples_out));
-}
-
-void Mixer::subsample_chroma(GLuint src_tex, GLuint dst_tex)
-{
- GLuint vao;
- glGenVertexArrays(1, &vao);
- check_error();
-
- glBindVertexArray(vao);
- check_error();
-
- // Extract Cb/Cr.
- GLuint fbo = resource_pool->create_fbo(dst_tex);
- glBindFramebuffer(GL_FRAMEBUFFER, fbo);
- glViewport(0, 0, WIDTH/2, HEIGHT/2);
- check_error();
-
- glUseProgram(cbcr_program_num);
- check_error();
-
- glActiveTexture(GL_TEXTURE0);
- check_error();
- glBindTexture(GL_TEXTURE_2D, src_tex);
- check_error();
- glTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_MIN_FILTER, GL_LINEAR);
- check_error();
- glTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_WRAP_S, GL_CLAMP_TO_EDGE);
- check_error();
- glTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_WRAP_T, GL_CLAMP_TO_EDGE);
- check_error();
-
- float chroma_offset_0[] = { -0.5f / WIDTH, 0.0f };
- set_uniform_vec2(cbcr_program_num, "foo", "chroma_offset_0", chroma_offset_0);
-
- 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();
-
- 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);