[FFmpeg-devel] [PATCH] libvorbis: split encoder from decoder
Paul B Mahol
onemda at gmail.com
Tue Jun 12 23:40:09 CEST 2012
Also fix build dependencies while here.
Signed-off-by: Paul B Mahol <onemda at gmail.com>
---
libavcodec/Makefile | 5 +-
libavcodec/libvorbis.c | 560 ---------------------------------------------
libavcodec/libvorbisdec.c | 200 ++++++++++++++++
libavcodec/libvorbisenc.c | 386 +++++++++++++++++++++++++++++++
4 files changed, 589 insertions(+), 562 deletions(-)
delete mode 100644 libavcodec/libvorbis.c
create mode 100644 libavcodec/libvorbisdec.c
create mode 100644 libavcodec/libvorbisenc.c
diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index 0bc6a0f..9e352a6 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -680,8 +680,9 @@ OBJS-$(CONFIG_LIBUTVIDEO_ENCODER) += libutvideoenc.o
OBJS-$(CONFIG_LIBVO_AACENC_ENCODER) += libvo-aacenc.o mpeg4audio.o \
audio_frame_queue.o
OBJS-$(CONFIG_LIBVO_AMRWBENC_ENCODER) += libvo-amrwbenc.o
-OBJS-$(CONFIG_LIBVORBIS_ENCODER) += libvorbis.o audio_frame_queue.o \
- vorbis_data.o vorbis_parser.o
+OBJS-$(CONFIG_LIBVORBIS_DECODER) += libvorbisdec.o
+OBJS-$(CONFIG_LIBVORBIS_ENCODER) += libvorbisenc.o audio_frame_queue.o \
+ vorbis_data.o vorbis_parser.o xiph.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
deleted file mode 100644
index 1324e49..0000000
--- a/libavcodec/libvorbis.c
+++ /dev/null
@@ -1,560 +0,0 @@
-/*
- * copyright (c) 2002 Mark Hills <mark at 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 at 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,
-} ;
diff --git a/libavcodec/libvorbisdec.c b/libavcodec/libvorbisdec.c
new file mode 100644
index 0000000..b9e5fe1
--- /dev/null
+++ b/libavcodec/libvorbisdec.c
@@ -0,0 +1,200 @@
+/*
+ * Copyright (c) 2002 Mark Hills <mark at 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 "avcodec.h"
+#include "bytestream.h"
+
+typedef struct OggVorbisDecContext {
+ 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 */
+ vorbis_comment vc; /**< VorbisComment info */
+ ogg_packet op; /**< ogg packet */
+} OggVorbisDecContext;
+
+static int oggvorbis_decode_init(AVCodecContext *avccontext) {
+ OggVorbisDecContext *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)
+{
+ OggVorbisDecContext *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) {
+ OggVorbisDecContext *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(OggVorbisDecContext),
+ .init = oggvorbis_decode_init,
+ .decode = oggvorbis_decode_frame,
+ .close = oggvorbis_decode_close,
+ .capabilities = CODEC_CAP_DELAY,
+ .long_name = NULL_IF_CONFIG_SMALL("libvorbis"),
+};
diff --git a/libavcodec/libvorbisenc.c b/libavcodec/libvorbisenc.c
new file mode 100644
index 0000000..7422a35
--- /dev/null
+++ b/libavcodec/libvorbisenc.c
@@ -0,0 +1,386 @@
+/*
+ * Copyright (c) 2002 Mark Hills <mark at 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/fifo.h"
+#include "libavutil/opt.h"
+#include "avcodec.h"
+#include "audio_frame_queue.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 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 = { "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)
+{
+ 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;
+ 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)
+{
+ 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 = 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,
+};
--
1.7.7
More information about the ffmpeg-devel
mailing list