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d46c1c72e4
FFmpeg/libavcodec/libfdk-aacenc.c
Michael Niedermayer d46c1c72e4 Merge commit 'e6153f173a49e5bfa70b0c04d2f82930533597b9' 
* commit 'e6153f173a49e5bfa70b0c04d2f82930533597b9':
  avopt: Store defaults for AV_OPT_TYPE_INT in the i64 union member

Conflicts:
	libavcodec/libopenjpegdec.c
	libavcodec/libopenjpegenc.c
	libavcodec/libx264.c
	libavcodec/mpeg12enc.c
	libavcodec/options_table.h
	libavcodec/snowenc.c
	libavcodec/tiffenc.c
	libavdevice/v4l2.c
	libavdevice/x11grab.c
	libavfilter/af_amix.c
	libavfilter/af_asyncts.c
	libavfilter/af_join.c
	libavfilter/buffersrc.c
	libavfilter/src_movie.c
	libavfilter/vf_delogo.c
	libavfilter/vf_drawtext.c
	libavformat/http.c
	libavformat/img2dec.c
	libavformat/img2enc.c
	libavformat/movenc.c
	libavformat/mpegenc.c
	libavformat/mpegtsenc.c
	libavformat/options_table.h
	libavformat/segment.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-09-05 14:33:32 +02:00

411 lines
15 KiB
C
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/*
* AAC encoder wrapper
* Copyright (c) 2012 Martin Storsjo
*
* 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 <fdk-aac/aacenc_lib.h>
#include "avcodec.h"
#include "audio_frame_queue.h"
#include "internal.h"
#include "libavutil/audioconvert.h"
#include "libavutil/common.h"
#include "libavutil/opt.h"
typedef struct AACContext {
const AVClass *class;
HANDLE_AACENCODER handle;
int afterburner;
int eld_sbr;
int signaling;
int latm;
int header_period;
AudioFrameQueue afq;
} AACContext;
static const AVOption aac_enc_options[] = {
{ "afterburner", "Afterburner (improved quality)", offsetof(AACContext, afterburner), AV_OPT_TYPE_INT, { .i64 = 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, { .i64 = 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, { .i64 = -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, { .i64 = -1 }, 0, 0, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM, "signaling" },
{ "implicit", "Implicit backwards compatible signaling", 0, AV_OPT_TYPE_CONST, { .i64 = 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, { .i64 = 1 }, 0, 0, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM, "signaling" },
{ "explicit_hierarchical", "Explicit hierarchical signaling", 0, AV_OPT_TYPE_CONST, { .i64 = 2 }, 0, 0, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM, "signaling" },
{ "latm", "Output LATM/LOAS encapsulated data", offsetof(AACContext, latm), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 1, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM },
{ "header_period", "StreamMuxConfig and PCE repetition period (in frames)", offsetof(AACContext, header_period), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 0xffff, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM },
{ 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 : s->latm ? 10 : 2)) != AACENC_OK) {
av_log(avctx, AV_LOG_ERROR, "Unable to set the transmux format: %s\n",
aac_get_error(err));
goto error;
}
if (s->latm && s->header_period) {
if ((err = aacEncoder_SetParam(s->handle, AACENC_HEADER_PERIOD,
s->header_period)) != AACENC_OK) {
av_log(avctx, AV_LOG_ERROR, "Unable to set header period: %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 (avctx->cutoff > 0) {
if (avctx->cutoff < (avctx->sample_rate + 255) >> 8) {
av_log(avctx, AV_LOG_ERROR, "cutoff valid range is %d-20000\n",
(avctx->sample_rate + 255) >> 8);
goto error;
}
if ((err = aacEncoder_SetParam(s->handle, AACENC_BANDWIDTH,
avctx->cutoff)) != AACENC_OK) {
av_log(avctx, AV_LOG_ERROR, "Unable to set the encoder bandwith to %d: %s\n",
avctx->cutoff, 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_packet2(avctx, avpkt, FFMAX(8192, 768 * avctx->channels))))
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 = AV_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,
};
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