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https://github.com/FFmpeg/FFmpeg.git
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libvorbis: K&R reformatting cosmetics
This commit is contained in:
Diego Biurrun
parent
c4db344664
commit
ca5ab8cd21
@ -37,63 +37,65 @@
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#define OGGVORBIS_FRAME_SIZE 64
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#define BUFFER_SIZE (1024*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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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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vorbis_comment vc;
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ogg_packet op;
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double iblock;
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} OggVorbisContext ;
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} OggVorbisContext;
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static const AVOption options[]={
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{"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},
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{NULL}
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static const AVOption options[] = {
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{ "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 },
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{ NULL }
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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 av_cold int oggvorbis_init_encoder(vorbis_info *vi, AVCodecContext *avccontext) {
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OggVorbisContext *context = avccontext->priv_data ;
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static av_cold int oggvorbis_init_encoder(vorbis_info *vi, AVCodecContext *avccontext)
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{
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OggVorbisContext *context = avccontext->priv_data;
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double cfreq;
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if(avccontext->flags & CODEC_FLAG_QSCALE) {
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if (avccontext->flags & CODEC_FLAG_QSCALE) {
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/* variable bitrate */
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if(vorbis_encode_setup_vbr(vi, avccontext->channels,
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avccontext->sample_rate,
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avccontext->global_quality / (float)FF_QP2LAMBDA / 10.0))
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if (vorbis_encode_setup_vbr(vi, avccontext->channels,
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avccontext->sample_rate,
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avccontext->global_quality / (float)FF_QP2LAMBDA / 10.0))
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return -1;
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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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/* constant bitrate */
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if(vorbis_encode_setup_managed(vi, avccontext->channels,
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avccontext->sample_rate, minrate, avccontext->bit_rate, maxrate))
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if (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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return -1;
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/* variable bitrate by estimate, disable slow rate management */
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if(minrate == -1 && maxrate == -1)
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if(vorbis_encode_ctl(vi, OV_ECTL_RATEMANAGE2_SET, NULL))
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if (minrate == -1 && maxrate == -1)
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if (vorbis_encode_ctl(vi, OV_ECTL_RATEMANAGE2_SET, NULL))
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return -1;
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}
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/* cutoff frequency */
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if(avccontext->cutoff > 0) {
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if (avccontext->cutoff > 0) {
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cfreq = avccontext->cutoff / 1000.0;
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if(vorbis_encode_ctl(vi, OV_ECTL_LOWPASS_SET, &cfreq))
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if (vorbis_encode_ctl(vi, OV_ECTL_LOWPASS_SET, &cfreq))
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return -1;
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}
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if(context->iblock){
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if (context->iblock) {
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vorbis_encode_ctl(vi, OV_ECTL_IBLOCK_SET, &context->iblock);
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}
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@ -101,35 +103,39 @@ static av_cold int oggvorbis_init_encoder(vorbis_info *vi, AVCodecContext *avcco
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}
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/* How many bytes are needed for a buffer of length 'l' */
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static int xiph_len(int l) { return (1 + l / 255 + l); }
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static int xiph_len(int l)
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{
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return (1 + l / 255 + l);
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}
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static av_cold int oggvorbis_encode_init(AVCodecContext *avccontext) {
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OggVorbisContext *context = avccontext->priv_data ;
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static av_cold int oggvorbis_encode_init(AVCodecContext *avccontext)
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{
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OggVorbisContext *context = avccontext->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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vorbis_info_init(&context->vi) ;
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if(oggvorbis_init_encoder(&context->vi, avccontext) < 0) {
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av_log(avccontext, AV_LOG_ERROR, "oggvorbis_encode_init: init_encoder failed\n") ;
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return -1 ;
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vorbis_info_init(&context->vi);
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if (oggvorbis_init_encoder(&context->vi, avccontext) < 0) {
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av_log(avccontext, AV_LOG_ERROR, "oggvorbis_encode_init: init_encoder failed\n");
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return -1;
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}
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vorbis_analysis_init(&context->vd, &context->vi) ;
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vorbis_block_init(&context->vd, &context->vb) ;
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vorbis_analysis_init(&context->vd, &context->vi);
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vorbis_block_init(&context->vd, &context->vb);
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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_add_tag(&context->vc, "encoder", LIBAVCODEC_IDENT);
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vorbis_analysis_headerout(&context->vd, &context->vc, &header,
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&header_comm, &header_code);
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&header_comm, &header_code);
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avccontext->extradata_size=
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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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p[0] = 2;
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offset = 1;
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av_malloc(avccontext->extradata_size + FF_INPUT_BUFFER_PADDING_SIZE);
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p[0] = 2;
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offset = 1;
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offset += av_xiphlacing(&p[offset], header.bytes);
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offset += av_xiphlacing(&p[offset], header_comm.bytes);
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memcpy(&p[offset], header.packet, header.bytes);
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@ -140,56 +146,57 @@ static av_cold int oggvorbis_encode_init(AVCodecContext *avccontext) {
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offset += header_code.bytes;
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assert(offset == avccontext->extradata_size);
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/* vorbis_block_clear(&context->vb);
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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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vorbis_info_clear(&context->vi);
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#endif
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vorbis_comment_clear(&context->vc);
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avccontext->frame_size = OGGVORBIS_FRAME_SIZE ;
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avccontext->frame_size = OGGVORBIS_FRAME_SIZE;
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avccontext->coded_frame= avcodec_alloc_frame();
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avccontext->coded_frame->key_frame= 1;
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avccontext->coded_frame = avcodec_alloc_frame();
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avccontext->coded_frame->key_frame = 1;
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return 0 ;
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return 0;
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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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int buf_size, void *data)
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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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ogg_packet op ;
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signed short *audio = data ;
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OggVorbisContext *context = avccontext->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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if(data) {
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if (data) {
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const int samples = avccontext->frame_size;
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float **buffer ;
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float **buffer;
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int c, channels = context->vi.channels;
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buffer = vorbis_analysis_buffer(&context->vd, samples) ;
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buffer = vorbis_analysis_buffer(&context->vd, samples);
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for (c = 0; c < channels; c++) {
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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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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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}
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vorbis_analysis_wrote(&context->vd, samples) ;
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vorbis_analysis_wrote(&context->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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if (!context->eof)
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vorbis_analysis_wrote(&context->vd, 0);
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context->eof = 1;
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}
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while(vorbis_analysis_blockout(&context->vd, &context->vb) == 1) {
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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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vorbis_bitrate_addblock(&context->vb);
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while(vorbis_bitrate_flushpacket(&context->vd, &op)) {
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while (vorbis_bitrate_flushpacket(&context->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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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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@ -203,13 +210,13 @@ static int oggvorbis_encode_frame(AVCodecContext *avccontext,
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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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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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l= op2->bytes;
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avccontext->coded_frame->pts= av_rescale_q(op2->granulepos, (AVRational){1, avccontext->sample_rate}, avccontext->time_base);
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l = op2->bytes;
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avccontext->coded_frame->pts = av_rescale_q(op2->granulepos, (AVRational) { 1, avccontext->sample_rate }, avccontext->time_base);
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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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@ -226,12 +233,12 @@ static int oggvorbis_encode_frame(AVCodecContext *avccontext,
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return l;
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}
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static av_cold int oggvorbis_encode_close(AVCodecContext *avccontext) {
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OggVorbisContext *context = avccontext->priv_data ;
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static av_cold int oggvorbis_encode_close(AVCodecContext *avccontext)
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{
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OggVorbisContext *context = avccontext->priv_data;
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/* ogg_packet op ; */
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vorbis_analysis_wrote(&context->vd, 0) ; /* notify vorbisenc this is EOF */
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vorbis_analysis_wrote(&context->vd, 0); /* notify vorbisenc this is EOF */
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vorbis_block_clear(&context->vb);
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vorbis_dsp_clear(&context->vd);
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@ -240,10 +247,9 @@ static av_cold int oggvorbis_encode_close(AVCodecContext *avccontext) {
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av_freep(&avccontext->coded_frame);
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av_freep(&avccontext->extradata);
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return 0 ;
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return 0;
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}
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AVCodec ff_libvorbis_encoder = {
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.name = "libvorbis",
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.type = AVMEDIA_TYPE_AUDIO,
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@ -253,7 +259,7 @@ 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, 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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