X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=audio_mixer.cpp;h=63758f9949d4f0b7c92008a179ec0d081b2329f6;hb=0556d8cb8416bdc7b432a432c3d58239a94358d2;hp=38d80c6bb4405e489e8e064ac49e749a19a9a0ba;hpb=bb7eef2994e16fece998bc2c5cf42579d485d67b;p=nageru

diff --git a/audio_mixer.cpp b/audio_mixer.cpp
index 38d80c6..63758f9 100644
--- a/audio_mixer.cpp
+++ b/audio_mixer.cpp
@@ -25,6 +25,7 @@
 
 using namespace bmusb;
 using namespace std;
+using namespace std::chrono;
 using namespace std::placeholders;
 
 namespace {
@@ -183,7 +184,18 @@ AudioMixer::AudioMixer(unsigned num_cards)
 	set_limiter_enabled(global_flags.limiter_enabled);
 	set_final_makeup_gain_auto(global_flags.final_makeup_gain_auto);
 
+	r128.init(2, OUTPUT_FREQUENCY);
+	r128.integr_start();
+
+	// 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);
+
+	global_audio_mixer = this;
+	alsa_pool.init();
+
 	if (!global_flags.input_mapping_filename.empty()) {
+		// Must happen after ALSAPool is initialized, as it needs to know the card list.
 		current_mapping_mode = MappingMode::MULTICHANNEL;
 		InputMapping new_input_mapping;
 		if (!load_input_mapping_from_file(get_devices(),
@@ -200,16 +212,6 @@ AudioMixer::AudioMixer(unsigned num_cards)
 			current_mapping_mode = MappingMode::MULTICHANNEL;
 		}
 	}
-
-	r128.init(2, OUTPUT_FREQUENCY);
-	r128.integr_start();
-
-	// 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);
-
-	global_audio_mixer = this;
-	alsa_pool.init();
 }
 
 void AudioMixer::reset_resampler(DeviceSpec device_spec)
@@ -227,12 +229,13 @@ void AudioMixer::reset_resampler_mutex_held(DeviceSpec device_spec)
 	} else {
 		// TODO: ResamplingQueue should probably take the full device spec.
 		// (It's only used for console output, though.)
-		device->resampling_queue.reset(new ResamplingQueue(device_spec.index, device->capture_frequency, OUTPUT_FREQUENCY, device->interesting_channels.size()));
+		device->resampling_queue.reset(new ResamplingQueue(
+			device_spec.index, device->capture_frequency, OUTPUT_FREQUENCY, device->interesting_channels.size(),
+			global_flags.audio_queue_length_ms * 0.001));
 	}
-	device->next_local_pts = 0;
 }
 
-bool AudioMixer::add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned num_samples, AudioFormat audio_format, int64_t frame_length)
+bool AudioMixer::add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned num_samples, AudioFormat audio_format, int64_t frame_length, steady_clock::time_point frame_time)
 {
 	AudioDevice *device = find_audio_device(device_spec);
 
@@ -272,9 +275,7 @@ bool AudioMixer::add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned
 	}
 
 	// Now add it.
-	int64_t local_pts = device->next_local_pts;
-	device->resampling_queue->add_input_samples(local_pts / double(TIMEBASE), audio.get(), num_samples);
-	device->next_local_pts = local_pts + frame_length;
+	device->resampling_queue->add_input_samples(frame_time, audio.get(), num_samples, ResamplingQueue::ADJUST_RATE);
 	return true;
 }
 
@@ -296,11 +297,7 @@ bool AudioMixer::add_silence(DeviceSpec device_spec, unsigned samples_per_frame,
 
 	vector<float> silence(samples_per_frame * num_channels, 0.0f);
 	for (unsigned i = 0; i < num_frames; ++i) {
-		device->resampling_queue->add_input_samples(device->next_local_pts / double(TIMEBASE), silence.data(), samples_per_frame);
-		// 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.
-		device->next_local_pts += frame_length;
+		device->resampling_queue->add_input_samples(steady_clock::now(), silence.data(), samples_per_frame, ResamplingQueue::DO_NOT_ADJUST_RATE);
 	}
 	return true;
 }
@@ -473,7 +470,7 @@ void apply_gain(float db, float last_db, vector<float> *samples)
 
 }  // namespace
 
-vector<float> AudioMixer::get_output(double pts, unsigned num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy)
+vector<float> AudioMixer::get_output(steady_clock::time_point ts, unsigned num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy)
 {
 	map<DeviceSpec, vector<float>> samples_card;
 	vector<float> samples_bus;
@@ -488,7 +485,7 @@ vector<float> AudioMixer::get_output(double pts, unsigned num_samples, Resamplin
 			memset(&samples_card[device_spec][0], 0, samples_card[device_spec].size() * sizeof(float));
 		} else {
 			device->resampling_queue->get_output_samples(
-				pts,
+				ts,
 				&samples_card[device_spec][0],
 				num_samples,
 				rate_adjustment_policy);