oak-editor/app/audio/audioprocessor.cpp

/*
 * Oak Video Editor - Non-Linear Video Editor
 * Copyright (C) 2025 Olive CE Team
 *
 * This program is free software: you can redistribute it and/or modify
 * it under the terms of the GNU General Public License as published by
 * the Free Software Foundation, either version 3 of the License, or
 * (at your option) any later version.
 *
 * This program is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 * GNU General Public License for more details.
 *
 * You should have received a copy of the GNU General Public License
 * along with this program.  If not, see <http://www.gnu.org/licenses/>.
 */

#include "audioprocessor.h"

extern "C" {
#include <libavfilter/buffersrc.h>
#include <libavfilter/buffersink.h>
}

#include <QDebug>

#include "common/ffmpegutils.h"

namespace olive
{

AudioProcessor::AudioProcessor()
{
	filter_graph_ = nullptr;
	in_frame_ = nullptr;
	out_frame_ = nullptr;
}

AudioProcessor::~AudioProcessor()
{
	Close();
}

bool AudioProcessor::Open(const AudioParams &from, const AudioParams &to,
						  double tempo)
{
	if (filter_graph_) {
		qWarning() << "Tried to open a processor that was already open";
		return false;
	}

	filter_graph_ = avfilter_graph_alloc();
	if (!filter_graph_) {
		qCritical() << "Failed to allocate filter graph";
		return false;
	}

	from_fmt_ = FFmpegUtils::GetFFmpegSampleFormat(from.format());
	to_fmt_ = FFmpegUtils::GetFFmpegSampleFormat(to.format());

	// Set up audio buffer args
	char filter_args[200];
	snprintf(
		filter_args, 200,
		"time_base=%d/%d:sample_rate=%d:sample_fmt=%d:channel_layout=0x%" PRIx64,
		1, from.sample_rate(), from.sample_rate(), from_fmt_,
		from.channel_layout().u.mask);

	int r;

	// Create buffersrc (input)
	r = avfilter_graph_create_filter(&buffersrc_ctx_,
									 avfilter_get_by_name("abuffer"), "in",
									 filter_args, nullptr, filter_graph_);
	if (r < 0) {
		qCritical() << "Failed to create buffersrc:" << r;
		Close();
		return false;
	}

	// Store "previous" filter for linking
	AVFilterContext *previous_filter = buffersrc_ctx_;

	// Create tempo
	bool create_tempo;
	if ((create_tempo = !qFuzzyCompare(tempo, 1.0))) {
		// Create audio tempo filters: FFmpeg's atempo can only be set between 0.5 and 2.0. If the requested speed is outside
		// those boundaries, we need to daisychain more than one together.
		double base = (tempo > 1.0) ? 2.0 : 0.5;
		double speed_log = log(tempo) / log(base);

		// This is the number of how many 0.5 or 2.0 tempos we need to daisychain
		int whole = std::floor(speed_log);

		// Set speed_log to the remainder
		speed_log -= whole;

		for (int i = 0; i <= whole; i++) {
			double filter_tempo = (i == whole) ? std::pow(base, speed_log) :
												 base;
			/*
      if (qFuzzyCompare(filter_tempo, 1.0)) {
        // This filter would do nothing
        continue;
      }*/

			previous_filter =
				CreateTempoFilter(filter_graph_, previous_filter, filter_tempo);

			if (!previous_filter) {
				qCritical() << "Failed to create audio tempo filter";
				Close();
				return false;
			}
		}
	}

	// Create conversion filter
	auto ch1 = from.channel_layout();
	auto ch2 = to.channel_layout();
	if (from.sample_rate() != to.sample_rate() ||
		av_channel_layout_compare(&ch1, &ch2) || from.format() != to.format() ||
		(to.format().is_planar() &&
		 create_tempo)) { // Tempo processor automatically converts to packed,
		// so if the desired output is planar, it'll need
		// to be converted
		snprintf(filter_args, 200,
				 "sample_fmts=%s:sample_rates=%d:channel_layouts=0x%" PRIx64,
				 av_get_sample_fmt_name(to_fmt_), to.sample_rate(),
				 to.channel_layout().u.mask);

		AVFilterContext *c;
		r = avfilter_graph_create_filter(&c, avfilter_get_by_name("aformat"),
										 "fmt", filter_args, nullptr,
										 filter_graph_);
		if (r < 0) {
			qCritical() << "Failed to create format conversion filter:" << r
						<< filter_args;
			Close();
			return false;
		}

		r = avfilter_link(previous_filter, 0, c, 0);
		if (r < 0) {
			qCritical() << "Failed to link filters:" << r;
			Close();
			return false;
		}

		previous_filter = c;
	}

	// Create buffersink (output)
	r = avfilter_graph_create_filter(&buffersink_ctx_,
									 avfilter_get_by_name("abuffersink"), "out",
									 nullptr, nullptr, filter_graph_);
	if (r < 0) {
		qCritical() << "Failed to create buffersink:" << r;
		Close();
		return false;
	}

	r = avfilter_link(previous_filter, 0, buffersink_ctx_, 0);
	if (r < 0) {
		qCritical() << "Failed to link filters:" << r;
		Close();
		return false;
	}
	char *dump = avfilter_graph_dump(filter_graph_, nullptr);
	qDebug() << dump;
	av_free(dump);
	r = avfilter_graph_config(filter_graph_, nullptr);
	if (r < 0) {
		qCritical() << "Failed to configure graph:" << r;
		Close();
		return false;
	}

	in_frame_ = av_frame_alloc();
	if (in_frame_) {
		in_frame_->sample_rate = from.sample_rate();
		in_frame_->format = from_fmt_;
		in_frame_->ch_layout = from.channel_layout();
		in_frame_->pts = 0;
	} else {
		qCritical() << "Failed to allocate input frame";
		Close();
		return false;
	}

	out_frame_ = av_frame_alloc();
	if (!out_frame_) {
		qCritical() << "Failed to allocate output frame";
		Close();
		return false;
	}

	from_ = from;
	to_ = to;

	return true;
}

void AudioProcessor::Close()
{
	if (filter_graph_) {
		avfilter_graph_free(&filter_graph_);
		filter_graph_ = nullptr;
		buffersrc_ctx_ = nullptr;
		buffersink_ctx_ = nullptr;
	}

	if (in_frame_) {
		av_frame_free(&in_frame_);
		in_frame_ = nullptr;
	}

	if (out_frame_) {
		av_frame_free(&out_frame_);
		out_frame_ = nullptr;
	}
}

int AudioProcessor::Convert(float **in, int nb_in_samples,
							AudioProcessor::Buffer *output)
{
	if (!IsOpen()) {
		qCritical() << "Tried to convert on closed processor";
		return -1;
	}

	int r = 0;

	if (in && nb_in_samples) {
		// Set frame parameters
		in_frame_->nb_samples = nb_in_samples;
		for (int i = 0; i < from_.channel_count(); i++) {
			in_frame_->data[i] = reinterpret_cast<uint8_t *>(in[i]);
			in_frame_->linesize[i] = from_.samples_to_bytes(nb_in_samples);
		}

		r = av_buffersrc_add_frame_flags(buffersrc_ctx_, in_frame_,
										 AV_BUFFERSRC_FLAG_KEEP_REF);
		if (r < 0) {
			qCritical() << "Failed to add frame to buffersrc:" << r;
			return r;
		}
	}

	if (output) {
		int nb_channels = to_.channel_count();

		if (to_.format().is_packed()) {
			nb_channels = 1;
		}

		AudioProcessor::Buffer &result = *output;
		result.resize(nb_channels);

		int byte_offset = 0;

		while (true) {
			av_frame_unref(out_frame_);
			r = av_buffersink_get_frame(buffersink_ctx_, out_frame_);
			if (r < 0) {
				if (r == AVERROR(EAGAIN)) {
					r = 0;
				} else {
					// Handle unexpected error
					qCritical() << "Failed to pull from buffersink:" << r;
				}
				break;
			}

			int nb_bytes =
				out_frame_->nb_samples * to_.bytes_per_sample_per_channel();
			if (to_.format().is_packed()) {
				nb_bytes *= to_.channel_count();
			}

			for (int i = 0; i < nb_channels; i++) {
				result[i].resize(byte_offset + nb_bytes);
				memcpy(result[i].data() + byte_offset, out_frame_->data[i],
					   nb_bytes);
			}
			byte_offset += nb_bytes;
		}
		av_frame_unref(out_frame_);
	}

	return r;
}

void AudioProcessor::Flush()
{
	int r = av_buffersrc_add_frame_flags(buffersrc_ctx_, nullptr,
										 AV_BUFFERSRC_FLAG_KEEP_REF);
	if (r < 0) {
		qCritical() << "Failed to flush:" << r;
	}
}

AVFilterContext *AudioProcessor::CreateTempoFilter(AVFilterGraph *graph,
												   AVFilterContext *link,
												   const double &tempo)
{
	// Set up tempo param, which is taken as a C string
	char speed_param[20];
	snprintf(speed_param, 20, "%f", tempo);

	AVFilterContext *tempo_ctx = nullptr;

	if (avfilter_graph_create_filter(&tempo_ctx, avfilter_get_by_name("atempo"),
									 "atempo", speed_param, nullptr,
									 graph) >= 0 &&
		avfilter_link(link, 0, tempo_ctx, 0) == 0) {
		return tempo_ctx;
	}

	return nullptr;
}

}