From d793b7c03ff7444b0aab485e5895573b4a4ff0a1 Mon Sep 17 00:00:00 2001 From: Max Kellermann Date: Mon, 1 Oct 2012 20:15:15 +0200 Subject: [PATCH] encoder/opus: new encoder plugin for the Opus codec --- Makefile.am | 8 + NEWS | 2 + configure.ac | 2 + src/encoder/OpusEncoderPlugin.cxx | 417 ++++++++++++++++++++++++++++++ src/encoder/OpusEncoderPlugin.hxx | 25 ++ src/encoder_list.c | 4 + 6 files changed, 458 insertions(+) create mode 100644 src/encoder/OpusEncoderPlugin.cxx create mode 100644 src/encoder/OpusEncoderPlugin.hxx diff --git a/Makefile.am b/Makefile.am index 469ac5330..f5fb2148f 100644 --- a/Makefile.am +++ b/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) \ $(VORBISENC_LIBS) libencoder_plugins_a_SOURCES = @@ -708,6 +710,12 @@ libencoder_plugins_a_SOURCES += \ src/encoder/VorbisEncoderPlugin.hxx endif +if HAVE_OPUS +libencoder_plugins_a_SOURCES += \ + src/encoder/OpusEncoderPlugin.cxx \ + src/encoder/OpusEncoderPlugin.hxx +endif + if ENABLE_LAME_ENCODER libencoder_plugins_a_SOURCES += src/encoder/lame_encoder.c endif diff --git a/NEWS b/NEWS index e3b7041f2..8db1acb2a 100644 --- a/NEWS +++ b/NEWS @@ -3,6 +3,8 @@ ver 0.18 (2012/??/??) - adplug: new decoder plugin using libadplug - opus: new decoder plugin for the Opus codec - vorbis: skip 16 bit quantisation, provide float samples +* encoder: + - opus: new encoder plugin for the Opus codec * output: - new option "tags" may be used to disable sending tags to output * improved decoder/output error reporting diff --git a/configure.ac b/configure.ac index f7599f4c4..b688a5f8a 100644 --- a/configure.ac +++ b/configure.ac @@ -1229,6 +1229,7 @@ fi dnl --------------------------- encoder plugins test -------------------------- if test x$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 diff --git a/src/encoder/OpusEncoderPlugin.cxx b/src/encoder/OpusEncoderPlugin.cxx new file mode 100644 index 000000000..c967a24a7 --- /dev/null +++ b/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 +#include + +#include + +#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, +}; diff --git a/src/encoder/OpusEncoderPlugin.hxx b/src/encoder/OpusEncoderPlugin.hxx new file mode 100644 index 000000000..f54377202 --- /dev/null +++ b/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 diff --git a/src/encoder_list.c b/src/encoder_list.c index 3bbf9330e..029b4be34 100644 --- a/src/encoder_list.c +++ b/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 @@ -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