encoder/opus: new encoder plugin for the Opus codec

This commit is contained in:
Max Kellermann 2012-10-01 20:15:15 +02:00
parent 9a715267ad
commit d793b7c03f
6 changed files with 458 additions and 0 deletions

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@ -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

2
NEWS
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@ -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

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@ -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

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@ -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,
};

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@ -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

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@ -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