X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Flibvorbis.c;h=86c1ed6a75a5360703480ace0627f202af15469e;hb=b805482b1fba1d82fbe47023a24c9261f18979b6;hp=1d7b7ef49b8edc3db3fee94aed09f615e8bd7e7d;hpb=e5aab2d7a47942d61f8c54141da5c6ec33f7ce48;p=ffmpeg diff --git a/libavcodec/libvorbis.c b/libavcodec/libvorbis.c index 1d7b7ef49b8..86c1ed6a75a 100644 --- a/libavcodec/libvorbis.c +++ b/libavcodec/libvorbis.c @@ -43,11 +43,11 @@ * an output packet will always start at the same point as one of the input * packets. */ -#define OGGVORBIS_FRAME_SIZE 64 +#define LIBVORBIS_FRAME_SIZE 64 #define BUFFER_SIZE (1024 * 64) -typedef struct OggVorbisContext { +typedef struct LibvorbisContext { AVClass *av_class; /**< class for AVOptions */ vorbis_info vi; /**< vorbis_info used during init */ vorbis_dsp_state vd; /**< DSP state used for analysis */ @@ -58,12 +58,12 @@ 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 */ + AVVorbisParseContext *vp; /**< parse context to get durations */ AudioFrameQueue afq; /**< frame queue for timestamps */ -} OggVorbisContext; +} LibvorbisContext; static const AVOption options[] = { - { "iblock", "Sets the impulse block bias", offsetof(OggVorbisContext, iblock), AV_OPT_TYPE_DOUBLE, { .dbl = 0 }, -15, 0, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM }, + { "iblock", "Sets the impulse block bias", offsetof(LibvorbisContext, iblock), AV_OPT_TYPE_DOUBLE, { .dbl = 0 }, -15, 0, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM }, { NULL } }; @@ -85,21 +85,20 @@ static int vorbis_error_to_averror(int ov_err) } } -static av_cold int oggvorbis_init_encoder(vorbis_info *vi, - AVCodecContext *avctx) +static av_cold int libvorbis_setup(vorbis_info *vi, AVCodecContext *avctx) { - OggVorbisContext *s = avctx->priv_data; + LibvorbisContext *s = avctx->priv_data; double cfreq; int ret; - if (avctx->flags & CODEC_FLAG_QSCALE || !avctx->bit_rate) { + if (avctx->flags & AV_CODEC_FLAG_QSCALE || !avctx->bit_rate) { /* variable bitrate * NOTE: we use the oggenc range of -1 to 10 for global_quality for * user convenience, but libvorbis uses -0.1 to 1.0. */ float q = avctx->global_quality / (float)FF_QP2LAMBDA; /* default to 3 if the user did not set quality or bitrate */ - if (!(avctx->flags & CODEC_FLAG_QSCALE)) + if (!(avctx->flags & AV_CODEC_FLAG_QSCALE)) q = 3.0; if ((ret = vorbis_encode_setup_vbr(vi, avctx->channels, avctx->sample_rate, @@ -148,9 +147,9 @@ static int xiph_len(int l) return 1 + l / 255 + l; } -static av_cold int oggvorbis_encode_close(AVCodecContext *avctx) +static av_cold int libvorbis_encode_close(AVCodecContext *avctx) { - OggVorbisContext *s = avctx->priv_data; + LibvorbisContext *s = avctx->priv_data; /* notify vorbisenc this is EOF */ if (s->dsp_initialized) @@ -162,24 +161,23 @@ static av_cold int oggvorbis_encode_close(AVCodecContext *avctx) 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); + av_vorbis_parse_free(&s->vp); + return 0; } -static av_cold int oggvorbis_encode_init(AVCodecContext *avctx) +static av_cold int libvorbis_encode_init(AVCodecContext *avctx) { - OggVorbisContext *s = avctx->priv_data; + LibvorbisContext *s = avctx->priv_data; ogg_packet header, header_comm, header_code; uint8_t *p; unsigned int offset; int ret; vorbis_info_init(&s->vi); - if ((ret = oggvorbis_init_encoder(&s->vi, avctx))) { + if ((ret = libvorbis_setup(&s->vi, avctx))) { av_log(avctx, AV_LOG_ERROR, "encoder setup failed\n"); goto error; } @@ -208,7 +206,7 @@ static av_cold int oggvorbis_encode_init(AVCodecContext *avctx) xiph_len(header_comm.bytes) + header_code.bytes; p = avctx->extradata = av_malloc(avctx->extradata_size + - FF_INPUT_BUFFER_PADDING_SIZE); + AV_INPUT_BUFFER_PADDING_SIZE); if (!p) { ret = AVERROR(ENOMEM); goto error; @@ -225,14 +223,15 @@ 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) { + s->vp = av_vorbis_parse_init(avctx->extradata, avctx->extradata_size); + if (!s->vp) { av_log(avctx, AV_LOG_ERROR, "invalid extradata\n"); return ret; } vorbis_comment_clear(&s->vc); - avctx->frame_size = OGGVORBIS_FRAME_SIZE; + avctx->frame_size = LIBVORBIS_FRAME_SIZE; ff_af_queue_init(avctx, &s->afq); s->pkt_fifo = av_fifo_alloc(BUFFER_SIZE); @@ -241,47 +240,37 @@ 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: - oggvorbis_encode_close(avctx); + libvorbis_encode_close(avctx); return ret; } -static int oggvorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, +static int libvorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr) { - OggVorbisContext *s = avctx->priv_data; + LibvorbisContext *s = avctx->priv_data; ogg_packet op; int ret, duration; /* send samples to libvorbis */ if (frame) { - const float *audio = (const float *)frame->data[0]; const int samples = frame->nb_samples; float **buffer; int c, channels = s->vi.channels; buffer = vorbis_analysis_buffer(&s->vd, samples); for (c = 0; c < channels; c++) { - int i; int co = (channels > 8) ? c : ff_vorbis_encoding_channel_layout_offsets[channels - 1][c]; - for (i = 0; i < samples; i++) - buffer[c][i] = audio[i * channels + co]; + memcpy(buffer[c], frame->extended_data[co], + samples * sizeof(*buffer[c])); } if ((ret = vorbis_analysis_wrote(&s->vd, samples)) < 0) { 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)) + if ((ret = ff_af_queue_add(&s->afq, frame)) < 0) return ret; } else { if (!s->eof) @@ -332,12 +321,12 @@ static int oggvorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, avpkt->pts = ff_samples_to_time_base(avctx, op.granulepos); - duration = avpriv_vorbis_parse_frame(&s->vp, avpkt->data, avpkt->size); + duration = av_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; + if (!avctx->initial_padding) { + avctx->initial_padding = duration; s->afq.remaining_delay += duration; s->afq.remaining_samples += duration; } @@ -350,16 +339,16 @@ static int oggvorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, AVCodec ff_libvorbis_encoder = { .name = "libvorbis", + .long_name = NULL_IF_CONFIG_SMALL("libvorbis Vorbis"), .type = AVMEDIA_TYPE_AUDIO, - .id = CODEC_ID_VORBIS, - .priv_data_size = sizeof(OggVorbisContext), - .init = oggvorbis_encode_init, - .encode2 = oggvorbis_encode_frame, - .close = oggvorbis_encode_close, - .capabilities = CODEC_CAP_DELAY, - .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLT, + .id = AV_CODEC_ID_VORBIS, + .priv_data_size = sizeof(LibvorbisContext), + .init = libvorbis_encode_init, + .encode2 = libvorbis_encode_frame, + .close = libvorbis_encode_close, + .capabilities = AV_CODEC_CAP_DELAY, + .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE }, - .long_name = NULL_IF_CONFIG_SMALL("libvorbis Vorbis"), .priv_class = &class, .defaults = defaults, };