a7f49ac9a1
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369 lines
11 KiB
C++
369 lines
11 KiB
C++
/*************************************************************************/
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/* audio_stream.cpp */
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/*************************************************************************/
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/* This file is part of: */
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/* GODOT ENGINE */
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/* https://godotengine.org */
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/*************************************************************************/
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/* Copyright (c) 2007-2020 Juan Linietsky, Ariel Manzur. */
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/* Copyright (c) 2014-2020 Godot Engine contributors (cf. AUTHORS.md). */
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/* */
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/* Permission is hereby granted, free of charge, to any person obtaining */
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/* a copy of this software and associated documentation files (the */
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/* "Software"), to deal in the Software without restriction, including */
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/* without limitation the rights to use, copy, modify, merge, publish, */
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/* distribute, sublicense, and/or sell copies of the Software, and to */
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/* permit persons to whom the Software is furnished to do so, subject to */
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/* the following conditions: */
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/* */
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/* The above copyright notice and this permission notice shall be */
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/* included in all copies or substantial portions of the Software. */
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/* */
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/* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, */
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/* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF */
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/* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.*/
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/* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY */
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/* CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, */
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/* TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE */
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/* SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. */
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/*************************************************************************/
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#include "audio_stream.h"
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#include "core/os/os.h"
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#include "core/project_settings.h"
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//////////////////////////////
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void AudioStreamPlaybackResampled::_begin_resample() {
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//clear cubic interpolation history
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internal_buffer[0] = AudioFrame(0.0, 0.0);
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internal_buffer[1] = AudioFrame(0.0, 0.0);
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internal_buffer[2] = AudioFrame(0.0, 0.0);
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internal_buffer[3] = AudioFrame(0.0, 0.0);
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//mix buffer
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_mix_internal(internal_buffer + 4, INTERNAL_BUFFER_LEN);
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mix_offset = 0;
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}
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void AudioStreamPlaybackResampled::mix(AudioFrame *p_buffer, float p_rate_scale, int p_frames) {
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float target_rate = AudioServer::get_singleton()->get_mix_rate();
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float global_rate_scale = AudioServer::get_singleton()->get_global_rate_scale();
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uint64_t mix_increment = uint64_t(((get_stream_sampling_rate() * p_rate_scale) / double(target_rate * global_rate_scale)) * double(FP_LEN));
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for (int i = 0; i < p_frames; i++) {
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uint32_t idx = CUBIC_INTERP_HISTORY + uint32_t(mix_offset >> FP_BITS);
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//standard cubic interpolation (great quality/performance ratio)
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//this used to be moved to a LUT for greater performance, but nowadays CPU speed is generally faster than memory.
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float mu = (mix_offset & FP_MASK) / float(FP_LEN);
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AudioFrame y0 = internal_buffer[idx - 3];
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AudioFrame y1 = internal_buffer[idx - 2];
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AudioFrame y2 = internal_buffer[idx - 1];
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AudioFrame y3 = internal_buffer[idx - 0];
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float mu2 = mu * mu;
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AudioFrame a0 = y3 - y2 - y0 + y1;
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AudioFrame a1 = y0 - y1 - a0;
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AudioFrame a2 = y2 - y0;
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AudioFrame a3 = y1;
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p_buffer[i] = (a0 * mu * mu2 + a1 * mu2 + a2 * mu + a3);
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mix_offset += mix_increment;
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while ((mix_offset >> FP_BITS) >= INTERNAL_BUFFER_LEN) {
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internal_buffer[0] = internal_buffer[INTERNAL_BUFFER_LEN + 0];
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internal_buffer[1] = internal_buffer[INTERNAL_BUFFER_LEN + 1];
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internal_buffer[2] = internal_buffer[INTERNAL_BUFFER_LEN + 2];
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internal_buffer[3] = internal_buffer[INTERNAL_BUFFER_LEN + 3];
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if (is_playing()) {
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_mix_internal(internal_buffer + 4, INTERNAL_BUFFER_LEN);
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} else {
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//fill with silence, not playing
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for (int j = 0; j < INTERNAL_BUFFER_LEN; ++j) {
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internal_buffer[j + 4] = AudioFrame(0, 0);
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}
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}
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mix_offset -= (INTERNAL_BUFFER_LEN << FP_BITS);
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}
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}
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}
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////////////////////////////////
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void AudioStream::_bind_methods() {
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ClassDB::bind_method(D_METHOD("get_length"), &AudioStream::get_length);
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}
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////////////////////////////////
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Ref<AudioStreamPlayback> AudioStreamMicrophone::instance_playback() {
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Ref<AudioStreamPlaybackMicrophone> playback;
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playback.instance();
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playbacks.insert(playback.ptr());
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playback->microphone = Ref<AudioStreamMicrophone>((AudioStreamMicrophone *)this);
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playback->active = false;
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return playback;
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}
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String AudioStreamMicrophone::get_stream_name() const {
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//if (audio_stream.is_valid()) {
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//return "Random: " + audio_stream->get_name();
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//}
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return "Microphone";
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}
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float AudioStreamMicrophone::get_length() const {
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return 0;
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}
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void AudioStreamMicrophone::_bind_methods() {
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}
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AudioStreamMicrophone::AudioStreamMicrophone() {
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}
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void AudioStreamPlaybackMicrophone::_mix_internal(AudioFrame *p_buffer, int p_frames) {
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AudioServer::get_singleton()->lock();
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PoolVector<int32_t> capture_buffer = AudioServer::get_singleton()->get_capture_buffer();
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unsigned int capture_size = AudioServer::get_singleton()->get_capture_size();
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int mix_rate = AudioServer::get_singleton()->get_mix_rate();
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unsigned int playback_delay = MIN(((50 * mix_rate) / 1000) * 2, capture_buffer.size() >> 1);
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#ifdef DEBUG_ENABLED
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unsigned int capture_position = AudioServer::get_singleton()->get_capture_position();
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#endif
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if (playback_delay > capture_size) {
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for (int i = 0; i < p_frames; i++) {
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p_buffer[i] = AudioFrame(0.0f, 0.0f);
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}
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capture_ofs = 0;
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} else {
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for (int i = 0; i < p_frames; i++) {
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if (capture_size > capture_ofs && (int)capture_ofs < capture_buffer.size()) {
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float l = (capture_buffer[capture_ofs++] >> 16) / 32768.f;
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if ((int)capture_ofs >= capture_buffer.size()) {
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capture_ofs = 0;
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}
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float r = (capture_buffer[capture_ofs++] >> 16) / 32768.f;
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if ((int)capture_ofs >= capture_buffer.size()) {
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capture_ofs = 0;
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}
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p_buffer[i] = AudioFrame(l, r);
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} else {
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p_buffer[i] = AudioFrame(0.0f, 0.0f);
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}
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}
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}
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#ifdef DEBUG_ENABLED
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if (capture_ofs > capture_position && (int)(capture_ofs - capture_position) < (p_frames * 2)) {
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print_verbose(String(get_class_name()) + " buffer underrun: capture_position=" + itos(capture_position) + " capture_ofs=" + itos(capture_ofs) + " capture_size=" + itos(capture_size));
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}
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#endif
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AudioServer::get_singleton()->unlock();
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}
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void AudioStreamPlaybackMicrophone::mix(AudioFrame *p_buffer, float p_rate_scale, int p_frames) {
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AudioStreamPlaybackResampled::mix(p_buffer, p_rate_scale, p_frames);
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}
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float AudioStreamPlaybackMicrophone::get_stream_sampling_rate() {
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return AudioDriver::get_singleton()->get_mix_rate();
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}
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void AudioStreamPlaybackMicrophone::start(float p_from_pos) {
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if (active) {
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return;
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}
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if (!GLOBAL_GET("audio/enable_audio_input")) {
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WARN_PRINTS("Need to enable Project settings > Audio > Enable Audio Input option to use capturing.");
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return;
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}
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capture_ofs = 0;
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if (AudioServer::get_singleton()->capture_start() == OK) {
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active = true;
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_begin_resample();
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}
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}
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void AudioStreamPlaybackMicrophone::stop() {
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if (active) {
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AudioServer::get_singleton()->capture_stop();
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active = false;
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}
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}
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bool AudioStreamPlaybackMicrophone::is_playing() const {
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return active;
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}
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int AudioStreamPlaybackMicrophone::get_loop_count() const {
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return 0;
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}
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float AudioStreamPlaybackMicrophone::get_playback_position() const {
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return 0;
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}
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void AudioStreamPlaybackMicrophone::seek(float p_time) {
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// Can't seek a microphone input
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}
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AudioStreamPlaybackMicrophone::~AudioStreamPlaybackMicrophone() {
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microphone->playbacks.erase(this);
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stop();
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}
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AudioStreamPlaybackMicrophone::AudioStreamPlaybackMicrophone() {
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}
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////////////////////////////////
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void AudioStreamRandomPitch::set_audio_stream(const Ref<AudioStream> &p_audio_stream) {
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audio_stream = p_audio_stream;
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if (audio_stream.is_valid()) {
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for (Set<AudioStreamPlaybackRandomPitch *>::Element *E = playbacks.front(); E; E = E->next()) {
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E->get()->playback = audio_stream->instance_playback();
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}
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}
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}
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Ref<AudioStream> AudioStreamRandomPitch::get_audio_stream() const {
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return audio_stream;
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}
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void AudioStreamRandomPitch::set_random_pitch(float p_pitch) {
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if (p_pitch < 1)
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p_pitch = 1;
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random_pitch = p_pitch;
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}
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float AudioStreamRandomPitch::get_random_pitch() const {
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return random_pitch;
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}
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Ref<AudioStreamPlayback> AudioStreamRandomPitch::instance_playback() {
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Ref<AudioStreamPlaybackRandomPitch> playback;
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playback.instance();
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if (audio_stream.is_valid())
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playback->playback = audio_stream->instance_playback();
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playbacks.insert(playback.ptr());
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playback->random_pitch = Ref<AudioStreamRandomPitch>((AudioStreamRandomPitch *)this);
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return playback;
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}
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String AudioStreamRandomPitch::get_stream_name() const {
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if (audio_stream.is_valid()) {
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return "Random: " + audio_stream->get_name();
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}
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return "RandomPitch";
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}
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float AudioStreamRandomPitch::get_length() const {
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if (audio_stream.is_valid()) {
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return audio_stream->get_length();
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}
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return 0;
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}
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void AudioStreamRandomPitch::_bind_methods() {
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ClassDB::bind_method(D_METHOD("set_audio_stream", "stream"), &AudioStreamRandomPitch::set_audio_stream);
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ClassDB::bind_method(D_METHOD("get_audio_stream"), &AudioStreamRandomPitch::get_audio_stream);
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ClassDB::bind_method(D_METHOD("set_random_pitch", "scale"), &AudioStreamRandomPitch::set_random_pitch);
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ClassDB::bind_method(D_METHOD("get_random_pitch"), &AudioStreamRandomPitch::get_random_pitch);
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ADD_PROPERTY(PropertyInfo(Variant::OBJECT, "audio_stream", PROPERTY_HINT_RESOURCE_TYPE, "AudioStream"), "set_audio_stream", "get_audio_stream");
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ADD_PROPERTY(PropertyInfo(Variant::REAL, "random_pitch", PROPERTY_HINT_RANGE, "1,16,0.01"), "set_random_pitch", "get_random_pitch");
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}
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AudioStreamRandomPitch::AudioStreamRandomPitch() {
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random_pitch = 1.1;
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}
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void AudioStreamPlaybackRandomPitch::start(float p_from_pos) {
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playing = playback;
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float range_from = 1.0 / random_pitch->random_pitch;
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float range_to = random_pitch->random_pitch;
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pitch_scale = range_from + Math::randf() * (range_to - range_from);
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if (playing.is_valid()) {
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playing->start(p_from_pos);
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}
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}
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void AudioStreamPlaybackRandomPitch::stop() {
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if (playing.is_valid()) {
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playing->stop();
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;
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}
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}
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bool AudioStreamPlaybackRandomPitch::is_playing() const {
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if (playing.is_valid()) {
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return playing->is_playing();
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}
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return false;
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}
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int AudioStreamPlaybackRandomPitch::get_loop_count() const {
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if (playing.is_valid()) {
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return playing->get_loop_count();
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}
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return 0;
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}
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float AudioStreamPlaybackRandomPitch::get_playback_position() const {
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if (playing.is_valid()) {
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return playing->get_playback_position();
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}
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return 0;
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}
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void AudioStreamPlaybackRandomPitch::seek(float p_time) {
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if (playing.is_valid()) {
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playing->seek(p_time);
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}
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}
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void AudioStreamPlaybackRandomPitch::mix(AudioFrame *p_buffer, float p_rate_scale, int p_frames) {
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if (playing.is_valid()) {
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playing->mix(p_buffer, p_rate_scale * pitch_scale, p_frames);
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} else {
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for (int i = 0; i < p_frames; i++) {
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p_buffer[i] = AudioFrame(0, 0);
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}
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}
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}
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AudioStreamPlaybackRandomPitch::~AudioStreamPlaybackRandomPitch() {
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random_pitch->playbacks.erase(this);
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}
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