Merge commit 'fd41cb43702498948ff14ba8c284fd5c15fc729d'

* commit 'fd41cb43702498948ff14ba8c284fd5c15fc729d':
  avconv: improve sample format negotiation for decoder request
  Opus encoder using libopus
  mpegts: Drop pointless casting of hex_dump_debug arguments
  avformat: const correctness for av_hex_dump / av_hex_dump_log
  wmadec: Adjust debug printf argument length modifier

Conflicts:
	Changelog
	ffmpeg.c
	libavcodec/libopusdec.c
	libavcodec/version.h

Merged-by: Michael Niedermayer <michaelni@gmx.at>

Changelog

@@ -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:

configure

@@ -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"

doc/general.texi

@@ -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

libavcodec/Makefile

@@ -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 \

libavcodec/allcodecs.c

@@ -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);

libavcodec/libopus.c

@@ -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);
+    }
+}

libavcodec/libopus.h

@@ -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 */

libavcodec/libopusdec.c

@@ -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

libavcodec/libopusenc.c

@@ -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,
+};

libavcodec/version.h

@@ -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, \

libavcodec/wmadec.c

@@ -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);
 

libavformat/avformat.h

@@ -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.

libavformat/mpegts.c

@@ -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;

libavformat/utils.c

@@ -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);
 }