encoder/opus: new encoder plugin for the Opus codec

Makefile.am

@@ -684,6 +684,7 @@ libencoder_plugins_a_CPPFLAGS = $(AM_CPPFLAGS) \
 	$(LAME_CFLAGS) \
 	$(TWOLAME_CFLAGS) \
 	$(patsubst -I%/FLAC,-I%,$(FLAC_CFLAGS)) \
+	$(OPUS_CFLAGS) \
 	$(VORBISENC_CFLAGS)
 
 ENCODER_LIBS = \
@@ -691,6 +692,7 @@ ENCODER_LIBS = \
 	$(LAME_LIBS) \
 	$(TWOLAME_LIBS) \
 	$(FLAC_LIBS) \
+	$(OPUS_LIBS) \
 	$(VORBISENC_LIBS)
 
 libencoder_plugins_a_SOURCES =
@@ -708,6 +710,12 @@ libencoder_plugins_a_SOURCES += \
 	src/encoder/VorbisEncoderPlugin.hxx
 endif
 
+if HAVE_OPUS
+libencoder_plugins_a_SOURCES += \
+	src/encoder/OpusEncoderPlugin.cxx \
+	src/encoder/OpusEncoderPlugin.hxx
+endif
+
 if ENABLE_LAME_ENCODER
 libencoder_plugins_a_SOURCES += src/encoder/lame_encoder.c
 endif

NEWS

@@ -3,6 +3,8 @@ ver 0.18 (2012/??/??)
   - adplug: new decoder plugin using libadplug
   - opus: new decoder plugin for the Opus codec
   - vorbis: skip 16 bit quantisation, provide float samples
+* encoder:
+  - opus: new encoder plugin for the Opus codec
 * output:
   - new option "tags" may be used to disable sending tags to output
 * improved decoder/output error reporting

configure.ac

@@ -1229,6 +1229,7 @@ fi
 
 dnl --------------------------- encoder plugins test --------------------------
 if test x$enable_vorbis_encoder != xno ||
+	test x$enable_opus != xno ||
 	test x$enable_lame_encoder != xno ||
 	test x$enable_twolame_encoder != xno ||
 	test x$enable_flac_encoder != xno ||
@@ -1657,6 +1658,7 @@ if
 		results(flac_encoder, [FLAC])
 		results(lame_encoder, [LAME])
 		results(vorbis_encoder, [Ogg Vorbis])
+		results(opus, [Opus])
 		results(twolame_encoder, [TwoLAME])
 		results(wave_encoder, [WAVE])
 fi

src/encoder/OpusEncoderPlugin.cxx

@@ -0,0 +1,417 @@
+/*
+ * Copyright (C) 2003-2012 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"
+
+extern "C" {
+#include "encoder_api.h"
+}
+
+#include "encoder_plugin.h"
+#include "audio_format.h"
+#include "mpd_error.h"
+
+#include <opus.h>
+#include <ogg/ogg.h>
+
+#include <assert.h>
+
+#undef G_LOG_DOMAIN
+#define G_LOG_DOMAIN "opus_encoder"
+
+struct opus_encoder {
+	/** the base class */
+	struct encoder encoder;
+
+	/* configuration */
+
+	opus_int32 bitrate;
+	int complexity;
+	int signal;
+
+	/* runtime information */
+
+	struct audio_format 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];
+
+	ogg_stream_state os;
+
+	ogg_int64_t packetno;
+
+	bool flush;
+};
+
+gcc_const
+static inline GQuark
+opus_encoder_quark(void)
+{
+	return g_quark_from_static_string("opus_encoder");
+}
+
+static bool
+opus_encoder_configure(struct opus_encoder *encoder,
+		       const struct config_param *param, GError **error_r)
+{
+	const char *value = config_get_block_string(param, "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) {
+			g_set_error(error_r, opus_encoder_quark(), 0,
+				    "Invalid bit rate");
+			return false;
+		}
+	}
+
+	encoder->complexity = config_get_block_unsigned(param, "complexity",
+							10);
+	if (encoder->complexity > 10) {
+		g_set_error(error_r, opus_encoder_quark(), 0,
+			    "Invalid complexity");
+		return false;
+	}
+
+	value = config_get_block_string(param, "signal", "auto");
+	if (strcmp(value, "auto") == 0)
+		encoder->bitrate = OPUS_AUTO;
+	else if (strcmp(value, "voice") == 0)
+		encoder->bitrate = OPUS_SIGNAL_VOICE;
+	else if (strcmp(value, "music") == 0)
+		encoder->bitrate = OPUS_SIGNAL_MUSIC;
+	else {
+		g_set_error(error_r, opus_encoder_quark(), 0,
+			    "Invalid signal");
+		return false;
+	}
+
+	return true;
+}
+
+static struct encoder *
+opus_encoder_init(const struct config_param *param, GError **error)
+{
+	struct opus_encoder *encoder;
+
+	encoder = g_new(struct opus_encoder, 1);
+	encoder_struct_init(&encoder->encoder, &opus_encoder_plugin);
+
+	/* load configuration from "param" */
+	if (!opus_encoder_configure(encoder, param, error)) {
+		/* configuration has failed, roll back and return error */
+		g_free(encoder);
+		return NULL;
+	}
+
+	return &encoder->encoder;
+}
+
+static void
+opus_encoder_finish(struct 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 */
+	g_free(encoder);
+}
+
+static bool
+opus_encoder_open(struct encoder *_encoder,
+		  struct audio_format *audio_format,
+		  GError **error_r)
+{
+	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 ((enum sample_format)audio_format->format) {
+	case SAMPLE_FORMAT_S16:
+	case SAMPLE_FORMAT_FLOAT:
+		break;
+
+	case SAMPLE_FORMAT_S8:
+		audio_format->format = SAMPLE_FORMAT_S16;
+		break;
+
+	default:
+		audio_format->format = SAMPLE_FORMAT_FLOAT;
+		break;
+	}
+
+	encoder->audio_format = *audio_format;
+	encoder->frame_size = audio_format_frame_size(audio_format);
+
+	int error;
+	encoder->enc = opus_encoder_create(audio_format->sample_rate,
+					   audio_format->channels,
+					   OPUS_APPLICATION_AUDIO,
+					   &error);
+	if (encoder->enc == nullptr) {
+		g_set_error_literal(error_r, opus_encoder_quark(), error,
+				    opus_strerror(error));
+		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));
+
+	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);
+
+	ogg_stream_init(&encoder->os, g_random_int());
+	encoder->packetno = 0;
+
+	/* set "flush" to true, so the caller gets the full headers on
+	   the first read() */
+	encoder->flush = true;
+
+	return true;
+}
+
+static void
+opus_encoder_close(struct encoder *_encoder)
+{
+	struct opus_encoder *encoder = (struct opus_encoder *)_encoder;
+
+	ogg_stream_clear(&encoder->os);
+	g_free(encoder->buffer);
+	opus_encoder_destroy(encoder->enc);
+}
+
+static bool
+opus_encoder_do_encode(struct opus_encoder *encoder, bool eos,
+		       GError **error_r)
+{
+	assert(encoder->buffer_position == encoder->buffer_size);
+
+	opus_int32 result =
+		encoder->audio_format.format == SAMPLE_FORMAT_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) {
+		g_set_error_literal(error_r, opus_encoder_quark(), 0,
+				    "Opus encoder error");
+		return false;
+	}
+
+	ogg_packet packet;
+	packet.packet = encoder->buffer2;
+	packet.bytes = result;
+	packet.b_o_s = false;
+	packet.e_o_s = eos;
+	packet.granulepos = 0; // TODO
+	packet.packetno = encoder->packetno++;
+	ogg_stream_packetin(&encoder->os, &packet);
+
+	encoder->buffer_position = 0;
+
+	return true;
+}
+
+static bool
+opus_encoder_end(struct encoder *_encoder, GError **error_r)
+{
+	struct opus_encoder *encoder = (struct opus_encoder *)_encoder;
+
+	encoder->flush = true;
+
+	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_r);
+}
+
+static bool
+opus_encoder_flush(struct encoder *_encoder, G_GNUC_UNUSED GError **error)
+{
+	struct opus_encoder *encoder = (struct opus_encoder *)_encoder;
+
+	encoder->flush = true;
+	return true;
+}
+
+static bool
+opus_encoder_write(struct encoder *_encoder,
+		   const void *_data, size_t length,
+		   GError **error_r)
+{
+	struct opus_encoder *encoder = (struct opus_encoder *)_encoder;
+	const uint8_t *data = (const uint8_t *)_data;
+
+	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_r))
+			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;
+	header[10] = 0;
+	header[11] = 0;
+	*(uint32_t *)(header + 12) =
+		GUINT32_TO_LE(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++;
+	ogg_stream_packetin(&encoder->os, &packet);
+
+	encoder->flush = true;
+}
+
+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) = GUINT32_TO_LE(version_length);
+	memcpy(comments + 12, version, version_length);
+	*(uint32_t *)(comments + 12 + version_length) = GUINT32_TO_LE(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++;
+	ogg_stream_packetin(&encoder->os, &packet);
+
+	g_free(comments);
+
+	encoder->flush = true;
+}
+
+static size_t
+opus_encoder_read(struct encoder *_encoder, void *_dest, size_t length)
+{
+	struct opus_encoder *encoder = (struct opus_encoder *)_encoder;
+	unsigned char *dest = (unsigned char *)_dest;
+
+	if (encoder->packetno == 0)
+		opus_encoder_generate_head(encoder);
+	else if (encoder->packetno == 1)
+		opus_encoder_generate_tags(encoder);
+
+	ogg_page page;
+	int result;
+	if (encoder->flush) {
+		encoder->flush = false;
+		result = ogg_stream_flush(&encoder->os, &page);
+	} else
+		result = ogg_stream_pageout(&encoder->os, &page);
+
+	if (result == 0)
+		return 0;
+
+	assert(page.header_len > 0 || page.body_len > 0);
+
+	size_t nbytes = (size_t)page.header_len + (size_t)page.body_len;
+
+	if (nbytes > length)
+		/* XXX better error handling */
+		MPD_ERROR("buffer too small");
+
+	memcpy(dest, page.header, page.header_len);
+	memcpy(dest + page.header_len, page.body, page.body_len);
+
+	return nbytes;
+}
+
+static const char *
+opus_encoder_get_mime_type(G_GNUC_UNUSED struct encoder *_encoder)
+{
+	return  "audio/ogg";
+}
+
+const struct encoder_plugin 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/OpusEncoderPlugin.hxx

@@ -0,0 +1,25 @@
+/*
+ * Copyright (C) 2003-2012 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 encoder_plugin opus_encoder_plugin;
+
+#endif

src/encoder_list.c

@@ -21,6 +21,7 @@
 #include "encoder_list.h"
 #include "encoder_plugin.h"
 #include "encoder/VorbisEncoderPlugin.hxx"
+#include "encoder/OpusEncoderPlugin.hxx"
 
 #include <string.h>
 
@@ -35,6 +36,9 @@ const struct encoder_plugin *const encoder_plugins[] = {
 #ifdef ENABLE_VORBIS_ENCODER
 	&vorbis_encoder_plugin,
 #endif
+#ifdef HAVE_OPUS
+	&opus_encoder_plugin,
+#endif
 #ifdef ENABLE_LAME_ENCODER
 	&lame_encoder_plugin,
 #endif