| /* |
| * 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 |
| */ |
| |
| /** |
| * @file |
| * Vorbis encoding support via libvorbisenc. |
| * @author Mark Hills <mark@pogo.org.uk> |
| */ |
| |
| #include <vorbis/vorbisenc.h> |
| |
| #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 <assert.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 OggVorbisContext { |
| 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 */ |
| 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[] = { |
| { "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 }, |
| { NULL } |
| }; |
| |
| static const AVCodecDefault defaults[] = { |
| { "b", "0" }, |
| { NULL }, |
| }; |
| |
| static const AVClass class = { "libvorbis", av_default_item_name, options, 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) |
| { |
| OggVorbisContext *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) |
| { |
| OggVorbisContext *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) |
| { |
| OggVorbisContext *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; |
| 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) { |
| 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) |
| { |
| OggVorbisContext *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 = 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, |
| AV_SAMPLE_FMT_NONE }, |
| .long_name = NULL_IF_CONFIG_SMALL("libvorbis Vorbis"), |
| .priv_class = &class, |
| .defaults = defaults, |
| }; |
| |
| static int oggvorbis_decode_init(AVCodecContext *avccontext) { |
| OggVorbisContext *context = avccontext->priv_data ; |
| uint8_t *p= avccontext->extradata; |
| int i, hsizes[3]; |
| unsigned char *headers[3], *extradata = avccontext->extradata; |
| |
| vorbis_info_init(&context->vi) ; |
| vorbis_comment_init(&context->vc) ; |
| |
| if(! avccontext->extradata_size || ! p) { |
| av_log(avccontext, AV_LOG_ERROR, "vorbis extradata absent\n"); |
| return -1; |
| } |
| |
| if(p[0] == 0 && p[1] == 30) { |
| for(i = 0; i < 3; i++){ |
| hsizes[i] = bytestream_get_be16(&p); |
| headers[i] = p; |
| p += hsizes[i]; |
| } |
| } else if(*p == 2) { |
| unsigned int offset = 1; |
| p++; |
| for(i=0; i<2; i++) { |
| hsizes[i] = 0; |
| while((*p == 0xFF) && (offset < avccontext->extradata_size)) { |
| hsizes[i] += 0xFF; |
| offset++; |
| p++; |
| } |
| if(offset >= avccontext->extradata_size - 1) { |
| av_log(avccontext, AV_LOG_ERROR, |
| "vorbis header sizes damaged\n"); |
| return -1; |
| } |
| hsizes[i] += *p; |
| offset++; |
| p++; |
| } |
| hsizes[2] = avccontext->extradata_size - hsizes[0]-hsizes[1]-offset; |
| #if 0 |
| av_log(avccontext, AV_LOG_DEBUG, |
| "vorbis header sizes: %d, %d, %d, / extradata_len is %d \n", |
| hsizes[0], hsizes[1], hsizes[2], avccontext->extradata_size); |
| #endif |
| headers[0] = extradata + offset; |
| headers[1] = extradata + offset + hsizes[0]; |
| headers[2] = extradata + offset + hsizes[0] + hsizes[1]; |
| } else { |
| av_log(avccontext, AV_LOG_ERROR, |
| "vorbis initial header len is wrong: %d\n", *p); |
| return -1; |
| } |
| |
| for(i=0; i<3; i++){ |
| context->op.b_o_s= i==0; |
| context->op.bytes = hsizes[i]; |
| context->op.packet = headers[i]; |
| if(vorbis_synthesis_headerin(&context->vi, &context->vc, &context->op)<0){ |
| av_log(avccontext, AV_LOG_ERROR, "%d. vorbis header damaged\n", i+1); |
| return -1; |
| } |
| } |
| |
| avccontext->channels = context->vi.channels; |
| avccontext->sample_rate = context->vi.rate; |
| avccontext->time_base= (AVRational){1, avccontext->sample_rate}; |
| |
| vorbis_synthesis_init(&context->vd, &context->vi); |
| vorbis_block_init(&context->vd, &context->vb); |
| |
| return 0 ; |
| } |
| |
| |
| static inline int conv(int samples, float **pcm, char *buf, int channels) { |
| int i, j; |
| ogg_int16_t *ptr, *data = (ogg_int16_t*)buf ; |
| float *mono ; |
| |
| for(i = 0 ; i < channels ; i++){ |
| ptr = &data[i]; |
| mono = pcm[i] ; |
| |
| for(j = 0 ; j < samples ; j++) { |
| *ptr = av_clip_int16(mono[j] * 32767.f); |
| ptr += channels; |
| } |
| } |
| |
| return 0 ; |
| } |
| |
| static int oggvorbis_decode_frame(AVCodecContext *avccontext, void *data, |
| int *got_frame_ptr, AVPacket *avpkt) |
| { |
| OggVorbisContext *context = avccontext->priv_data ; |
| float **pcm ; |
| ogg_packet *op= &context->op; |
| int samples, total_samples, total_bytes; |
| int ret; |
| int16_t *output; |
| |
| if(!avpkt->size){ |
| //FIXME flush |
| return 0; |
| } |
| |
| context->frame.nb_samples = 8192*4; |
| if ((ret = avccontext->get_buffer(avccontext, &context->frame)) < 0) { |
| av_log(avccontext, AV_LOG_ERROR, "get_buffer() failed\n"); |
| return ret; |
| } |
| output = (int16_t *)context->frame.data[0]; |
| |
| |
| op->packet = avpkt->data; |
| op->bytes = avpkt->size; |
| |
| // av_log(avccontext, AV_LOG_DEBUG, "%d %d %d %"PRId64" %"PRId64" %d %d\n", op->bytes, op->b_o_s, op->e_o_s, op->granulepos, op->packetno, buf_size, context->vi.rate); |
| |
| /* for(i=0; i<op->bytes; i++) |
| av_log(avccontext, AV_LOG_DEBUG, "%02X ", op->packet[i]); |
| av_log(avccontext, AV_LOG_DEBUG, "\n");*/ |
| |
| if(vorbis_synthesis(&context->vb, op) == 0) |
| vorbis_synthesis_blockin(&context->vd, &context->vb) ; |
| |
| total_samples = 0 ; |
| total_bytes = 0 ; |
| |
| while((samples = vorbis_synthesis_pcmout(&context->vd, &pcm)) > 0) { |
| conv(samples, pcm, (char*)output + total_bytes, context->vi.channels) ; |
| total_bytes += samples * 2 * context->vi.channels ; |
| total_samples += samples ; |
| vorbis_synthesis_read(&context->vd, samples) ; |
| } |
| |
| context->frame.nb_samples = total_samples; |
| *got_frame_ptr = 1; |
| *(AVFrame *)data = context->frame; |
| return avpkt->size; |
| } |
| |
| |
| static int oggvorbis_decode_close(AVCodecContext *avccontext) { |
| OggVorbisContext *context = avccontext->priv_data ; |
| |
| vorbis_info_clear(&context->vi) ; |
| vorbis_comment_clear(&context->vc) ; |
| |
| return 0 ; |
| } |
| |
| |
| AVCodec ff_libvorbis_decoder = { |
| .name = "libvorbis", |
| .type = AVMEDIA_TYPE_AUDIO, |
| .id = CODEC_ID_VORBIS, |
| .priv_data_size = sizeof(OggVorbisContext), |
| .init = oggvorbis_decode_init, |
| .decode = oggvorbis_decode_frame, |
| .close = oggvorbis_decode_close, |
| .capabilities = CODEC_CAP_DELAY, |
| } ; |
| |