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/*
* AAC encoder wrapper
* Copyright (c) 2010 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 <vo-aacenc/voAAC.h>
#include <vo-aacenc/cmnMemory.h>
#include "avcodec.h"
#include "audio_frame_queue.h"
#include "internal.h"
#include "mpeg4audio.h"
#define FRAME_SIZE 1024
#define ENC_DELAY 1600
typedef struct AACContext {
VO_AUDIO_CODECAPI codec_api;
VO_HANDLE handle;
VO_MEM_OPERATOR mem_operator;
VO_CODEC_INIT_USERDATA user_data;
VO_PBYTE end_buffer;
AudioFrameQueue afq;
int last_frame;
int last_samples;
} AACContext;
static int aac_encode_close(AVCodecContext *avctx)
{
AACContext *s = avctx->priv_data;
s->codec_api.Uninit(s->handle);
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
av_freep(&avctx->extradata);
ff_af_queue_close(&s->afq);
av_freep(&s->end_buffer);
return 0;
}
static av_cold int aac_encode_init(AVCodecContext *avctx)
{
AACContext *s = avctx->priv_data;
AACENC_PARAM params = { 0 };
int index, ret;
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame = avcodec_alloc_frame();
if (!avctx->coded_frame)
return AVERROR(ENOMEM);
#endif
avctx->frame_size = FRAME_SIZE;
avctx->delay = ENC_DELAY;
s->last_frame = 2;
ff_af_queue_init(avctx, &s->afq);
s->end_buffer = av_mallocz(avctx->frame_size * avctx->channels * 2);
if (!s->end_buffer) {
ret = AVERROR(ENOMEM);
goto error;
}
voGetAACEncAPI(&s->codec_api);
s->mem_operator.Alloc = cmnMemAlloc;
s->mem_operator.Copy = cmnMemCopy;
s->mem_operator.Free = cmnMemFree;
s->mem_operator.Set = cmnMemSet;
s->mem_operator.Check = cmnMemCheck;
s->user_data.memflag = VO_IMF_USERMEMOPERATOR;
s->user_data.memData = &s->mem_operator;
s->codec_api.Init(&s->handle, VO_AUDIO_CodingAAC, &s->user_data);
params.sampleRate = avctx->sample_rate;
params.bitRate = avctx->bit_rate;
params.nChannels = avctx->channels;
params.adtsUsed = !(avctx->flags & CODEC_FLAG_GLOBAL_HEADER);
if (s->codec_api.SetParam(s->handle, VO_PID_AAC_ENCPARAM, ¶ms)
!= VO_ERR_NONE) {
av_log(avctx, AV_LOG_ERROR, "Unable to set encoding parameters\n");
ret = AVERROR(EINVAL);
goto error;
}
for (index = 0; index < 16; index++)
if (avctx->sample_rate == avpriv_mpeg4audio_sample_rates[index])
break;
if (index == 16) {
av_log(avctx, AV_LOG_ERROR, "Unsupported sample rate %d\n",
avctx->sample_rate);
ret = AVERROR(ENOSYS);
goto error;
}
if (avctx->flags & CODEC_FLAG_GLOBAL_HEADER) {
avctx->extradata_size = 2;
avctx->extradata = av_mallocz(avctx->extradata_size +
FF_INPUT_BUFFER_PADDING_SIZE);
if (!avctx->extradata) {
ret = AVERROR(ENOMEM);
goto error;
}
avctx->extradata[0] = 0x02 << 3 | index >> 1;
avctx->extradata[1] = (index & 0x01) << 7 | avctx->channels << 3;
}
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;
VO_CODECBUFFER input = { 0 }, output = { 0 };
VO_AUDIO_OUTPUTINFO output_info = { { 0 } };
VO_PBYTE samples;
int ret;
/* handle end-of-stream small frame and flushing */
if (!frame) {
if (s->last_frame <= 0)
return 0;
if (s->last_samples > 0 && s->last_samples < ENC_DELAY - FRAME_SIZE) {
s->last_samples = 0;
s->last_frame--;
}
s->last_frame--;
memset(s->end_buffer, 0, 2 * avctx->channels * avctx->frame_size);
samples = s->end_buffer;
} else {
if (frame->nb_samples < avctx->frame_size) {
s->last_samples = frame->nb_samples;
memcpy(s->end_buffer, frame->data[0], 2 * avctx->channels * frame->nb_samples);
samples = s->end_buffer;
} else {
samples = (VO_PBYTE)frame->data[0];
}
/* add current frame to the queue */
if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
return ret;
}
if ((ret = ff_alloc_packet2(avctx, avpkt, FFMAX(8192, 768 * avctx->channels))) < 0)
return ret;
input.Buffer = samples;
input.Length = 2 * avctx->channels * avctx->frame_size;
output.Buffer = avpkt->data;
output.Length = avpkt->size;
s->codec_api.SetInputData(s->handle, &input);
if (s->codec_api.GetOutputData(s->handle, &output, &output_info)
!= VO_ERR_NONE) {
av_log(avctx, AV_LOG_ERROR, "Unable to encode frame\n");
return AVERROR(EINVAL);
}
/* Get the next frame pts/duration */
ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
&avpkt->duration);
avpkt->size = output.Length;
*got_packet_ptr = 1;
return 0;
}
/* duplicated from avpriv_mpeg4audio_sample_rates to avoid shared build
* failures */
static const int mpeg4audio_sample_rates[16] = {
96000, 88200, 64000, 48000, 44100, 32000,
24000, 22050, 16000, 12000, 11025, 8000, 7350
};
AVCodec ff_libvo_aacenc_encoder = {
.name = "libvo_aacenc",
.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,
.supported_samplerates = mpeg4audio_sample_rates,
.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("Android VisualOn AAC (Advanced Audio Coding)"),
};
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