/*************************************************************************/ /* audio_stream.cpp */ /*************************************************************************/ /* This file is part of: */ /* GODOT ENGINE */ /* https://godotengine.org */ /*************************************************************************/ /* Copyright (c) 2007-2021 Juan Linietsky, Ariel Manzur. */ /* Copyright (c) 2014-2021 Godot Engine contributors (cf. AUTHORS.md). */ /* */ /* Permission is hereby granted, free of charge, to any person obtaining */ /* a copy of this software and associated documentation files (the */ /* "Software"), to deal in the Software without restriction, including */ /* without limitation the rights to use, copy, modify, merge, publish, */ /* distribute, sublicense, and/or sell copies of the Software, and to */ /* permit persons to whom the Software is furnished to do so, subject to */ /* the following conditions: */ /* */ /* The above copyright notice and this permission notice shall be */ /* included in all copies or substantial portions of the Software. */ /* */ /* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, */ /* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF */ /* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.*/ /* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY */ /* CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, */ /* TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE */ /* SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. */ /*************************************************************************/ #include "audio_stream.h" #include "core/config/project_settings.h" #include "core/os/os.h" ////////////////////////////// void AudioStreamPlaybackResampled::_begin_resample() { //clear cubic interpolation history internal_buffer[0] = AudioFrame(0.0, 0.0); internal_buffer[1] = AudioFrame(0.0, 0.0); internal_buffer[2] = AudioFrame(0.0, 0.0); internal_buffer[3] = AudioFrame(0.0, 0.0); //mix buffer _mix_internal(internal_buffer + 4, INTERNAL_BUFFER_LEN); mix_offset = 0; } void AudioStreamPlaybackResampled::mix(AudioFrame *p_buffer, float p_rate_scale, int p_frames) { float target_rate = AudioServer::get_singleton()->get_mix_rate(); float global_rate_scale = AudioServer::get_singleton()->get_global_rate_scale(); uint64_t mix_increment = uint64_t(((get_stream_sampling_rate() * p_rate_scale) / double(target_rate * global_rate_scale)) * double(FP_LEN)); for (int i = 0; i < p_frames; i++) { uint32_t idx = CUBIC_INTERP_HISTORY + uint32_t(mix_offset >> FP_BITS); //standard cubic interpolation (great quality/performance ratio) //this used to be moved to a LUT for greater performance, but nowadays CPU speed is generally faster than memory. float mu = (mix_offset & FP_MASK) / float(FP_LEN); AudioFrame y0 = internal_buffer[idx - 3]; AudioFrame y1 = internal_buffer[idx - 2]; AudioFrame y2 = internal_buffer[idx - 1]; AudioFrame y3 = internal_buffer[idx - 0]; float mu2 = mu * mu; AudioFrame a0 = y3 - y2 - y0 + y1; AudioFrame a1 = y0 - y1 - a0; AudioFrame a2 = y2 - y0; AudioFrame a3 = y1; p_buffer[i] = (a0 * mu * mu2 + a1 * mu2 + a2 * mu + a3); mix_offset += mix_increment; while ((mix_offset >> FP_BITS) >= INTERNAL_BUFFER_LEN) { internal_buffer[0] = internal_buffer[INTERNAL_BUFFER_LEN + 0]; internal_buffer[1] = internal_buffer[INTERNAL_BUFFER_LEN + 1]; internal_buffer[2] = internal_buffer[INTERNAL_BUFFER_LEN + 2]; internal_buffer[3] = internal_buffer[INTERNAL_BUFFER_LEN + 3]; if (is_playing()) { _mix_internal(internal_buffer + 4, INTERNAL_BUFFER_LEN); } else { //fill with silence, not playing for (int j = 0; j < INTERNAL_BUFFER_LEN; ++j) { internal_buffer[j + 4] = AudioFrame(0, 0); } } mix_offset -= (INTERNAL_BUFFER_LEN << FP_BITS); } } } //////////////////////////////// void AudioStream::_bind_methods() { ClassDB::bind_method(D_METHOD("get_length"), &AudioStream::get_length); } //////////////////////////////// Ref AudioStreamMicrophone::instance_playback() { Ref playback; playback.instance(); playbacks.insert(playback.ptr()); playback->microphone = Ref((AudioStreamMicrophone *)this); playback->active = false; return playback; } String AudioStreamMicrophone::get_stream_name() const { //if (audio_stream.is_valid()) { //return "Random: " + audio_stream->get_name(); //} return "Microphone"; } float AudioStreamMicrophone::get_length() const { return 0; } void AudioStreamMicrophone::_bind_methods() { } AudioStreamMicrophone::AudioStreamMicrophone() { } void AudioStreamPlaybackMicrophone::_mix_internal(AudioFrame *p_buffer, int p_frames) { AudioDriver::get_singleton()->lock(); Vector buf = AudioDriver::get_singleton()->get_input_buffer(); unsigned int input_size = AudioDriver::get_singleton()->get_input_size(); int mix_rate = AudioDriver::get_singleton()->get_mix_rate(); unsigned int playback_delay = MIN(((50 * mix_rate) / 1000) * 2, buf.size() >> 1); #ifdef DEBUG_ENABLED unsigned int input_position = AudioDriver::get_singleton()->get_input_position(); #endif if (playback_delay > input_size) { for (int i = 0; i < p_frames; i++) { p_buffer[i] = AudioFrame(0.0f, 0.0f); } input_ofs = 0; } else { for (int i = 0; i < p_frames; i++) { if (input_size > input_ofs && (int)input_ofs < buf.size()) { float l = (buf[input_ofs++] >> 16) / 32768.f; if ((int)input_ofs >= buf.size()) { input_ofs = 0; } float r = (buf[input_ofs++] >> 16) / 32768.f; if ((int)input_ofs >= buf.size()) { input_ofs = 0; } p_buffer[i] = AudioFrame(l, r); } else { p_buffer[i] = AudioFrame(0.0f, 0.0f); } } } #ifdef DEBUG_ENABLED if (input_ofs > input_position && (int)(input_ofs - input_position) < (p_frames * 2)) { print_verbose(String(get_class_name()) + " buffer underrun: input_position=" + itos(input_position) + " input_ofs=" + itos(input_ofs) + " input_size=" + itos(input_size)); } #endif AudioDriver::get_singleton()->unlock(); } void AudioStreamPlaybackMicrophone::mix(AudioFrame *p_buffer, float p_rate_scale, int p_frames) { AudioStreamPlaybackResampled::mix(p_buffer, p_rate_scale, p_frames); } float AudioStreamPlaybackMicrophone::get_stream_sampling_rate() { return AudioDriver::get_singleton()->get_mix_rate(); } void AudioStreamPlaybackMicrophone::start(float p_from_pos) { if (active) { return; } if (!GLOBAL_GET("audio/enable_audio_input")) { WARN_PRINT("Need to enable Project settings > Audio > Enable Audio Input option to use capturing."); return; } input_ofs = 0; if (AudioDriver::get_singleton()->capture_start() == OK) { active = true; _begin_resample(); } } void AudioStreamPlaybackMicrophone::stop() { if (active) { AudioDriver::get_singleton()->capture_stop(); active = false; } } bool AudioStreamPlaybackMicrophone::is_playing() const { return active; } int AudioStreamPlaybackMicrophone::get_loop_count() const { return 0; } float AudioStreamPlaybackMicrophone::get_playback_position() const { return 0; } void AudioStreamPlaybackMicrophone::seek(float p_time) { // Can't seek a microphone input } AudioStreamPlaybackMicrophone::~AudioStreamPlaybackMicrophone() { microphone->playbacks.erase(this); stop(); } AudioStreamPlaybackMicrophone::AudioStreamPlaybackMicrophone() { } //////////////////////////////// void AudioStreamRandomPitch::set_audio_stream(const Ref &p_audio_stream) { audio_stream = p_audio_stream; if (audio_stream.is_valid()) { for (Set::Element *E = playbacks.front(); E; E = E->next()) { E->get()->playback = audio_stream->instance_playback(); } } } Ref AudioStreamRandomPitch::get_audio_stream() const { return audio_stream; } void AudioStreamRandomPitch::set_random_pitch(float p_pitch) { if (p_pitch < 1) { p_pitch = 1; } random_pitch = p_pitch; } float AudioStreamRandomPitch::get_random_pitch() const { return random_pitch; } Ref AudioStreamRandomPitch::instance_playback() { Ref playback; playback.instance(); if (audio_stream.is_valid()) { playback->playback = audio_stream->instance_playback(); } playbacks.insert(playback.ptr()); playback->random_pitch = Ref((AudioStreamRandomPitch *)this); return playback; } String AudioStreamRandomPitch::get_stream_name() const { if (audio_stream.is_valid()) { return "Random: " + audio_stream->get_name(); } return "RandomPitch"; } float AudioStreamRandomPitch::get_length() const { if (audio_stream.is_valid()) { return audio_stream->get_length(); } return 0; } void AudioStreamRandomPitch::_bind_methods() { ClassDB::bind_method(D_METHOD("set_audio_stream", "stream"), &AudioStreamRandomPitch::set_audio_stream); ClassDB::bind_method(D_METHOD("get_audio_stream"), &AudioStreamRandomPitch::get_audio_stream); ClassDB::bind_method(D_METHOD("set_random_pitch", "scale"), &AudioStreamRandomPitch::set_random_pitch); ClassDB::bind_method(D_METHOD("get_random_pitch"), &AudioStreamRandomPitch::get_random_pitch); ADD_PROPERTY(PropertyInfo(Variant::OBJECT, "audio_stream", PROPERTY_HINT_RESOURCE_TYPE, "AudioStream"), "set_audio_stream", "get_audio_stream"); ADD_PROPERTY(PropertyInfo(Variant::FLOAT, "random_pitch", PROPERTY_HINT_RANGE, "1,16,0.01"), "set_random_pitch", "get_random_pitch"); } AudioStreamRandomPitch::AudioStreamRandomPitch() { random_pitch = 1.1; } void AudioStreamPlaybackRandomPitch::start(float p_from_pos) { playing = playback; float range_from = 1.0 / random_pitch->random_pitch; float range_to = random_pitch->random_pitch; pitch_scale = range_from + Math::randf() * (range_to - range_from); if (playing.is_valid()) { playing->start(p_from_pos); } } void AudioStreamPlaybackRandomPitch::stop() { if (playing.is_valid()) { playing->stop(); ; } } bool AudioStreamPlaybackRandomPitch::is_playing() const { if (playing.is_valid()) { return playing->is_playing(); } return false; } int AudioStreamPlaybackRandomPitch::get_loop_count() const { if (playing.is_valid()) { return playing->get_loop_count(); } return 0; } float AudioStreamPlaybackRandomPitch::get_playback_position() const { if (playing.is_valid()) { return playing->get_playback_position(); } return 0; } void AudioStreamPlaybackRandomPitch::seek(float p_time) { if (playing.is_valid()) { playing->seek(p_time); } } void AudioStreamPlaybackRandomPitch::mix(AudioFrame *p_buffer, float p_rate_scale, int p_frames) { if (playing.is_valid()) { playing->mix(p_buffer, p_rate_scale * pitch_scale, p_frames); } else { for (int i = 0; i < p_frames; i++) { p_buffer[i] = AudioFrame(0, 0); } } } AudioStreamPlaybackRandomPitch::~AudioStreamPlaybackRandomPitch() { random_pitch->playbacks.erase(this); }