workinf_Blender_Wasm/blender-5.2.0/extern/audaspace/plugins/libsndfile/SndFileWriter.cpp

/*******************************************************************************
 * Copyright 2009-2016 Jörg Müller
 *
 * Licensed under the Apache License, Version 2.0 (the "License");
 * you may not use this file except in compliance with the License.
 * You may obtain a copy of the License at
 *
 *   http://www.apache.org/licenses/LICENSE-2.0
 *
 * Unless required by applicable law or agreed to in writing, software
 * distributed under the License is distributed on an "AS IS" BASIS,
 * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
 * See the License for the specific language governing permissions and
 * limitations under the License.
 ******************************************************************************/

#include "SndFileWriter.h"
#include "Exception.h"

#include <cstring>

AUD_NAMESPACE_BEGIN

SndFileWriter::SndFileWriter(const std::string &filename, DeviceSpecs specs,
									 Container format, Codec codec, unsigned int bitrate) :
	m_position(0), m_specs(specs)
{
	SF_INFO sfinfo{};

	sfinfo.channels = specs.channels;
	sfinfo.samplerate = int(specs.rate);

	switch(format)
	{
#ifdef __EMSCRIPTEN__
	case CONTAINER_MP3:
		if(codec != CODEC_MP3)
			AUD_THROW(FileException, "The MP3 container requires the MP3 codec.");
		if(sfinfo.channels < 1 || sfinfo.channels > 2)
			AUD_THROW(FileException, "MP3 supports mono or stereo; use split channels for multichannel output.");
		switch(sfinfo.samplerate)
		{
		case 8000: case 11025: case 12000:
		case 16000: case 22050: case 24000:
		case 32000: case 44100: case 48000:
			break;
		default:
			AUD_THROW(FileException, "MP3 requires an 8, 11.025, 12, 16, 22.05, 24, 32, 44.1 or 48 kHz sample rate.");
		}
		{
			// CBR frame rates accepted by LAME. Reject unsupported requests rather
			// than letting the encoder silently round Blender's bitrate setting.
			const unsigned int mpeg1[] = {32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320};
			const unsigned int mpeg2[] = {8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160};
			const bool high_rate = sfinfo.samplerate >= 32000;
			const unsigned int* rates = high_rate ? mpeg1 : mpeg2;
			const unsigned int max_rate = high_rate ? 320 : sfinfo.samplerate >= 16000 ? 160 : 64;
			bool supported = false;
			for(unsigned int i = 0; i < 14; ++i)
				if(bitrate == rates[i] * 1000 && rates[i] <= max_rate)
					supported = true;
			if(!supported)
				AUD_THROW(FileException, "Unsupported MP3 constant bitrate for the selected sample rate (32-320 kbit/s at 32-48 kHz, up to 160 at 16-24 kHz, up to 64 below 16 kHz).");
		}
		sfinfo.format = SF_FORMAT_MPEG | SF_FORMAT_MPEG_LAYER_III;
		break;
#endif
	case CONTAINER_FLAC:
		sfinfo.format = SF_FORMAT_FLAC;
		switch(specs.format)
		{
		case FORMAT_S16:
			sfinfo.format |= SF_FORMAT_PCM_16;
			break;
		case FORMAT_S24:
			sfinfo.format |= SF_FORMAT_PCM_24;
			break;
		case FORMAT_S32:
			sfinfo.format |= SF_FORMAT_PCM_32;
			break;
		case FORMAT_FLOAT32:
			sfinfo.format |= SF_FORMAT_FLOAT;
			break;
		case FORMAT_FLOAT64:
			sfinfo.format |= SF_FORMAT_DOUBLE;
			break;
		default:
			sfinfo.format = 0;
			break;
		}
		break;
	case CONTAINER_OGG:
		if(codec == CODEC_VORBIS)
			sfinfo.format = SF_FORMAT_OGG | SF_FORMAT_VORBIS;
#ifdef __EMSCRIPTEN__
		else if(codec == CODEC_OPUS)
			sfinfo.format = SF_FORMAT_OGG | SF_FORMAT_OPUS;
#endif
		else
			sfinfo.format = 0;
		break;
	case CONTAINER_WAV:
		sfinfo.format = SF_FORMAT_WAV;
		switch(specs.format)
		{
		case FORMAT_U8:
			sfinfo.format |= SF_FORMAT_PCM_U8;
			break;
		case FORMAT_S16:
			sfinfo.format |= SF_FORMAT_PCM_16;
			break;
		case FORMAT_S24:
			sfinfo.format |= SF_FORMAT_PCM_24;
			break;
		case FORMAT_S32:
			sfinfo.format |= SF_FORMAT_PCM_32;
			break;
		case FORMAT_FLOAT32:
			sfinfo.format |= SF_FORMAT_FLOAT;
			break;
		case FORMAT_FLOAT64:
			sfinfo.format |= SF_FORMAT_DOUBLE;
			break;
		default:
			sfinfo.format = 0;
			break;
		}
		break;
	default:
		sfinfo.format = 0;
		break;
	}

	if(sfinfo.format == 0 || !sf_format_check(&sfinfo))
		AUD_THROW(FileException, "This format couldn't be written with libsndfile.");

	m_sndfile = sf_open(filename.c_str(), SFM_WRITE, &sfinfo);

	if(!m_sndfile)
		AUD_THROW(FileException, std::string("The file couldn't be written with libsndfile: ") + sf_strerror(nullptr));

#ifdef __EMSCRIPTEN__
	if(format == CONTAINER_OGG && codec == CODEC_VORBIS)
	{
		// Preserve the operator's bits/s for this output stream (each mono
		// stream in split-channel mode). libvorbis selects its real managed
		// average bitrate mode and rejects unsupported rate/channel tuples.
		int target_bitrate = int(bitrate);
		if(sf_command(m_sndfile, SFC_WEB_SET_VORBIS_BITRATE,
		              &target_bitrate, sizeof(target_bitrate)) != SF_TRUE)
		{
			sf_close(m_sndfile);
			m_sndfile = nullptr;
			AUD_THROW(FileException, "The native Vorbis encoder cannot use the requested average bitrate for this sample rate and channel count. Adjust Bitrate, Sample Rate or Channels (split channels encodes each file as mono).");
		}
	}
	if(format == CONTAINER_MP3)
	{
		// libsndfile exposes LAME bitrate through compression, with an integer
		// truncation in mpeg_l3_encoder_set_quality. Target the interior of that
		// integer interval so floating point rounding cannot subtract 1 kbit/s.
		const double maximum = sfinfo.samplerate >= 32000 ? 320.0 :
		                       sfinfo.samplerate >= 16000 ? 160.0 : 64.0;
		const double minimum = sfinfo.samplerate >= 32000 ? 32.0 : 8.0;
		const double kbps = bitrate / 1000;
		double compression = kbps == maximum ? 0.0 :
		                     (maximum - kbps - 0.25) / (maximum - minimum);
		int mode = SF_BITRATE_MODE_CONSTANT;
		// Set a valid compression first: changing mode reuses that value.
		if(sf_command(m_sndfile, SFC_SET_COMPRESSION_LEVEL, &compression, sizeof(compression)) != SF_TRUE ||
		   sf_command(m_sndfile, SFC_SET_BITRATE_MODE, &mode, sizeof(mode)) != SF_TRUE)
		{
			sf_close(m_sndfile);
			m_sndfile = nullptr;
			AUD_THROW(FileException, "The native MP3 encoder could not set the requested constant bitrate.");
		}
	}
	if(format == CONTAINER_OGG && codec == CODEC_OPUS)
	{
		// libsndfile 1.2.2 maps compression to 6..256 kbit/s per channel.
		// Invert that native mapping to preserve Blender's target bitrate.
		double per_channel = double(bitrate) / sfinfo.channels;
		double compression = 1.0 - (per_channel - 6000.0) / 250000.0;
		if(per_channel < 6000.0 || per_channel > 256000.0 ||
		   sf_command(m_sndfile, SFC_SET_COMPRESSION_LEVEL, &compression, sizeof(compression)) != SF_TRUE)
		{
			sf_close(m_sndfile);
			m_sndfile = nullptr;
			AUD_THROW(FileException, "Opus bitrate must be between 6 and 256 kbit/s per output channel.");
		}
	}
#endif
}

SndFileWriter::~SndFileWriter()
{
	if(m_sndfile)
		sf_close(m_sndfile);
}

int SndFileWriter::getPosition() const
{
	return m_position;
}

DeviceSpecs SndFileWriter::getSpecs() const
{
	return m_specs;
}

void SndFileWriter::write(unsigned int length, sample_t* buffer)
{
	if(!m_sndfile)
		AUD_THROW(FileException, "Cannot write to a finished audio file.");
	sf_count_t written = sf_writef_float(m_sndfile, buffer, length);
	if(written > 0)
		m_position += written;
	if(written != length)
		AUD_THROW(FileException, std::string("Audio file write failed: ") + sf_strerror(m_sndfile));
}

void SndFileWriter::finish()
{
	if(!m_sndfile)
		return;
	SNDFILE* file = m_sndfile;
	m_sndfile = nullptr;
	int error = sf_close(file);
	if(error)
		AUD_THROW(FileException, std::string("Audio encoding finalization failed: ") + sf_error_number(error));
}

AUD_NAMESPACE_END