libfaac: use AVCodec.encode2()
Encoder output is delayed by several frames, so we keep a queue of input frame timing info to match up with corresponding output packets.
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				@ -581,7 +581,7 @@ OBJS-$(CONFIG_WTV_DEMUXER)             += mpeg4audio.o mpegaudiodata.o
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# external codec libraries
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OBJS-$(CONFIG_LIBDIRAC_DECODER)           += libdiracdec.o
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OBJS-$(CONFIG_LIBFAAC_ENCODER)            += libfaac.o
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OBJS-$(CONFIG_LIBFAAC_ENCODER)            += libfaac.o audio_frame_queue.o
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OBJS-$(CONFIG_LIBGSM_DECODER)             += libgsm.o
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OBJS-$(CONFIG_LIBGSM_ENCODER)             += libgsm.o
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OBJS-$(CONFIG_LIBGSM_MS_DECODER)          += libgsm.o
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@ -24,11 +24,19 @@
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 * Interface to libfaac for aac encoding.
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 */
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#include "avcodec.h"
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#include <faac.h>
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#include "avcodec.h"
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#include "audio_frame_queue.h"
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#include "internal.h"
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/* libfaac has an encoder delay of 1024 samples */
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#define FAAC_DELAY_SAMPLES 1024
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typedef struct FaacAudioContext {
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    faacEncHandle faac_handle;
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    AudioFrameQueue afq;
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} FaacAudioContext;
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@ -36,11 +44,15 @@ static av_cold int Faac_encode_close(AVCodecContext *avctx)
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{
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    FaacAudioContext *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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    ff_af_queue_close(&s->afq);
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    if (s->faac_handle)
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        faacEncClose(s->faac_handle);
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    return 0;
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}
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@ -109,11 +121,13 @@ static av_cold int Faac_encode_init(AVCodecContext *avctx)
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    avctx->frame_size = 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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    if (!avctx->coded_frame) {
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        ret = AVERROR(ENOMEM);
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        goto error;
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    }
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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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@ -144,26 +158,52 @@ static av_cold int Faac_encode_init(AVCodecContext *avctx)
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        goto error;
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    }
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    avctx->delay = FAAC_DELAY_SAMPLES;
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    ff_af_queue_init(avctx, &s->afq);
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    return 0;
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error:
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    Faac_encode_close(avctx);
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    return ret;
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}
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static int Faac_encode_frame(AVCodecContext *avctx,
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                             unsigned char *frame, int buf_size, void *data)
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static int Faac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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                             const AVFrame *frame, int *got_packet_ptr)
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{
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    FaacAudioContext *s = avctx->priv_data;
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    int bytes_written;
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    int num_samples = data ? avctx->frame_size : 0;
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    int bytes_written, ret;
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    int num_samples  = frame ? frame->nb_samples : 0;
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    void *samples    = frame ? frame->data[0]    : NULL;
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    bytes_written = faacEncEncode(s->faac_handle,
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                                  data,
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    if ((ret = ff_alloc_packet(avpkt, (7 + 768) * avctx->channels))) {
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        av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
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        return ret;
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    }
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    bytes_written = faacEncEncode(s->faac_handle, samples,
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                                  num_samples * avctx->channels,
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                                  frame,
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                                  buf_size);
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                                  avpkt->data, avpkt->size);
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    if (bytes_written < 0) {
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        av_log(avctx, AV_LOG_ERROR, "faacEncEncode() error\n");
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        return bytes_written;
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    }
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    return bytes_written;
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    /* add current frame to the queue */
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    if (frame) {
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        if ((ret = ff_af_queue_add(&s->afq, frame) < 0))
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            return ret;
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    }
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    if (!bytes_written)
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        return 0;
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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->duration);
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    avpkt->size = bytes_written;
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    *got_packet_ptr = 1;
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    return 0;
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}
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static const AVProfile profiles[] = {
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@ -180,7 +220,7 @@ AVCodec ff_libfaac_encoder = {
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    .id             = CODEC_ID_AAC,
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    .priv_data_size = sizeof(FaacAudioContext),
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    .init           = Faac_encode_init,
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    .encode         = Faac_encode_frame,
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    .encode2        = Faac_encode_frame,
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    .close          = Faac_encode_close,
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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,AV_SAMPLE_FMT_NONE},
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