diff options
Diffstat (limited to 'libavcodec/libvorbisenc.c')
| -rw-r--r-- | libavcodec/libvorbisenc.c | 389 |
1 files changed, 389 insertions, 0 deletions
diff --git a/libavcodec/libvorbisenc.c b/libavcodec/libvorbisenc.c new file mode 100644 index 0000000..76a912d --- /dev/null +++ b/libavcodec/libvorbisenc.c @@ -0,0 +1,389 @@ +/* + * Copyright (c) 2002 Mark Hills <mark@pogo.org.uk> + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include <vorbis/vorbisenc.h> + +#include "libavutil/avassert.h" +#include "libavutil/fifo.h" +#include "libavutil/opt.h" +#include "avcodec.h" +#include "audio_frame_queue.h" +#include "internal.h" +#include "vorbis.h" +#include "vorbis_parser.h" + + +/* Number of samples the user should send in each call. + * This value is used because it is the LCD of all possible frame sizes, so + * an output packet will always start at the same point as one of the input + * packets. + */ +#define OGGVORBIS_FRAME_SIZE 64 + +#define BUFFER_SIZE (1024 * 64) + +typedef struct OggVorbisEncContext { + AVClass *av_class; /**< class for AVOptions */ + AVFrame frame; + vorbis_info vi; /**< vorbis_info used during init */ + vorbis_dsp_state vd; /**< DSP state used for analysis */ + vorbis_block vb; /**< vorbis_block used for analysis */ + AVFifoBuffer *pkt_fifo; /**< output packet buffer */ + int eof; /**< end-of-file flag */ + int dsp_initialized; /**< vd has been initialized */ + vorbis_comment vc; /**< VorbisComment info */ + double iblock; /**< impulse block bias option */ + VorbisParseContext vp; /**< parse context to get durations */ + AudioFrameQueue afq; /**< frame queue for timestamps */ +} OggVorbisEncContext; + +static const AVOption options[] = { + { "iblock", "Sets the impulse block bias", offsetof(OggVorbisEncContext, iblock), AV_OPT_TYPE_DOUBLE, { .dbl = 0 }, -15, 0, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM }, + { NULL } +}; + +static const AVCodecDefault defaults[] = { + { "b", "0" }, + { NULL }, +}; + +static const AVClass class = { + .class_name = "libvorbis", + .item_name = av_default_item_name, + .option = options, + .version = LIBAVUTIL_VERSION_INT, +}; + +static int vorbis_error_to_averror(int ov_err) +{ + switch (ov_err) { + case OV_EFAULT: return AVERROR_BUG; + case OV_EINVAL: return AVERROR(EINVAL); + case OV_EIMPL: return AVERROR(EINVAL); + default: return AVERROR_UNKNOWN; + } +} + +static av_cold int oggvorbis_init_encoder(vorbis_info *vi, + AVCodecContext *avctx) +{ + OggVorbisEncContext *s = avctx->priv_data; + double cfreq; + int ret; + + if (avctx->flags & 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)) + q = 3.0; + if ((ret = vorbis_encode_setup_vbr(vi, avctx->channels, + avctx->sample_rate, + q / 10.0))) + goto error; + } else { + int minrate = avctx->rc_min_rate > 0 ? avctx->rc_min_rate : -1; + int maxrate = avctx->rc_max_rate > 0 ? avctx->rc_max_rate : -1; + + /* average bitrate */ + if ((ret = vorbis_encode_setup_managed(vi, avctx->channels, + avctx->sample_rate, maxrate, + avctx->bit_rate, minrate))) + goto error; + + /* variable bitrate by estimate, disable slow rate management */ + if (minrate == -1 && maxrate == -1) + if ((ret = vorbis_encode_ctl(vi, OV_ECTL_RATEMANAGE2_SET, NULL))) + goto error; /* should not happen */ + } + + /* cutoff frequency */ + if (avctx->cutoff > 0) { + cfreq = avctx->cutoff / 1000.0; + if ((ret = vorbis_encode_ctl(vi, OV_ECTL_LOWPASS_SET, &cfreq))) + goto error; /* should not happen */ + } + + /* impulse block bias */ + if (s->iblock) { + if ((ret = vorbis_encode_ctl(vi, OV_ECTL_IBLOCK_SET, &s->iblock))) + goto error; + } + + if (avctx->channels == 3 && + avctx->channel_layout != (AV_CH_LAYOUT_STEREO|AV_CH_FRONT_CENTER) || + avctx->channels == 4 && + avctx->channel_layout != AV_CH_LAYOUT_2_2 && + avctx->channel_layout != AV_CH_LAYOUT_QUAD || + avctx->channels == 5 && + avctx->channel_layout != AV_CH_LAYOUT_5POINT0 && + avctx->channel_layout != AV_CH_LAYOUT_5POINT0_BACK || + avctx->channels == 6 && + avctx->channel_layout != AV_CH_LAYOUT_5POINT1 && + avctx->channel_layout != AV_CH_LAYOUT_5POINT1_BACK || + avctx->channels == 7 && + avctx->channel_layout != (AV_CH_LAYOUT_5POINT1|AV_CH_BACK_CENTER) || + avctx->channels == 8 && + avctx->channel_layout != AV_CH_LAYOUT_7POINT1) { + if (avctx->channel_layout) { + char name[32]; + av_get_channel_layout_string(name, sizeof(name), avctx->channels, + avctx->channel_layout); + av_log(avctx, AV_LOG_ERROR, "%s not supported by Vorbis: " + "output stream will have incorrect " + "channel layout.\n", name); + } else { + av_log(avctx, AV_LOG_WARNING, "No channel layout specified. The encoder " + "will use Vorbis channel layout for " + "%d channels.\n", avctx->channels); + } + } + + if ((ret = vorbis_encode_setup_init(vi))) + goto error; + + return 0; +error: + return vorbis_error_to_averror(ret); +} + +/* How many bytes are needed for a buffer of length 'l' */ +static int xiph_len(int l) +{ + return 1 + l / 255 + l; +} + +static av_cold int oggvorbis_encode_close(AVCodecContext *avctx) +{ + OggVorbisEncContext *s = avctx->priv_data; + + /* notify vorbisenc this is EOF */ + if (s->dsp_initialized) + vorbis_analysis_wrote(&s->vd, 0); + + vorbis_block_clear(&s->vb); + vorbis_dsp_clear(&s->vd); + 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; +} + +static av_cold int oggvorbis_encode_init(AVCodecContext *avctx) +{ + OggVorbisEncContext *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))) { + av_log(avctx, AV_LOG_ERROR, "encoder setup failed\n"); + goto error; + } + if ((ret = vorbis_analysis_init(&s->vd, &s->vi))) { + av_log(avctx, AV_LOG_ERROR, "analysis init failed\n"); + ret = vorbis_error_to_averror(ret); + goto error; + } + s->dsp_initialized = 1; + if ((ret = vorbis_block_init(&s->vd, &s->vb))) { + av_log(avctx, AV_LOG_ERROR, "dsp init failed\n"); + ret = vorbis_error_to_averror(ret); + goto error; + } + + vorbis_comment_init(&s->vc); + if (!(avctx->flags & CODEC_FLAG_BITEXACT)) + vorbis_comment_add_tag(&s->vc, "encoder", LIBAVCODEC_IDENT); + + if ((ret = vorbis_analysis_headerout(&s->vd, &s->vc, &header, &header_comm, + &header_code))) { + ret = vorbis_error_to_averror(ret); + goto error; + } + + avctx->extradata_size = 1 + xiph_len(header.bytes) + + xiph_len(header_comm.bytes) + + header_code.bytes; + p = avctx->extradata = av_malloc(avctx->extradata_size + + FF_INPUT_BUFFER_PADDING_SIZE); + if (!p) { + ret = AVERROR(ENOMEM); + goto error; + } + p[0] = 2; + offset = 1; + offset += av_xiphlacing(&p[offset], header.bytes); + offset += av_xiphlacing(&p[offset], header_comm.bytes); + memcpy(&p[offset], header.packet, header.bytes); + offset += header.bytes; + memcpy(&p[offset], header_comm.packet, header_comm.bytes); + offset += header_comm.bytes; + memcpy(&p[offset], header_code.packet, header_code.bytes); + offset += header_code.bytes; + av_assert0(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) { + ret = AVERROR(ENOMEM); + 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); + return ret; +} + +static int oggvorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, + const AVFrame *frame, int *got_packet_ptr) +{ + OggVorbisEncContext *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]; + } + 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)) + return ret; + } else { + if (!s->eof) + if ((ret = vorbis_analysis_wrote(&s->vd, 0)) < 0) { + av_log(avctx, AV_LOG_ERROR, "error in vorbis_analysis_wrote()\n"); + return vorbis_error_to_averror(ret); + } + s->eof = 1; + } + + /* retrieve available packets from libvorbis */ + while ((ret = vorbis_analysis_blockout(&s->vd, &s->vb)) == 1) { + if ((ret = vorbis_analysis(&s->vb, NULL)) < 0) + break; + if ((ret = vorbis_bitrate_addblock(&s->vb)) < 0) + break; + + /* add any available packets to the output packet buffer */ + while ((ret = vorbis_bitrate_flushpacket(&s->vd, &op)) == 1) { + if (av_fifo_space(s->pkt_fifo) < sizeof(ogg_packet) + op.bytes) { + av_log(avctx, AV_LOG_ERROR, "packet buffer is too small\n"); + return AVERROR_BUG; + } + av_fifo_generic_write(s->pkt_fifo, &op, sizeof(ogg_packet), NULL); + av_fifo_generic_write(s->pkt_fifo, op.packet, op.bytes, NULL); + } + if (ret < 0) { + av_log(avctx, AV_LOG_ERROR, "error getting available packets\n"); + break; + } + } + if (ret < 0) { + av_log(avctx, AV_LOG_ERROR, "error getting available packets\n"); + return vorbis_error_to_averror(ret); + } + + /* 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_packet2(avctx, avpkt, op.bytes))) + 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; + } + ff_af_queue_remove(&s->afq, duration, &avpkt->pts, &avpkt->duration); + } + + *got_packet_ptr = 1; + return 0; +} + +AVCodec ff_libvorbis_encoder = { + .name = "libvorbis", + .type = AVMEDIA_TYPE_AUDIO, + .id = AV_CODEC_ID_VORBIS, + .priv_data_size = sizeof(OggVorbisEncContext), + .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, + AV_SAMPLE_FMT_NONE }, + .long_name = NULL_IF_CONFIG_SMALL("libvorbis"), + .priv_class = &class, + .defaults = defaults, +}; |
