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https://github.com/xenia-project/FFmpeg.git
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libaacplus: switch to encode2()
Signed-off-by: Paul B Mahol <onemda@gmail.com> Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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@ -24,18 +24,21 @@
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* Interface to libaacplus for aac+ (sbr+ps) encoding.
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*/
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#include "avcodec.h"
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#include <aacplus.h>
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#include "avcodec.h"
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#include "internal.h"
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typedef struct aacPlusAudioContext {
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aacplusEncHandle aacplus_handle;
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unsigned long max_output_bytes;
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unsigned long samples_input;
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} aacPlusAudioContext;
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static av_cold int aacPlus_encode_init(AVCodecContext *avctx)
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{
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aacPlusAudioContext *s = avctx->priv_data;
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aacplusEncConfiguration *aacplus_cfg;
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unsigned long samples_input, max_bytes_output;
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/* number of channels */
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if (avctx->channels < 1 || avctx->channels > 2) {
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@ -43,9 +46,8 @@ static av_cold int aacPlus_encode_init(AVCodecContext *avctx)
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return -1;
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}
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s->aacplus_handle = aacplusEncOpen(avctx->sample_rate,
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avctx->channels,
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&samples_input, &max_bytes_output);
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s->aacplus_handle = aacplusEncOpen(avctx->sample_rate, avctx->channels,
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&s->samples_input, &s->max_output_bytes);
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if(!s->aacplus_handle) {
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av_log(avctx, AV_LOG_ERROR, "can't open encoder\n");
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return -1;
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@ -70,10 +72,12 @@ static av_cold int aacPlus_encode_init(AVCodecContext *avctx)
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return -1;
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}
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avctx->frame_size = samples_input / avctx->channels;
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avctx->frame_size = s->samples_input / avctx->channels;
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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->coded_frame->key_frame= 1;
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#endif
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/* Set decoder specific info */
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avctx->extradata_size = 0;
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@ -95,26 +99,30 @@ static av_cold int aacPlus_encode_init(AVCodecContext *avctx)
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return 0;
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}
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static int aacPlus_encode_frame(AVCodecContext *avctx,
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unsigned char *frame, int buf_size, void *data)
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static int aacPlus_encode_frame(AVCodecContext *avctx, AVPacket *pkt,
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const AVFrame *frame, int *got_packet)
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{
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aacPlusAudioContext *s = avctx->priv_data;
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int bytes_written;
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int32_t *input_buffer = (int32_t *)frame->data[0];
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int ret;
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bytes_written = aacplusEncEncode(s->aacplus_handle,
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data,
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avctx->frame_size * avctx->channels,
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frame,
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buf_size);
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if ((ret = ff_alloc_packet2(avctx, pkt, s->max_output_bytes)))
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return ret;
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return bytes_written;
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pkt->size = aacplusEncEncode(s->aacplus_handle, input_buffer,
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s->samples_input, pkt->data, pkt->size);
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*got_packet = 1;
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pkt->pts = frame->pts;
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return 0;
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}
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static av_cold int aacPlus_encode_close(AVCodecContext *avctx)
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{
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aacPlusAudioContext *s = avctx->priv_data;
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#if FF_API_OLD_ENCODE_AUDIO
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av_freep(&avctx->coded_frame);
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#endif
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av_freep(&avctx->extradata);
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aacplusEncClose(s->aacplus_handle);
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@ -127,7 +135,7 @@ AVCodec ff_libaacplus_encoder = {
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.id = CODEC_ID_AAC,
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.priv_data_size = sizeof(aacPlusAudioContext),
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.init = aacPlus_encode_init,
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.encode = aacPlus_encode_frame,
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.encode2 = aacPlus_encode_frame,
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.close = aacPlus_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("libaacplus AAC+ (Advanced Audio Codec with SBR+PS)"),
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