godot/servers/audio/audio_rb_resampler.cpp

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/*  audio_rb_resampler.cpp                                               */
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/* Copyright (c) 2007-2018 Juan Linietsky, Ariel Manzur.                 */
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#include "audio_rb_resampler.h"
#include "core/math/math_funcs.h"
#include "os/os.h"
#include "servers/audio_server.h"

int AudioRBResampler::get_channel_count() const {

	if (!rb)
		return 0;

	return channels;
}

// Linear interpolation based sample rate conversion (low quality)
// Note that AudioStreamPlaybackResampled::mix has better algorithm,
// but it wasn't obvious to integrate that with VideoPlayer
template <int C>
uint32_t AudioRBResampler::_resample(AudioFrame *p_dest, int p_todo, int32_t p_increment) {

	uint32_t read = offset & MIX_FRAC_MASK;

	for (int i = 0; i < p_todo; i++) {

		offset = (offset + p_increment) & (((1 << (rb_bits + MIX_FRAC_BITS)) - 1));
		read += p_increment;
		uint32_t pos = offset >> MIX_FRAC_BITS;
		float frac = float(offset & MIX_FRAC_MASK) / float(MIX_FRAC_LEN);
		ERR_FAIL_COND_V(pos >= rb_len, 0);
		uint32_t pos_next = (pos + 1) & rb_mask;

		// since this is a template with a known compile time value (C), conditionals go away when compiling.
		if (C == 1) {

			float v0 = rb[pos];
			float v0n = rb[pos_next];
			v0 += (v0n - v0) * frac;
			p_dest[i] = AudioFrame(v0, v0);
		}

		if (C == 2) {

			float v0 = rb[(pos << 1) + 0];
			float v1 = rb[(pos << 1) + 1];
			float v0n = rb[(pos_next << 1) + 0];
			float v1n = rb[(pos_next << 1) + 1];

			v0 += (v0n - v0) * frac;
			v1 += (v1n - v1) * frac;
			p_dest[i] = AudioFrame(v0, v1);
		}

		// For now, channels higher than stereo are almost ignored
		if (C == 4) {

			float v0 = rb[(pos << 2) + 0];
			float v1 = rb[(pos << 2) + 1];
			float v2 = rb[(pos << 2) + 2];
			float v3 = rb[(pos << 2) + 3];
			float v0n = rb[(pos_next << 2) + 0];
			float v1n = rb[(pos_next << 2) + 1];
			float v2n = rb[(pos_next << 2) + 2];
			float v3n = rb[(pos_next << 2) + 3];

			v0 += (v0n - v0) * frac;
			v1 += (v1n - v1) * frac;
			v2 += (v2n - v2) * frac;
			v3 += (v3n - v3) * frac;
			p_dest[i] = AudioFrame(v0, v1);
		}

		if (C == 6) {

			// FIXME: Lot of unused assignments here, but it seems like intermediate calculations
			// should be done as for C == 2 (C == 4 also has some unused assignments).
			float v0 = rb[(pos * 6) + 0];
			float v1 = rb[(pos * 6) + 1];
			float v2 = rb[(pos * 6) + 2];
			float v3 = rb[(pos * 6) + 3];
			float v4 = rb[(pos * 6) + 4];
			float v5 = rb[(pos * 6) + 5];
			float v0n = rb[(pos_next * 6) + 0];
			float v1n = rb[(pos_next * 6) + 1];
			float v2n = rb[(pos_next * 6) + 2];
			float v3n = rb[(pos_next * 6) + 3];
			float v4n = rb[(pos_next * 6) + 4];
			float v5n = rb[(pos_next * 6) + 5];

			p_dest[i] = AudioFrame(v0, v1);
		}
	}

	return read >> MIX_FRAC_BITS; //rb_read_pos = offset >> MIX_FRAC_BITS;
}

bool AudioRBResampler::mix(AudioFrame *p_dest, int p_frames) {

	if (!rb)
		return false;

	int32_t increment = (src_mix_rate * MIX_FRAC_LEN) / target_mix_rate;
	int read_space = get_reader_space();
	int target_todo = MIN(get_num_of_ready_frames(), p_frames);

	{
		int src_read = 0;
		switch (channels) {
			case 1: src_read = _resample<1>(p_dest, target_todo, increment); break;
			case 2: src_read = _resample<2>(p_dest, target_todo, increment); break;
			case 4: src_read = _resample<4>(p_dest, target_todo, increment); break;
			case 6: src_read = _resample<6>(p_dest, target_todo, increment); break;
		}

		if (src_read > read_space)
			src_read = read_space;

		rb_read_pos = (rb_read_pos + src_read) & rb_mask;

		// Create fadeout effect for the end of stream (note that it can be because of slow writer)
		if (p_frames - target_todo > 0) {
			for (int i = 0; i < target_todo; i++) {
				p_dest[i] = p_dest[i] * float(target_todo - i) / float(target_todo);
			}
		}

		// Fill zeros (silence) for the rest of frames
		for (uint32_t i = target_todo; i < p_frames; i++) {
			p_dest[i] = AudioFrame(0, 0);
		}
	}

	return true;
}

int AudioRBResampler::get_num_of_ready_frames() {
	if (!is_ready())
		return 0;
	int32_t increment = (src_mix_rate * MIX_FRAC_LEN) / target_mix_rate;
	int read_space = get_reader_space();
	return (int64_t(read_space) << MIX_FRAC_BITS) / increment;
}

Error AudioRBResampler::setup(int p_channels, int p_src_mix_rate, int p_target_mix_rate, int p_buffer_msec, int p_minbuff_needed) {

	ERR_FAIL_COND_V(p_channels != 1 && p_channels != 2 && p_channels != 4 && p_channels != 6, ERR_INVALID_PARAMETER);

	int desired_rb_bits = nearest_shift(MAX((p_buffer_msec / 1000.0) * p_src_mix_rate, p_minbuff_needed));

	bool recreate = !rb;

	if (rb && (uint32_t(desired_rb_bits) != rb_bits || channels != uint32_t(p_channels))) {

		memdelete_arr(rb);
		memdelete_arr(read_buf);
		recreate = true;
	}

	if (recreate) {

		channels = p_channels;
		rb_bits = desired_rb_bits;
		rb_len = (1 << rb_bits);
		rb_mask = rb_len - 1;
		rb = memnew_arr(float, rb_len *p_channels);
		read_buf = memnew_arr(float, rb_len *p_channels);
	}

	src_mix_rate = p_src_mix_rate;
	target_mix_rate = p_target_mix_rate;
	offset = 0;
	rb_read_pos = 0;
	rb_write_pos = 0;

	//avoid maybe strange noises upon load
	for (unsigned int i = 0; i < (rb_len * channels); i++) {

		rb[i] = 0;
		read_buf[i] = 0;
	}

	return OK;
}

void AudioRBResampler::clear() {

	if (!rb)
		return;

	//should be stopped at this point but just in case
	if (rb) {
		memdelete_arr(rb);
		memdelete_arr(read_buf);
	}
	rb = NULL;
	offset = 0;
	rb_read_pos = 0;
	rb_write_pos = 0;
	read_buf = NULL;
}

AudioRBResampler::AudioRBResampler() {

	rb = NULL;
	offset = 0;
	read_buf = NULL;
	rb_read_pos = 0;
	rb_write_pos = 0;

	rb_bits = 0;
	rb_len = 0;
	rb_mask = 0;
	read_buff_len = 0;
	channels = 0;
	src_mix_rate = 0;
	target_mix_rate = 0;
}

AudioRBResampler::~AudioRBResampler() {

	if (rb) {
		memdelete_arr(rb);
		memdelete_arr(read_buf);
	}
}