/*
  * copyright (c) 2002 Mark Hills <mark@pogo.org.uk>
  *

  * 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

  * Vorbis encoding support via libvorbisenc.

  * @author Mark Hills <mark@pogo.org.uk>

  */
 
 #include <vorbis/vorbisenc.h>
 

 #include "libavutil/fifo.h"

 #include "libavutil/opt.h"

 #include "avcodec.h"

 #include "audio_frame_queue.h"

 #include "bytestream.h"

 #include "internal.h"

 #include "vorbis.h"

 #include "vorbis_parser.h"

 #undef NDEBUG
 #include <assert.h>
 

 /* Number of samples the user should send in each call.
  * This value is used because it is the LCD of all possible frame sizes, so
  * an output packet will always start at the same point as one of the input
  * packets.
  */

 #define OGGVORBIS_FRAME_SIZE 64

 #define BUFFER_SIZE (1024 * 64)

 
 typedef struct OggVorbisContext {

     AVClass *av_class;                  /**< class for AVOptions            */

     AVFrame frame;

     vorbis_info vi;                     /**< vorbis_info used during init   */
     vorbis_dsp_state vd;                /**< DSP state used for analysis    */
     vorbis_block vb;                    /**< vorbis_block used for analysis */

     AVFifoBuffer *pkt_fifo;             /**< output packet buffer           */

     int eof;                            /**< end-of-file flag               */

     int dsp_initialized;                /**< vd has been initialized        */

     vorbis_comment vc;                  /**< VorbisComment info             */
     ogg_packet op;                      /**< ogg packet                     */
     double iblock;                      /**< impulse block bias option      */

     VorbisParseContext vp;              /**< parse context to get durations */
     AudioFrameQueue afq;                /**< frame queue for timestamps     */

 } OggVorbisContext;

 static const AVOption options[] = {
     { "iblock", "Sets the impulse block bias", offsetof(OggVorbisContext, iblock), AV_OPT_TYPE_DOUBLE, { .dbl = 0 }, -15, 0, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM },
     { NULL }

 };

 
 static const AVCodecDefault defaults[] = {
     { "b",  "0" },
     { NULL },
 };
 

 static const AVClass class = { "libvorbis", av_default_item_name, options, LIBAVUTIL_VERSION_INT };

 static int vorbis_error_to_averror(int ov_err)

 {

     switch (ov_err) {
     case OV_EFAULT: return AVERROR_BUG;
     case OV_EINVAL: return AVERROR(EINVAL);
     case OV_EIMPL:  return AVERROR(EINVAL);
     default:        return AVERROR_UNKNOWN;

     }
 }
 

 static av_cold int oggvorbis_init_encoder(vorbis_info *vi,
                                           AVCodecContext *avctx)

 {

     OggVorbisContext *s = avctx->priv_data;

     double cfreq;

     int ret;

     if (avctx->flags & CODEC_FLAG_QSCALE || !avctx->bit_rate) {

         /* variable bitrate
          * NOTE: we use the oggenc range of -1 to 10 for global_quality for

          *       user convenience, but libvorbis uses -0.1 to 1.0.

          */
         float q = avctx->global_quality / (float)FF_QP2LAMBDA;

         /* default to 3 if the user did not set quality or bitrate */
         if (!(avctx->flags & CODEC_FLAG_QSCALE))
             q = 3.0;

         if ((ret = vorbis_encode_setup_vbr(vi, avctx->channels,
                                            avctx->sample_rate,

                                            q / 10.0)))
             goto error;

     } else {

         int minrate = avctx->rc_min_rate > 0 ? avctx->rc_min_rate : -1;

         int maxrate = avctx->rc_max_rate > 0 ? avctx->rc_max_rate : -1;

         /* average bitrate */
         if ((ret = vorbis_encode_setup_managed(vi, avctx->channels,

                                                avctx->sample_rate, maxrate,
                                                avctx->bit_rate, minrate)))

             goto error;

         /* variable bitrate by estimate, disable slow rate management */

         if (minrate == -1 && maxrate == -1)

             if ((ret = vorbis_encode_ctl(vi, OV_ECTL_RATEMANAGE2_SET, NULL)))

                 goto error; /* should not happen */

     }

     /* cutoff frequency */

     if (avctx->cutoff > 0) {
         cfreq = avctx->cutoff / 1000.0;

         if ((ret = vorbis_encode_ctl(vi, OV_ECTL_LOWPASS_SET, &cfreq)))

             goto error; /* should not happen */

     }

     /* impulse block bias */
     if (s->iblock) {
         if ((ret = vorbis_encode_ctl(vi, OV_ECTL_IBLOCK_SET, &s->iblock)))

             goto error;

     }
 

     if (avctx->channels == 3 &&
             avctx->channel_layout != (AV_CH_LAYOUT_STEREO|AV_CH_FRONT_CENTER) ||
         avctx->channels == 4 &&
             avctx->channel_layout != AV_CH_LAYOUT_2_2 &&
             avctx->channel_layout != AV_CH_LAYOUT_QUAD ||
         avctx->channels == 5 &&
             avctx->channel_layout != AV_CH_LAYOUT_5POINT0 &&
             avctx->channel_layout != AV_CH_LAYOUT_5POINT0_BACK ||
         avctx->channels == 6 &&
             avctx->channel_layout != AV_CH_LAYOUT_5POINT1 &&
             avctx->channel_layout != AV_CH_LAYOUT_5POINT1_BACK ||
         avctx->channels == 7 &&
             avctx->channel_layout != (AV_CH_LAYOUT_5POINT1|AV_CH_BACK_CENTER) ||
         avctx->channels == 8 &&
             avctx->channel_layout != AV_CH_LAYOUT_7POINT1) {
         if (avctx->channel_layout) {

             char name[32];

             av_get_channel_layout_string(name, sizeof(name), avctx->channels,
                                          avctx->channel_layout);
             av_log(avctx, AV_LOG_ERROR, "%s not supported by Vorbis: "

                                              "output stream will have incorrect "
                                              "channel layout.\n", name);
         } else {

             av_log(avctx, AV_LOG_WARNING, "No channel layout specified. The encoder "

                                                "will use Vorbis channel layout for "

                                                "%d channels.\n", avctx->channels);

         }
     }
 

     if ((ret = vorbis_encode_setup_init(vi)))
         goto error;
 
     return 0;
 error:
     return vorbis_error_to_averror(ret);

 }
 

 /* How many bytes are needed for a buffer of length 'l' */

 static int xiph_len(int l)
 {

     return 1 + l / 255 + l;

 }

 static av_cold int oggvorbis_encode_close(AVCodecContext *avctx)

 {

     OggVorbisContext *s = avctx->priv_data;

     /* notify vorbisenc this is EOF */

     if (s->dsp_initialized)
         vorbis_analysis_wrote(&s->vd, 0);

     vorbis_block_clear(&s->vb);
     vorbis_dsp_clear(&s->vd);
     vorbis_info_clear(&s->vi);

     av_fifo_free(s->pkt_fifo);

     ff_af_queue_close(&s->afq);
 #if FF_API_OLD_ENCODE_AUDIO

     av_freep(&avctx->coded_frame);

 #endif

     av_freep(&avctx->extradata);

 
     return 0;
 }
 

 static av_cold int oggvorbis_encode_init(AVCodecContext *avctx)

 {

     OggVorbisContext *s = avctx->priv_data;

     ogg_packet header, header_comm, header_code;
     uint8_t *p;

     unsigned int offset;

     int ret;

     vorbis_info_init(&s->vi);
     if ((ret = oggvorbis_init_encoder(&s->vi, avctx))) {

         av_log(avctx, AV_LOG_ERROR, "encoder setup failed\n");

         goto error;
     }

     if ((ret = vorbis_analysis_init(&s->vd, &s->vi))) {

         av_log(avctx, AV_LOG_ERROR, "analysis init failed\n");

         ret = vorbis_error_to_averror(ret);
         goto error;
     }

     s->dsp_initialized = 1;

     if ((ret = vorbis_block_init(&s->vd, &s->vb))) {

         av_log(avctx, AV_LOG_ERROR, "dsp init failed\n");

         ret = vorbis_error_to_averror(ret);
         goto error;

     }
 

     vorbis_comment_init(&s->vc);

     if (!(avctx->flags & CODEC_FLAG_BITEXACT))
         vorbis_comment_add_tag(&s->vc, "encoder", LIBAVCODEC_IDENT);

     if ((ret = vorbis_analysis_headerout(&s->vd, &s->vc, &header, &header_comm,
                                          &header_code))) {

         ret = vorbis_error_to_averror(ret);
         goto error;
     }

     avctx->extradata_size = 1 + xiph_len(header.bytes)      +
                                 xiph_len(header_comm.bytes) +
                                 header_code.bytes;
     p = avctx->extradata = av_malloc(avctx->extradata_size +
                                      FF_INPUT_BUFFER_PADDING_SIZE);

     if (!p) {
         ret = AVERROR(ENOMEM);
         goto error;
     }

     p[0]    = 2;
     offset  = 1;

     offset += av_xiphlacing(&p[offset], header.bytes);
     offset += av_xiphlacing(&p[offset], header_comm.bytes);
     memcpy(&p[offset], header.packet, header.bytes);
     offset += header.bytes;
     memcpy(&p[offset], header_comm.packet, header_comm.bytes);
     offset += header_comm.bytes;
     memcpy(&p[offset], header_code.packet, header_code.bytes);
     offset += header_code.bytes;

     assert(offset == avctx->extradata_size);

     if ((ret = avpriv_vorbis_parse_extradata(avctx, &s->vp)) < 0) {
         av_log(avctx, AV_LOG_ERROR, "invalid extradata\n");
         return ret;
     }
 

     vorbis_comment_clear(&s->vc);

     avctx->frame_size = OGGVORBIS_FRAME_SIZE;

     ff_af_queue_init(avctx, &s->afq);

     s->pkt_fifo = av_fifo_alloc(BUFFER_SIZE);
     if (!s->pkt_fifo) {
         ret = AVERROR(ENOMEM);
         goto error;
     }

 #if FF_API_OLD_ENCODE_AUDIO

     avctx->coded_frame = avcodec_alloc_frame();
     if (!avctx->coded_frame) {

         ret = AVERROR(ENOMEM);
         goto error;
     }

 #endif

     return 0;

 error:

     oggvorbis_encode_close(avctx);

     return ret;

 }
 

 static int oggvorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
                                   const AVFrame *frame, int *got_packet_ptr)

 {

     OggVorbisContext *s = avctx->priv_data;

     ogg_packet op;

     int ret, duration;

     /* send samples to libvorbis */

     if (frame) {
         const float *audio = (const float *)frame->data[0];
         const int samples = frame->nb_samples;

         float **buffer;

         int c, channels = s->vi.channels;

         buffer = vorbis_analysis_buffer(&s->vd, samples);

         for (c = 0; c < channels; c++) {

             int i;

             int co = (channels > 8) ? c :

                      ff_vorbis_encoding_channel_layout_offsets[channels - 1][c];

             for (i = 0; i < samples; i++)

                 buffer[c][i] = audio[i * channels + co];

         }

         if ((ret = vorbis_analysis_wrote(&s->vd, samples)) < 0) {
             av_log(avctx, AV_LOG_ERROR, "error in vorbis_analysis_wrote()\n");

             return vorbis_error_to_averror(ret);

         }

         if ((ret = ff_af_queue_add(&s->afq, frame) < 0))
             return ret;

     } else {

         if (!s->eof)

             if ((ret = vorbis_analysis_wrote(&s->vd, 0)) < 0) {
                 av_log(avctx, AV_LOG_ERROR, "error in vorbis_analysis_wrote()\n");

                 return vorbis_error_to_averror(ret);

             }

         s->eof = 1;

     }

     /* retrieve available packets from libvorbis */

     while ((ret = vorbis_analysis_blockout(&s->vd, &s->vb)) == 1) {
         if ((ret = vorbis_analysis(&s->vb, NULL)) < 0)
             break;
         if ((ret = vorbis_bitrate_addblock(&s->vb)) < 0)
             break;

         /* add any available packets to the output packet buffer */

         while ((ret = vorbis_bitrate_flushpacket(&s->vd, &op)) == 1) {

             if (av_fifo_space(s->pkt_fifo) < sizeof(ogg_packet) + op.bytes) {

                 av_log(avctx, AV_LOG_ERROR, "packet buffer is too small\n");

                 return AVERROR_BUG;

             }

             av_fifo_generic_write(s->pkt_fifo, &op, sizeof(ogg_packet), NULL);
             av_fifo_generic_write(s->pkt_fifo, op.packet, op.bytes, NULL);

         }

         if (ret < 0) {
             av_log(avctx, AV_LOG_ERROR, "error getting available packets\n");

             break;

         }

     }

     if (ret < 0) {
         av_log(avctx, AV_LOG_ERROR, "error getting available packets\n");

         return vorbis_error_to_averror(ret);

     }

     /* check for available packets */
     if (av_fifo_size(s->pkt_fifo) < sizeof(ogg_packet))
         return 0;
 
     av_fifo_generic_read(s->pkt_fifo, &op, sizeof(ogg_packet), NULL);
 

     if ((ret = ff_alloc_packet2(avctx, avpkt, op.bytes)))

         return ret;
     av_fifo_generic_read(s->pkt_fifo, avpkt->data, op.bytes, NULL);
 
     avpkt->pts = ff_samples_to_time_base(avctx, op.granulepos);
 
     duration = avpriv_vorbis_parse_frame(&s->vp, avpkt->data, avpkt->size);
     if (duration > 0) {
         /* we do not know encoder delay until we get the first packet from
          * libvorbis, so we have to update the AudioFrameQueue counts */
         if (!avctx->delay) {
             avctx->delay              = duration;
             s->afq.remaining_delay   += duration;
             s->afq.remaining_samples += duration;

         }

         ff_af_queue_remove(&s->afq, duration, &avpkt->pts, &avpkt->duration);

     }
 

     *got_packet_ptr = 1;
     return 0;

 }
 

 AVCodec ff_libvorbis_encoder = {

     .name           = "libvorbis",
     .type           = AVMEDIA_TYPE_AUDIO,
     .id             = CODEC_ID_VORBIS,
     .priv_data_size = sizeof(OggVorbisContext),
     .init           = oggvorbis_encode_init,

     .encode2        = oggvorbis_encode_frame,

     .close          = oggvorbis_encode_close,
     .capabilities   = CODEC_CAP_DELAY,

     .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLT,

                                                       AV_SAMPLE_FMT_NONE },

     .long_name      = NULL_IF_CONFIG_SMALL("libvorbis Vorbis"),
     .priv_class     = &class,

     .defaults       = defaults,

 };

 
 static int oggvorbis_decode_init(AVCodecContext *avccontext) {
     OggVorbisContext *context = avccontext->priv_data ;
     uint8_t *p= avccontext->extradata;
     int i, hsizes[3];
     unsigned char *headers[3], *extradata = avccontext->extradata;
 
     vorbis_info_init(&context->vi) ;
     vorbis_comment_init(&context->vc) ;
 
     if(! avccontext->extradata_size || ! p) {
         av_log(avccontext, AV_LOG_ERROR, "vorbis extradata absent\n");
         return -1;
     }
 
     if(p[0] == 0 && p[1] == 30) {
         for(i = 0; i < 3; i++){
             hsizes[i] = bytestream_get_be16(&p);
             headers[i] = p;
             p += hsizes[i];
         }
     } else if(*p == 2) {
         unsigned int offset = 1;
         p++;
         for(i=0; i<2; i++) {
             hsizes[i] = 0;
             while((*p == 0xFF) && (offset < avccontext->extradata_size)) {
                 hsizes[i] += 0xFF;
                 offset++;
                 p++;
             }
             if(offset >= avccontext->extradata_size - 1) {
                 av_log(avccontext, AV_LOG_ERROR,
                        "vorbis header sizes damaged\n");
                 return -1;
             }
             hsizes[i] += *p;
             offset++;
             p++;
         }
         hsizes[2] = avccontext->extradata_size - hsizes[0]-hsizes[1]-offset;
 #if 0
         av_log(avccontext, AV_LOG_DEBUG,
                "vorbis header sizes: %d, %d, %d, / extradata_len is %d \n",
                hsizes[0], hsizes[1], hsizes[2], avccontext->extradata_size);
 #endif
         headers[0] = extradata + offset;
         headers[1] = extradata + offset + hsizes[0];
         headers[2] = extradata + offset + hsizes[0] + hsizes[1];
     } else {
         av_log(avccontext, AV_LOG_ERROR,
                "vorbis initial header len is wrong: %d\n", *p);
         return -1;
     }
 
     for(i=0; i<3; i++){
         context->op.b_o_s= i==0;
         context->op.bytes = hsizes[i];
         context->op.packet = headers[i];
         if(vorbis_synthesis_headerin(&context->vi, &context->vc, &context->op)<0){
             av_log(avccontext, AV_LOG_ERROR, "%d. vorbis header damaged\n", i+1);
             return -1;
         }
     }
 
     avccontext->channels = context->vi.channels;
     avccontext->sample_rate = context->vi.rate;
     avccontext->time_base= (AVRational){1, avccontext->sample_rate};
 
     vorbis_synthesis_init(&context->vd, &context->vi);
     vorbis_block_init(&context->vd, &context->vb);
 
     return 0 ;
 }
 
 
 static inline int conv(int samples, float **pcm, char *buf, int channels) {
     int i, j;
     ogg_int16_t *ptr, *data = (ogg_int16_t*)buf ;
     float *mono ;
 
     for(i = 0 ; i < channels ; i++){
         ptr = &data[i];
         mono = pcm[i] ;
 
         for(j = 0 ; j < samples ; j++) {
             *ptr = av_clip_int16(mono[j] * 32767.f);
             ptr += channels;
         }
     }
 
     return 0 ;
 }
 
 static int oggvorbis_decode_frame(AVCodecContext *avccontext, void *data,
                         int *got_frame_ptr, AVPacket *avpkt)
 {
     OggVorbisContext *context = avccontext->priv_data ;
     float **pcm ;
     ogg_packet *op= &context->op;
     int samples, total_samples, total_bytes;
     int ret;
     int16_t *output;
 
     if(!avpkt->size){
     //FIXME flush
         return 0;
     }
 
     context->frame.nb_samples = 8192*4;
     if ((ret = avccontext->get_buffer(avccontext, &context->frame)) < 0) {
         av_log(avccontext, AV_LOG_ERROR, "get_buffer() failed\n");
         return ret;
     }
     output = (int16_t *)context->frame.data[0];
 
 
     op->packet = avpkt->data;
     op->bytes  = avpkt->size;
 
 //    av_log(avccontext, AV_LOG_DEBUG, "%d %d %d %"PRId64" %"PRId64" %d %d\n", op->bytes, op->b_o_s, op->e_o_s, op->granulepos, op->packetno, buf_size, context->vi.rate);
 
 /*    for(i=0; i<op->bytes; i++)
       av_log(avccontext, AV_LOG_DEBUG, "%02X ", op->packet[i]);
     av_log(avccontext, AV_LOG_DEBUG, "\n");*/
 
     if(vorbis_synthesis(&context->vb, op) == 0)
         vorbis_synthesis_blockin(&context->vd, &context->vb) ;
 
     total_samples = 0 ;
     total_bytes = 0 ;
 
     while((samples = vorbis_synthesis_pcmout(&context->vd, &pcm)) > 0) {
         conv(samples, pcm, (char*)output + total_bytes, context->vi.channels) ;
         total_bytes += samples * 2 * context->vi.channels ;
         total_samples += samples ;
         vorbis_synthesis_read(&context->vd, samples) ;
     }
 
     context->frame.nb_samples = total_samples;
     *got_frame_ptr   = 1;
     *(AVFrame *)data = context->frame;
     return avpkt->size;
 }
 
 
 static int oggvorbis_decode_close(AVCodecContext *avccontext) {
     OggVorbisContext *context = avccontext->priv_data ;
 
     vorbis_info_clear(&context->vi) ;
     vorbis_comment_clear(&context->vc) ;
 
     return 0 ;
 }
 
 
 AVCodec ff_libvorbis_decoder = {
   .name           = "libvorbis",
   .type           = AVMEDIA_TYPE_AUDIO,
   .id             = CODEC_ID_VORBIS,
   .priv_data_size = sizeof(OggVorbisContext),
   .init           = oggvorbis_decode_init,
   .decode         = oggvorbis_decode_frame,
   .close          = oggvorbis_decode_close,
   .capabilities   = CODEC_CAP_DELAY,
 } ;