libvorbis: cosmetics: renaming/pretty-printing/comments/unused code
This commit is contained in:
Justin Ruggles
parent
4e99501f62
commit
eb35ef2932
@ -20,7 +20,7 @@
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/**
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* @file
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* Ogg Vorbis codec support via libvorbisenc.
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* Vorbis encoding support via libvorbisenc.
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* @author Mark Hills <mark@pogo.org.uk>
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*/
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@ -35,24 +35,26 @@
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#undef NDEBUG
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#include <assert.h>
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/* Number of samples the user should send in each call.
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* This value is used because it is the LCD of all possible frame sizes, so
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* an output packet will always start at the same point as one of the input
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* packets.
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*/
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#define OGGVORBIS_FRAME_SIZE 64
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#define BUFFER_SIZE (1024 * 64)
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typedef struct OggVorbisContext {
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AVClass *av_class;
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vorbis_info vi;
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vorbis_dsp_state vd;
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vorbis_block vb;
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uint8_t buffer[BUFFER_SIZE];
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int buffer_index;
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int eof;
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/* decoder */
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vorbis_comment vc;
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ogg_packet op;
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double iblock;
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AVClass *av_class; /**< class for AVOptions */
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vorbis_info vi; /**< vorbis_info used during init */
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vorbis_dsp_state vd; /**< DSP state used for analysis */
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vorbis_block vb; /**< vorbis_block used for analysis */
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uint8_t buffer[BUFFER_SIZE]; /**< output packet buffer */
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int buffer_index; /**< current buffer position */
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int eof; /**< end-of-file flag */
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vorbis_comment vc; /**< VorbisComment info */
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ogg_packet op; /**< ogg packet */
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double iblock; /**< impulse block bias option */
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} OggVorbisContext;
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static const AVOption options[] = {
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@ -61,6 +63,7 @@ static const AVOption options[] = {
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};
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static const AVClass class = { "libvorbis", av_default_item_name, options, LIBAVUTIL_VERSION_INT };
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static int vorbis_error_to_averror(int ov_err)
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{
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switch (ov_err) {
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@ -71,27 +74,31 @@ static int vorbis_error_to_averror(int ov_err)
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}
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}
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static av_cold int oggvorbis_init_encoder(vorbis_info *vi, AVCodecContext *avccontext)
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static av_cold int oggvorbis_init_encoder(vorbis_info *vi,
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AVCodecContext *avctx)
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{
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OggVorbisContext *context = avccontext->priv_data;
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OggVorbisContext *s = avctx->priv_data;
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double cfreq;
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int ret;
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if (avccontext->flags & CODEC_FLAG_QSCALE) {
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/* variable bitrate */
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float q = avccontext->global_quality / (float)FF_QP2LAMBDA;
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if ((ret = vorbis_encode_setup_vbr(vi, avccontext->channels,
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avccontext->sample_rate,
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if (avctx->flags & CODEC_FLAG_QSCALE) {
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/* variable bitrate
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* NOTE: we use the oggenc range of -1 to 10 for global_quality for
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* user convenience, but libvorbis uses -0.1 to 1.0
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*/
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float q = avctx->global_quality / (float)FF_QP2LAMBDA;
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if ((ret = vorbis_encode_setup_vbr(vi, avctx->channels,
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avctx->sample_rate,
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q / 10.0)))
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goto error;
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} else {
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int minrate = avccontext->rc_min_rate > 0 ? avccontext->rc_min_rate : -1;
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int maxrate = avccontext->rc_min_rate > 0 ? avccontext->rc_max_rate : -1;
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int minrate = avctx->rc_min_rate > 0 ? avctx->rc_min_rate : -1;
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int maxrate = avctx->rc_min_rate > 0 ? avctx->rc_max_rate : -1;
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/* constant bitrate */
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if ((ret = vorbis_encode_setup_managed(vi, avccontext->channels,
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avccontext->sample_rate, minrate,
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avccontext->bit_rate, maxrate)))
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/* average bitrate */
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if ((ret = vorbis_encode_setup_managed(vi, avctx->channels,
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avctx->sample_rate, minrate,
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avctx->bit_rate, maxrate)))
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goto error;
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/* variable bitrate by estimate, disable slow rate management */
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@ -101,14 +108,15 @@ static av_cold int oggvorbis_init_encoder(vorbis_info *vi, AVCodecContext *avcco
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}
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/* cutoff frequency */
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if (avccontext->cutoff > 0) {
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cfreq = avccontext->cutoff / 1000.0;
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if (avctx->cutoff > 0) {
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cfreq = avctx->cutoff / 1000.0;
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if ((ret = vorbis_encode_ctl(vi, OV_ECTL_LOWPASS_SET, &cfreq)))
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goto error;
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}
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if (context->iblock) {
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if ((ret = vorbis_encode_ctl(vi, OV_ECTL_IBLOCK_SET, &context->iblock)))
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/* impulse block bias */
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if (s->iblock) {
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if ((ret = vorbis_encode_ctl(vi, OV_ECTL_IBLOCK_SET, &s->iblock)))
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goto error;
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}
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@ -126,59 +134,59 @@ static int xiph_len(int l)
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return 1 + l / 255 + l;
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}
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static av_cold int oggvorbis_encode_close(AVCodecContext *avccontext)
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static av_cold int oggvorbis_encode_close(AVCodecContext *avctx)
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{
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OggVorbisContext *context = avccontext->priv_data;
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/* ogg_packet op ; */
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OggVorbisContext *s = avctx->priv_data;
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vorbis_analysis_wrote(&context->vd, 0); /* notify vorbisenc this is EOF */
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/* notify vorbisenc this is EOF */
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vorbis_analysis_wrote(&s->vd, 0);
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vorbis_block_clear(&context->vb);
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vorbis_dsp_clear(&context->vd);
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vorbis_info_clear(&context->vi);
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vorbis_block_clear(&s->vb);
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vorbis_dsp_clear(&s->vd);
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vorbis_info_clear(&s->vi);
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av_freep(&avccontext->coded_frame);
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av_freep(&avccontext->extradata);
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av_freep(&avctx->coded_frame);
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av_freep(&avctx->extradata);
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return 0;
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}
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static av_cold int oggvorbis_encode_init(AVCodecContext *avccontext)
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static av_cold int oggvorbis_encode_init(AVCodecContext *avctx)
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{
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OggVorbisContext *context = avccontext->priv_data;
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OggVorbisContext *s = avctx->priv_data;
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ogg_packet header, header_comm, header_code;
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uint8_t *p;
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unsigned int offset;
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int ret;
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vorbis_info_init(&context->vi);
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if ((ret = oggvorbis_init_encoder(&context->vi, avccontext))) {
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av_log(avccontext, AV_LOG_ERROR, "oggvorbis_encode_init: init_encoder failed\n");
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vorbis_info_init(&s->vi);
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if ((ret = oggvorbis_init_encoder(&s->vi, avctx))) {
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av_log(avctx, AV_LOG_ERROR, "oggvorbis_encode_init: init_encoder failed\n");
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goto error;
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}
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if ((ret = vorbis_analysis_init(&context->vd, &context->vi))) {
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if ((ret = vorbis_analysis_init(&s->vd, &s->vi))) {
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ret = vorbis_error_to_averror(ret);
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goto error;
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}
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if ((ret = vorbis_block_init(&context->vd, &context->vb))) {
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if ((ret = vorbis_block_init(&s->vd, &s->vb))) {
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ret = vorbis_error_to_averror(ret);
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goto error;
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}
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vorbis_comment_init(&context->vc);
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vorbis_comment_add_tag(&context->vc, "encoder", LIBAVCODEC_IDENT);
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vorbis_comment_init(&s->vc);
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vorbis_comment_add_tag(&s->vc, "encoder", LIBAVCODEC_IDENT);
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if ((ret = vorbis_analysis_headerout(&context->vd, &context->vc, &header,
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&header_comm, &header_code))) {
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if ((ret = vorbis_analysis_headerout(&s->vd, &s->vc, &header, &header_comm,
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&header_code))) {
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ret = vorbis_error_to_averror(ret);
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goto error;
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}
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avccontext->extradata_size =
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1 + xiph_len(header.bytes) + xiph_len(header_comm.bytes) +
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header_code.bytes;
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p = avccontext->extradata =
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av_malloc(avccontext->extradata_size + FF_INPUT_BUFFER_PADDING_SIZE);
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avctx->extradata_size = 1 + xiph_len(header.bytes) +
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xiph_len(header_comm.bytes) +
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header_code.bytes;
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p = avctx->extradata = av_malloc(avctx->extradata_size +
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FF_INPUT_BUFFER_PADDING_SIZE);
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if (!p) {
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ret = AVERROR(ENOMEM);
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goto error;
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@ -193,100 +201,97 @@ static av_cold int oggvorbis_encode_init(AVCodecContext *avccontext)
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offset += header_comm.bytes;
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memcpy(&p[offset], header_code.packet, header_code.bytes);
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offset += header_code.bytes;
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assert(offset == avccontext->extradata_size);
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assert(offset == avctx->extradata_size);
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#if 0
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vorbis_block_clear(&context->vb);
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vorbis_dsp_clear(&context->vd);
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vorbis_info_clear(&context->vi);
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#endif
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vorbis_comment_clear(&context->vc);
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vorbis_comment_clear(&s->vc);
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avccontext->frame_size = OGGVORBIS_FRAME_SIZE;
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avctx->frame_size = OGGVORBIS_FRAME_SIZE;
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avccontext->coded_frame = avcodec_alloc_frame();
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if (!avccontext->coded_frame) {
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avctx->coded_frame = avcodec_alloc_frame();
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if (!avctx->coded_frame) {
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ret = AVERROR(ENOMEM);
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goto error;
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}
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return 0;
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error:
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oggvorbis_encode_close(avccontext);
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oggvorbis_encode_close(avctx);
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return ret;
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}
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static int oggvorbis_encode_frame(AVCodecContext *avccontext,
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unsigned char *packets,
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static int oggvorbis_encode_frame(AVCodecContext *avctx, unsigned char *packets,
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int buf_size, void *data)
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{
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OggVorbisContext *context = avccontext->priv_data;
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OggVorbisContext *s = avctx->priv_data;
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ogg_packet op;
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signed short *audio = data;
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int l;
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int pkt_size;
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/* send samples to libvorbis */
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if (data) {
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const int samples = avccontext->frame_size;
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const int samples = avctx->frame_size;
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float **buffer;
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int c, channels = context->vi.channels;
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int c, channels = s->vi.channels;
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buffer = vorbis_analysis_buffer(&context->vd, samples);
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buffer = vorbis_analysis_buffer(&s->vd, samples);
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for (c = 0; c < channels; c++) {
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int i;
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int co = (channels > 8) ? c :
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ff_vorbis_encoding_channel_layout_offsets[channels - 1][c];
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for (l = 0; l < samples; l++)
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buffer[c][l] = audio[l * channels + co] / 32768.f;
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for (i = 0; i < samples; i++)
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buffer[c][i] = audio[i * channels + co] / 32768.f;
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}
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vorbis_analysis_wrote(&context->vd, samples);
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vorbis_analysis_wrote(&s->vd, samples);
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} else {
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if (!context->eof)
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vorbis_analysis_wrote(&context->vd, 0);
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context->eof = 1;
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if (!s->eof)
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vorbis_analysis_wrote(&s->vd, 0);
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s->eof = 1;
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}
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while (vorbis_analysis_blockout(&context->vd, &context->vb) == 1) {
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vorbis_analysis(&context->vb, NULL);
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vorbis_bitrate_addblock(&context->vb);
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/* retrieve available packets from libvorbis */
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while (vorbis_analysis_blockout(&s->vd, &s->vb) == 1) {
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vorbis_analysis(&s->vb, NULL);
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vorbis_bitrate_addblock(&s->vb);
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while (vorbis_bitrate_flushpacket(&context->vd, &op)) {
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/* add any available packets to the output packet buffer */
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while (vorbis_bitrate_flushpacket(&s->vd, &op)) {
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/* i'd love to say the following line is a hack, but sadly it's
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* not, apparently the end of stream decision is in libogg. */
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if (op.bytes == 1 && op.e_o_s)
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continue;
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if (context->buffer_index + sizeof(ogg_packet) + op.bytes > BUFFER_SIZE) {
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av_log(avccontext, AV_LOG_ERROR, "libvorbis: buffer overflow.");
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if (s->buffer_index + sizeof(ogg_packet) + op.bytes > BUFFER_SIZE) {
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av_log(avctx, AV_LOG_ERROR, "libvorbis: buffer overflow.");
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return -1;
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}
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memcpy(context->buffer + context->buffer_index, &op, sizeof(ogg_packet));
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context->buffer_index += sizeof(ogg_packet);
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memcpy(context->buffer + context->buffer_index, op.packet, op.bytes);
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context->buffer_index += op.bytes;
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// av_log(avccontext, AV_LOG_DEBUG, "e%d / %d\n", context->buffer_index, op.bytes);
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memcpy(s->buffer + s->buffer_index, &op, sizeof(ogg_packet));
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s->buffer_index += sizeof(ogg_packet);
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memcpy(s->buffer + s->buffer_index, op.packet, op.bytes);
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s->buffer_index += op.bytes;
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}
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}
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l = 0;
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if (context->buffer_index) {
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ogg_packet *op2 = (ogg_packet *)context->buffer;
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op2->packet = context->buffer + sizeof(ogg_packet);
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/* output then next packet from the output buffer, if available */
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pkt_size = 0;
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if (s->buffer_index) {
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ogg_packet *op2 = (ogg_packet *)s->buffer;
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op2->packet = s->buffer + sizeof(ogg_packet);
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l = op2->bytes;
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avccontext->coded_frame->pts = ff_samples_to_time_base(avccontext,
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op2->granulepos);
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//FIXME we should reorder the user supplied pts and not assume that they are spaced by 1/sample_rate
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if (l > buf_size) {
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av_log(avccontext, AV_LOG_ERROR, "libvorbis: buffer overflow.");
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pkt_size = op2->bytes;
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// FIXME: we should use the user-supplied pts and duration
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avctx->coded_frame->pts = ff_samples_to_time_base(avctx,
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op2->granulepos);
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if (pkt_size > buf_size) {
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av_log(avctx, AV_LOG_ERROR, "libvorbis: buffer overflow.");
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return -1;
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}
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memcpy(packets, op2->packet, l);
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context->buffer_index -= l + sizeof(ogg_packet);
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memmove(context->buffer, context->buffer + l + sizeof(ogg_packet), context->buffer_index);
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// av_log(avccontext, AV_LOG_DEBUG, "E%d\n", l);
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memcpy(packets, op2->packet, pkt_size);
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s->buffer_index -= pkt_size + sizeof(ogg_packet);
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memmove(s->buffer, s->buffer + pkt_size + sizeof(ogg_packet),
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s->buffer_index);
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}
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return l;
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return pkt_size;
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}
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AVCodec ff_libvorbis_encoder = {
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@ -298,7 +303,8 @@ AVCodec ff_libvorbis_encoder = {
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.encode = oggvorbis_encode_frame,
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.close = oggvorbis_encode_close,
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.capabilities = CODEC_CAP_DELAY,
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.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE },
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.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16,
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AV_SAMPLE_FMT_NONE },
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.long_name = NULL_IF_CONFIG_SMALL("libvorbis Vorbis"),
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.priv_class = &class,
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};
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