/*
  * 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, &params)
         != 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)"),

 };