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https://github.com/xenia-project/FFmpeg.git
synced 2024-11-28 14:01:27 +00:00
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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