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author | Justin Ruggles <justin.ruggles@gmail.com> | 2012-02-07 14:28:50 -0500 |
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committer | Justin Ruggles <justin.ruggles@gmail.com> | 2012-03-20 18:55:36 -0400 |
commit | d1afb2f94e7070cca7c62dcbc258efc85f267249 (patch) | |
tree | 889ebe166c8b8d05c7e81746caedbb1d971b11fd /libavcodec | |
parent | 59041fd053d11d26772ac45e21b6ac3eea39c738 (diff) | |
download | ffmpeg-streaming-d1afb2f94e7070cca7c62dcbc258efc85f267249.zip ffmpeg-streaming-d1afb2f94e7070cca7c62dcbc258efc85f267249.tar.gz |
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.
Diffstat (limited to 'libavcodec')
-rw-r--r-- | libavcodec/Makefile | 2 | ||||
-rw-r--r-- | libavcodec/libfaac.c | 62 |
2 files changed, 52 insertions, 12 deletions
diff --git a/libavcodec/Makefile b/libavcodec/Makefile index a9b9a7e..65385aa 100644 --- a/libavcodec/Makefile +++ b/libavcodec/Makefile @@ -581,7 +581,7 @@ OBJS-$(CONFIG_WTV_DEMUXER) += mpeg4audio.o mpegaudiodata.o # external codec libraries OBJS-$(CONFIG_LIBDIRAC_DECODER) += libdiracdec.o -OBJS-$(CONFIG_LIBFAAC_ENCODER) += libfaac.o +OBJS-$(CONFIG_LIBFAAC_ENCODER) += libfaac.o audio_frame_queue.o OBJS-$(CONFIG_LIBGSM_DECODER) += libgsm.o OBJS-$(CONFIG_LIBGSM_ENCODER) += libgsm.o OBJS-$(CONFIG_LIBGSM_MS_DECODER) += libgsm.o diff --git a/libavcodec/libfaac.c b/libavcodec/libfaac.c index 514d2a5..6a74465 100644 --- a/libavcodec/libfaac.c +++ b/libavcodec/libfaac.c @@ -24,11 +24,19 @@ * Interface to libfaac for aac encoding. */ -#include "avcodec.h" #include <faac.h> +#include "avcodec.h" +#include "audio_frame_queue.h" +#include "internal.h" + + +/* libfaac has an encoder delay of 1024 samples */ +#define FAAC_DELAY_SAMPLES 1024 + typedef struct FaacAudioContext { faacEncHandle faac_handle; + AudioFrameQueue afq; } FaacAudioContext; @@ -36,11 +44,15 @@ static av_cold int Faac_encode_close(AVCodecContext *avctx) { FaacAudioContext *s = avctx->priv_data; +#if FF_API_OLD_ENCODE_AUDIO av_freep(&avctx->coded_frame); +#endif av_freep(&avctx->extradata); + ff_af_queue_close(&s->afq); if (s->faac_handle) faacEncClose(s->faac_handle); + return 0; } @@ -109,11 +121,13 @@ static av_cold int Faac_encode_init(AVCodecContext *avctx) avctx->frame_size = samples_input / avctx->channels; +#if FF_API_OLD_ENCODE_AUDIO avctx->coded_frame= avcodec_alloc_frame(); if (!avctx->coded_frame) { ret = AVERROR(ENOMEM); goto error; } +#endif /* Set decoder specific info */ avctx->extradata_size = 0; @@ -144,26 +158,52 @@ static av_cold int Faac_encode_init(AVCodecContext *avctx) goto error; } + avctx->delay = FAAC_DELAY_SAMPLES; + ff_af_queue_init(avctx, &s->afq); + return 0; error: Faac_encode_close(avctx); return ret; } -static int Faac_encode_frame(AVCodecContext *avctx, - unsigned char *frame, int buf_size, void *data) +static int Faac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, + const AVFrame *frame, int *got_packet_ptr) { FaacAudioContext *s = avctx->priv_data; - int bytes_written; - int num_samples = data ? avctx->frame_size : 0; + int bytes_written, ret; + int num_samples = frame ? frame->nb_samples : 0; + void *samples = frame ? frame->data[0] : NULL; - bytes_written = faacEncEncode(s->faac_handle, - data, + if ((ret = ff_alloc_packet(avpkt, (7 + 768) * avctx->channels))) { + av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n"); + return ret; + } + + bytes_written = faacEncEncode(s->faac_handle, samples, num_samples * avctx->channels, - frame, - buf_size); + avpkt->data, avpkt->size); + if (bytes_written < 0) { + av_log(avctx, AV_LOG_ERROR, "faacEncEncode() error\n"); + return bytes_written; + } + + /* add current frame to the queue */ + if (frame) { + if ((ret = ff_af_queue_add(&s->afq, frame) < 0)) + return ret; + } - return bytes_written; + if (!bytes_written) + return 0; + + /* Get the next frame pts/duration */ + ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts, + &avpkt->duration); + + avpkt->size = bytes_written; + *got_packet_ptr = 1; + return 0; } static const AVProfile profiles[] = { @@ -180,7 +220,7 @@ AVCodec ff_libfaac_encoder = { .id = CODEC_ID_AAC, .priv_data_size = sizeof(FaacAudioContext), .init = Faac_encode_init, - .encode = Faac_encode_frame, + .encode2 = Faac_encode_frame, .close = Faac_encode_close, .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY, .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE}, |