[libav-commits] Support AAC encoding via the external library fdk-aac

Martin Storsjö git at libav.org
Thu Jul 12 10:20:31 CEST 2012


Module: libav
Branch: master
Commit: 37eeb5e273cdea19a7c9979e0d032dbc0868df88

Author:    Martin Storsjö <martin at martin.st>
Committer: Martin Storsjö <martin at martin.st>
Date:      Thu Jun 28 16:46:24 2012 +0300

Support AAC encoding via the external library fdk-aac

Signed-off-by: Martin Storsjö <martin at martin.st>

---

 Changelog                  |    1 +
 configure                  |    5 +
 doc/general.texi           |   12 ++-
 libavcodec/Makefile        |    1 +
 libavcodec/allcodecs.c     |    1 +
 libavcodec/libfdk-aacenc.c |  384 ++++++++++++++++++++++++++++++++++++++++++++
 libavcodec/version.h       |    2 +-
 7 files changed, 403 insertions(+), 3 deletions(-)

diff --git a/Changelog b/Changelog
index 2fb5e3d..c56740c 100644
--- a/Changelog
+++ b/Changelog
@@ -33,6 +33,7 @@ version <next>:
 - Microsoft ATC Screen decoder
 - RTSP listen mode
 - TechSmith Screen Codec 2 decoder
+- AAC encoding via libfdk-aac
 
 
 version 0.8:
diff --git a/configure b/configure
index 4d83d4b..397be73 100755
--- a/configure
+++ b/configure
@@ -170,6 +170,7 @@ External library support:
   --enable-libdc1394       enable IIDC-1394 grabbing using libdc1394
                            and libraw1394 [no]
   --enable-libfaac         enable FAAC support via libfaac [no]
+  --enable-libfdk-aac      enable AAC support via libfdk-aac [no]
   --enable-libfreetype     enable libfreetype [no]
   --enable-libgsm          enable GSM support via libgsm [no]
   --enable-libilbc         enable iLBC de/encoding via libilbc [no]
@@ -943,6 +944,7 @@ CONFIG_LIST="
     libcdio
     libdc1394
     libfaac
+    libfdk_aac
     libfreetype
     libgsm
     libilbc
@@ -1448,6 +1450,7 @@ h264_parser_select="golomb h264dsp h264pred"
 
 # external libraries
 libfaac_encoder_deps="libfaac"
+libfdk_aac_encoder_deps="libfdk_aac"
 libgsm_decoder_deps="libgsm"
 libgsm_encoder_deps="libgsm"
 libgsm_ms_decoder_deps="libgsm"
@@ -2968,6 +2971,7 @@ enabled avisynth   && require2 vfw32 "windows.h vfw.h" AVIFileInit -lavifil32
 enabled frei0r     && { check_header frei0r.h || die "ERROR: frei0r.h header not found"; }
 enabled gnutls     && require_pkg_config gnutls gnutls/gnutls.h gnutls_global_init
 enabled libfaac    && require2 libfaac "stdint.h faac.h" faacEncGetVersion -lfaac
+enabled libfdk_aac && require  libfdk_aac fdk-aac/aacenc_lib.h aacEncOpen -lfdk-aac
 enabled libfreetype && require_pkg_config freetype2 "ft2build.h freetype/freetype.h" FT_Init_FreeType
 enabled libgsm     && require  libgsm gsm/gsm.h gsm_create -lgsm
 enabled libilbc    && require  libilbc ilbc.h WebRtcIlbcfix_InitDecode -lilbc
@@ -3259,6 +3263,7 @@ echo "gnutls enabled            ${gnutls-no}"
 echo "libcdio support           ${libcdio-no}"
 echo "libdc1394 support         ${libdc1394-no}"
 echo "libfaac enabled           ${libfaac-no}"
+echo "libfdk-aac enabled        ${libfdk_aac-no}"
 echo "libgsm enabled            ${libgsm-no}"
 echo "libilbc enabled           ${libilbc-no}"
 echo "libmp3lame enabled        ${libmp3lame-no}"
diff --git a/doc/general.texi b/doc/general.texi
index 7e9cfaf..fcac114 100644
--- a/doc/general.texi
+++ b/doc/general.texi
@@ -18,8 +18,8 @@ explicitly requested by passing the appropriate flags to
 
 @section OpenCORE and VisualOn libraries
 
-Spun off Google Android sources, OpenCore and VisualOn libraries provide
-encoders for a number of audio codecs.
+Spun off Google Android sources, OpenCore, VisualOn and Fraunhofer
+libraries provide encoders for a number of audio codecs.
 
 @float NOTE
 OpenCORE and VisualOn libraries are under the Apache License 2.0
@@ -55,6 +55,14 @@ Go to @url{http://sourceforge.net/projects/opencore-amr/} and follow the
 instructions for installing the library.
 Then pass @code{--enable-libvo-amrwbenc} to configure to enable it.
 
+ at subsection Fraunhofer AAC library
+
+Libav can make use of the Fraunhofer AAC library for AAC encoding.
+
+Go to @url{http://sourceforge.net/projects/opencore-amr/} and follow the
+instructions for installing the library.
+Then pass @code{--enable-libfdk-aac} to configure to enable it.
+
 @section LAME
 
 Libav can make use of the LAME library for MP3 encoding.
diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index ac97d34..8d38ca2 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -595,6 +595,7 @@ OBJS-$(CONFIG_WTV_DEMUXER)             += mpeg4audio.o mpegaudiodata.o
 
 # external codec libraries
 OBJS-$(CONFIG_LIBFAAC_ENCODER)            += libfaac.o audio_frame_queue.o
+OBJS-$(CONFIG_LIBFDK_AAC_ENCODER)         += libfdk-aacenc.o audio_frame_queue.o
 OBJS-$(CONFIG_LIBGSM_DECODER)             += libgsm.o
 OBJS-$(CONFIG_LIBGSM_ENCODER)             += libgsm.o
 OBJS-$(CONFIG_LIBGSM_MS_DECODER)          += libgsm.o
diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c
index 068f191..bd48728 100644
--- a/libavcodec/allcodecs.c
+++ b/libavcodec/allcodecs.c
@@ -380,6 +380,7 @@ void avcodec_register_all(void)
 
     /* external libraries */
     REGISTER_ENCODER (LIBFAAC, libfaac);
+    REGISTER_ENCODER (LIBFDK_AAC, libfdk_aac);
     REGISTER_ENCDEC  (LIBGSM, libgsm);
     REGISTER_ENCDEC  (LIBGSM_MS, libgsm_ms);
     REGISTER_ENCDEC  (LIBILBC, libilbc);
diff --git a/libavcodec/libfdk-aacenc.c b/libavcodec/libfdk-aacenc.c
new file mode 100644
index 0000000..6fda53c
--- /dev/null
+++ b/libavcodec/libfdk-aacenc.c
@@ -0,0 +1,384 @@
+/*
+ * AAC encoder wrapper
+ * Copyright (c) 2012 Martin Storsjo
+ *
+ * This file is part of Libav.
+ *
+ * Libav 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.
+ *
+ * Libav 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 Libav; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include <fdk-aac/aacenc_lib.h>
+
+#include "avcodec.h"
+#include "audio_frame_queue.h"
+#include "internal.h"
+#include "libavutil/audioconvert.h"
+#include "libavutil/opt.h"
+
+typedef struct AACContext {
+    const AVClass *class;
+    HANDLE_AACENCODER handle;
+    int afterburner;
+    int eld_sbr;
+    int signaling;
+
+    AudioFrameQueue afq;
+} AACContext;
+
+static const AVOption aac_enc_options[] = {
+    { "afterburner", "Afterburner (improved quality)", offsetof(AACContext, afterburner), AV_OPT_TYPE_INT, { 1 }, 0, 1, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM },
+    { "eld_sbr", "Enable SBR for ELD (for SBR in other configurations, use the -profile parameter)", offsetof(AACContext, eld_sbr), AV_OPT_TYPE_INT, { 0 }, 0, 1, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM },
+    { "signaling", "SBR/PS signaling style", offsetof(AACContext, signaling), AV_OPT_TYPE_INT, { -1 }, -1, 2, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM, "signaling" },
+    { "default", "Choose signaling implicitly (explicit hierarchical by default, implicit if global header is disabled)", 0, AV_OPT_TYPE_CONST, { -1 }, 0, 0, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM, "signaling" },
+    { "implicit", "Implicit backwards compatible signaling", 0, AV_OPT_TYPE_CONST, { 0 }, 0, 0, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM, "signaling" },
+    { "explicit_sbr", "Explicit SBR, implicit PS signaling", 0, AV_OPT_TYPE_CONST, { 1 }, 0, 0, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM, "signaling" },
+    { "explicit_hierarchical", "Explicit hierarchical signaling", 0, AV_OPT_TYPE_CONST, { 2 }, 0, 0, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM, "signaling" },
+    { NULL }
+};
+
+static const AVClass aac_enc_class = {
+    "libfdk_aac", av_default_item_name, aac_enc_options, LIBAVUTIL_VERSION_INT
+};
+
+static const char *aac_get_error(AACENC_ERROR err)
+{
+    switch (err) {
+    case AACENC_OK:
+        return "No error";
+    case AACENC_INVALID_HANDLE:
+        return "Invalid handle";
+    case AACENC_MEMORY_ERROR:
+        return "Memory allocation error";
+    case AACENC_UNSUPPORTED_PARAMETER:
+        return "Unsupported parameter";
+    case AACENC_INVALID_CONFIG:
+        return "Invalid config";
+    case AACENC_INIT_ERROR:
+        return "Initialization error";
+    case AACENC_INIT_AAC_ERROR:
+        return "AAC library initialization error";
+    case AACENC_INIT_SBR_ERROR:
+        return "SBR library initialization error";
+    case AACENC_INIT_TP_ERROR:
+        return "Transport library initialization error";
+    case AACENC_INIT_META_ERROR:
+        return "Metadata library initialization error";
+    case AACENC_ENCODE_ERROR:
+        return "Encoding error";
+    case AACENC_ENCODE_EOF:
+        return "End of file";
+    default:
+        return "Unknown error";
+    }
+}
+
+static int aac_encode_close(AVCodecContext *avctx)
+{
+    AACContext *s = avctx->priv_data;
+
+    if (s->handle)
+        aacEncClose(&s->handle);
+#if FF_API_OLD_ENCODE_AUDIO
+    av_freep(&avctx->coded_frame);
+#endif
+    av_freep(&avctx->extradata);
+    ff_af_queue_close(&s->afq);
+
+    return 0;
+}
+
+static av_cold int aac_encode_init(AVCodecContext *avctx)
+{
+    AACContext *s = avctx->priv_data;
+    int ret = AVERROR(EINVAL);
+    AACENC_InfoStruct info = { 0 };
+    CHANNEL_MODE mode;
+    AACENC_ERROR err;
+    int aot = FF_PROFILE_AAC_LOW + 1;
+    int sce = 0, cpe = 0;
+
+    if ((err = aacEncOpen(&s->handle, 0, avctx->channels)) != AACENC_OK) {
+        av_log(avctx, AV_LOG_ERROR, "Unable to open the encoder: %s\n",
+               aac_get_error(err));
+        goto error;
+    }
+
+    if (avctx->profile != FF_PROFILE_UNKNOWN)
+        aot = avctx->profile + 1;
+
+    if ((err = aacEncoder_SetParam(s->handle, AACENC_AOT, aot)) != AACENC_OK) {
+        av_log(avctx, AV_LOG_ERROR, "Unable to set the AOT %d: %s\n",
+               aot, aac_get_error(err));
+        goto error;
+    }
+
+    if (aot == FF_PROFILE_AAC_ELD + 1 && s->eld_sbr) {
+        if ((err = aacEncoder_SetParam(s->handle, AACENC_SBR_MODE,
+                                       1)) != AACENC_OK) {
+            av_log(avctx, AV_LOG_ERROR, "Unable to enable SBR for ELD: %s\n",
+                   aac_get_error(err));
+            goto error;
+        }
+    }
+
+    if ((err = aacEncoder_SetParam(s->handle, AACENC_SAMPLERATE,
+                                   avctx->sample_rate)) != AACENC_OK) {
+        av_log(avctx, AV_LOG_ERROR, "Unable to set the sample rate %d: %s\n",
+               avctx->sample_rate, aac_get_error(err));
+        goto error;
+    }
+
+    switch (avctx->channels) {
+    case 1: mode = MODE_1;       sce = 1; cpe = 0; break;
+    case 2: mode = MODE_2;       sce = 0; cpe = 1; break;
+    case 3: mode = MODE_1_2;     sce = 1; cpe = 1; break;
+    case 4: mode = MODE_1_2_1;   sce = 2; cpe = 1; break;
+    case 5: mode = MODE_1_2_2;   sce = 1; cpe = 2; break;
+    case 6: mode = MODE_1_2_2_1; sce = 2; cpe = 2; break;
+    default:
+        av_log(avctx, AV_LOG_ERROR,
+               "Unsupported number of channels %d\n", avctx->channels);
+        goto error;
+    }
+
+    if ((err = aacEncoder_SetParam(s->handle, AACENC_CHANNELMODE,
+                                   mode)) != AACENC_OK) {
+        av_log(avctx, AV_LOG_ERROR,
+               "Unable to set channel mode %d: %s\n", mode, aac_get_error(err));
+        goto error;
+    }
+
+    if ((err = aacEncoder_SetParam(s->handle, AACENC_CHANNELORDER,
+                                   1)) != AACENC_OK) {
+        av_log(avctx, AV_LOG_ERROR,
+               "Unable to set wav channel order %d: %s\n",
+               mode, aac_get_error(err));
+        goto error;
+    }
+
+    if (avctx->flags & CODEC_FLAG_QSCALE) {
+        int mode = avctx->global_quality;
+        if (mode <  1 || mode > 5) {
+            av_log(avctx, AV_LOG_WARNING,
+                   "VBR quality %d out of range, should be 1-5\n", mode);
+            mode = av_clip(mode, 1, 5);
+        }
+        if ((err = aacEncoder_SetParam(s->handle, AACENC_BITRATEMODE,
+                                       mode)) != AACENC_OK) {
+            av_log(avctx, AV_LOG_ERROR, "Unable to set the VBR bitrate mode %d: %s\n",
+                   mode, aac_get_error(err));
+            goto error;
+        }
+    } else {
+        if (avctx->bit_rate <= 0) {
+            if (avctx->profile == FF_PROFILE_AAC_HE_V2) {
+                sce = 1;
+                cpe = 0;
+            }
+            avctx->bit_rate = (96*sce + 128*cpe) * avctx->sample_rate / 44;
+            if (avctx->profile == FF_PROFILE_AAC_HE ||
+                avctx->profile == FF_PROFILE_AAC_HE_V2 ||
+                s->eld_sbr)
+                avctx->bit_rate /= 2;
+        }
+        if ((err = aacEncoder_SetParam(s->handle, AACENC_BITRATE,
+                                       avctx->bit_rate)) != AACENC_OK) {
+            av_log(avctx, AV_LOG_ERROR, "Unable to set the bitrate %d: %s\n",
+                   avctx->bit_rate, aac_get_error(err));
+            goto error;
+        }
+    }
+
+    /* Choose bitstream format - if global header is requested, use
+     * raw access units, otherwise use ADTS. */
+    if ((err = aacEncoder_SetParam(s->handle, AACENC_TRANSMUX,
+                                   avctx->flags & CODEC_FLAG_GLOBAL_HEADER ? 0 : 2)) != AACENC_OK) {
+        av_log(avctx, AV_LOG_ERROR, "Unable to set the transmux format: %s\n",
+               aac_get_error(err));
+        goto error;
+    }
+
+    /* If no signaling mode is chosen, use explicit hierarchical signaling
+     * if using mp4 mode (raw access units, with global header) and
+     * implicit signaling if using ADTS. */
+    if (s->signaling < 0)
+        s->signaling = avctx->flags & CODEC_FLAG_GLOBAL_HEADER ? 2 : 0;
+
+    if ((err = aacEncoder_SetParam(s->handle, AACENC_SIGNALING_MODE,
+                                   s->signaling)) != AACENC_OK) {
+        av_log(avctx, AV_LOG_ERROR, "Unable to set signaling mode %d: %s\n",
+               s->signaling, aac_get_error(err));
+        goto error;
+    }
+
+    if ((err = aacEncoder_SetParam(s->handle, AACENC_AFTERBURNER,
+                                   s->afterburner)) != AACENC_OK) {
+        av_log(avctx, AV_LOG_ERROR, "Unable to set afterburner to %d: %s\n",
+               s->afterburner, aac_get_error(err));
+        goto error;
+    }
+
+    if ((err = aacEncEncode(s->handle, NULL, NULL, NULL, NULL)) != AACENC_OK) {
+        av_log(avctx, AV_LOG_ERROR, "Unable to initialize the encoder: %s\n",
+               aac_get_error(err));
+        return AVERROR(EINVAL);
+    }
+
+    if ((err = aacEncInfo(s->handle, &info)) != AACENC_OK) {
+        av_log(avctx, AV_LOG_ERROR, "Unable to get encoder info: %s\n",
+               aac_get_error(err));
+        goto error;
+    }
+
+#if FF_API_OLD_ENCODE_AUDIO
+    avctx->coded_frame = avcodec_alloc_frame();
+    if (!avctx->coded_frame) {
+        ret = AVERROR(ENOMEM);
+        goto error;
+    }
+#endif
+    avctx->frame_size = info.frameLength;
+    avctx->delay      = info.encoderDelay;
+    ff_af_queue_init(avctx, &s->afq);
+
+    if (avctx->flags & CODEC_FLAG_GLOBAL_HEADER) {
+        avctx->extradata_size = info.confSize;
+        avctx->extradata      = av_mallocz(avctx->extradata_size +
+                                           FF_INPUT_BUFFER_PADDING_SIZE);
+        if (!avctx->extradata) {
+            ret = AVERROR(ENOMEM);
+            goto error;
+        }
+
+        memcpy(avctx->extradata, info.confBuf, info.confSize);
+    }
+    return 0;
+error:
+    aac_encode_close(avctx);
+    return ret;
+}
+
+static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
+                            const AVFrame *frame, int *got_packet_ptr)
+{
+    AACContext    *s        = avctx->priv_data;
+    AACENC_BufDesc in_buf   = { 0 }, out_buf = { 0 };
+    AACENC_InArgs  in_args  = { 0 };
+    AACENC_OutArgs out_args = { 0 };
+    int in_buffer_identifier = IN_AUDIO_DATA;
+    int in_buffer_size, in_buffer_element_size;
+    int out_buffer_identifier = OUT_BITSTREAM_DATA;
+    int out_buffer_size, out_buffer_element_size;
+    void *in_ptr, *out_ptr;
+    int ret;
+    AACENC_ERROR err;
+
+    /* handle end-of-stream small frame and flushing */
+    if (!frame) {
+        in_args.numInSamples = -1;
+    } else {
+        in_ptr                   = frame->data[0];
+        in_buffer_size           = 2 * avctx->channels * frame->nb_samples;
+        in_buffer_element_size   = 2;
+
+        in_args.numInSamples     = avctx->channels * frame->nb_samples;
+        in_buf.numBufs           = 1;
+        in_buf.bufs              = &in_ptr;
+        in_buf.bufferIdentifiers = &in_buffer_identifier;
+        in_buf.bufSizes          = &in_buffer_size;
+        in_buf.bufElSizes        = &in_buffer_element_size;
+
+        /* add current frame to the queue */
+        if ((ret = ff_af_queue_add(&s->afq, frame) < 0))
+            return ret;
+    }
+
+    /* The maximum packet size is 6144 bits aka 768 bytes per channel. */
+    if ((ret = ff_alloc_packet(avpkt, FFMAX(8192, 768 * avctx->channels)))) {
+        av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
+        return ret;
+    }
+
+    out_ptr                   = avpkt->data;
+    out_buffer_size           = avpkt->size;
+    out_buffer_element_size   = 1;
+    out_buf.numBufs           = 1;
+    out_buf.bufs              = &out_ptr;
+    out_buf.bufferIdentifiers = &out_buffer_identifier;
+    out_buf.bufSizes          = &out_buffer_size;
+    out_buf.bufElSizes        = &out_buffer_element_size;
+
+    if ((err = aacEncEncode(s->handle, &in_buf, &out_buf, &in_args,
+                            &out_args)) != AACENC_OK) {
+        if (!frame && err == AACENC_ENCODE_EOF)
+            return 0;
+        av_log(avctx, AV_LOG_ERROR, "Unable to encode frame: %s\n",
+               aac_get_error(err));
+        return AVERROR(EINVAL);
+    }
+
+    if (!out_args.numOutBytes)
+        return 0;
+
+    /* Get the next frame pts & duration */
+    ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
+                       &avpkt->duration);
+
+    avpkt->size     = out_args.numOutBytes;
+    *got_packet_ptr = 1;
+    return 0;
+}
+
+static const AVProfile profiles[] = {
+    { FF_PROFILE_AAC_LOW,   "LC"       },
+    { FF_PROFILE_AAC_HE,    "HE-AAC"   },
+    { FF_PROFILE_AAC_HE_V2, "HE-AACv2" },
+    { FF_PROFILE_AAC_LD,    "LD"       },
+    { FF_PROFILE_AAC_ELD,   "ELD"      },
+    { FF_PROFILE_UNKNOWN },
+};
+
+static const AVCodecDefault aac_encode_defaults[] = {
+    { "b", "0" },
+    { NULL }
+};
+
+static const uint64_t aac_channel_layout[] = {
+    AV_CH_LAYOUT_MONO,
+    AV_CH_LAYOUT_STEREO,
+    AV_CH_LAYOUT_SURROUND,
+    AV_CH_LAYOUT_4POINT0,
+    AV_CH_LAYOUT_5POINT0_BACK,
+    AV_CH_LAYOUT_5POINT1_BACK,
+    0,
+};
+
+AVCodec ff_libfdk_aac_encoder = {
+    .name            = "libfdk_aac",
+    .type            = AVMEDIA_TYPE_AUDIO,
+    .id              = CODEC_ID_AAC,
+    .priv_data_size  = sizeof(AACContext),
+    .init            = aac_encode_init,
+    .encode2         = aac_encode_frame,
+    .close           = aac_encode_close,
+    .capabilities    = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY,
+    .sample_fmts     = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
+                                                      AV_SAMPLE_FMT_NONE },
+    .long_name       = NULL_IF_CONFIG_SMALL("Fraunhofer FDK AAC"),
+    .priv_class      = &aac_enc_class,
+    .defaults        = aac_encode_defaults,
+    .profiles        = profiles,
+    .channel_layouts = aac_channel_layout,
+};
diff --git a/libavcodec/version.h b/libavcodec/version.h
index 6f47df9..48db12e 100644
--- a/libavcodec/version.h
+++ b/libavcodec/version.h
@@ -27,7 +27,7 @@
  */
 
 #define LIBAVCODEC_VERSION_MAJOR 54
-#define LIBAVCODEC_VERSION_MINOR 18
+#define LIBAVCODEC_VERSION_MINOR 19
 #define LIBAVCODEC_VERSION_MICRO  0
 
 #define LIBAVCODEC_VERSION_INT  AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \



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