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FFmpeg/libavcodec/libfaac.c
Paul B Mahol ae2c33b0c2 cosmetics: remove superfluous curly brackets
Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-03-23 03:09:07 +01:00

237 lines
7.3 KiB
C

/*
* Interface to libfaac for aac encoding
* Copyright (c) 2002 Gildas Bazin <gbazin@netcourrier.com>
*
* 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
*/
/**
* @file
* Interface to libfaac for aac encoding.
*/
#include <faac.h>
#include "avcodec.h"
#include "audio_frame_queue.h"
#include "internal.h"
/* libfaac has an encoder delay of 1024 samples */
#define FAAC_DELAY_SAMPLES 1024
typedef struct FaacAudioContext {
faacEncHandle faac_handle;
AudioFrameQueue afq;
} FaacAudioContext;
static const int channel_maps[][6] = {
{ 2, 0, 1 }, //< C L R
{ 2, 0, 1, 3 }, //< C L R Cs
{ 2, 0, 1, 3, 4 }, //< C L R Ls Rs
{ 2, 0, 1, 4, 5, 3 }, //< C L R Ls Rs LFE
};
static av_cold int Faac_encode_close(AVCodecContext *avctx)
{
FaacAudioContext *s = avctx->priv_data;
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
av_freep(&avctx->extradata);
ff_af_queue_close(&s->afq);
if (s->faac_handle)
faacEncClose(s->faac_handle);
return 0;
}
static av_cold int Faac_encode_init(AVCodecContext *avctx)
{
FaacAudioContext *s = avctx->priv_data;
faacEncConfigurationPtr faac_cfg;
unsigned long samples_input, max_bytes_output;
int ret;
/* number of channels */
if (avctx->channels < 1 || avctx->channels > 6) {
av_log(avctx, AV_LOG_ERROR, "encoding %d channel(s) is not allowed\n", avctx->channels);
ret = AVERROR(EINVAL);
goto error;
}
s->faac_handle = faacEncOpen(avctx->sample_rate,
avctx->channels,
&samples_input, &max_bytes_output);
if (!s->faac_handle) {
av_log(avctx, AV_LOG_ERROR, "error in faacEncOpen()\n");
ret = AVERROR_UNKNOWN;
goto error;
}
/* check faac version */
faac_cfg = faacEncGetCurrentConfiguration(s->faac_handle);
if (faac_cfg->version != FAAC_CFG_VERSION) {
av_log(avctx, AV_LOG_ERROR, "wrong libfaac version (compiled for: %d, using %d)\n", FAAC_CFG_VERSION, faac_cfg->version);
ret = AVERROR(EINVAL);
goto error;
}
/* put the options in the configuration struct */
switch(avctx->profile) {
case FF_PROFILE_AAC_MAIN:
faac_cfg->aacObjectType = MAIN;
break;
case FF_PROFILE_UNKNOWN:
case FF_PROFILE_AAC_LOW:
faac_cfg->aacObjectType = LOW;
break;
case FF_PROFILE_AAC_SSR:
faac_cfg->aacObjectType = SSR;
break;
case FF_PROFILE_AAC_LTP:
faac_cfg->aacObjectType = LTP;
break;
default:
av_log(avctx, AV_LOG_ERROR, "invalid AAC profile\n");
ret = AVERROR(EINVAL);
goto error;
}
faac_cfg->mpegVersion = MPEG4;
faac_cfg->useTns = 0;
faac_cfg->allowMidside = 1;
faac_cfg->bitRate = avctx->bit_rate / avctx->channels;
faac_cfg->bandWidth = avctx->cutoff;
if(avctx->flags & CODEC_FLAG_QSCALE) {
faac_cfg->bitRate = 0;
faac_cfg->quantqual = avctx->global_quality / FF_QP2LAMBDA;
}
faac_cfg->outputFormat = 1;
faac_cfg->inputFormat = FAAC_INPUT_16BIT;
if (avctx->channels > 2)
memcpy(faac_cfg->channel_map, channel_maps[avctx->channels-3],
avctx->channels * sizeof(int));
avctx->frame_size = samples_input / avctx->channels;
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame= avcodec_alloc_frame();
if (!avctx->coded_frame) {
ret = AVERROR(ENOMEM);
goto error;
}
#endif
/* Set decoder specific info */
avctx->extradata_size = 0;
if (avctx->flags & CODEC_FLAG_GLOBAL_HEADER) {
unsigned char *buffer = NULL;
unsigned long decoder_specific_info_size;
if (!faacEncGetDecoderSpecificInfo(s->faac_handle, &buffer,
&decoder_specific_info_size)) {
avctx->extradata = av_malloc(decoder_specific_info_size + FF_INPUT_BUFFER_PADDING_SIZE);
if (!avctx->extradata) {
ret = AVERROR(ENOMEM);
goto error;
}
avctx->extradata_size = decoder_specific_info_size;
memcpy(avctx->extradata, buffer, avctx->extradata_size);
faac_cfg->outputFormat = 0;
}
#undef free
free(buffer);
#define free please_use_av_free
}
if (!faacEncSetConfiguration(s->faac_handle, faac_cfg)) {
av_log(avctx, AV_LOG_ERROR, "libfaac doesn't support this output format!\n");
ret = AVERROR(EINVAL);
goto error;
}
avctx->delay = FAAC_DELAY_SAMPLES;
ff_af_queue_init(avctx, &s->afq);
return 0;
error:
Faac_encode_close(avctx);
return ret;
}
static int Faac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
FaacAudioContext *s = avctx->priv_data;
int bytes_written, ret;
int num_samples = frame ? frame->nb_samples : 0;
void *samples = frame ? frame->data[0] : NULL;
if ((ret = ff_alloc_packet2(avctx, avpkt, (7 + 768) * avctx->channels)))
return ret;
bytes_written = faacEncEncode(s->faac_handle, samples,
num_samples * avctx->channels,
avpkt->data, avpkt->size);
if (bytes_written < 0) {
av_log(avctx, AV_LOG_ERROR, "faacEncEncode() error\n");
return bytes_written;
}
/* add current frame to the queue */
if (frame) {
if ((ret = ff_af_queue_add(&s->afq, frame) < 0))
return ret;
}
if (!bytes_written)
return 0;
/* Get the next frame pts/duration */
ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
&avpkt->duration);
avpkt->size = bytes_written;
*got_packet_ptr = 1;
return 0;
}
static const AVProfile profiles[] = {
{ FF_PROFILE_AAC_MAIN, "Main" },
{ FF_PROFILE_AAC_LOW, "LC" },
{ FF_PROFILE_AAC_SSR, "SSR" },
{ FF_PROFILE_AAC_LTP, "LTP" },
{ FF_PROFILE_UNKNOWN },
};
AVCodec ff_libfaac_encoder = {
.name = "libfaac",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_AAC,
.priv_data_size = sizeof(FaacAudioContext),
.init = Faac_encode_init,
.encode2 = Faac_encode_frame,
.close = Faac_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("libfaac AAC (Advanced Audio Codec)"),
.profiles = NULL_IF_CONFIG_SMALL(profiles),
};