From e5aab2d7a47942d61f8c54141da5c6ec33f7ce48 Mon Sep 17 00:00:00 2001 From: Justin Ruggles Date: Wed, 29 Feb 2012 01:02:54 -0500 Subject: [PATCH] libvorbis: use AVCodec.encode2() --- libavcodec/Makefile | 3 +- libavcodec/libvorbis.c | 69 ++++++++++++++++++++++++++++++------------ 2 files changed, 51 insertions(+), 21 deletions(-) diff --git a/libavcodec/Makefile b/libavcodec/Makefile index cb97bb6a8e..23270544e9 100644 --- a/libavcodec/Makefile +++ b/libavcodec/Makefile @@ -605,7 +605,8 @@ OBJS-$(CONFIG_LIBSPEEX_ENCODER) += libspeexenc.o audio_frame_queue.o OBJS-$(CONFIG_LIBTHEORA_ENCODER) += libtheoraenc.o OBJS-$(CONFIG_LIBVO_AACENC_ENCODER) += libvo-aacenc.o mpeg4audio.o OBJS-$(CONFIG_LIBVO_AMRWBENC_ENCODER) += libvo-amrwbenc.o -OBJS-$(CONFIG_LIBVORBIS_ENCODER) += libvorbis.o vorbis_data.o +OBJS-$(CONFIG_LIBVORBIS_ENCODER) += libvorbis.o audio_frame_queue.o \ + vorbis_data.o vorbis_parser.o OBJS-$(CONFIG_LIBVPX_DECODER) += libvpxdec.o OBJS-$(CONFIG_LIBVPX_ENCODER) += libvpxenc.o OBJS-$(CONFIG_LIBX264_ENCODER) += libx264.o diff --git a/libavcodec/libvorbis.c b/libavcodec/libvorbis.c index 991c64f81d..1d7b7ef49b 100644 --- a/libavcodec/libvorbis.c +++ b/libavcodec/libvorbis.c @@ -29,9 +29,11 @@ #include "libavutil/fifo.h" #include "libavutil/opt.h" #include "avcodec.h" +#include "audio_frame_queue.h" #include "bytestream.h" #include "internal.h" #include "vorbis.h" +#include "vorbis_parser.h" #undef NDEBUG #include @@ -56,6 +58,8 @@ typedef struct OggVorbisContext { vorbis_comment vc; /**< VorbisComment info */ ogg_packet op; /**< ogg packet */ double iblock; /**< impulse block bias option */ + VorbisParseContext vp; /**< parse context to get durations */ + AudioFrameQueue afq; /**< frame queue for timestamps */ } OggVorbisContext; static const AVOption options[] = { @@ -157,7 +161,10 @@ static av_cold int oggvorbis_encode_close(AVCodecContext *avctx) vorbis_info_clear(&s->vi); av_fifo_free(s->pkt_fifo); + ff_af_queue_close(&s->afq); +#if FF_API_OLD_ENCODE_AUDIO av_freep(&avctx->coded_frame); +#endif av_freep(&avctx->extradata); return 0; @@ -218,9 +225,15 @@ static av_cold int oggvorbis_encode_init(AVCodecContext *avctx) offset += header_code.bytes; assert(offset == avctx->extradata_size); + if ((ret = avpriv_vorbis_parse_extradata(avctx, &s->vp)) < 0) { + av_log(avctx, AV_LOG_ERROR, "invalid extradata\n"); + return ret; + } + vorbis_comment_clear(&s->vc); avctx->frame_size = OGGVORBIS_FRAME_SIZE; + ff_af_queue_init(avctx, &s->afq); s->pkt_fifo = av_fifo_alloc(BUFFER_SIZE); if (!s->pkt_fifo) { @@ -228,11 +241,13 @@ static av_cold int oggvorbis_encode_init(AVCodecContext *avctx) goto error; } +#if FF_API_OLD_ENCODE_AUDIO avctx->coded_frame = avcodec_alloc_frame(); if (!avctx->coded_frame) { ret = AVERROR(ENOMEM); goto error; } +#endif return 0; error: @@ -240,17 +255,17 @@ error: return ret; } -static int oggvorbis_encode_frame(AVCodecContext *avctx, unsigned char *packets, - int buf_size, void *data) +static int oggvorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, + const AVFrame *frame, int *got_packet_ptr) { OggVorbisContext *s = avctx->priv_data; ogg_packet op; - float *audio = data; - int pkt_size, ret; + int ret, duration; /* send samples to libvorbis */ - if (data) { - const int samples = avctx->frame_size; + if (frame) { + const float *audio = (const float *)frame->data[0]; + const int samples = frame->nb_samples; float **buffer; int c, channels = s->vi.channels; @@ -266,6 +281,8 @@ static int oggvorbis_encode_frame(AVCodecContext *avctx, unsigned char *packets, av_log(avctx, AV_LOG_ERROR, "error in vorbis_analysis_wrote()\n"); return vorbis_error_to_averror(ret); } + if ((ret = ff_af_queue_add(&s->afq, frame) < 0)) + return ret; } else { if (!s->eof) if ((ret = vorbis_analysis_wrote(&s->vd, 0)) < 0) { @@ -301,22 +318,34 @@ static int oggvorbis_encode_frame(AVCodecContext *avctx, unsigned char *packets, return vorbis_error_to_averror(ret); } - /* output then next packet from the output buffer, if available */ - pkt_size = 0; - if (av_fifo_size(s->pkt_fifo) >= sizeof(ogg_packet)) { - av_fifo_generic_read(s->pkt_fifo, &op, sizeof(ogg_packet), NULL); - pkt_size = op.bytes; - // FIXME: we should use the user-supplied pts and duration - avctx->coded_frame->pts = ff_samples_to_time_base(avctx, - op.granulepos); - if (pkt_size > buf_size) { - av_log(avctx, AV_LOG_ERROR, "output buffer is too small"); - return AVERROR(EINVAL); + /* check for available packets */ + if (av_fifo_size(s->pkt_fifo) < sizeof(ogg_packet)) + return 0; + + av_fifo_generic_read(s->pkt_fifo, &op, sizeof(ogg_packet), NULL); + + if ((ret = ff_alloc_packet(avpkt, op.bytes))) { + av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n"); + return ret; + } + av_fifo_generic_read(s->pkt_fifo, avpkt->data, op.bytes, NULL); + + avpkt->pts = ff_samples_to_time_base(avctx, op.granulepos); + + duration = avpriv_vorbis_parse_frame(&s->vp, avpkt->data, avpkt->size); + if (duration > 0) { + /* we do not know encoder delay until we get the first packet from + * libvorbis, so we have to update the AudioFrameQueue counts */ + if (!avctx->delay) { + avctx->delay = duration; + s->afq.remaining_delay += duration; + s->afq.remaining_samples += duration; } - av_fifo_generic_read(s->pkt_fifo, packets, pkt_size, NULL); + ff_af_queue_remove(&s->afq, duration, &avpkt->pts, &avpkt->duration); } - return pkt_size; + *got_packet_ptr = 1; + return 0; } AVCodec ff_libvorbis_encoder = { @@ -325,7 +354,7 @@ AVCodec ff_libvorbis_encoder = { .id = CODEC_ID_VORBIS, .priv_data_size = sizeof(OggVorbisContext), .init = oggvorbis_encode_init, - .encode = oggvorbis_encode_frame, + .encode2 = oggvorbis_encode_frame, .close = oggvorbis_encode_close, .capabilities = CODEC_CAP_DELAY, .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLT,