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-rw-r--r--src/encoder/plugins/FlacEncoderPlugin.cxx325
-rw-r--r--src/encoder/plugins/FlacEncoderPlugin.hxx25
-rw-r--r--src/encoder/plugins/LameEncoderPlugin.cxx293
-rw-r--r--src/encoder/plugins/LameEncoderPlugin.hxx25
-rw-r--r--src/encoder/plugins/NullEncoderPlugin.cxx105
-rw-r--r--src/encoder/plugins/NullEncoderPlugin.hxx25
-rw-r--r--src/encoder/plugins/OggSerial.cxx43
-rw-r--r--src/encoder/plugins/OggSerial.hxx29
-rw-r--r--src/encoder/plugins/OggStream.hxx128
-rw-r--r--src/encoder/plugins/OpusEncoderPlugin.cxx419
-rw-r--r--src/encoder/plugins/OpusEncoderPlugin.hxx25
-rw-r--r--src/encoder/plugins/ShineEncoderPlugin.cxx271
-rw-r--r--src/encoder/plugins/ShineEncoderPlugin.hxx25
-rw-r--r--src/encoder/plugins/TwolameEncoderPlugin.cxx314
-rw-r--r--src/encoder/plugins/TwolameEncoderPlugin.hxx25
-rw-r--r--src/encoder/plugins/VorbisEncoderPlugin.cxx364
-rw-r--r--src/encoder/plugins/VorbisEncoderPlugin.hxx25
-rw-r--r--src/encoder/plugins/WaveEncoderPlugin.cxx265
-rw-r--r--src/encoder/plugins/WaveEncoderPlugin.hxx25
19 files changed, 2756 insertions, 0 deletions
diff --git a/src/encoder/plugins/FlacEncoderPlugin.cxx b/src/encoder/plugins/FlacEncoderPlugin.cxx
new file mode 100644
index 000000000..26987fe99
--- /dev/null
+++ b/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 "config/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,
+};
+
diff --git a/src/encoder/plugins/FlacEncoderPlugin.hxx b/src/encoder/plugins/FlacEncoderPlugin.hxx
new file mode 100644
index 000000000..0cdc01600
--- /dev/null
+++ b/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
diff --git a/src/encoder/plugins/LameEncoderPlugin.cxx b/src/encoder/plugins/LameEncoderPlugin.cxx
new file mode 100644
index 000000000..3878b52bb
--- /dev/null
+++ b/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 "config/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,
+};
diff --git a/src/encoder/plugins/LameEncoderPlugin.hxx b/src/encoder/plugins/LameEncoderPlugin.hxx
new file mode 100644
index 000000000..03e398f67
--- /dev/null
+++ b/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
diff --git a/src/encoder/plugins/NullEncoderPlugin.cxx b/src/encoder/plugins/NullEncoderPlugin.cxx
new file mode 100644
index 000000000..1d571d465
--- /dev/null
+++ b/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,
+};
diff --git a/src/encoder/plugins/NullEncoderPlugin.hxx b/src/encoder/plugins/NullEncoderPlugin.hxx
new file mode 100644
index 000000000..6acf88e49
--- /dev/null
+++ b/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
diff --git a/src/encoder/plugins/OggSerial.cxx b/src/encoder/plugins/OggSerial.cxx
new file mode 100644
index 000000000..677829439
--- /dev/null
+++ b/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;
+}
+
diff --git a/src/encoder/plugins/OggSerial.hxx b/src/encoder/plugins/OggSerial.hxx
new file mode 100644
index 000000000..ceba8ebf9
--- /dev/null
+++ b/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
diff --git a/src/encoder/plugins/OggStream.hxx b/src/encoder/plugins/OggStream.hxx
new file mode 100644
index 000000000..805238c1d
--- /dev/null
+++ b/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
diff --git a/src/encoder/plugins/OpusEncoderPlugin.cxx b/src/encoder/plugins/OpusEncoderPlugin.cxx
new file mode 100644
index 000000000..27b614b86
--- /dev/null
+++ b/src/encoder/plugins/OpusEncoderPlugin.cxx
@@ -0,0 +1,419 @@
+/*
+ * 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 "config/ConfigError.hxx"
+#include "util/Alloc.hxx"
+#include "util/Error.hxx"
+#include "util/Domain.hxx"
+#include "system/ByteOrder.hxx"
+
+#include <opus.h>
+#include <ogg/ogg.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 nullptr;
+ }
+
+ 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 *)xalloc(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();
+ 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 *)xalloc(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();
+
+ 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,
+};
diff --git a/src/encoder/plugins/OpusEncoderPlugin.hxx b/src/encoder/plugins/OpusEncoderPlugin.hxx
new file mode 100644
index 000000000..4e71694b9
--- /dev/null
+++ b/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
diff --git a/src/encoder/plugins/ShineEncoderPlugin.cxx b/src/encoder/plugins/ShineEncoderPlugin.cxx
new file mode 100644
index 000000000..61cb8609e
--- /dev/null
+++ b/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 "config/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) {
+ if (flush) {
+ /* fill remaining with 0s */
+ for (; input_pos < frame_size; input_pos++) {
+ stereo[0][input_pos] = stereo[1][input_pos] = 0;
+ }
+ }
+
+ int written;
+ 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;
+
+ /* flush buffers and flush shine */
+ encoder->WriteChunk(true);
+
+ int written;
+ 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,
+};
diff --git a/src/encoder/plugins/ShineEncoderPlugin.hxx b/src/encoder/plugins/ShineEncoderPlugin.hxx
new file mode 100644
index 000000000..8b1520a74
--- /dev/null
+++ b/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
diff --git a/src/encoder/plugins/TwolameEncoderPlugin.cxx b/src/encoder/plugins/TwolameEncoderPlugin.cxx
new file mode 100644
index 000000000..2eb6b2b1c
--- /dev/null
+++ b/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 "config/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,
+};
diff --git a/src/encoder/plugins/TwolameEncoderPlugin.hxx b/src/encoder/plugins/TwolameEncoderPlugin.hxx
new file mode 100644
index 000000000..531dd3e90
--- /dev/null
+++ b/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
diff --git a/src/encoder/plugins/VorbisEncoderPlugin.cxx b/src/encoder/plugins/VorbisEncoderPlugin.cxx
new file mode 100644
index 000000000..ecc784a47
--- /dev/null
+++ b/src/encoder/plugins/VorbisEncoderPlugin.cxx
@@ -0,0 +1,364 @@
+/*
+ * 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 "config/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 (const auto &item : *tag) {
+ 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,
+};
diff --git a/src/encoder/plugins/VorbisEncoderPlugin.hxx b/src/encoder/plugins/VorbisEncoderPlugin.hxx
new file mode 100644
index 000000000..80703bf88
--- /dev/null
+++ b/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
diff --git a/src/encoder/plugins/WaveEncoderPlugin.cxx b/src/encoder/plugins/WaveEncoderPlugin.cxx
new file mode 100644
index 000000000..97a26e821
--- /dev/null
+++ b/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,
+};
diff --git a/src/encoder/plugins/WaveEncoderPlugin.hxx b/src/encoder/plugins/WaveEncoderPlugin.hxx
new file mode 100644
index 000000000..341b98adc
--- /dev/null
+++ b/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