@@ -4,6 +4,7 @@ releases are sorted from youngest to oldest.

 version next:

 - stream disposition information printing in ffprobe

 - filter for loudness analysis following EBU R128

+- Opus encoder using libopus

 version 1.0:

@@ -1768,6 +1768,7 @@ libopencore_amrwb_decoder_deps="libopencore_amrwb"

 libopenjpeg_decoder_deps="libopenjpeg"

 libopenjpeg_encoder_deps="libopenjpeg"

 libopus_decoder_deps="libopus"

+libopus_encoder_deps="libopus"

 libschroedinger_decoder_deps="libschroedinger"

 libschroedinger_encoder_deps="libschroedinger"

 libspeex_decoder_deps="libspeex"

@@ -813,7 +813,7 @@ following image formats are supported:

 @item Musepack SV7           @tab     @tab  X

 @item Musepack SV8           @tab     @tab  X

 @item Nellymoser Asao        @tab  X  @tab  X

-@item Opus                   @tab     @tab  E

+@item Opus                   @tab  E  @tab  E

     @tab supported through external library libopus

 @item PCM A-law              @tab  X  @tab  X

 @item PCM mu-law             @tab  X  @tab  X

@@ -657,7 +657,10 @@ OBJS-$(CONFIG_LIBOPENCORE_AMRNB_ENCODER)  += libopencore-amr.o \

 OBJS-$(CONFIG_LIBOPENCORE_AMRWB_DECODER)  += libopencore-amr.o

 OBJS-$(CONFIG_LIBOPENJPEG_DECODER)        += libopenjpegdec.o

 OBJS-$(CONFIG_LIBOPENJPEG_ENCODER)        += libopenjpegenc.o

-OBJS-$(CONFIG_LIBOPUS_DECODER)            += libopusdec.o vorbis_data.o

+OBJS-$(CONFIG_LIBOPUS_DECODER)            += libopusdec.o libopus.o     \

+                                             vorbis_data.o

+OBJS-$(CONFIG_LIBOPUS_ENCODER)            += libopusenc.o libopus.o     \

+                                             vorbis_data.o audio_frame_queue.o

 OBJS-$(CONFIG_LIBSCHROEDINGER_DECODER)    += libschroedingerdec.o \

                                              libschroedinger.o

 OBJS-$(CONFIG_LIBSCHROEDINGER_ENCODER)    += libschroedingerenc.o \

@@ -436,7 +436,7 @@ void avcodec_register_all(void)

     REGISTER_ENCDEC  (LIBOPENCORE_AMRNB, libopencore_amrnb);

     REGISTER_DECODER (LIBOPENCORE_AMRWB, libopencore_amrwb);

     REGISTER_ENCDEC  (LIBOPENJPEG, libopenjpeg);

-    REGISTER_DECODER (LIBOPUS, libopus);

+    REGISTER_ENCDEC  (LIBOPUS, libopus);

     REGISTER_ENCDEC  (LIBSCHROEDINGER, libschroedinger);

     REGISTER_ENCDEC  (LIBSPEEX, libspeex);

     REGISTER_DECODER (LIBSTAGEFRIGHT_H264, libstagefright_h264);

new file mode 100644

@@ -0,0 +1,48 @@

+/*

+ * libopus encoder/decoder common code

+ * Copyright (c) 2012 Nicolas George

+ *

+ * This file is part of FFmpeg.

+ *

+ * FFmpeg is free software; you can redistribute it and/or

+ * modify it under the terms of the GNU Lesser General Public

+ * License as published by the Free Software Foundation; either

+ * version 2.1 of the License, or (at your option) any later version.

+ *

+ * FFmpeg 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

+ * Lesser General Public License for more details.

+ *

+ * You should have received a copy of the GNU Lesser General Public

+ * License along with FFmpeg; if not, write to the Free Software

+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA

+ */

+

+#include <opus_defines.h>

+

+#include "libavutil/common.h"

+#include "libavutil/error.h"

+#include "libopus.h"

+

+int ff_opus_error_to_averror(int err)

+{

+    switch (err) {

+    case OPUS_BAD_ARG:

+        return AVERROR(EINVAL);

+    case OPUS_BUFFER_TOO_SMALL:

+        return AVERROR_UNKNOWN;

+    case OPUS_INTERNAL_ERROR:

+        return AVERROR(EFAULT);

+    case OPUS_INVALID_PACKET:

+        return AVERROR_INVALIDDATA;

+    case OPUS_UNIMPLEMENTED:

+        return AVERROR(ENOSYS);

+    case OPUS_INVALID_STATE:

+        return AVERROR_UNKNOWN;

+    case OPUS_ALLOC_FAIL:

+        return AVERROR(ENOMEM);

+    default:

+        return AVERROR(EINVAL);

+    }

+}

new file mode 100644

@@ -0,0 +1,27 @@

+/*

+ * libopus encoder/decoder common code

+ * Copyright (c) 2012 Nicolas George

+ *

+ * This file is part of FFmpeg.

+ *

+ * FFmpeg is free software; you can redistribute it and/or

+ * modify it under the terms of the GNU Lesser General Public

+ * License as published by the Free Software Foundation; either

+ * version 2.1 of the License, or (at your option) any later version.

+ *

+ * FFmpeg 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

+ * Lesser General Public License for more details.

+ *

+ * You should have received a copy of the GNU Lesser General Public

+ * License along with FFmpeg; if not, write to the Free Software

+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA

+ */

+

+#ifndef AVCODEC_LIBOPUS_H

+#define AVCODEC_LIBOPUS_H

+

+int ff_opus_error_to_averror(int err);

+

+#endif /* AVCODEC_LIBOPUS_H */

@@ -22,13 +22,13 @@

 #include <opus.h>

 #include <opus_multistream.h>

-#include "libavutil/common.h"

 #include "libavutil/avassert.h"

 #include "libavutil/intreadwrite.h"

 #include "avcodec.h"

 #include "internal.h"

 #include "vorbis.h"

 #include "mathops.h"

+#include "libopus.h"

 struct libopus_context {

     OpusMSDecoder *dec;

@@ -39,20 +39,6 @@ struct libopus_context {

 #endif

 };

-static int opus_error_to_averror(int err)

-{

-    switch (err) {

-        case OPUS_BAD_ARG:          return AVERROR(EINVAL);

-        case OPUS_BUFFER_TOO_SMALL: return AVERROR_BUFFER_TOO_SMALL;

-        case OPUS_INTERNAL_ERROR:   return AVERROR(EFAULT);

-        case OPUS_INVALID_PACKET:   return AVERROR_INVALIDDATA;

-        case OPUS_UNIMPLEMENTED:    return AVERROR(ENOSYS);

-        case OPUS_INVALID_STATE:    return AVERROR_EXTERNAL;

-        case OPUS_ALLOC_FAIL:       return AVERROR(ENOMEM);

-        default:                    return AVERROR(EINVAL);

-    }

-}

-

 #define OPUS_HEAD_SIZE 19

 static av_cold int libopus_decode_init(AVCodecContext *avc)

@@ -105,7 +91,7 @@ static av_cold int libopus_decode_init(AVCodecContext *avc)

     if (!opus->dec) {

         av_log(avc, AV_LOG_ERROR, "Unable to create decoder: %s\n",

                opus_strerror(ret));

-        return opus_error_to_averror(ret);

+        return ff_opus_error_to_averror(ret);

     }

 #ifdef OPUS_SET_GAIN

@@ -165,7 +151,7 @@ static int libopus_decode(AVCodecContext *avc, void *frame,

     if (nb_samples < 0) {

         av_log(avc, AV_LOG_ERROR, "Decoding error: %s\n",

                opus_strerror(nb_samples));

-        return opus_error_to_averror(nb_samples);

+        return ff_opus_error_to_averror(nb_samples);

     }

 #ifndef OPUS_SET_GAIN

new file mode 100644

@@ -0,0 +1,421 @@

+/*

+ * Opus encoder using libopus

+ * Copyright (c) 2012 Nathan Caldwell

+ *

+ * This file is part of FFmpeg.

+ *

+ * FFmpeg is free software; you can redistribute it and/or

+ * modify it under the terms of the GNU Lesser General Public

+ * License as published by the Free Software Foundation; either

+ * version 2.1 of the License, or (at your option) any later version.

+ *

+ * FFmpeg 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

+ * Lesser General Public License for more details.

+ *

+ * You should have received a copy of the GNU Lesser General Public

+ * License along with FFmpeg; if not, write to the Free Software

+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA

+ */

+

+#include <opus.h>

+#include <opus_multistream.h>

+

+#include "libavutil/opt.h"

+#include "avcodec.h"

+#include "bytestream.h"

+#include "internal.h"

+#include "libopus.h"

+#include "vorbis.h"

+#include "audio_frame_queue.h"

+

+typedef struct LibopusEncOpts {

+    int vbr;

+    int application;

+    int packet_loss;

+    int complexity;

+    float frame_duration;

+    int packet_size;

+    int max_bandwidth;

+} LibopusEncOpts;

+

+typedef struct LibopusEncContext {

+    AVClass *class;

+    OpusMSEncoder *enc;

+    int stream_count;

+    uint8_t *samples;

+    LibopusEncOpts opts;

+    AudioFrameQueue afq;

+} LibopusEncContext;

+

+static const uint8_t opus_coupled_streams[8] = {

+    0, 1, 1, 2, 2, 2, 2, 3

+};

+

+/* Opus internal to Vorbis channel order mapping written in the header */

+static const uint8_t opus_vorbis_channel_map[8][8] = {

+    { 0 },

+    { 0, 1 },

+    { 0, 2, 1 },

+    { 0, 1, 2, 3 },

+    { 0, 4, 1, 2, 3 },

+    { 0, 4, 1, 2, 3, 5 },

+    { 0, 4, 1, 2, 3, 5, 6 },

+    { 0, 6, 1, 2, 3, 4, 5, 7 },

+};

+

+/* libav to libopus channel order mapping, passed to libopus */

+static const uint8_t libav_libopus_channel_map[8][8] = {

+    { 0 },

+    { 0, 1 },

+    { 0, 1, 2 },

+    { 0, 1, 2, 3 },

+    { 0, 1, 3, 4, 2 },

+    { 0, 1, 4, 5, 2, 3 },

+    { 0, 1, 5, 6, 2, 4, 3 },

+    { 0, 1, 6, 7, 4, 5, 2, 3 },

+};

+

+static void libopus_write_header(AVCodecContext *avctx, int stream_count,

+                                 int coupled_stream_count,

+                                 const uint8_t *channel_mapping)

+{

+    uint8_t *p   = avctx->extradata;

+    int channels = avctx->channels;

+

+    bytestream_put_buffer(&p, "OpusHead", 8);

+    bytestream_put_byte(&p, 1); /* Version */

+    bytestream_put_byte(&p, channels);

+    bytestream_put_le16(&p, avctx->delay); /* Lookahead samples at 48kHz */

+    bytestream_put_le32(&p, avctx->sample_rate); /* Original sample rate */

+    bytestream_put_le16(&p, 0); /* Gain of 0dB is recommended. */

+

+    /* Channel mapping */

+    if (channels > 2) {

+        bytestream_put_byte(&p, channels <= 8 ? 1 : 255);

+        bytestream_put_byte(&p, stream_count);

+        bytestream_put_byte(&p, coupled_stream_count);

+        bytestream_put_buffer(&p, channel_mapping, channels);

+    } else {

+        bytestream_put_byte(&p, 0);

+    }

+}

+

+static int libopus_configure_encoder(AVCodecContext *avctx, OpusMSEncoder *enc,

+                                     LibopusEncOpts *opts)

+{

+    int ret;

+

+    ret = opus_multistream_encoder_ctl(enc, OPUS_SET_BITRATE(avctx->bit_rate));

+    if (ret != OPUS_OK) {

+        av_log(avctx, AV_LOG_ERROR,

+               "Failed to set bitrate: %s\n", opus_strerror(ret));

+        return ret;

+    }

+

+    ret = opus_multistream_encoder_ctl(enc,

+                                       OPUS_SET_COMPLEXITY(opts->complexity));

+    if (ret != OPUS_OK)

+        av_log(avctx, AV_LOG_WARNING,

+               "Unable to set complexity: %s\n", opus_strerror(ret));

+

+    ret = opus_multistream_encoder_ctl(enc, OPUS_SET_VBR(!!opts->vbr));

+    if (ret != OPUS_OK)

+        av_log(avctx, AV_LOG_WARNING,

+               "Unable to set VBR: %s\n", opus_strerror(ret));

+

+    ret = opus_multistream_encoder_ctl(enc,

+                                       OPUS_SET_VBR_CONSTRAINT(opts->vbr == 2));

+    if (ret != OPUS_OK)

+        av_log(avctx, AV_LOG_WARNING,

+               "Unable to set constrained VBR: %s\n", opus_strerror(ret));

+

+    ret = opus_multistream_encoder_ctl(enc,

+                                       OPUS_SET_PACKET_LOSS_PERC(opts->packet_loss));

+    if (ret != OPUS_OK)

+        av_log(avctx, AV_LOG_WARNING,

+               "Unable to set expected packet loss percentage: %s\n",

+               opus_strerror(ret));

+

+    if (avctx->cutoff) {

+        ret = opus_multistream_encoder_ctl(enc,

+                                           OPUS_SET_MAX_BANDWIDTH(opts->max_bandwidth));

+        if (ret != OPUS_OK)

+            av_log(avctx, AV_LOG_WARNING,

+                   "Unable to set maximum bandwidth: %s\n", opus_strerror(ret));

+    }

+

+    return OPUS_OK;

+}

+

+static int av_cold libopus_encode_init(AVCodecContext *avctx)

+{

+    LibopusEncContext *opus = avctx->priv_data;

+    const uint8_t *channel_mapping;

+    OpusMSEncoder *enc;

+    int ret = OPUS_OK;

+    int coupled_stream_count, header_size, frame_size;

+

+    coupled_stream_count = opus_coupled_streams[avctx->channels - 1];

+    opus->stream_count   = avctx->channels - coupled_stream_count;

+    channel_mapping      = libav_libopus_channel_map[avctx->channels - 1];

+

+    /* FIXME: Opus can handle up to 255 channels. However, the mapping for

+     * anything greater than 8 is undefined. */

+    if (avctx->channels > 8)

+        av_log(avctx, AV_LOG_WARNING,

+               "Channel layout undefined for %d channels.\n", avctx->channels);

+

+    if (!avctx->bit_rate) {

+        /* Sane default copied from opusenc */

+        avctx->bit_rate = 64000 * opus->stream_count +

+                          32000 * coupled_stream_count;

+        av_log(avctx, AV_LOG_WARNING,

+               "No bit rate set. Defaulting to %d bps.\n", avctx->bit_rate);

+    }

+

+    if (avctx->bit_rate < 500 || avctx->bit_rate > 256000 * avctx->channels) {

+        av_log(avctx, AV_LOG_ERROR, "The bit rate %d bps is unsupported. "

+               "Please choose a value between 500 and %d.\n", avctx->bit_rate,

+               256000 * avctx->channels);

+        return AVERROR(EINVAL);

+    }

+

+    frame_size = opus->opts.frame_duration * 48000 / 1000;

+    switch (frame_size) {

+    case 120:

+    case 240:

+        if (opus->opts.application != OPUS_APPLICATION_RESTRICTED_LOWDELAY)

+            av_log(avctx, AV_LOG_WARNING,

+                   "LPC mode cannot be used with a frame duration of less "

+                   "than 10ms. Enabling restricted low-delay mode.\n"

+                   "Use a longer frame duration if this is not what you want.\n");

+        /* Frame sizes less than 10 ms can only use MDCT mode, so switching to

+         * RESTRICTED_LOWDELAY avoids an unnecessary extra 2.5ms lookahead. */

+        opus->opts.application = OPUS_APPLICATION_RESTRICTED_LOWDELAY;

+    case 480:

+    case 960:

+    case 1920:

+    case 2880:

+        opus->opts.packet_size =

+        avctx->frame_size      = frame_size * avctx->sample_rate / 48000;

+        break;

+    default:

+        av_log(avctx, AV_LOG_ERROR, "Invalid frame duration: %g.\n"

+               "Frame duration must be exactly one of: 2.5, 5, 10, 20, 40 or 60.\n",

+               opus->opts.frame_duration);

+        return AVERROR(EINVAL);

+    }

+

+    if (avctx->compression_level < 0 || avctx->compression_level > 10) {

+        av_log(avctx, AV_LOG_WARNING,

+               "Compression level must be in the range 0 to 10. "

+               "Defaulting to 10.\n");

+        opus->opts.complexity = 10;

+    } else {

+        opus->opts.complexity = avctx->compression_level;

+    }

+

+    if (avctx->cutoff) {

+        switch (avctx->cutoff) {

+        case  4000:

+            opus->opts.max_bandwidth = OPUS_BANDWIDTH_NARROWBAND;

+            break;

+        case  6000:

+            opus->opts.max_bandwidth = OPUS_BANDWIDTH_MEDIUMBAND;

+            break;

+        case  8000:

+            opus->opts.max_bandwidth = OPUS_BANDWIDTH_WIDEBAND;

+            break;

+        case 12000:

+            opus->opts.max_bandwidth = OPUS_BANDWIDTH_SUPERWIDEBAND;

+            break;

+        case 20000:

+            opus->opts.max_bandwidth = OPUS_BANDWIDTH_FULLBAND;

+            break;

+        default:

+            av_log(avctx, AV_LOG_WARNING,

+                   "Invalid frequency cutoff: %d. Using default maximum bandwidth.\n"

+                   "Cutoff frequency must be exactly one of: 4000, 6000, 8000, 12000 or 20000.\n",

+                   avctx->cutoff);

+            avctx->cutoff = 0;

+        }

+    }

+

+    enc = opus_multistream_encoder_create(avctx->sample_rate, avctx->channels,

+                                          opus->stream_count,

+                                          coupled_stream_count,

+                                          channel_mapping,

+                                          opus->opts.application, &ret);

+    if (ret != OPUS_OK) {

+        av_log(avctx, AV_LOG_ERROR,

+               "Failed to create encoder: %s\n", opus_strerror(ret));

+        return ff_opus_error_to_averror(ret);

+    }

+

+    ret = libopus_configure_encoder(avctx, enc, &opus->opts);

+    if (ret != OPUS_OK) {

+        ret = ff_opus_error_to_averror(ret);

+        goto fail;

+    }

+

+    header_size = 19 + (avctx->channels > 2 ? 2 + avctx->channels : 0);

+    avctx->extradata = av_malloc(header_size + FF_INPUT_BUFFER_PADDING_SIZE);

+    if (!avctx->extradata) {

+        av_log(avctx, AV_LOG_ERROR, "Failed to allocate extradata.\n");

+        ret = AVERROR(ENOMEM);

+        goto fail;

+    }

+    avctx->extradata_size = header_size;

+

+    opus->samples = av_mallocz(frame_size * avctx->channels *

+                               av_get_bytes_per_sample(avctx->sample_fmt));

+    if (!opus->samples) {

+        av_log(avctx, AV_LOG_ERROR, "Failed to allocate samples buffer.\n");

+        ret = AVERROR(ENOMEM);

+        goto fail;

+    }

+

+    ret = opus_multistream_encoder_ctl(enc, OPUS_GET_LOOKAHEAD(&avctx->delay));

+    if (ret != OPUS_OK)

+        av_log(avctx, AV_LOG_WARNING,

+               "Unable to get number of lookahead samples: %s\n",

+               opus_strerror(ret));

+

+    libopus_write_header(avctx, opus->stream_count, coupled_stream_count,

+                         opus_vorbis_channel_map[avctx->channels - 1]);

+

+    ff_af_queue_init(avctx, &opus->afq);

+

+    opus->enc = enc;

+

+    return 0;

+

+fail:

+    opus_multistream_encoder_destroy(enc);

+    av_freep(&avctx->extradata);

+    return ret;

+}

+

+static int libopus_encode(AVCodecContext *avctx, AVPacket *avpkt,

+                          const AVFrame *frame, int *got_packet_ptr)

+{

+    LibopusEncContext *opus = avctx->priv_data;

+    const int sample_size   = avctx->channels *

+                              av_get_bytes_per_sample(avctx->sample_fmt);

+    uint8_t *audio;

+    int ret;

+

+    if (frame) {

+        ff_af_queue_add(&opus->afq, frame);

+        if (frame->nb_samples < opus->opts.packet_size) {

+            audio = opus->samples;

+            memcpy(audio, frame->data[0], frame->nb_samples * sample_size);

+        } else

+            audio = frame->data[0];

+    } else {

+        if (!opus->afq.remaining_samples)

+            return 0;

+        audio = opus->samples;

+        memset(audio, 0, opus->opts.packet_size * sample_size);

+    }

+

+    /* Maximum packet size taken from opusenc in opus-tools. 60ms packets

+     * consist of 3 frames in one packet. The maximum frame size is 1275

+     * bytes along with the largest possible packet header of 7 bytes. */

+    if (ret = ff_alloc_packet(avpkt, (1275 * 3 + 7) * opus->stream_count)) {

+        av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");

+        return ret;

+    }

+

+    if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT)

+        ret = opus_multistream_encode_float(opus->enc, (float *)audio,

+                                            opus->opts.packet_size,

+                                            avpkt->data, avpkt->size);

+    else

+        ret = opus_multistream_encode(opus->enc, (opus_int16 *)audio,

+                                      opus->opts.packet_size,

+                                      avpkt->data, avpkt->size);

+

+    if (ret < 0) {

+        av_log(avctx, AV_LOG_ERROR,

+               "Error encoding frame: %s\n", opus_strerror(ret));

+        return ff_opus_error_to_averror(ret);

+    }

+

+    av_shrink_packet(avpkt, ret);

+

+    ff_af_queue_remove(&opus->afq, opus->opts.packet_size,

+                       &avpkt->pts, &avpkt->duration);

+

+    *got_packet_ptr = 1;

+

+    return 0;

+}

+

+static int av_cold libopus_encode_close(AVCodecContext *avctx)

+{

+    LibopusEncContext *opus = avctx->priv_data;

+

+    opus_multistream_encoder_destroy(opus->enc);

+

+    ff_af_queue_close(&opus->afq);

+

+    av_freep(&opus->samples);

+    av_freep(&avctx->extradata);

+

+    return 0;

+}

+

+#define OFFSET(x) offsetof(LibopusEncContext, opts.x)

+#define FLAGS AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM

+static const AVOption libopus_options[] = {

+    { "application",    "Intended application type",           OFFSET(application),    AV_OPT_TYPE_INT,   { .i64 = OPUS_APPLICATION_AUDIO }, OPUS_APPLICATION_VOIP, OPUS_APPLICATION_RESTRICTED_LOWDELAY, FLAGS, "application" },

+        { "voip",           "Favor improved speech intelligibility",   0, AV_OPT_TYPE_CONST, { .i64 = OPUS_APPLICATION_VOIP },                0, 0, FLAGS, "application" },

+        { "audio",          "Favor faithfulness to the input",         0, AV_OPT_TYPE_CONST, { .i64 = OPUS_APPLICATION_AUDIO },               0, 0, FLAGS, "application" },

+        { "lowdelay",       "Restrict to only the lowest delay modes", 0, AV_OPT_TYPE_CONST, { .i64 = OPUS_APPLICATION_RESTRICTED_LOWDELAY }, 0, 0, FLAGS, "application" },

+    { "frame_duration", "Duration of a frame in milliseconds", OFFSET(frame_duration), AV_OPT_TYPE_FLOAT, { .dbl = 10.0 }, 2.5, 60.0, FLAGS },

+    { "packet_loss",    "Expected packet loss percentage",     OFFSET(packet_loss),    AV_OPT_TYPE_INT,   { .i64 = 0 },    0,   100,  FLAGS },

+    { "vbr",            "Variable bit rate mode",              OFFSET(vbr),            AV_OPT_TYPE_INT,   { .i64 = 1 },    0,   2,    FLAGS, "vbr" },

+        { "off",            "Use constant bit rate", 0, AV_OPT_TYPE_CONST, { .i64 = 0 }, 0, 0, FLAGS, "vbr" },

+        { "on",             "Use variable bit rate", 0, AV_OPT_TYPE_CONST, { .i64 = 1 }, 0, 0, FLAGS, "vbr" },

+        { "constrained",    "Use constrained VBR",   0, AV_OPT_TYPE_CONST, { .i64 = 2 }, 0, 0, FLAGS, "vbr" },

+    { NULL },

+};

+

+static const AVClass libopus_class = {

+    .class_name = "libopus",

+    .item_name  = av_default_item_name,

+    .option     = libopus_options,

+    .version    = LIBAVUTIL_VERSION_INT,

+};

+

+static const AVCodecDefault libopus_defaults[] = {

+    { "b",                 "0" },

+    { "compression_level", "10" },

+    { NULL },

+};

+

+static const int libopus_sample_rates[] = {

+    48000, 24000, 16000, 12000, 8000, 0,

+};

+

+AVCodec ff_libopus_encoder = {

+    .name            = "libopus",

+    .type            = AVMEDIA_TYPE_AUDIO,

+    .id              = AV_CODEC_ID_OPUS,

+    .priv_data_size  = sizeof(LibopusEncContext),

+    .init            = libopus_encode_init,

+    .encode2         = libopus_encode,

+    .close           = libopus_encode_close,

+    .capabilities    = CODEC_CAP_DELAY | CODEC_CAP_SMALL_LAST_FRAME,

+    .sample_fmts     = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,

+                                                      AV_SAMPLE_FMT_FLT,

+                                                      AV_SAMPLE_FMT_NONE },

+    .channel_layouts = ff_vorbis_channel_layouts,

+    .supported_samplerates = libopus_sample_rates,

+    .long_name       = NULL_IF_CONFIG_SMALL("libopus Opus"),

+    .priv_class      = &libopus_class,

+    .defaults        = libopus_defaults,

+};

@@ -29,7 +29,7 @@

 #include "libavutil/avutil.h"

 #define LIBAVCODEC_VERSION_MAJOR 54

-#define LIBAVCODEC_VERSION_MINOR 62

+#define LIBAVCODEC_VERSION_MINOR 63

 #define LIBAVCODEC_VERSION_MICRO 100

 #define LIBAVCODEC_VERSION_INT  AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \

@@ -926,7 +926,7 @@ static int wma_decode_superframe(AVCodecContext *avctx, void *data,

         samples += s->nb_channels * s->frame_len;

     }

-    av_dlog(s->avctx, "%d %d %d %d outbytes:%d eaten:%d\n",

+    av_dlog(s->avctx, "%d %d %d %d outbytes:%td eaten:%d\n",

             s->frame_len_bits, s->block_len_bits, s->frame_len, s->block_len,

             (int8_t *)samples - (int8_t *)data, s->block_align);

@@ -1813,7 +1813,7 @@ int av_get_output_timestamp(struct AVFormatContext *s, int stream,

  *

  * @see av_hex_dump_log, av_pkt_dump2, av_pkt_dump_log2

  */

-void av_hex_dump(FILE *f, uint8_t *buf, int size);

+void av_hex_dump(FILE *f, const uint8_t *buf, int size);

 /**

  * Send a nice hexadecimal dump of a buffer to the log.

@@ -1827,7 +1827,7 @@ void av_hex_dump(FILE *f, uint8_t *buf, int size);

  *

  * @see av_hex_dump, av_pkt_dump2, av_pkt_dump_log2

  */

-void av_hex_dump_log(void *avcl, int level, uint8_t *buf, int size);

+void av_hex_dump_log(void *avcl, int level, const uint8_t *buf, int size);

 /**

  * Send a nice dump of a packet to the specified file stream.

@@ -1414,7 +1414,7 @@ static void pmt_cb(MpegTSFilter *filter, const uint8_t *section, int section_len

     int i;

     av_dlog(ts->stream, "PMT: len %i\n", section_len);

-    hex_dump_debug(ts->stream, (uint8_t *)section, section_len);

+    hex_dump_debug(ts->stream, section, section_len);

     p_end = section + section_len - 4;

     p = section;

@@ -1553,7 +1553,7 @@ static void pat_cb(MpegTSFilter *filter, const uint8_t *section, int section_len

     AVProgram *program;

     av_dlog(ts->stream, "PAT:\n");

-    hex_dump_debug(ts->stream, (uint8_t *)section, section_len);

+    hex_dump_debug(ts->stream, section, section_len);

     p_end = section + section_len - 4;

     p = section;

@@ -1601,7 +1601,7 @@ static void sdt_cb(MpegTSFilter *filter, const uint8_t *section, int section_len

     char *name, *provider_name;

     av_dlog(ts->stream, "SDT:\n");

-    hex_dump_debug(ts->stream, (uint8_t *)section, section_len);

+    hex_dump_debug(ts->stream, section, section_len);

     p_end = section + section_len - 4;

     p = section;

@@ -4146,7 +4146,8 @@ int av_get_frame_filename(char *buf, int buf_size,

     return -1;

 }

-static void hex_dump_internal(void *avcl, FILE *f, int level, uint8_t *buf, int size)

+static void hex_dump_internal(void *avcl, FILE *f, int level,

+                              const uint8_t *buf, int size)

 {

     int len, i, j, c;

 #undef fprintf

@@ -4175,12 +4176,12 @@ static void hex_dump_internal(void *avcl, FILE *f, int level, uint8_t *buf, int

 #undef PRINT

 }

-void av_hex_dump(FILE *f, uint8_t *buf, int size)

+void av_hex_dump(FILE *f, const uint8_t *buf, int size)

 {

     hex_dump_internal(NULL, f, 0, buf, size);

 }

-void av_hex_dump_log(void *avcl, int level, uint8_t *buf, int size)

+void av_hex_dump_log(void *avcl, int level, const uint8_t *buf, int size)

 {

     hex_dump_internal(avcl, NULL, level, buf, size);

 }