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#include <vo-aacenc/cmnMemory.h>
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#include "avcodec.h"
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#include "audio_frame_queue.h"
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#include "mpeg4audio.h"
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#define FRAME_SIZE 1024
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#define ENC_DELAY 1600
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typedef struct AACContext {
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VO_AUDIO_CODECAPI codec_api;
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VO_MEM_OPERATOR mem_operator;
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VO_CODEC_INIT_USERDATA user_data;
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static int aac_encode_close(AVCodecContext *avctx)
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AACContext *s = avctx->priv_data;
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s->codec_api.Uninit(s->handle);
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#if FF_API_OLD_ENCODE_AUDIO
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av_freep(&avctx->coded_frame);
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av_freep(&avctx->extradata);
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ff_af_queue_close(&s->afq);
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av_freep(&s->end_buffer);
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static av_cold int aac_encode_init(AVCodecContext *avctx)
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AACContext *s = avctx->priv_data;
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AACENC_PARAM params = { 0 };
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#if FF_API_OLD_ENCODE_AUDIO
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avctx->coded_frame = avcodec_alloc_frame();
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avctx->frame_size = 1024;
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if (!avctx->coded_frame)
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return AVERROR(ENOMEM);
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avctx->frame_size = FRAME_SIZE;
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avctx->delay = ENC_DELAY;
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ff_af_queue_init(avctx, &s->afq);
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s->end_buffer = av_mallocz(avctx->frame_size * avctx->channels * 2);
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ret = AVERROR(ENOMEM);
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voGetAACEncAPI(&s->codec_api);
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if (index == 16) {
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av_log(avctx, AV_LOG_ERROR, "Unsupported sample rate %d\n",
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avctx->sample_rate);
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return AVERROR(ENOSYS);
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ret = AVERROR(ENOSYS);
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if (avctx->flags & CODEC_FLAG_GLOBAL_HEADER) {
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avctx->extradata_size = 2;
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avctx->extradata = av_mallocz(avctx->extradata_size +
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FF_INPUT_BUFFER_PADDING_SIZE);
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if (!avctx->extradata)
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return AVERROR(ENOMEM);
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if (!avctx->extradata) {
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ret = AVERROR(ENOMEM);
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avctx->extradata[0] = 0x02 << 3 | index >> 1;
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avctx->extradata[1] = (index & 0x01) << 7 | avctx->channels << 3;
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static int aac_encode_close(AVCodecContext *avctx)
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AACContext *s = avctx->priv_data;
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s->codec_api.Uninit(s->handle);
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av_freep(&avctx->coded_frame);
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static int aac_encode_frame(AVCodecContext *avctx,
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unsigned char *frame/*out*/,
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int buf_size, void *data/*in*/)
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aac_encode_close(avctx);
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static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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const AVFrame *frame, int *got_packet_ptr)
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AACContext *s = avctx->priv_data;
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VO_CODECBUFFER input = { 0 }, output = { 0 };
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VO_AUDIO_OUTPUTINFO output_info = { { 0 } };
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input.Length = 2 * avctx->channels * avctx->frame_size;
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output.Buffer = frame;
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output.Length = buf_size;
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/* handle end-of-stream small frame and flushing */
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if (s->last_frame <= 0)
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if (s->last_samples > 0 && s->last_samples < ENC_DELAY - FRAME_SIZE) {
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memset(s->end_buffer, 0, 2 * avctx->channels * avctx->frame_size);
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samples = s->end_buffer;
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if (frame->nb_samples < avctx->frame_size) {
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s->last_samples = frame->nb_samples;
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memcpy(s->end_buffer, frame->data[0], 2 * avctx->channels * frame->nb_samples);
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samples = s->end_buffer;
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samples = (VO_PBYTE)frame->data[0];
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/* add current frame to the queue */
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if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
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if ((ret = ff_alloc_packet(avpkt, FFMAX(8192, 768 * avctx->channels)))) {
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av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
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input.Buffer = samples;
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input.Length = 2 * avctx->channels * avctx->frame_size;
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output.Buffer = avpkt->data;
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output.Length = avpkt->size;
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s->codec_api.SetInputData(s->handle, &input);
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if (s->codec_api.GetOutputData(s->handle, &output, &output_info)
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av_log(avctx, AV_LOG_ERROR, "Unable to encode frame\n");
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return AVERROR(EINVAL);
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return output.Length;
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/* Get the next frame pts/duration */
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ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
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avpkt->size = output.Length;
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AVCodec ff_libvo_aacenc_encoder = {
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.name = "libvo_aacenc",
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.type = AVMEDIA_TYPE_AUDIO,
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.id = AV_CODEC_ID_AAC,
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.priv_data_size = sizeof(AACContext),
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.init = aac_encode_init,
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.encode = aac_encode_frame,
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.encode2 = aac_encode_frame,
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.close = aac_encode_close,
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.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
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.long_name = NULL_IF_CONFIG_SMALL("Android VisualOn AAC"),
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.capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY,
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.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
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AV_SAMPLE_FMT_NONE },
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.long_name = NULL_IF_CONFIG_SMALL("Android VisualOn AAC (Advanced Audio Coding)"),