#include <utility>
#include "decibel.h"
+#include "delay_analyzer.h"
#include "flags.h"
#include "shared/metrics.h"
#include "state.pb.h"
}
}
+double get_delay_seconds(double extra_delay_ms)
+{
+ // Make sure we never get negative delay. Even 1 ms is probably way less than we
+ // could ever hope to actually have; this is just a failsafe.
+ double delay_ms = max(global_flags.audio_queue_length_ms + extra_delay_ms, 1.0);
+ return delay_ms * 0.001;
+}
+
} // namespace
AudioMixer::AudioMixer(unsigned num_capture_cards, unsigned num_ffmpeg_inputs)
// 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(),
+ if (!load_input_mapping_from_file(get_devices(HOLD_ALSA_DEVICES),
global_flags.input_mapping_filename,
&new_input_mapping)) {
fprintf(stderr, "Failed to load input mapping from '%s', exiting.\n",
if (device->interesting_channels.empty()) {
device->resampling_queue.reset();
} else {
- // Make sure we never get negative delay. Even 1 ms is probably way less than we
- // could ever hope to actually have; this is just a failsafe.
- double delay_ms = max(global_flags.audio_queue_length_ms + device->extra_delay_ms, 1.0);
-
device->resampling_queue.reset(new ResamplingQueue(
device_spec, device->capture_frequency, OUTPUT_FREQUENCY, device->interesting_channels.size(),
- delay_ms * 0.001));
+ get_delay_seconds(device->extra_delay_ms)));
}
}
bool AudioMixer::add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned num_samples, AudioFormat audio_format, steady_clock::time_point frame_time)
{
+ if (delay_analyzer != nullptr && delay_analyzer->is_grabbing()) {
+ delay_analyzer->add_audio(device_spec, data, num_samples, audio_format, frame_time);
+ }
+
AudioDevice *device = find_audio_device(device_spec);
unique_lock<timed_mutex> lock(audio_mutex, defer_lock);
unique_ptr<float[]> audio(new float[num_samples * num_channels]);
unsigned channel_index = 0;
for (auto channel_it = device->interesting_channels.cbegin(); channel_it != device->interesting_channels.end(); ++channel_it, ++channel_index) {
- switch (audio_format.bits_per_sample) {
- case 0:
- assert(num_samples == 0);
- break;
- case 16:
- convert_fixed16_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
- break;
- case 24:
- convert_fixed24_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
- break;
- case 32:
- convert_fixed32_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, 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);
- }
+ convert_audio_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format, num_samples);
}
// If we changed frequency since last frame, we'll need to reset the resampler.
return true;
}
+// Converts all channels.
vector<int32_t> convert_audio_to_fixed32(const uint8_t *data, unsigned num_samples, bmusb::AudioFormat audio_format, unsigned num_channels)
{
vector<int32_t> audio;
return audio;
}
+// Converts only one channel.
+void convert_audio_to_fp32(float *dst, size_t out_channel, size_t out_num_channels,
+ const uint8_t *src, size_t in_channel, bmusb::AudioFormat in_audio_format,
+ size_t num_samples)
+{
+ switch (in_audio_format.bits_per_sample) {
+ case 0:
+ assert(num_samples == 0);
+ break;
+ case 16:
+ convert_fixed16_to_fp32(dst, out_channel, out_num_channels, src, in_channel, in_audio_format.num_channels, num_samples);
+ break;
+ case 24:
+ convert_fixed24_to_fp32(dst, out_channel, out_num_channels, src, in_channel, in_audio_format.num_channels, num_samples);
+ break;
+ case 32:
+ convert_fixed32_to_fp32(dst, out_channel, out_num_channels, src, in_channel, in_audio_format.num_channels, num_samples);
+ break;
+ default:
+ fprintf(stderr, "Cannot handle audio with %u bits per sample\n", in_audio_format.bits_per_sample);
+ assert(false);
+ }
+}
+
bool AudioMixer::add_silence(DeviceSpec device_spec, unsigned samples_per_frame, unsigned num_frames)
{
AudioDevice *device = find_audio_device(device_spec);
correlation.get_correlation());
}
-map<DeviceSpec, DeviceInfo> AudioMixer::get_devices()
+map<DeviceSpec, DeviceInfo> AudioMixer::get_devices(HoldDevices hold_devices)
{
lock_guard<timed_mutex> lock(audio_mutex);
info.num_channels = 8;
devices.insert(make_pair(spec, info));
}
- vector<ALSAPool::Device> available_alsa_devices = alsa_pool.get_devices();
+ vector<ALSAPool::Device> available_alsa_devices = alsa_pool.get_devices(hold_devices);
for (unsigned card_index = 0; card_index < available_alsa_devices.size(); ++card_index) {
const DeviceSpec spec{ InputSourceType::ALSA_INPUT, card_index };
const ALSAPool::Device &device = available_alsa_devices[card_index];
new_input_mapping.buses.push_back(input);
+ // NOTE: Delay is implicitly at 0.0 ms, since none has been set in the mapping.
+
lock_guard<timed_mutex> lock(audio_mutex);
current_mapping_mode = MappingMode::SIMPLE;
set_input_mapping_lock_held(new_input_mapping);
for (unsigned card_index = 0; card_index < MAX_VIDEO_CARDS; ++card_index) {
const DeviceSpec device_spec{InputSourceType::CAPTURE_CARD, card_index};
AudioDevice *device = find_audio_device(device_spec);
- if (device->interesting_channels != interesting_channels[device_spec] ||
- device->extra_delay_ms != new_extra_delay_ms[device_spec]) {
+ double extra_delay_ms = new_extra_delay_ms[device_spec];
+ if (device->interesting_channels != interesting_channels[device_spec]) {
device->interesting_channels = interesting_channels[device_spec];
- device->extra_delay_ms = new_extra_delay_ms[device_spec];
+ device->extra_delay_ms = extra_delay_ms;
reset_resampler_mutex_held(device_spec);
+ } else if (device->extra_delay_ms != extra_delay_ms &&
+ device->resampling_queue != nullptr) {
+ device->extra_delay_ms = extra_delay_ms;
+ device->resampling_queue->change_expected_delay(get_delay_seconds(extra_delay_ms));
}
}
for (unsigned card_index = 0; card_index < MAX_ALSA_CARDS; ++card_index) {
const DeviceSpec device_spec{InputSourceType::ALSA_INPUT, card_index};
AudioDevice *device = find_audio_device(device_spec);
+ double extra_delay_ms = new_extra_delay_ms[device_spec];
if (interesting_channels[device_spec].empty()) {
alsa_pool.release_device(card_index);
} else {
alsa_pool.hold_device(card_index);
}
- if (device->interesting_channels != interesting_channels[device_spec] ||
- device->extra_delay_ms != new_extra_delay_ms[device_spec]) {
+ if (device->interesting_channels != interesting_channels[device_spec]) {
device->interesting_channels = interesting_channels[device_spec];
- device->extra_delay_ms = new_extra_delay_ms[device_spec];
+ device->extra_delay_ms = extra_delay_ms;
alsa_pool.reset_device(device_spec.index);
reset_resampler_mutex_held(device_spec);
+ } else if (device->extra_delay_ms != extra_delay_ms &&
+ device->resampling_queue != nullptr) {
+ device->extra_delay_ms = extra_delay_ms;
+ device->resampling_queue->change_expected_delay(get_delay_seconds(extra_delay_ms));
}
}
for (unsigned card_index = 0; card_index < num_ffmpeg_inputs; ++card_index) {
const DeviceSpec device_spec{InputSourceType::FFMPEG_VIDEO_INPUT, card_index};
AudioDevice *device = find_audio_device(device_spec);
- if (device->interesting_channels != interesting_channels[device_spec] ||
- device->extra_delay_ms != new_extra_delay_ms[device_spec]) {
+ double extra_delay_ms = new_extra_delay_ms[device_spec];
+ if (device->interesting_channels != interesting_channels[device_spec]) {
device->interesting_channels = interesting_channels[device_spec];
- device->extra_delay_ms = new_extra_delay_ms[device_spec];
+ device->extra_delay_ms = extra_delay_ms;
reset_resampler_mutex_held(device_spec);
+ } else if (device->extra_delay_ms != extra_delay_ms &&
+ device->resampling_queue != nullptr) {
+ device->extra_delay_ms = extra_delay_ms;
+ device->resampling_queue->change_expected_delay(get_delay_seconds(extra_delay_ms));
}
}
projects / nageru / blobdiff