[deb_ffmpeg.git] / ffmpeg / libavcodec / libvorbisenc.c

/*
 * 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 LIBVORBIS_FRAME_SIZE 64

#define BUFFER_SIZE (1024 * 64)

typedef struct LibvorbisEncContext {
    AVClass *av_class;                  /**< class for AVOptions            */
    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      */
    AVVorbisParseContext *vp;           /**< parse context to get durations */
    AudioFrameQueue afq;                /**< frame queue for timestamps     */
} LibvorbisEncContext;

static const AVOption options[] = {
    { "iblock", "Sets the impulse block bias", offsetof(LibvorbisEncContext, 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 vorbis_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 libvorbis_setup(vorbis_info *vi, AVCodecContext *avctx)
{
    LibvorbisEncContext *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 libvorbis_encode_close(AVCodecContext *avctx)
{
    LibvorbisEncContext *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_freep(&s->pkt_fifo);
    ff_af_queue_close(&s->afq);
    av_freep(&avctx->extradata);

    av_vorbis_parse_free(&s->vp);

    return 0;
}

static av_cold int libvorbis_encode_init(AVCodecContext *avctx)
{
    LibvorbisEncContext *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 = libvorbis_setup(&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);

    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 = LIBVORBIS_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;
    }

    return 0;
error:
    libvorbis_encode_close(avctx);
    return ret;
}

static int libvorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
                                  const AVFrame *frame, int *got_packet_ptr)
{
    LibvorbisEncContext *s = avctx->priv_data;
    ogg_packet op;
    int ret, duration;

    /* send samples to libvorbis */
    if (frame) {
        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 co = (channels > 8) ? c :
                     ff_vorbis_encoding_channel_layout_offsets[channels - 1][c];
            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)
            return ret;
    } else {
        if (!s->eof && s->afq.frame_alloc)
            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)) < 0)
        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 = 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->initial_padding && s->afq.frames) {
            avctx->initial_padding    = duration;
            av_assert0(!s->afq.remaining_delay);
            s->afq.frames->duration  += duration;
            if (s->afq.frames->pts != AV_NOPTS_VALUE)
                s->afq.frames->pts       -= 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",
    .long_name      = NULL_IF_CONFIG_SMALL("libvorbis"),
    .type           = AVMEDIA_TYPE_AUDIO,
    .id             = AV_CODEC_ID_VORBIS,
    .priv_data_size = sizeof(LibvorbisEncContext),
    .init           = libvorbis_encode_init,
    .encode2        = libvorbis_encode_frame,
    .close          = libvorbis_encode_close,
    .capabilities   = CODEC_CAP_DELAY | CODEC_CAP_SMALL_LAST_FRAME,
    .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
                                                      AV_SAMPLE_FMT_NONE },
    .priv_class     = &vorbis_class,
    .defaults       = defaults,
};
Commit	Line	Data
2ba45a60 DM	1	/*
	2	* Copyright (c) 2002 Mark Hills <mark@pogo.org.uk>
	3	*
	4	* This file is part of FFmpeg.
	5	*
	6	* FFmpeg is free software; you can redistribute it and/or
	7	* modify it under the terms of the GNU Lesser General Public
	8	* License as published by the Free Software Foundation; either
	9	* version 2.1 of the License, or (at your option) any later version.
	10	*
	11	* FFmpeg is distributed in the hope that it will be useful,
	12	* but WITHOUT ANY WARRANTY; without even the implied warranty of
	13	* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
	14	* Lesser General Public License for more details.
	15	*
	16	* You should have received a copy of the GNU Lesser General Public
	17	* License along with FFmpeg; if not, write to the Free Software
	18	* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
	19	*/
	20
	21	#include <vorbis/vorbisenc.h>
	22
	23	#include "libavutil/avassert.h"
	24	#include "libavutil/fifo.h"
	25	#include "libavutil/opt.h"
	26	#include "avcodec.h"
	27	#include "audio_frame_queue.h"
	28	#include "internal.h"
	29	#include "vorbis.h"
	30	#include "vorbis_parser.h"
	31
	32
	33	/* Number of samples the user should send in each call.
	34	* This value is used because it is the LCD of all possible frame sizes, so
	35	* an output packet will always start at the same point as one of the input
	36	* packets.
	37	*/
	38	#define LIBVORBIS_FRAME_SIZE 64
	39
	40	#define BUFFER_SIZE (1024 * 64)
	41
	42	typedef struct LibvorbisEncContext {
	43	AVClass av_class; /< class for AVOptions /
	44	vorbis_info vi; /*< vorbis_info used during init /
	45	vorbis_dsp_state vd; /*< DSP state used for analysis /
	46	vorbis_block vb; /*< vorbis_block used for analysis /
	47	AVFifoBuffer pkt_fifo; /< output packet buffer /
	48	int eof; /*< end-of-file flag /
	49	int dsp_initialized; /*< vd has been initialized /
	50	vorbis_comment vc; /*< VorbisComment info /
	51	double iblock; /*< impulse block bias option /
f6fa7814	52	AVVorbisParseContext vp; /< parse context to get durations /
2ba45a60 DM	53	AudioFrameQueue afq; /*< frame queue for timestamps /
	54	} LibvorbisEncContext;
	55
	56	static const AVOption options[] = {
	57	{ "iblock", "Sets the impulse block bias", offsetof(LibvorbisEncContext, iblock), AV_OPT_TYPE_DOUBLE, { .dbl = 0 }, -15, 0, AV_OPT_FLAG_AUDIO_PARAM \| AV_OPT_FLAG_ENCODING_PARAM },
	58	{ NULL }
	59	};
	60
	61	static const AVCodecDefault defaults[] = {
	62	{ "b", "0" },
	63	{ NULL },
	64	};
	65
	66	static const AVClass vorbis_class = {
	67	.class_name = "libvorbis",
	68	.item_name = av_default_item_name,
	69	.option = options,
	70	.version = LIBAVUTIL_VERSION_INT,
	71	};
	72
	73	static int vorbis_error_to_averror(int ov_err)
	74	{
	75	switch (ov_err) {
	76	case OV_EFAULT: return AVERROR_BUG;
	77	case OV_EINVAL: return AVERROR(EINVAL);
	78	case OV_EIMPL: return AVERROR(EINVAL);
	79	default: return AVERROR_UNKNOWN;
	80	}
	81	}
	82
	83	static av_cold int libvorbis_setup(vorbis_info vi, AVCodecContext avctx)
	84	{
	85	LibvorbisEncContext *s = avctx->priv_data;
	86	double cfreq;
	87	int ret;
	88
	89	if (avctx->flags & CODEC_FLAG_QSCALE \|\| !avctx->bit_rate) {
	90	/* variable bitrate
	91	* NOTE: we use the oggenc range of -1 to 10 for global_quality for
	92	* user convenience, but libvorbis uses -0.1 to 1.0.
	93	*/
	94	float q = avctx->global_quality / (float)FF_QP2LAMBDA;
	95	/* default to 3 if the user did not set quality or bitrate */
	96	if (!(avctx->flags & CODEC_FLAG_QSCALE))
	97	q = 3.0;
	98	if ((ret = vorbis_encode_setup_vbr(vi, avctx->channels,
	99	avctx->sample_rate,
	100	q / 10.0)))
	101	goto error;
	102	} else {
	103	int minrate = avctx->rc_min_rate > 0 ? avctx->rc_min_rate : -1;
	104	int maxrate = avctx->rc_max_rate > 0 ? avctx->rc_max_rate : -1;
	105
	106	/* average bitrate */
	107	if ((ret = vorbis_encode_setup_managed(vi, avctx->channels,
	108	avctx->sample_rate, maxrate,
	109	avctx->bit_rate, minrate)))
	110	goto error;
	111
	112	/* variable bitrate by estimate, disable slow rate management */
	113	if (minrate == -1 && maxrate == -1)
	114	if ((ret = vorbis_encode_ctl(vi, OV_ECTL_RATEMANAGE2_SET, NULL)))
	115	goto error; /* should not happen */
	116	}
117
118	/* cutoff frequency */
119	if (avctx->cutoff > 0) {
120	cfreq = avctx->cutoff / 1000.0;
121	if ((ret = vorbis_encode_ctl(vi, OV_ECTL_LOWPASS_SET, &cfreq)))
122	goto error; /* should not happen */
123	}
124
125	/* impulse block bias */
126	if (s->iblock) {
127	if ((ret = vorbis_encode_ctl(vi, OV_ECTL_IBLOCK_SET, &s->iblock)))
128	goto error;
129	}
130
131	if (avctx->channels == 3 &&
132	avctx->channel_layout != (AV_CH_LAYOUT_STEREO\|AV_CH_FRONT_CENTER) \|\|
133	avctx->channels == 4 &&
134	avctx->channel_layout != AV_CH_LAYOUT_2_2 &&
135	avctx->channel_layout != AV_CH_LAYOUT_QUAD \|\|
136	avctx->channels == 5 &&
137	avctx->channel_layout != AV_CH_LAYOUT_5POINT0 &&
138	avctx->channel_layout != AV_CH_LAYOUT_5POINT0_BACK \|\|
139	avctx->channels == 6 &&
140	avctx->channel_layout != AV_CH_LAYOUT_5POINT1 &&
141	avctx->channel_layout != AV_CH_LAYOUT_5POINT1_BACK \|\|
142	avctx->channels == 7 &&
143	avctx->channel_layout != (AV_CH_LAYOUT_5POINT1\|AV_CH_BACK_CENTER) \|\|
144	avctx->channels == 8 &&
145	avctx->channel_layout != AV_CH_LAYOUT_7POINT1) {
146	if (avctx->channel_layout) {
147	char name[32];
148	av_get_channel_layout_string(name, sizeof(name), avctx->channels,
149	avctx->channel_layout);
150	av_log(avctx, AV_LOG_ERROR, "%s not supported by Vorbis: "
151	"output stream will have incorrect "
152	"channel layout.\n", name);
153	} else {
154	av_log(avctx, AV_LOG_WARNING, "No channel layout specified. The encoder "
155	"will use Vorbis channel layout for "
156	"%d channels.\n", avctx->channels);
157	}
158	}
159
160	if ((ret = vorbis_encode_setup_init(vi)))
161	goto error;
162
163	return 0;
164	error:
165	return vorbis_error_to_averror(ret);
166	}
167
168	/* How many bytes are needed for a buffer of length 'l' */
169	static int xiph_len(int l)
170	{
171	return 1 + l / 255 + l;
172	}
173
174	static av_cold int libvorbis_encode_close(AVCodecContext *avctx)
175	{
176	LibvorbisEncContext *s = avctx->priv_data;
177
178	/* notify vorbisenc this is EOF */
179	if (s->dsp_initialized)
180	vorbis_analysis_wrote(&s->vd, 0);
181
182	vorbis_block_clear(&s->vb);
183	vorbis_dsp_clear(&s->vd);
184	vorbis_info_clear(&s->vi);
185
186	av_fifo_freep(&s->pkt_fifo);
187	ff_af_queue_close(&s->afq);
188	av_freep(&avctx->extradata);
189
f6fa7814 DM	190	av_vorbis_parse_free(&s->vp);
f6fa7814 DM	191
2ba45a60 DM	192	return 0;
	193	}
	194
	195	static av_cold int libvorbis_encode_init(AVCodecContext *avctx)
	196	{
	197	LibvorbisEncContext *s = avctx->priv_data;
	198	ogg_packet header, header_comm, header_code;
	199	uint8_t *p;
	200	unsigned int offset;
	201	int ret;
	202
	203	vorbis_info_init(&s->vi);
	204	if ((ret = libvorbis_setup(&s->vi, avctx))) {
	205	av_log(avctx, AV_LOG_ERROR, "encoder setup failed\n");
	206	goto error;
	207	}
	208	if ((ret = vorbis_analysis_init(&s->vd, &s->vi))) {
	209	av_log(avctx, AV_LOG_ERROR, "analysis init failed\n");
	210	ret = vorbis_error_to_averror(ret);
	211	goto error;
	212	}
	213	s->dsp_initialized = 1;
	214	if ((ret = vorbis_block_init(&s->vd, &s->vb))) {
	215	av_log(avctx, AV_LOG_ERROR, "dsp init failed\n");
	216	ret = vorbis_error_to_averror(ret);
	217	goto error;
	218	}
	219
	220	vorbis_comment_init(&s->vc);
	221	if (!(avctx->flags & CODEC_FLAG_BITEXACT))
	222	vorbis_comment_add_tag(&s->vc, "encoder", LIBAVCODEC_IDENT);
	223
	224	if ((ret = vorbis_analysis_headerout(&s->vd, &s->vc, &header, &header_comm,
	225	&header_code))) {
	226	ret = vorbis_error_to_averror(ret);
	227	goto error;
	228	}
	229
	230	avctx->extradata_size = 1 + xiph_len(header.bytes) +
	231	xiph_len(header_comm.bytes) +
	232	header_code.bytes;
	233	p = avctx->extradata = av_malloc(avctx->extradata_size +
	234	FF_INPUT_BUFFER_PADDING_SIZE);
	235	if (!p) {
	236	ret = AVERROR(ENOMEM);
	237	goto error;
	238	}
	239	p[0] = 2;
	240	offset = 1;
	241	offset += av_xiphlacing(&p[offset], header.bytes);
	242	offset += av_xiphlacing(&p[offset], header_comm.bytes);
	243	memcpy(&p[offset], header.packet, header.bytes);
	244	offset += header.bytes;
	245	memcpy(&p[offset], header_comm.packet, header_comm.bytes);
	246	offset += header_comm.bytes;
	247	memcpy(&p[offset], header_code.packet, header_code.bytes);
	248	offset += header_code.bytes;
	249	av_assert0(offset == avctx->extradata_size);
	250
f6fa7814 DM	251	s->vp = av_vorbis_parse_init(avctx->extradata, avctx->extradata_size);
f6fa7814 DM	252	if (!s->vp) {
2ba45a60 DM	253	av_log(avctx, AV_LOG_ERROR, "invalid extradata\n");
	254	return ret;
	255	}
	256
	257	vorbis_comment_clear(&s->vc);
	258
	259	avctx->frame_size = LIBVORBIS_FRAME_SIZE;
	260	ff_af_queue_init(avctx, &s->afq);
	261
	262	s->pkt_fifo = av_fifo_alloc(BUFFER_SIZE);
	263	if (!s->pkt_fifo) {
	264	ret = AVERROR(ENOMEM);
	265	goto error;
	266	}
	267
	268	return 0;
	269	error:
	270	libvorbis_encode_close(avctx);
	271	return ret;
	272	}
	273
	274	static int libvorbis_encode_frame(AVCodecContext avctx, AVPacket avpkt,
	275	const AVFrame frame, int got_packet_ptr)
	276	{
	277	LibvorbisEncContext *s = avctx->priv_data;
	278	ogg_packet op;
	279	int ret, duration;
	280
	281	/* send samples to libvorbis */
	282	if (frame) {
	283	const int samples = frame->nb_samples;
	284	float **buffer;
	285	int c, channels = s->vi.channels;
	286
	287	buffer = vorbis_analysis_buffer(&s->vd, samples);
	288	for (c = 0; c < channels; c++) {
	289	int co = (channels > 8) ? c :
	290	ff_vorbis_encoding_channel_layout_offsets[channels - 1][c];
	291	memcpy(buffer[c], frame->extended_data[co],
	292	samples * sizeof(*buffer[c]));
	293	}
	294	if ((ret = vorbis_analysis_wrote(&s->vd, samples)) < 0) {
	295	av_log(avctx, AV_LOG_ERROR, "error in vorbis_analysis_wrote()\n");
	296	return vorbis_error_to_averror(ret);
	297	}
	298	if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
	299	return ret;
	300	} else {
	301	if (!s->eof && s->afq.frame_alloc)
	302	if ((ret = vorbis_analysis_wrote(&s->vd, 0)) < 0) {
	303	av_log(avctx, AV_LOG_ERROR, "error in vorbis_analysis_wrote()\n");
	304	return vorbis_error_to_averror(ret);
	305	}
	306	s->eof = 1;
	307	}
	308
	309	/* retrieve available packets from libvorbis */
	310	while ((ret = vorbis_analysis_blockout(&s->vd, &s->vb)) == 1) {
	311	if ((ret = vorbis_analysis(&s->vb, NULL)) < 0)
	312	break;
	313	if ((ret = vorbis_bitrate_addblock(&s->vb)) < 0)
	314	break;
	315
	316	/* add any available packets to the output packet buffer */
317	while ((ret = vorbis_bitrate_flushpacket(&s->vd, &op)) == 1) {
318	if (av_fifo_space(s->pkt_fifo) < sizeof(ogg_packet) + op.bytes) {
319	av_log(avctx, AV_LOG_ERROR, "packet buffer is too small\n");
320	return AVERROR_BUG;
321	}
322	av_fifo_generic_write(s->pkt_fifo, &op, sizeof(ogg_packet), NULL);
323	av_fifo_generic_write(s->pkt_fifo, op.packet, op.bytes, NULL);
324	}
325	if (ret < 0) {
326	av_log(avctx, AV_LOG_ERROR, "error getting available packets\n");
327	break;
328	}
329	}
330	if (ret < 0) {
331	av_log(avctx, AV_LOG_ERROR, "error getting available packets\n");
332	return vorbis_error_to_averror(ret);
333	}
334
335	/* check for available packets */
336	if (av_fifo_size(s->pkt_fifo) < sizeof(ogg_packet))
337	return 0;
338
339	av_fifo_generic_read(s->pkt_fifo, &op, sizeof(ogg_packet), NULL);
340
341	if ((ret = ff_alloc_packet2(avctx, avpkt, op.bytes)) < 0)
342	return ret;
343	av_fifo_generic_read(s->pkt_fifo, avpkt->data, op.bytes, NULL);
344
345	avpkt->pts = ff_samples_to_time_base(avctx, op.granulepos);
346
f6fa7814	347	duration = av_vorbis_parse_frame(s->vp, avpkt->data, avpkt->size);
2ba45a60 DM	348	if (duration > 0) {
	349	/* we do not know encoder delay until we get the first packet from
	350	* libvorbis, so we have to update the AudioFrameQueue counts */
f6fa7814 DM	351	if (!avctx->initial_padding && s->afq.frames) {
f6fa7814 DM	352	avctx->initial_padding = duration;
2ba45a60 DM	353	av_assert0(!s->afq.remaining_delay);
	354	s->afq.frames->duration += duration;
	355	if (s->afq.frames->pts != AV_NOPTS_VALUE)
	356	s->afq.frames->pts -= duration;
	357	s->afq.remaining_samples += duration;
	358	}
	359	ff_af_queue_remove(&s->afq, duration, &avpkt->pts, &avpkt->duration);
	360	}
	361
	362	*got_packet_ptr = 1;
	363	return 0;
	364	}
	365
	366	AVCodec ff_libvorbis_encoder = {
	367	.name = "libvorbis",
	368	.long_name = NULL_IF_CONFIG_SMALL("libvorbis"),
	369	.type = AVMEDIA_TYPE_AUDIO,
	370	.id = AV_CODEC_ID_VORBIS,
	371	.priv_data_size = sizeof(LibvorbisEncContext),
	372	.init = libvorbis_encode_init,
	373	.encode2 = libvorbis_encode_frame,
	374	.close = libvorbis_encode_close,
f6fa7814	375	.capabilities = CODEC_CAP_DELAY \| CODEC_CAP_SMALL_LAST_FRAME,
2ba45a60 DM	376	.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
	377	AV_SAMPLE_FMT_NONE },
	378	.priv_class = &vorbis_class,
	379	.defaults = defaults,
	380	};