[FFmpeg-cvslog] Support AAC encoding via the external library fdk-aac
Martin Storsjö
git at videolan.org
Fri Jul 13 00:56:59 CEST 2012
ffmpeg | branch: master | Martin Storsjö <martin at martin.st> | Thu Jun 28 16:46:24 2012 +0300| [37eeb5e273cdea19a7c9979e0d032dbc0868df88] | committer: Martin Storsjö
Support AAC encoding via the external library fdk-aac
Signed-off-by: Martin Storsjö <martin at martin.st>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=37eeb5e273cdea19a7c9979e0d032dbc0868df88
---
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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