Encoder*: move to src/encoder

.. and move the plugins to src/encoder/plugins/.

src/encoder/plugins/FlacEncoderPlugin.cxx

@@ -0,0 +1,325 @@
+/*
+ * Copyright (C) 2003-2014 The Music Player Daemon Project
+ * http://www.musicpd.org
+ *
+ * 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 2 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, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include "config.h"
+#include "FlacEncoderPlugin.hxx"
+#include "../EncoderAPI.hxx"
+#include "AudioFormat.hxx"
+#include "pcm/PcmBuffer.hxx"
+#include "ConfigError.hxx"
+#include "util/Manual.hxx"
+#include "util/DynamicFifoBuffer.hxx"
+#include "util/Error.hxx"
+#include "util/Domain.hxx"
+
+#include <FLAC/stream_encoder.h>
+
+#if !defined(FLAC_API_VERSION_CURRENT) || FLAC_API_VERSION_CURRENT <= 7
+#error libFLAC is too old
+#endif
+
+struct flac_encoder {
+	Encoder encoder;
+
+	AudioFormat audio_format;
+	unsigned compression;
+
+	FLAC__StreamEncoder *fse;
+
+	PcmBuffer expand_buffer;
+
+	/**
+	 * This buffer will hold encoded data from libFLAC until it is
+	 * picked up with flac_encoder_read().
+	 */
+	Manual<DynamicFifoBuffer<uint8_t>> output_buffer;
+
+	flac_encoder():encoder(flac_encoder_plugin) {}
+};
+
+static constexpr Domain flac_encoder_domain("vorbis_encoder");
+
+static bool
+flac_encoder_configure(struct flac_encoder *encoder, const config_param &param,
+		       gcc_unused Error &error)
+{
+	encoder->compression = param.GetBlockValue("compression", 5u);
+
+	return true;
+}
+
+static Encoder *
+flac_encoder_init(const config_param &param, Error &error)
+{
+	flac_encoder *encoder = new flac_encoder();
+
+	/* load configuration from "param" */
+	if (!flac_encoder_configure(encoder, param, error)) {
+		/* configuration has failed, roll back and return error */
+		delete encoder;
+		return nullptr;
+	}
+
+	return &encoder->encoder;
+}
+
+static void
+flac_encoder_finish(Encoder *_encoder)
+{
+	struct flac_encoder *encoder = (struct flac_encoder *)_encoder;
+
+	/* the real libFLAC cleanup was already performed by
+	   flac_encoder_close(), so no real work here */
+	delete encoder;
+}
+
+static bool
+flac_encoder_setup(struct flac_encoder *encoder, unsigned bits_per_sample,
+		   Error &error)
+{
+	if ( !FLAC__stream_encoder_set_compression_level(encoder->fse,
+					encoder->compression)) {
+		error.Format(config_domain,
+			     "error setting flac compression to %d",
+			     encoder->compression);
+		return false;
+	}
+
+	if ( !FLAC__stream_encoder_set_channels(encoder->fse,
+					encoder->audio_format.channels)) {
+		error.Format(config_domain,
+			     "error setting flac channels num to %d",
+			     encoder->audio_format.channels);
+		return false;
+	}
+	if ( !FLAC__stream_encoder_set_bits_per_sample(encoder->fse,
+							bits_per_sample)) {
+		error.Format(config_domain,
+			     "error setting flac bit format to %d",
+			     bits_per_sample);
+		return false;
+	}
+	if ( !FLAC__stream_encoder_set_sample_rate(encoder->fse,
+					encoder->audio_format.sample_rate)) {
+		error.Format(config_domain,
+			     "error setting flac sample rate to %d",
+			     encoder->audio_format.sample_rate);
+		return false;
+	}
+	return true;
+}
+
+static FLAC__StreamEncoderWriteStatus
+flac_write_callback(gcc_unused const FLAC__StreamEncoder *fse,
+		    const FLAC__byte data[],
+		    size_t bytes,
+		    gcc_unused unsigned samples,
+		    gcc_unused unsigned current_frame, void *client_data)
+{
+	struct flac_encoder *encoder = (struct flac_encoder *) client_data;
+
+	//transfer data to buffer
+	encoder->output_buffer->Append((const uint8_t *)data, bytes);
+
+	return FLAC__STREAM_ENCODER_WRITE_STATUS_OK;
+}
+
+static void
+flac_encoder_close(Encoder *_encoder)
+{
+	struct flac_encoder *encoder = (struct flac_encoder *)_encoder;
+
+	FLAC__stream_encoder_delete(encoder->fse);
+
+	encoder->expand_buffer.Clear();
+	encoder->output_buffer.Destruct();
+}
+
+static bool
+flac_encoder_open(Encoder *_encoder, AudioFormat &audio_format, Error &error)
+{
+	struct flac_encoder *encoder = (struct flac_encoder *)_encoder;
+	unsigned bits_per_sample;
+
+	encoder->audio_format = audio_format;
+
+	/* FIXME: flac should support 32bit as well */
+	switch (audio_format.format) {
+	case SampleFormat::S8:
+		bits_per_sample = 8;
+		break;
+
+	case SampleFormat::S16:
+		bits_per_sample = 16;
+		break;
+
+	case SampleFormat::S24_P32:
+		bits_per_sample = 24;
+		break;
+
+	default:
+		bits_per_sample = 24;
+		audio_format.format = SampleFormat::S24_P32;
+	}
+
+	/* allocate the encoder */
+	encoder->fse = FLAC__stream_encoder_new();
+	if (encoder->fse == nullptr) {
+		error.Set(flac_encoder_domain, "flac_new() failed");
+		return false;
+	}
+
+	if (!flac_encoder_setup(encoder, bits_per_sample, error)) {
+		FLAC__stream_encoder_delete(encoder->fse);
+		return false;
+	}
+
+	encoder->output_buffer.Construct(8192);
+
+	/* this immediately outputs data through callback */
+
+	{
+		FLAC__StreamEncoderInitStatus init_status;
+
+		init_status = FLAC__stream_encoder_init_stream(encoder->fse,
+			    flac_write_callback,
+			    nullptr, nullptr, nullptr, encoder);
+
+		if(init_status != FLAC__STREAM_ENCODER_INIT_STATUS_OK) {
+			error.Format(flac_encoder_domain,
+				     "failed to initialize encoder: %s\n",
+				     FLAC__StreamEncoderInitStatusString[init_status]);
+			flac_encoder_close(_encoder);
+			return false;
+		}
+	}
+
+	return true;
+}
+
+
+static bool
+flac_encoder_flush(Encoder *_encoder, gcc_unused Error &error)
+{
+	struct flac_encoder *encoder = (struct flac_encoder *)_encoder;
+
+	(void) FLAC__stream_encoder_finish(encoder->fse);
+	return true;
+}
+
+static inline void
+pcm8_to_flac(int32_t *out, const int8_t *in, unsigned num_samples)
+{
+	while (num_samples > 0) {
+		*out++ = *in++;
+		--num_samples;
+	}
+}
+
+static inline void
+pcm16_to_flac(int32_t *out, const int16_t *in, unsigned num_samples)
+{
+	while (num_samples > 0) {
+		*out++ = *in++;
+		--num_samples;
+	}
+}
+
+static bool
+flac_encoder_write(Encoder *_encoder,
+		   const void *data, size_t length,
+		   gcc_unused Error &error)
+{
+	struct flac_encoder *encoder = (struct flac_encoder *)_encoder;
+	unsigned num_frames, num_samples;
+	void *exbuffer;
+	const void *buffer = nullptr;
+
+	/* format conversion */
+
+	num_frames = length / encoder->audio_format.GetFrameSize();
+	num_samples = num_frames * encoder->audio_format.channels;
+
+	switch (encoder->audio_format.format) {
+	case SampleFormat::S8:
+		exbuffer = encoder->expand_buffer.Get(length * 4);
+		pcm8_to_flac((int32_t *)exbuffer, (const int8_t *)data,
+			     num_samples);
+		buffer = exbuffer;
+		break;
+
+	case SampleFormat::S16:
+		exbuffer = encoder->expand_buffer.Get(length * 2);
+		pcm16_to_flac((int32_t *)exbuffer, (const int16_t *)data,
+			      num_samples);
+		buffer = exbuffer;
+		break;
+
+	case SampleFormat::S24_P32:
+	case SampleFormat::S32:
+		/* nothing need to be done; format is the same for
+		   both mpd and libFLAC */
+		buffer = data;
+		break;
+
+	default:
+		gcc_unreachable();
+	}
+
+	/* feed samples to encoder */
+
+	if (!FLAC__stream_encoder_process_interleaved(encoder->fse,
+						      (const FLAC__int32 *)buffer,
+						      num_frames)) {
+		error.Set(flac_encoder_domain, "flac encoder process failed");
+		return false;
+	}
+
+	return true;
+}
+
+static size_t
+flac_encoder_read(Encoder *_encoder, void *dest, size_t length)
+{
+	struct flac_encoder *encoder = (struct flac_encoder *)_encoder;
+
+	return encoder->output_buffer->Read((uint8_t *)dest, length);
+}
+
+static const char *
+flac_encoder_get_mime_type(gcc_unused Encoder *_encoder)
+{
+	return "audio/flac";
+}
+
+const EncoderPlugin flac_encoder_plugin = {
+	"flac",
+	flac_encoder_init,
+	flac_encoder_finish,
+	flac_encoder_open,
+	flac_encoder_close,
+	flac_encoder_flush,
+	flac_encoder_flush,
+	nullptr,
+	nullptr,
+	flac_encoder_write,
+	flac_encoder_read,
+	flac_encoder_get_mime_type,
+};
+

src/encoder/plugins/FlacEncoderPlugin.hxx

@@ -0,0 +1,25 @@
+/*
+ * Copyright (C) 2003-2014 The Music Player Daemon Project
+ * http://www.musicpd.org
+ *
+ * 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 2 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, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#ifndef MPD_ENCODER_FLAC_HXX
+#define MPD_ENCODER_FLAC_HXX
+
+extern const struct EncoderPlugin flac_encoder_plugin;
+
+#endif

src/encoder/plugins/LameEncoderPlugin.cxx

@@ -0,0 +1,293 @@
+/*
+ * Copyright (C) 2003-2014 The Music Player Daemon Project
+ * http://www.musicpd.org
+ *
+ * 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 2 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, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include "config.h"
+#include "LameEncoderPlugin.hxx"
+#include "../EncoderAPI.hxx"
+#include "AudioFormat.hxx"
+#include "ConfigError.hxx"
+#include "util/NumberParser.hxx"
+#include "util/ReusableArray.hxx"
+#include "util/Manual.hxx"
+#include "util/Error.hxx"
+#include "util/Domain.hxx"
+
+#include <lame/lame.h>
+
+#include <assert.h>
+#include <string.h>
+
+struct LameEncoder final {
+	Encoder encoder;
+
+	AudioFormat audio_format;
+	float quality;
+	int bitrate;
+
+	lame_global_flags *gfp;
+
+	Manual<ReusableArray<unsigned char, 32768>> output_buffer;
+	unsigned char *output_begin, *output_end;
+
+	LameEncoder():encoder(lame_encoder_plugin) {}
+
+	bool Configure(const config_param &param, Error &error);
+};
+
+static constexpr Domain lame_encoder_domain("lame_encoder");
+
+bool
+LameEncoder::Configure(const config_param &param, Error &error)
+{
+	const char *value;
+	char *endptr;
+
+	value = param.GetBlockValue("quality");
+	if (value != nullptr) {
+		/* a quality was configured (VBR) */
+
+		quality = ParseDouble(value, &endptr);
+
+		if (*endptr != '\0' || quality < -1.0 || quality > 10.0) {
+			error.Format(config_domain,
+				     "quality \"%s\" is not a number in the "
+				     "range -1 to 10",
+				     value);
+			return false;
+		}
+
+		if (param.GetBlockValue("bitrate") != nullptr) {
+			error.Set(config_domain,
+				  "quality and bitrate are both defined");
+			return false;
+		}
+	} else {
+		/* a bit rate was configured */
+
+		value = param.GetBlockValue("bitrate");
+		if (value == nullptr) {
+			error.Set(config_domain,
+				  "neither bitrate nor quality defined");
+			return false;
+		}
+
+		quality = -2.0;
+		bitrate = ParseInt(value, &endptr);
+
+		if (*endptr != '\0' || bitrate <= 0) {
+			error.Set(config_domain,
+				  "bitrate should be a positive integer");
+			return false;
+		}
+	}
+
+	return true;
+}
+
+static Encoder *
+lame_encoder_init(const config_param &param, Error &error)
+{
+	LameEncoder *encoder = new LameEncoder();
+
+	/* load configuration from "param" */
+	if (!encoder->Configure(param, error)) {
+		/* configuration has failed, roll back and return error */
+		delete encoder;
+		return nullptr;
+	}
+
+	return &encoder->encoder;
+}
+
+static void
+lame_encoder_finish(Encoder *_encoder)
+{
+	LameEncoder *encoder = (LameEncoder *)_encoder;
+
+	/* the real liblame cleanup was already performed by
+	   lame_encoder_close(), so no real work here */
+	delete encoder;
+}
+
+static bool
+lame_encoder_setup(LameEncoder *encoder, Error &error)
+{
+	if (encoder->quality >= -1.0) {
+		/* a quality was configured (VBR) */
+
+		if (0 != lame_set_VBR(encoder->gfp, vbr_rh)) {
+			error.Set(lame_encoder_domain,
+				  "error setting lame VBR mode");
+			return false;
+		}
+		if (0 != lame_set_VBR_q(encoder->gfp, encoder->quality)) {
+			error.Set(lame_encoder_domain,
+				  "error setting lame VBR quality");
+			return false;
+		}
+	} else {
+		/* a bit rate was configured */
+
+		if (0 != lame_set_brate(encoder->gfp, encoder->bitrate)) {
+			error.Set(lame_encoder_domain,
+				  "error setting lame bitrate");
+			return false;
+		}
+	}
+
+	if (0 != lame_set_num_channels(encoder->gfp,
+				       encoder->audio_format.channels)) {
+		error.Set(lame_encoder_domain,
+			  "error setting lame num channels");
+		return false;
+	}
+
+	if (0 != lame_set_in_samplerate(encoder->gfp,
+					encoder->audio_format.sample_rate)) {
+		error.Set(lame_encoder_domain,
+			  "error setting lame sample rate");
+		return false;
+	}
+
+	if (0 != lame_set_out_samplerate(encoder->gfp,
+					 encoder->audio_format.sample_rate)) {
+		error.Set(lame_encoder_domain,
+			  "error setting lame out sample rate");
+		return false;
+	}
+
+	if (0 > lame_init_params(encoder->gfp)) {
+		error.Set(lame_encoder_domain,
+			  "error initializing lame params");
+		return false;
+	}
+
+	return true;
+}
+
+static bool
+lame_encoder_open(Encoder *_encoder, AudioFormat &audio_format, Error &error)
+{
+	LameEncoder *encoder = (LameEncoder *)_encoder;
+
+	audio_format.format = SampleFormat::S16;
+	audio_format.channels = 2;
+
+	encoder->audio_format = audio_format;
+
+	encoder->gfp = lame_init();
+	if (encoder->gfp == nullptr) {
+		error.Set(lame_encoder_domain, "lame_init() failed");
+		return false;
+	}
+
+	if (!lame_encoder_setup(encoder, error)) {
+		lame_close(encoder->gfp);
+		return false;
+	}
+
+	encoder->output_buffer.Construct();
+	encoder->output_begin = encoder->output_end = nullptr;
+
+	return true;
+}
+
+static void
+lame_encoder_close(Encoder *_encoder)
+{
+	LameEncoder *encoder = (LameEncoder *)_encoder;
+
+	lame_close(encoder->gfp);
+	encoder->output_buffer.Destruct();
+}
+
+static bool
+lame_encoder_write(Encoder *_encoder,
+		   const void *data, size_t length,
+		   gcc_unused Error &error)
+{
+	LameEncoder *encoder = (LameEncoder *)_encoder;
+	const int16_t *src = (const int16_t*)data;
+
+	assert(encoder->output_begin == encoder->output_end);
+
+	const unsigned num_frames =
+		length / encoder->audio_format.GetFrameSize();
+	const unsigned num_samples =
+		length / encoder->audio_format.GetSampleSize();
+
+	/* worst-case formula according to LAME documentation */
+	const size_t output_buffer_size = 5 * num_samples / 4 + 7200;
+	const auto output_buffer = encoder->output_buffer->Get(output_buffer_size);
+
+	/* this is for only 16-bit audio */
+
+	int bytes_out = lame_encode_buffer_interleaved(encoder->gfp,
+						       const_cast<short *>(src),
+						       num_frames,
+						       output_buffer,
+						       output_buffer_size);
+
+	if (bytes_out < 0) {
+		error.Set(lame_encoder_domain, "lame encoder failed");
+		return false;
+	}
+
+	encoder->output_begin = output_buffer;
+	encoder->output_end = output_buffer + bytes_out;
+	return true;
+}
+
+static size_t
+lame_encoder_read(Encoder *_encoder, void *dest, size_t length)
+{
+	LameEncoder *encoder = (LameEncoder *)_encoder;
+
+	const auto begin = encoder->output_begin;
+	assert(begin <= encoder->output_end);
+	const size_t remainning = encoder->output_end - begin;
+	if (length > remainning)
+		length = remainning;
+
+	memcpy(dest, begin, length);
+
+	encoder->output_begin = begin + length;
+	return length;
+}
+
+static const char *
+lame_encoder_get_mime_type(gcc_unused Encoder *_encoder)
+{
+	return "audio/mpeg";
+}
+
+const EncoderPlugin lame_encoder_plugin = {
+	"lame",
+	lame_encoder_init,
+	lame_encoder_finish,
+	lame_encoder_open,
+	lame_encoder_close,
+	nullptr,
+	nullptr,
+	nullptr,
+	nullptr,
+	lame_encoder_write,
+	lame_encoder_read,
+	lame_encoder_get_mime_type,
+};

src/encoder/plugins/LameEncoderPlugin.hxx

@@ -0,0 +1,25 @@
+/*
+ * Copyright (C) 2003-2014 The Music Player Daemon Project
+ * http://www.musicpd.org
+ *
+ * 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 2 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, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#ifndef MPD_ENCODER_LAME_HXX
+#define MPD_ENCODER_LAME_HXX
+
+extern const struct EncoderPlugin lame_encoder_plugin;
+
+#endif

src/encoder/plugins/NullEncoderPlugin.cxx

@@ -0,0 +1,105 @@
+/*
+ * Copyright (C) 2003-2014 The Music Player Daemon Project
+ * http://www.musicpd.org
+ *
+ * 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 2 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, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include "config.h"
+#include "NullEncoderPlugin.hxx"
+#include "../EncoderAPI.hxx"
+#include "util/Manual.hxx"
+#include "util/DynamicFifoBuffer.hxx"
+#include "Compiler.h"
+
+#include <assert.h>
+
+struct NullEncoder final {
+	Encoder encoder;
+
+	Manual<DynamicFifoBuffer<uint8_t>> buffer;
+
+	NullEncoder()
+		:encoder(null_encoder_plugin) {}
+};
+
+static Encoder *
+null_encoder_init(gcc_unused const config_param &param,
+		  gcc_unused Error &error)
+{
+	NullEncoder *encoder = new NullEncoder();
+	return &encoder->encoder;
+}
+
+static void
+null_encoder_finish(Encoder *_encoder)
+{
+	NullEncoder *encoder = (NullEncoder *)_encoder;
+
+	delete encoder;
+}
+
+static void
+null_encoder_close(Encoder *_encoder)
+{
+	NullEncoder *encoder = (NullEncoder *)_encoder;
+
+	encoder->buffer.Destruct();
+}
+
+
+static bool
+null_encoder_open(Encoder *_encoder,
+		  gcc_unused AudioFormat &audio_format,
+		  gcc_unused Error &error)
+{
+	NullEncoder *encoder = (NullEncoder *)_encoder;
+	encoder->buffer.Construct(8192);
+	return true;
+}
+
+static bool
+null_encoder_write(Encoder *_encoder,
+		   const void *data, size_t length,
+		   gcc_unused Error &error)
+{
+	NullEncoder *encoder = (NullEncoder *)_encoder;
+
+	encoder->buffer->Append((const uint8_t *)data, length);
+	return length;
+}
+
+static size_t
+null_encoder_read(Encoder *_encoder, void *dest, size_t length)
+{
+	NullEncoder *encoder = (NullEncoder *)_encoder;
+
+	return encoder->buffer->Read((uint8_t *)dest, length);
+}
+
+const EncoderPlugin null_encoder_plugin = {
+	"null",
+	null_encoder_init,
+	null_encoder_finish,
+	null_encoder_open,
+	null_encoder_close,
+	nullptr,
+	nullptr,
+	nullptr,
+	nullptr,
+	null_encoder_write,
+	null_encoder_read,
+	nullptr,
+};

src/encoder/plugins/NullEncoderPlugin.hxx

@@ -0,0 +1,25 @@
+/*
+ * Copyright (C) 2003-2014 The Music Player Daemon Project
+ * http://www.musicpd.org
+ *
+ * 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 2 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, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#ifndef MPD_ENCODER_NULL_HXX
+#define MPD_ENCODER_NULL_HXX
+
+extern const struct EncoderPlugin null_encoder_plugin;
+
+#endif

src/encoder/plugins/OggSerial.cxx

@@ -0,0 +1,43 @@
+/*
+ * Copyright (C) 2003-2014 The Music Player Daemon Project
+ * http://www.musicpd.org
+ *
+ * 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 2 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, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include "OggSerial.hxx"
+#include "system/Clock.hxx"
+#include "Compiler.h"
+
+#include <atomic>
+
+static std::atomic_uint next_ogg_serial;
+
+int
+GenerateOggSerial()
+{
+	unsigned serial = ++next_ogg_serial;
+	if (gcc_unlikely(serial < 16)) {
+		/* first-time initialization: seed with a clock value,
+		   which is random enough for our use */
+
+		/* this code is not race-free, but good enough */
+		const unsigned seed = MonotonicClockMS();
+		next_ogg_serial = serial = seed;
+	}
+
+	return serial;
+}
+

src/encoder/plugins/OggSerial.hxx

@@ -0,0 +1,29 @@
+/*
+ * Copyright (C) 2003-2014 The Music Player Daemon Project
+ * http://www.musicpd.org
+ *
+ * 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 2 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, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#ifndef MPD_OGG_SERIAL_HXX
+#define MPD_OGG_SERIAL_HXX
+
+/**
+ * Generate the next pseudo-random Ogg serial.
+ */
+int
+GenerateOggSerial();
+
+#endif

src/encoder/plugins/OggStream.hxx

@@ -0,0 +1,128 @@
+/*
+ * Copyright (C) 2003-2014 The Music Player Daemon Project
+ * http://www.musicpd.org
+ *
+ * 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 2 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, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#ifndef MPD_OGG_STREAM_HXX
+#define MPD_OGG_STREAM_HXX
+
+#include "check.h"
+
+#include <ogg/ogg.h>
+
+#include <assert.h>
+#include <string.h>
+#include <stdint.h>
+
+class OggStream {
+	ogg_stream_state state;
+
+	bool flush;
+
+#ifndef NDEBUG
+	bool initialized;
+#endif
+
+public:
+#ifndef NDEBUG
+	OggStream():initialized(false) {}
+	~OggStream() {
+		assert(!initialized);
+	}
+#endif
+
+	void Initialize(int serialno) {
+		assert(!initialized);
+
+		ogg_stream_init(&state, serialno);
+
+		/* set "flush" to true, so the caller gets the full
+		   headers on the first read() */
+		flush = true;
+
+#ifndef NDEBUG
+		initialized = true;
+#endif
+	}
+
+	void Reinitialize(int serialno) {
+		assert(initialized);
+
+		ogg_stream_reset_serialno(&state, serialno);
+
+		/* set "flush" to true, so the caller gets the full
+		   headers on the first read() */
+		flush = true;
+	}
+
+	void Deinitialize() {
+		assert(initialized);
+
+		ogg_stream_clear(&state);
+
+#ifndef NDEBUG
+		initialized = false;
+#endif
+	}
+
+	void Flush() {
+		assert(initialized);
+
+		flush = true;
+	}
+
+	void PacketIn(const ogg_packet &packet) {
+		assert(initialized);
+
+		ogg_stream_packetin(&state,
+				    const_cast<ogg_packet *>(&packet));
+	}
+
+	bool PageOut(ogg_page &page) {
+		int result = ogg_stream_pageout(&state, &page);
+		if (result == 0 && flush) {
+			flush = false;
+			result = ogg_stream_flush(&state, &page);
+		}
+
+		return result != 0;
+	}
+
+	size_t PageOut(void *_buffer, size_t size) {
+		ogg_page page;
+		if (!PageOut(page))
+			return 0;
+
+		assert(page.header_len > 0 || page.body_len > 0);
+
+		size_t header_len = (size_t)page.header_len;
+		size_t body_len = (size_t)page.body_len;
+		assert(header_len <= size);
+
+		if (header_len + body_len > size)
+			/* TODO: better overflow handling */
+			body_len = size - header_len;
+
+		uint8_t *buffer = (uint8_t *)_buffer;
+		memcpy(buffer, page.header, header_len);
+		memcpy(buffer + header_len, page.body, body_len);
+
+		return header_len + body_len;
+	}
+};
+
+#endif

src/encoder/plugins/OpusEncoderPlugin.cxx

@@ -0,0 +1,420 @@
+/*
+ * Copyright (C) 2003-2014 The Music Player Daemon Project
+ * http://www.musicpd.org
+ *
+ * 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 2 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, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include "config.h"
+#include "OpusEncoderPlugin.hxx"
+#include "OggStream.hxx"
+#include "OggSerial.hxx"
+#include "../EncoderAPI.hxx"
+#include "AudioFormat.hxx"
+#include "ConfigError.hxx"
+#include "util/Error.hxx"
+#include "util/Domain.hxx"
+#include "system/ByteOrder.hxx"
+
+#include <opus.h>
+#include <ogg/ogg.h>
+
+#include <glib.h>
+
+#include <assert.h>
+#include <stdlib.h>
+
+struct opus_encoder {
+	/** the base class */
+	Encoder encoder;
+
+	/* configuration */
+
+	opus_int32 bitrate;
+	int complexity;
+	int signal;
+
+	/* runtime information */
+
+	AudioFormat audio_format;
+
+	size_t frame_size;
+
+	size_t buffer_frames, buffer_size, buffer_position;
+	uint8_t *buffer;
+
+	OpusEncoder *enc;
+
+	unsigned char buffer2[1275 * 3 + 7];
+
+	OggStream stream;
+
+	int lookahead;
+
+	ogg_int64_t packetno;
+
+	ogg_int64_t granulepos;
+
+	opus_encoder():encoder(opus_encoder_plugin) {}
+};
+
+static constexpr Domain opus_encoder_domain("opus_encoder");
+
+static bool
+opus_encoder_configure(struct opus_encoder *encoder,
+		       const config_param &param, Error &error)
+{
+	const char *value = param.GetBlockValue("bitrate", "auto");
+	if (strcmp(value, "auto") == 0)
+		encoder->bitrate = OPUS_AUTO;
+	else if (strcmp(value, "max") == 0)
+		encoder->bitrate = OPUS_BITRATE_MAX;
+	else {
+		char *endptr;
+		encoder->bitrate = strtoul(value, &endptr, 10);
+		if (endptr == value || *endptr != 0 ||
+		    encoder->bitrate < 500 || encoder->bitrate > 512000) {
+			error.Set(config_domain, "Invalid bit rate");
+			return false;
+		}
+	}
+
+	encoder->complexity = param.GetBlockValue("complexity", 10u);
+	if (encoder->complexity > 10) {
+		error.Format(config_domain, "Invalid complexity");
+		return false;
+	}
+
+	value = param.GetBlockValue("signal", "auto");
+	if (strcmp(value, "auto") == 0)
+		encoder->signal = OPUS_AUTO;
+	else if (strcmp(value, "voice") == 0)
+		encoder->signal = OPUS_SIGNAL_VOICE;
+	else if (strcmp(value, "music") == 0)
+		encoder->signal = OPUS_SIGNAL_MUSIC;
+	else {
+		error.Format(config_domain, "Invalid signal");
+		return false;
+	}
+
+	return true;
+}
+
+static Encoder *
+opus_encoder_init(const config_param &param, Error &error)
+{
+	opus_encoder *encoder = new opus_encoder();
+
+	/* load configuration from "param" */
+	if (!opus_encoder_configure(encoder, param, error)) {
+		/* configuration has failed, roll back and return error */
+		delete encoder;
+		return NULL;
+	}
+
+	return &encoder->encoder;
+}
+
+static void
+opus_encoder_finish(Encoder *_encoder)
+{
+	struct opus_encoder *encoder = (struct opus_encoder *)_encoder;
+
+	/* the real libopus cleanup was already performed by
+	   opus_encoder_close(), so no real work here */
+	delete encoder;
+}
+
+static bool
+opus_encoder_open(Encoder *_encoder,
+		  AudioFormat &audio_format,
+		  Error &error)
+{
+	struct opus_encoder *encoder = (struct opus_encoder *)_encoder;
+
+	/* libopus supports only 48 kHz */
+	audio_format.sample_rate = 48000;
+
+	if (audio_format.channels > 2)
+		audio_format.channels = 1;
+
+	switch (audio_format.format) {
+	case SampleFormat::S16:
+	case SampleFormat::FLOAT:
+		break;
+
+	case SampleFormat::S8:
+		audio_format.format = SampleFormat::S16;
+		break;
+
+	default:
+		audio_format.format = SampleFormat::FLOAT;
+		break;
+	}
+
+	encoder->audio_format = audio_format;
+	encoder->frame_size = audio_format.GetFrameSize();
+
+	int error_code;
+	encoder->enc = opus_encoder_create(audio_format.sample_rate,
+					   audio_format.channels,
+					   OPUS_APPLICATION_AUDIO,
+					   &error_code);
+	if (encoder->enc == nullptr) {
+		error.Set(opus_encoder_domain, error_code,
+			  opus_strerror(error_code));
+		return false;
+	}
+
+	opus_encoder_ctl(encoder->enc, OPUS_SET_BITRATE(encoder->bitrate));
+	opus_encoder_ctl(encoder->enc,
+			 OPUS_SET_COMPLEXITY(encoder->complexity));
+	opus_encoder_ctl(encoder->enc, OPUS_SET_SIGNAL(encoder->signal));
+
+	opus_encoder_ctl(encoder->enc, OPUS_GET_LOOKAHEAD(&encoder->lookahead));
+
+	encoder->buffer_frames = audio_format.sample_rate / 50;
+	encoder->buffer_size = encoder->frame_size * encoder->buffer_frames;
+	encoder->buffer_position = 0;
+	encoder->buffer = (unsigned char *)g_malloc(encoder->buffer_size);
+
+	encoder->stream.Initialize(GenerateOggSerial());
+	encoder->packetno = 0;
+
+	return true;
+}
+
+static void
+opus_encoder_close(Encoder *_encoder)
+{
+	struct opus_encoder *encoder = (struct opus_encoder *)_encoder;
+
+	encoder->stream.Deinitialize();
+	g_free(encoder->buffer);
+	opus_encoder_destroy(encoder->enc);
+}
+
+static bool
+opus_encoder_do_encode(struct opus_encoder *encoder, bool eos,
+		       Error &error)
+{
+	assert(encoder->buffer_position == encoder->buffer_size);
+
+	opus_int32 result =
+		encoder->audio_format.format == SampleFormat::S16
+		? opus_encode(encoder->enc,
+			      (const opus_int16 *)encoder->buffer,
+			      encoder->buffer_frames,
+			      encoder->buffer2,
+			      sizeof(encoder->buffer2))
+		: opus_encode_float(encoder->enc,
+				    (const float *)encoder->buffer,
+				    encoder->buffer_frames,
+				    encoder->buffer2,
+				    sizeof(encoder->buffer2));
+	if (result < 0) {
+		error.Set(opus_encoder_domain, "Opus encoder error");
+		return false;
+	}
+
+	encoder->granulepos += encoder->buffer_frames;
+
+	ogg_packet packet;
+	packet.packet = encoder->buffer2;
+	packet.bytes = result;
+	packet.b_o_s = false;
+	packet.e_o_s = eos;
+	packet.granulepos = encoder->granulepos;
+	packet.packetno = encoder->packetno++;
+	encoder->stream.PacketIn(packet);
+
+	encoder->buffer_position = 0;
+
+	return true;
+}
+
+static bool
+opus_encoder_end(Encoder *_encoder, Error &error)
+{
+	struct opus_encoder *encoder = (struct opus_encoder *)_encoder;
+
+	encoder->stream.Flush();
+
+	memset(encoder->buffer + encoder->buffer_position, 0,
+	       encoder->buffer_size - encoder->buffer_position);
+	encoder->buffer_position = encoder->buffer_size;
+
+	return opus_encoder_do_encode(encoder, true, error);
+}
+
+static bool
+opus_encoder_flush(Encoder *_encoder, gcc_unused Error &error)
+{
+	struct opus_encoder *encoder = (struct opus_encoder *)_encoder;
+
+	encoder->stream.Flush();
+	return true;
+}
+
+static bool
+opus_encoder_write_silence(struct opus_encoder *encoder, unsigned fill_frames,
+			   Error &error)
+{
+	size_t fill_bytes = fill_frames * encoder->frame_size;
+
+	while (fill_bytes > 0) {
+		size_t nbytes =
+			encoder->buffer_size - encoder->buffer_position;
+		if (nbytes > fill_bytes)
+			nbytes = fill_bytes;
+
+		memset(encoder->buffer + encoder->buffer_position,
+		       0, nbytes);
+		encoder->buffer_position += nbytes;
+		fill_bytes -= nbytes;
+
+		if (encoder->buffer_position == encoder->buffer_size &&
+		    !opus_encoder_do_encode(encoder, false, error))
+			return false;
+	}
+
+	return true;
+}
+
+static bool
+opus_encoder_write(Encoder *_encoder,
+		   const void *_data, size_t length,
+		   Error &error)
+{
+	struct opus_encoder *encoder = (struct opus_encoder *)_encoder;
+	const uint8_t *data = (const uint8_t *)_data;
+
+	if (encoder->lookahead > 0) {
+		/* generate some silence at the beginning of the
+		   stream */
+
+		assert(encoder->buffer_position == 0);
+
+		if (!opus_encoder_write_silence(encoder, encoder->lookahead,
+						error))
+			return false;
+
+		encoder->lookahead = 0;
+	}
+
+	while (length > 0) {
+		size_t nbytes =
+			encoder->buffer_size - encoder->buffer_position;
+		if (nbytes > length)
+			nbytes = length;
+
+		memcpy(encoder->buffer + encoder->buffer_position,
+		       data, nbytes);
+		data += nbytes;
+		length -= nbytes;
+		encoder->buffer_position += nbytes;
+
+		if (encoder->buffer_position == encoder->buffer_size &&
+		    !opus_encoder_do_encode(encoder, false, error))
+			return false;
+	}
+
+	return true;
+}
+
+static void
+opus_encoder_generate_head(struct opus_encoder *encoder)
+{
+	unsigned char header[19];
+	memcpy(header, "OpusHead", 8);
+	header[8] = 1;
+	header[9] = encoder->audio_format.channels;
+	*(uint16_t *)(header + 10) = ToLE16(encoder->lookahead);
+	*(uint32_t *)(header + 12) =
+		ToLE32(encoder->audio_format.sample_rate);
+	header[16] = 0;
+	header[17] = 0;
+	header[18] = 0;
+
+	ogg_packet packet;
+	packet.packet = header;
+	packet.bytes = 19;
+	packet.b_o_s = true;
+	packet.e_o_s = false;
+	packet.granulepos = 0;
+	packet.packetno = encoder->packetno++;
+	encoder->stream.PacketIn(packet);
+	encoder->stream.Flush();
+}
+
+static void
+opus_encoder_generate_tags(struct opus_encoder *encoder)
+{
+	const char *version = opus_get_version_string();
+	size_t version_length = strlen(version);
+
+	size_t comments_size = 8 + 4 + version_length + 4;
+	unsigned char *comments = (unsigned char *)g_malloc(comments_size);
+	memcpy(comments, "OpusTags", 8);
+	*(uint32_t *)(comments + 8) = ToLE32(version_length);
+	memcpy(comments + 12, version, version_length);
+	*(uint32_t *)(comments + 12 + version_length) = ToLE32(0);
+
+	ogg_packet packet;
+	packet.packet = comments;
+	packet.bytes = comments_size;
+	packet.b_o_s = false;
+	packet.e_o_s = false;
+	packet.granulepos = 0;
+	packet.packetno = encoder->packetno++;
+	encoder->stream.PacketIn(packet);
+	encoder->stream.Flush();
+
+	g_free(comments);
+}
+
+static size_t
+opus_encoder_read(Encoder *_encoder, void *dest, size_t length)
+{
+	struct opus_encoder *encoder = (struct opus_encoder *)_encoder;
+
+	if (encoder->packetno == 0)
+		opus_encoder_generate_head(encoder);
+	else if (encoder->packetno == 1)
+		opus_encoder_generate_tags(encoder);
+
+	return encoder->stream.PageOut(dest, length);
+}
+
+static const char *
+opus_encoder_get_mime_type(gcc_unused Encoder *_encoder)
+{
+	return  "audio/ogg";
+}
+
+const EncoderPlugin opus_encoder_plugin = {
+	"opus",
+	opus_encoder_init,
+	opus_encoder_finish,
+	opus_encoder_open,
+	opus_encoder_close,
+	opus_encoder_end,
+	opus_encoder_flush,
+	nullptr,
+	nullptr,
+	opus_encoder_write,
+	opus_encoder_read,
+	opus_encoder_get_mime_type,
+};

src/encoder/plugins/OpusEncoderPlugin.hxx

@@ -0,0 +1,25 @@
+/*
+ * Copyright (C) 2003-2014 The Music Player Daemon Project
+ * http://www.musicpd.org
+ *
+ * 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 2 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, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#ifndef MPD_ENCODER_OPUS_H
+#define MPD_ENCODER_OPUS_H
+
+extern const struct EncoderPlugin opus_encoder_plugin;
+
+#endif

src/encoder/plugins/ShineEncoderPlugin.cxx

@@ -0,0 +1,271 @@
+/*
+ * Copyright (C) 2003-2014 The Music Player Daemon Project
+ * http://www.musicpd.org
+ *
+ * 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 2 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, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include "ShineEncoderPlugin.hxx"
+#include "config.h"
+#include "../EncoderAPI.hxx"
+#include "AudioFormat.hxx"
+#include "ConfigError.hxx"
+#include "util/Manual.hxx"
+#include "util/NumberParser.hxx"
+#include "util/DynamicFifoBuffer.hxx"
+#include "util/Error.hxx"
+#include "util/Domain.hxx"
+
+extern "C"
+{
+#include <shine/layer3.h>
+}
+
+static constexpr size_t BUFFER_INIT_SIZE = 8192;
+static constexpr unsigned CHANNELS = 2;
+
+struct ShineEncoder {
+	Encoder encoder;
+
+	AudioFormat audio_format;
+
+	shine_t shine;
+
+	shine_config_t config;
+
+	size_t frame_size;
+	size_t input_pos;
+	int16_t *stereo[CHANNELS];
+
+	Manual<DynamicFifoBuffer<uint8_t>> output_buffer;
+
+	ShineEncoder():encoder(shine_encoder_plugin){}
+
+	bool Configure(const config_param &param, Error &error);
+
+	bool Setup(Error &error);
+
+	bool WriteChunk(bool flush);
+};
+
+static constexpr Domain shine_encoder_domain("shine_encoder");
+
+inline bool
+ShineEncoder::Configure(const config_param &param,
+			 gcc_unused Error &error)
+{
+	shine_set_config_mpeg_defaults(&config.mpeg);
+	config.mpeg.bitr = param.GetBlockValue("bitrate", 128);
+
+	return true;
+}
+
+static Encoder *
+shine_encoder_init(const config_param &param, Error &error)
+{
+	ShineEncoder *encoder = new ShineEncoder();
+
+	/* load configuration from "param" */
+	if (!encoder->Configure(param, error)) {
+		/* configuration has failed, roll back and return error */
+		delete encoder;
+		return nullptr;
+	}
+
+	return &encoder->encoder;
+}
+
+static void
+shine_encoder_finish(Encoder *_encoder)
+{
+	ShineEncoder *encoder = (ShineEncoder *)_encoder;
+
+	delete encoder;
+}
+
+inline bool
+ShineEncoder::Setup(Error &error)
+{
+	config.mpeg.mode = audio_format.channels == 2 ? STEREO : MONO;
+	config.wave.samplerate = audio_format.sample_rate;
+	config.wave.channels =
+		audio_format.channels == 2 ? PCM_STEREO : PCM_MONO;
+
+	if (shine_check_config(config.wave.samplerate, config.mpeg.bitr) < 0) {
+		error.Format(config_domain,
+			     "error configuring shine. "
+			     "samplerate %d and bitrate %d configuration"
+			     " not supported.",
+			     config.wave.samplerate,
+			     config.mpeg.bitr);
+
+		return false;
+	}
+
+	shine = shine_initialise(&config);
+
+	if (!shine) {
+		error.Format(config_domain,
+			     "error initializing shine.");
+
+		return false;
+	}
+
+	frame_size = shine_samples_per_pass(shine);
+
+	return true;
+}
+
+static bool
+shine_encoder_open(Encoder *_encoder, AudioFormat &audio_format, Error &error)
+{
+	ShineEncoder *encoder = (ShineEncoder *)_encoder;
+
+	audio_format.format = SampleFormat::S16;
+	audio_format.channels = CHANNELS;
+	encoder->audio_format = audio_format;
+
+	if (!encoder->Setup(error))
+		return false;
+
+	encoder->stereo[0] = new int16_t[encoder->frame_size];
+	encoder->stereo[1] = new int16_t[encoder->frame_size];
+	/* workaround for bug:
+	   https://github.com/savonet/shine/issues/11 */
+	encoder->input_pos = SHINE_MAX_SAMPLES + 1;
+
+	encoder->output_buffer.Construct(BUFFER_INIT_SIZE);
+
+	return true;
+}
+
+static void
+shine_encoder_close(Encoder *_encoder)
+{
+	ShineEncoder *encoder = (ShineEncoder *)_encoder;
+
+	if (encoder->input_pos > SHINE_MAX_SAMPLES) {
+		/* write zero chunk */
+		encoder->input_pos = 0;
+		encoder->WriteChunk(true);
+	}
+
+	shine_close(encoder->shine);
+	delete[] encoder->stereo[0];
+	delete[] encoder->stereo[1];
+	encoder->output_buffer.Destruct();
+}
+
+bool
+ShineEncoder::WriteChunk(bool flush)
+{
+	if (flush || input_pos == frame_size) {
+		long written;
+
+		if (flush) {
+			/* fill remaining with 0s */
+			for (; input_pos < frame_size; input_pos++) {
+				stereo[0][input_pos] = stereo[1][input_pos] = 0;
+			}
+		}
+
+		const uint8_t *out =
+			shine_encode_buffer(shine, stereo, &written);
+
+		if (written > 0)
+			output_buffer->Append(out, written);
+
+		input_pos = 0;
+	}
+
+	return true;
+}
+
+static bool
+shine_encoder_write(Encoder *_encoder,
+		    const void *_data, size_t length,
+		    gcc_unused Error &error)
+{
+	ShineEncoder *encoder = (ShineEncoder *)_encoder;
+	const int16_t *data = (const int16_t*)_data;
+	length /= sizeof(*data) * encoder->audio_format.channels;
+	size_t written = 0;
+
+	if (encoder->input_pos > SHINE_MAX_SAMPLES) {
+		encoder->input_pos = 0;
+	}
+
+	/* write all data to de-interleaved buffers */
+	while (written < length) {
+		for (;
+		     written < length
+			     && encoder->input_pos < encoder->frame_size;
+		     written++, encoder->input_pos++) {
+			const size_t base =
+				written * encoder->audio_format.channels;
+			encoder->stereo[0][encoder->input_pos] = data[base];
+			encoder->stereo[1][encoder->input_pos] = data[base + 1];
+		}
+		/* write if chunk is filled */
+		encoder->WriteChunk(false);
+	}
+
+	return true;
+}
+
+static bool
+shine_encoder_flush(Encoder *_encoder, gcc_unused Error &error)
+{
+	ShineEncoder *encoder = (ShineEncoder *)_encoder;
+	long written;
+
+	/* flush buffers and flush shine */
+	encoder->WriteChunk(true);
+	const uint8_t *data = shine_flush(encoder->shine, &written);
+
+	if (written > 0)
+		encoder->output_buffer->Append(data, written);
+
+	return true;
+}
+
+static size_t
+shine_encoder_read(Encoder *_encoder, void *dest, size_t length)
+{
+	ShineEncoder *encoder = (ShineEncoder *)_encoder;
+
+	return encoder->output_buffer->Read((uint8_t *)dest, length);
+}
+
+static const char *
+shine_encoder_get_mime_type(gcc_unused Encoder *_encoder)
+{
+	return "audio/mpeg";
+}
+
+const EncoderPlugin shine_encoder_plugin = {
+	"shine",
+	shine_encoder_init,
+	shine_encoder_finish,
+	shine_encoder_open,
+	shine_encoder_close,
+	shine_encoder_flush,
+	shine_encoder_flush,
+	nullptr,
+	nullptr,
+	shine_encoder_write,
+	shine_encoder_read,
+	shine_encoder_get_mime_type,
+};

src/encoder/plugins/ShineEncoderPlugin.hxx

@@ -0,0 +1,25 @@
+/*
+ * Copyright (C) 2003-2014 The Music Player Daemon Project
+ * http://www.musicpd.org
+ *
+ * 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 2 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, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#ifndef MPD_ENCODER_SHINE_HXX
+#define MPD_ENCODER_SHINE_HXX
+
+extern const struct EncoderPlugin shine_encoder_plugin;
+
+#endif

src/encoder/plugins/TwolameEncoderPlugin.cxx

@@ -0,0 +1,314 @@
+/*
+ * Copyright (C) 2003-2014 The Music Player Daemon Project
+ * http://www.musicpd.org
+ *
+ * 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 2 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, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include "config.h"
+#include "TwolameEncoderPlugin.hxx"
+#include "../EncoderAPI.hxx"
+#include "AudioFormat.hxx"
+#include "ConfigError.hxx"
+#include "util/NumberParser.hxx"
+#include "util/Error.hxx"
+#include "util/Domain.hxx"
+#include "Log.hxx"
+
+#include <twolame.h>
+
+#include <assert.h>
+#include <string.h>
+
+struct TwolameEncoder final {
+	Encoder encoder;
+
+	AudioFormat audio_format;
+	float quality;
+	int bitrate;
+
+	twolame_options *options;
+
+	unsigned char output_buffer[32768];
+	size_t output_buffer_length;
+	size_t output_buffer_position;
+
+	/**
+	 * Call libtwolame's flush function when the output_buffer is
+	 * empty?
+	 */
+	bool flush;
+
+	TwolameEncoder():encoder(twolame_encoder_plugin) {}
+
+	bool Configure(const config_param &param, Error &error);
+};
+
+static constexpr Domain twolame_encoder_domain("twolame_encoder");
+
+bool
+TwolameEncoder::Configure(const config_param &param, Error &error)
+{
+	const char *value;
+	char *endptr;
+
+	value = param.GetBlockValue("quality");
+	if (value != nullptr) {
+		/* a quality was configured (VBR) */
+
+		quality = ParseDouble(value, &endptr);
+
+		if (*endptr != '\0' || quality < -1.0 || quality > 10.0) {
+			error.Format(config_domain,
+				     "quality \"%s\" is not a number in the "
+				     "range -1 to 10",
+				     value);
+			return false;
+		}
+
+		if (param.GetBlockValue("bitrate") != nullptr) {
+			error.Set(config_domain,
+				  "quality and bitrate are both defined");
+			return false;
+		}
+	} else {
+		/* a bit rate was configured */
+
+		value = param.GetBlockValue("bitrate");
+		if (value == nullptr) {
+			error.Set(config_domain,
+				  "neither bitrate nor quality defined");
+			return false;
+		}
+
+		quality = -2.0;
+		bitrate = ParseInt(value, &endptr);
+
+		if (*endptr != '\0' || bitrate <= 0) {
+			error.Set(config_domain,
+				  "bitrate should be a positive integer");
+			return false;
+		}
+	}
+
+	return true;
+}
+
+static Encoder *
+twolame_encoder_init(const config_param &param, Error &error_r)
+{
+	FormatDebug(twolame_encoder_domain,
+		    "libtwolame version %s", get_twolame_version());
+
+	TwolameEncoder *encoder = new TwolameEncoder();
+
+	/* load configuration from "param" */
+	if (!encoder->Configure(param, error_r)) {
+		/* configuration has failed, roll back and return error */
+		delete encoder;
+		return nullptr;
+	}
+
+	return &encoder->encoder;
+}
+
+static void
+twolame_encoder_finish(Encoder *_encoder)
+{
+	TwolameEncoder *encoder = (TwolameEncoder *)_encoder;
+
+	/* the real libtwolame cleanup was already performed by
+	   twolame_encoder_close(), so no real work here */
+	delete encoder;
+}
+
+static bool
+twolame_encoder_setup(TwolameEncoder *encoder, Error &error)
+{
+	if (encoder->quality >= -1.0) {
+		/* a quality was configured (VBR) */
+
+		if (0 != twolame_set_VBR(encoder->options, true)) {
+			error.Set(twolame_encoder_domain,
+				  "error setting twolame VBR mode");
+			return false;
+		}
+		if (0 != twolame_set_VBR_q(encoder->options, encoder->quality)) {
+			error.Set(twolame_encoder_domain,
+				  "error setting twolame VBR quality");
+			return false;
+		}
+	} else {
+		/* a bit rate was configured */
+
+		if (0 != twolame_set_brate(encoder->options, encoder->bitrate)) {
+			error.Set(twolame_encoder_domain,
+				  "error setting twolame bitrate");
+			return false;
+		}
+	}
+
+	if (0 != twolame_set_num_channels(encoder->options,
+					  encoder->audio_format.channels)) {
+		error.Set(twolame_encoder_domain,
+			  "error setting twolame num channels");
+		return false;
+	}
+
+	if (0 != twolame_set_in_samplerate(encoder->options,
+					   encoder->audio_format.sample_rate)) {
+		error.Set(twolame_encoder_domain,
+			  "error setting twolame sample rate");
+		return false;
+	}
+
+	if (0 > twolame_init_params(encoder->options)) {
+		error.Set(twolame_encoder_domain,
+			  "error initializing twolame params");
+		return false;
+	}
+
+	return true;
+}
+
+static bool
+twolame_encoder_open(Encoder *_encoder, AudioFormat &audio_format,
+		     Error &error)
+{
+	TwolameEncoder *encoder = (TwolameEncoder *)_encoder;
+
+	audio_format.format = SampleFormat::S16;
+	audio_format.channels = 2;
+
+	encoder->audio_format = audio_format;
+
+	encoder->options = twolame_init();
+	if (encoder->options == nullptr) {
+		error.Set(twolame_encoder_domain, "twolame_init() failed");
+		return false;
+	}
+
+	if (!twolame_encoder_setup(encoder, error)) {
+		twolame_close(&encoder->options);
+		return false;
+	}
+
+	encoder->output_buffer_length = 0;
+	encoder->output_buffer_position = 0;
+	encoder->flush = false;
+
+	return true;
+}
+
+static void
+twolame_encoder_close(Encoder *_encoder)
+{
+	TwolameEncoder *encoder = (TwolameEncoder *)_encoder;
+
+	twolame_close(&encoder->options);
+}
+
+static bool
+twolame_encoder_flush(Encoder *_encoder, gcc_unused Error &error)
+{
+	TwolameEncoder *encoder = (TwolameEncoder *)_encoder;
+
+	encoder->flush = true;
+	return true;
+}
+
+static bool
+twolame_encoder_write(Encoder *_encoder,
+		      const void *data, size_t length,
+		      gcc_unused Error &error)
+{
+	TwolameEncoder *encoder = (TwolameEncoder *)_encoder;
+	const int16_t *src = (const int16_t*)data;
+
+	assert(encoder->output_buffer_position ==
+	       encoder->output_buffer_length);
+
+	const unsigned num_frames =
+		length / encoder->audio_format.GetFrameSize();
+
+	int bytes_out = twolame_encode_buffer_interleaved(encoder->options,
+							  src, num_frames,
+							  encoder->output_buffer,
+							  sizeof(encoder->output_buffer));
+	if (bytes_out < 0) {
+		error.Set(twolame_encoder_domain, "twolame encoder failed");
+		return false;
+	}
+
+	encoder->output_buffer_length = (size_t)bytes_out;
+	encoder->output_buffer_position = 0;
+	return true;
+}
+
+static size_t
+twolame_encoder_read(Encoder *_encoder, void *dest, size_t length)
+{
+	TwolameEncoder *encoder = (TwolameEncoder *)_encoder;
+
+	assert(encoder->output_buffer_position <=
+	       encoder->output_buffer_length);
+
+	if (encoder->output_buffer_position == encoder->output_buffer_length &&
+	    encoder->flush) {
+		int ret = twolame_encode_flush(encoder->options,
+					       encoder->output_buffer,
+					       sizeof(encoder->output_buffer));
+		if (ret > 0) {
+			encoder->output_buffer_length = (size_t)ret;
+			encoder->output_buffer_position = 0;
+		}
+
+		encoder->flush = false;
+	}
+
+
+	const size_t remainning = encoder->output_buffer_length
+		- encoder->output_buffer_position;
+	if (length > remainning)
+		length = remainning;
+
+	memcpy(dest, encoder->output_buffer + encoder->output_buffer_position,
+	       length);
+
+	encoder->output_buffer_position += length;
+
+	return length;
+}
+
+static const char *
+twolame_encoder_get_mime_type(gcc_unused Encoder *_encoder)
+{
+	return "audio/mpeg";
+}
+
+const EncoderPlugin twolame_encoder_plugin = {
+	"twolame",
+	twolame_encoder_init,
+	twolame_encoder_finish,
+	twolame_encoder_open,
+	twolame_encoder_close,
+	twolame_encoder_flush,
+	twolame_encoder_flush,
+	nullptr,
+	nullptr,
+	twolame_encoder_write,
+	twolame_encoder_read,
+	twolame_encoder_get_mime_type,
+};

src/encoder/plugins/TwolameEncoderPlugin.hxx

@@ -0,0 +1,25 @@
+/*
+ * Copyright (C) 2003-2014 The Music Player Daemon Project
+ * http://www.musicpd.org
+ *
+ * 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 2 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, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#ifndef MPD_ENCODER_TWOLAME_HXX
+#define MPD_ENCODER_TWOLAME_HXX
+
+extern const struct EncoderPlugin twolame_encoder_plugin;
+
+#endif

src/encoder/plugins/VorbisEncoderPlugin.cxx

@@ -0,0 +1,365 @@
+/*
+ * Copyright (C) 2003-2014 The Music Player Daemon Project
+ * http://www.musicpd.org
+ *
+ * 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 2 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, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include "config.h"
+#include "VorbisEncoderPlugin.hxx"
+#include "OggStream.hxx"
+#include "OggSerial.hxx"
+#include "../EncoderAPI.hxx"
+#include "tag/Tag.hxx"
+#include "AudioFormat.hxx"
+#include "ConfigError.hxx"
+#include "util/NumberParser.hxx"
+#include "util/Error.hxx"
+#include "util/Domain.hxx"
+
+#include <vorbis/vorbisenc.h>
+
+#include <glib.h>
+
+struct vorbis_encoder {
+	/** the base class */
+	Encoder encoder;
+
+	/* configuration */
+
+	float quality;
+	int bitrate;
+
+	/* runtime information */
+
+	AudioFormat audio_format;
+
+	vorbis_dsp_state vd;
+	vorbis_block vb;
+	vorbis_info vi;
+
+	OggStream stream;
+
+	vorbis_encoder():encoder(vorbis_encoder_plugin) {}
+};
+
+static constexpr Domain vorbis_encoder_domain("vorbis_encoder");
+
+static bool
+vorbis_encoder_configure(struct vorbis_encoder *encoder,
+			 const config_param &param, Error &error)
+{
+	const char *value = param.GetBlockValue("quality");
+	if (value != nullptr) {
+		/* a quality was configured (VBR) */
+
+		char *endptr;
+		encoder->quality = ParseDouble(value, &endptr);
+
+		if (*endptr != '\0' || encoder->quality < -1.0 ||
+		    encoder->quality > 10.0) {
+			error.Format(config_domain,
+				     "quality \"%s\" is not a number in the "
+				     "range -1 to 10",
+				     value);
+			return false;
+		}
+
+		if (param.GetBlockValue("bitrate") != nullptr) {
+			error.Set(config_domain,
+				  "quality and bitrate are both defined");
+			return false;
+		}
+	} else {
+		/* a bit rate was configured */
+
+		value = param.GetBlockValue("bitrate");
+		if (value == nullptr) {
+			error.Set(config_domain,
+				  "neither bitrate nor quality defined");
+			return false;
+		}
+
+		encoder->quality = -2.0;
+
+		char *endptr;
+		encoder->bitrate = ParseInt(value, &endptr);
+		if (*endptr != '\0' || encoder->bitrate <= 0) {
+			error.Set(config_domain,
+				  "bitrate should be a positive integer");
+			return false;
+		}
+	}
+
+	return true;
+}
+
+static Encoder *
+vorbis_encoder_init(const config_param &param, Error &error)
+{
+	vorbis_encoder *encoder = new vorbis_encoder();
+
+	/* load configuration from "param" */
+	if (!vorbis_encoder_configure(encoder, param, error)) {
+		/* configuration has failed, roll back and return error */
+		delete encoder;
+		return nullptr;
+	}
+
+	return &encoder->encoder;
+}
+
+static void
+vorbis_encoder_finish(Encoder *_encoder)
+{
+	struct vorbis_encoder *encoder = (struct vorbis_encoder *)_encoder;
+
+	/* the real libvorbis/libogg cleanup was already performed by
+	   vorbis_encoder_close(), so no real work here */
+	delete encoder;
+}
+
+static bool
+vorbis_encoder_reinit(struct vorbis_encoder *encoder, Error &error)
+{
+	vorbis_info_init(&encoder->vi);
+
+	if (encoder->quality >= -1.0) {
+		/* a quality was configured (VBR) */
+
+		if (0 != vorbis_encode_init_vbr(&encoder->vi,
+						encoder->audio_format.channels,
+						encoder->audio_format.sample_rate,
+						encoder->quality * 0.1)) {
+			error.Set(vorbis_encoder_domain,
+				  "error initializing vorbis vbr");
+			vorbis_info_clear(&encoder->vi);
+			return false;
+		}
+	} else {
+		/* a bit rate was configured */
+
+		if (0 != vorbis_encode_init(&encoder->vi,
+					    encoder->audio_format.channels,
+					    encoder->audio_format.sample_rate, -1.0,
+					    encoder->bitrate * 1000, -1.0)) {
+			error.Set(vorbis_encoder_domain,
+				  "error initializing vorbis encoder");
+			vorbis_info_clear(&encoder->vi);
+			return false;
+		}
+	}
+
+	vorbis_analysis_init(&encoder->vd, &encoder->vi);
+	vorbis_block_init(&encoder->vd, &encoder->vb);
+	encoder->stream.Initialize(GenerateOggSerial());
+
+	return true;
+}
+
+static void
+vorbis_encoder_headerout(struct vorbis_encoder *encoder, vorbis_comment *vc)
+{
+	ogg_packet packet, comments, codebooks;
+
+	vorbis_analysis_headerout(&encoder->vd, vc,
+				  &packet, &comments, &codebooks);
+
+	encoder->stream.PacketIn(packet);
+	encoder->stream.PacketIn(comments);
+	encoder->stream.PacketIn(codebooks);
+}
+
+static void
+vorbis_encoder_send_header(struct vorbis_encoder *encoder)
+{
+	vorbis_comment vc;
+
+	vorbis_comment_init(&vc);
+	vorbis_encoder_headerout(encoder, &vc);
+	vorbis_comment_clear(&vc);
+}
+
+static bool
+vorbis_encoder_open(Encoder *_encoder,
+		    AudioFormat &audio_format,
+		    Error &error)
+{
+	struct vorbis_encoder *encoder = (struct vorbis_encoder *)_encoder;
+
+	audio_format.format = SampleFormat::FLOAT;
+
+	encoder->audio_format = audio_format;
+
+	if (!vorbis_encoder_reinit(encoder, error))
+		return false;
+
+	vorbis_encoder_send_header(encoder);
+
+	return true;
+}
+
+static void
+vorbis_encoder_clear(struct vorbis_encoder *encoder)
+{
+	encoder->stream.Deinitialize();
+	vorbis_block_clear(&encoder->vb);
+	vorbis_dsp_clear(&encoder->vd);
+	vorbis_info_clear(&encoder->vi);
+}
+
+static void
+vorbis_encoder_close(Encoder *_encoder)
+{
+	struct vorbis_encoder *encoder = (struct vorbis_encoder *)_encoder;
+
+	vorbis_encoder_clear(encoder);
+}
+
+static void
+vorbis_encoder_blockout(struct vorbis_encoder *encoder)
+{
+	while (vorbis_analysis_blockout(&encoder->vd, &encoder->vb) == 1) {
+		vorbis_analysis(&encoder->vb, nullptr);
+		vorbis_bitrate_addblock(&encoder->vb);
+
+		ogg_packet packet;
+		while (vorbis_bitrate_flushpacket(&encoder->vd, &packet))
+			encoder->stream.PacketIn(packet);
+	}
+}
+
+static bool
+vorbis_encoder_flush(Encoder *_encoder, gcc_unused Error &error)
+{
+	struct vorbis_encoder *encoder = (struct vorbis_encoder *)_encoder;
+
+	encoder->stream.Flush();
+	return true;
+}
+
+static bool
+vorbis_encoder_pre_tag(Encoder *_encoder, gcc_unused Error &error)
+{
+	struct vorbis_encoder *encoder = (struct vorbis_encoder *)_encoder;
+
+	vorbis_analysis_wrote(&encoder->vd, 0);
+	vorbis_encoder_blockout(encoder);
+
+	/* reinitialize vorbis_dsp_state and vorbis_block to reset the
+	   end-of-stream marker */
+	vorbis_block_clear(&encoder->vb);
+	vorbis_dsp_clear(&encoder->vd);
+	vorbis_analysis_init(&encoder->vd, &encoder->vi);
+	vorbis_block_init(&encoder->vd, &encoder->vb);
+
+	encoder->stream.Flush();
+	return true;
+}
+
+static void
+copy_tag_to_vorbis_comment(vorbis_comment *vc, const Tag *tag)
+{
+	for (unsigned i = 0; i < tag->num_items; i++) {
+		const TagItem &item = *tag->items[i];
+		char *name = g_ascii_strup(tag_item_names[item.type], -1);
+		vorbis_comment_add_tag(vc, name, item.value);
+		g_free(name);
+	}
+}
+
+static bool
+vorbis_encoder_tag(Encoder *_encoder, const Tag *tag,
+		   gcc_unused Error &error)
+{
+	struct vorbis_encoder *encoder = (struct vorbis_encoder *)_encoder;
+	vorbis_comment comment;
+
+	/* write the vorbis_comment object */
+
+	vorbis_comment_init(&comment);
+	copy_tag_to_vorbis_comment(&comment, tag);
+
+	/* reset ogg_stream_state and begin a new stream */
+
+	encoder->stream.Reinitialize(GenerateOggSerial());
+
+	/* send that vorbis_comment to the ogg_stream_state */
+
+	vorbis_encoder_headerout(encoder, &comment);
+	vorbis_comment_clear(&comment);
+
+	return true;
+}
+
+static void
+interleaved_to_vorbis_buffer(float **dest, const float *src,
+			     unsigned num_frames, unsigned num_channels)
+{
+	for (unsigned i = 0; i < num_frames; i++)
+		for (unsigned j = 0; j < num_channels; j++)
+			dest[j][i] = *src++;
+}
+
+static bool
+vorbis_encoder_write(Encoder *_encoder,
+		     const void *data, size_t length,
+		     gcc_unused Error &error)
+{
+	struct vorbis_encoder *encoder = (struct vorbis_encoder *)_encoder;
+
+	unsigned num_frames = length / encoder->audio_format.GetFrameSize();
+
+	/* this is for only 16-bit audio */
+
+	interleaved_to_vorbis_buffer(vorbis_analysis_buffer(&encoder->vd,
+							    num_frames),
+				     (const float *)data,
+				     num_frames,
+				     encoder->audio_format.channels);
+
+	vorbis_analysis_wrote(&encoder->vd, num_frames);
+	vorbis_encoder_blockout(encoder);
+	return true;
+}
+
+static size_t
+vorbis_encoder_read(Encoder *_encoder, void *dest, size_t length)
+{
+	struct vorbis_encoder *encoder = (struct vorbis_encoder *)_encoder;
+
+	return encoder->stream.PageOut(dest, length);
+}
+
+static const char *
+vorbis_encoder_get_mime_type(gcc_unused Encoder *_encoder)
+{
+	return  "audio/ogg";
+}
+
+const EncoderPlugin vorbis_encoder_plugin = {
+	"vorbis",
+	vorbis_encoder_init,
+	vorbis_encoder_finish,
+	vorbis_encoder_open,
+	vorbis_encoder_close,
+	vorbis_encoder_pre_tag,
+	vorbis_encoder_flush,
+	vorbis_encoder_pre_tag,
+	vorbis_encoder_tag,
+	vorbis_encoder_write,
+	vorbis_encoder_read,
+	vorbis_encoder_get_mime_type,
+};

src/encoder/plugins/VorbisEncoderPlugin.hxx

@@ -0,0 +1,25 @@
+/*
+ * Copyright (C) 2003-2014 The Music Player Daemon Project
+ * http://www.musicpd.org
+ *
+ * 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 2 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, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#ifndef MPD_ENCODER_VORBIS_H
+#define MPD_ENCODER_VORBIS_H
+
+extern const struct EncoderPlugin vorbis_encoder_plugin;
+
+#endif

src/encoder/plugins/WaveEncoderPlugin.cxx

@@ -0,0 +1,265 @@
+/*
+ * Copyright (C) 2003-2014 The Music Player Daemon Project
+ * http://www.musicpd.org
+ *
+ * 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 2 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, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include "config.h"
+#include "WaveEncoderPlugin.hxx"
+#include "../EncoderAPI.hxx"
+#include "system/ByteOrder.hxx"
+#include "util/Manual.hxx"
+#include "util/DynamicFifoBuffer.hxx"
+
+#include <assert.h>
+#include <string.h>
+
+struct WaveEncoder {
+	Encoder encoder;
+	unsigned bits;
+
+	Manual<DynamicFifoBuffer<uint8_t>> buffer;
+
+	WaveEncoder():encoder(wave_encoder_plugin) {}
+};
+
+struct wave_header {
+	uint32_t id_riff;
+	uint32_t riff_size;
+	uint32_t id_wave;
+	uint32_t id_fmt;
+	uint32_t fmt_size;
+	uint16_t format;
+	uint16_t channels;
+	uint32_t freq;
+	uint32_t byterate;
+	uint16_t blocksize;
+	uint16_t bits;
+	uint32_t id_data;
+	uint32_t data_size;
+};
+
+static void
+fill_wave_header(struct wave_header *header, int channels, int bits,
+		int freq, int block_size)
+{
+	int data_size = 0x0FFFFFFF;
+
+	/* constants */
+	header->id_riff = ToLE32(0x46464952);
+	header->id_wave = ToLE32(0x45564157);
+	header->id_fmt = ToLE32(0x20746d66);
+	header->id_data = ToLE32(0x61746164);
+
+        /* wave format */
+	header->format = ToLE16(1); // PCM_FORMAT
+	header->channels = ToLE16(channels);
+	header->bits = ToLE16(bits);
+	header->freq = ToLE32(freq);
+	header->blocksize = ToLE16(block_size);
+	header->byterate = ToLE32(freq * block_size);
+
+        /* chunk sizes (fake data length) */
+	header->fmt_size = ToLE32(16);
+	header->data_size = ToLE32(data_size);
+	header->riff_size = ToLE32(4 + (8 + 16) + (8 + data_size));
+}
+
+static Encoder *
+wave_encoder_init(gcc_unused const config_param &param,
+		  gcc_unused Error &error)
+{
+	WaveEncoder *encoder = new WaveEncoder();
+	return &encoder->encoder;
+}
+
+static void
+wave_encoder_finish(Encoder *_encoder)
+{
+	WaveEncoder *encoder = (WaveEncoder *)_encoder;
+
+	delete encoder;
+}
+
+static bool
+wave_encoder_open(Encoder *_encoder,
+		  AudioFormat &audio_format,
+		  gcc_unused Error &error)
+{
+	WaveEncoder *encoder = (WaveEncoder *)_encoder;
+
+	assert(audio_format.IsValid());
+
+	switch (audio_format.format) {
+	case SampleFormat::S8:
+		encoder->bits = 8;
+		break;
+
+	case SampleFormat::S16:
+		encoder->bits = 16;
+		break;
+
+	case SampleFormat::S24_P32:
+		encoder->bits = 24;
+		break;
+
+	case SampleFormat::S32:
+		encoder->bits = 32;
+		break;
+
+	default:
+		audio_format.format = SampleFormat::S16;
+		encoder->bits = 16;
+		break;
+	}
+
+	encoder->buffer.Construct(8192);
+
+	auto range = encoder->buffer->Write();
+	assert(range.size >= sizeof(wave_header));
+	wave_header *header = (wave_header *)range.data;
+
+	/* create PCM wave header in initial buffer */
+	fill_wave_header(header,
+			 audio_format.channels,
+			 encoder->bits,
+			 audio_format.sample_rate,
+			 (encoder->bits / 8) * audio_format.channels);
+
+	encoder->buffer->Append(sizeof(*header));
+
+	return true;
+}
+
+static void
+wave_encoder_close(Encoder *_encoder)
+{
+	WaveEncoder *encoder = (WaveEncoder *)_encoder;
+
+	encoder->buffer.Destruct();
+}
+
+static size_t
+pcm16_to_wave(uint16_t *dst16, const uint16_t *src16, size_t length)
+{
+	size_t cnt = length >> 1;
+	while (cnt > 0) {
+		*dst16++ = ToLE16(*src16++);
+		cnt--;
+	}
+	return length;
+}
+
+static size_t
+pcm32_to_wave(uint32_t *dst32, const uint32_t *src32, size_t length)
+{
+	size_t cnt = length >> 2;
+	while (cnt > 0){
+		*dst32++ = ToLE32(*src32++);
+		cnt--;
+	}
+	return length;
+}
+
+static size_t
+pcm24_to_wave(uint8_t *dst8, const uint32_t *src32, size_t length)
+{
+	uint32_t value;
+	uint8_t *dst_old = dst8;
+
+	length = length >> 2;
+	while (length > 0){
+		value = *src32++;
+		*dst8++ = (value) & 0xFF;
+		*dst8++ = (value >> 8) & 0xFF;
+		*dst8++ = (value >> 16) & 0xFF;
+		length--;
+	}
+	//correct buffer length
+	return (dst8 - dst_old);
+}
+
+static bool
+wave_encoder_write(Encoder *_encoder,
+		   const void *src, size_t length,
+		   gcc_unused Error &error)
+{
+	WaveEncoder *encoder = (WaveEncoder *)_encoder;
+
+	uint8_t *dst = encoder->buffer->Write(length);
+
+	if (IsLittleEndian()) {
+		switch (encoder->bits) {
+		case 8:
+		case 16:
+		case 32:// optimized cases
+			memcpy(dst, src, length);
+			break;
+		case 24:
+			length = pcm24_to_wave(dst, (const uint32_t *)src, length);
+			break;
+		}
+	} else {
+		switch (encoder->bits) {
+		case 8:
+			memcpy(dst, src, length);
+			break;
+		case 16:
+			length = pcm16_to_wave((uint16_t *)dst,
+					       (const uint16_t *)src, length);
+			break;
+		case 24:
+			length = pcm24_to_wave(dst, (const uint32_t *)src, length);
+			break;
+		case 32:
+			length = pcm32_to_wave((uint32_t *)dst,
+					       (const uint32_t *)src, length);
+			break;
+		}
+	}
+
+	encoder->buffer->Append(length);
+	return true;
+}
+
+static size_t
+wave_encoder_read(Encoder *_encoder, void *dest, size_t length)
+{
+	WaveEncoder *encoder = (WaveEncoder *)_encoder;
+
+	return encoder->buffer->Read((uint8_t *)dest, length);
+}
+
+static const char *
+wave_encoder_get_mime_type(gcc_unused Encoder *_encoder)
+{
+	return "audio/wav";
+}
+
+const EncoderPlugin wave_encoder_plugin = {
+	"wave",
+	wave_encoder_init,
+	wave_encoder_finish,
+	wave_encoder_open,
+	wave_encoder_close,
+	nullptr,
+	nullptr,
+	nullptr,
+	nullptr,
+	wave_encoder_write,
+	wave_encoder_read,
+	wave_encoder_get_mime_type,
+};

src/encoder/plugins/WaveEncoderPlugin.hxx

@@ -0,0 +1,25 @@
+/*
+ * Copyright (C) 2003-2014 The Music Player Daemon Project
+ * http://www.musicpd.org
+ *
+ * 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 2 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, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#ifndef MPD_ENCODER_WAVE_HXX
+#define MPD_ENCODER_WAVE_HXX
+
+extern const struct EncoderPlugin wave_encoder_plugin;
+
+#endif