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ffmpeg.wasm-core
Commit ca5ab8cd authored by Diego Biurrun's avatar Diego Biurrun
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libvorbis: K&R reformatting cosmetics

parent c4db3446
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Showing with 79 additions and 73 deletions
libavcodec/libvorbis.c
View file @ ca5ab8cd
......@@ -37,63 +37,65 @@
#define OGGVORBIS_FRAME_SIZE 64
#define BUFFER_SIZE (1024*64)
#define BUFFER_SIZE (1024 * 64)
typedef struct OggVorbisContext {
AVClass *av_class;
vorbis_info vi ;
vorbis_dsp_state vd ;
vorbis_block vb ;
vorbis_info vi;
vorbis_dsp_state vd;
vorbis_block vb;
uint8_t buffer[BUFFER_SIZE];
int buffer_index;
int eof;
/* decoder */
vorbis_comment vc ;
vorbis_comment vc;
ogg_packet op;
double iblock;
} OggVorbisContext ;
} 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 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 AVClass class = { "libvorbis", av_default_item_name, options, LIBAVUTIL_VERSION_INT };
static av_cold int oggvorbis_init_encoder(vorbis_info *vi, AVCodecContext *avccontext) {
OggVorbisContext *context = avccontext->priv_data ;
static av_cold int oggvorbis_init_encoder(vorbis_info *vi, AVCodecContext *avccontext)
{
OggVorbisContext *context = avccontext->priv_data;
double cfreq;
if(avccontext->flags & CODEC_FLAG_QSCALE) {
if (avccontext->flags & CODEC_FLAG_QSCALE) {
/* variable bitrate */
if(vorbis_encode_setup_vbr(vi, avccontext->channels,
avccontext->sample_rate,
avccontext->global_quality / (float)FF_QP2LAMBDA / 10.0))
if (vorbis_encode_setup_vbr(vi, avccontext->channels,
avccontext->sample_rate,
avccontext->global_quality / (float)FF_QP2LAMBDA / 10.0))
return -1;
} else {
int minrate = avccontext->rc_min_rate > 0 ? avccontext->rc_min_rate : -1;
int maxrate = avccontext->rc_min_rate > 0 ? avccontext->rc_max_rate : -1;
/* constant bitrate */
if(vorbis_encode_setup_managed(vi, avccontext->channels,
avccontext->sample_rate, minrate, avccontext->bit_rate, maxrate))
if (vorbis_encode_setup_managed(vi, avccontext->channels,
avccontext->sample_rate, minrate,
avccontext->bit_rate, maxrate))
return -1;
/* variable bitrate by estimate, disable slow rate management */
if(minrate == -1 && maxrate == -1)
if(vorbis_encode_ctl(vi, OV_ECTL_RATEMANAGE2_SET, NULL))
if (minrate == -1 && maxrate == -1)
if (vorbis_encode_ctl(vi, OV_ECTL_RATEMANAGE2_SET, NULL))
return -1;
}
/* cutoff frequency */
if(avccontext->cutoff > 0) {
if (avccontext->cutoff > 0) {
cfreq = avccontext->cutoff / 1000.0;
if(vorbis_encode_ctl(vi, OV_ECTL_LOWPASS_SET, &cfreq))
if (vorbis_encode_ctl(vi, OV_ECTL_LOWPASS_SET, &cfreq))
return -1;
}
if(context->iblock){
if (context->iblock) {
vorbis_encode_ctl(vi, OV_ECTL_IBLOCK_SET, &context->iblock);
}
......@@ -101,35 +103,39 @@ static av_cold int oggvorbis_init_encoder(vorbis_info *vi, AVCodecContext *avcco
}
/* How many bytes are needed for a buffer of length 'l' */
static int xiph_len(int l) { return (1 + l / 255 + l); }
static int xiph_len(int l)
{
return (1 + l / 255 + l);
}
static av_cold int oggvorbis_encode_init(AVCodecContext *avccontext) {
OggVorbisContext *context = avccontext->priv_data ;
static av_cold int oggvorbis_encode_init(AVCodecContext *avccontext)
{
OggVorbisContext *context = avccontext->priv_data;
ogg_packet header, header_comm, header_code;
uint8_t *p;
unsigned int offset;
vorbis_info_init(&context->vi) ;
if(oggvorbis_init_encoder(&context->vi, avccontext) < 0) {
av_log(avccontext, AV_LOG_ERROR, "oggvorbis_encode_init: init_encoder failed\n") ;
return -1 ;
vorbis_info_init(&context->vi);
if (oggvorbis_init_encoder(&context->vi, avccontext) < 0) {
av_log(avccontext, AV_LOG_ERROR, "oggvorbis_encode_init: init_encoder failed\n");
return -1;
}
vorbis_analysis_init(&context->vd, &context->vi) ;
vorbis_block_init(&context->vd, &context->vb) ;
vorbis_analysis_init(&context->vd, &context->vi);
vorbis_block_init(&context->vd, &context->vb);
vorbis_comment_init(&context->vc);
vorbis_comment_add_tag(&context->vc, "encoder", LIBAVCODEC_IDENT) ;
vorbis_comment_add_tag(&context->vc, "encoder", LIBAVCODEC_IDENT);
vorbis_analysis_headerout(&context->vd, &context->vc, &header,
&header_comm, &header_code);
&header_comm, &header_code);
avccontext->extradata_size=
avccontext->extradata_size =
1 + xiph_len(header.bytes) + xiph_len(header_comm.bytes) +
header_code.bytes;
p = avccontext->extradata =
av_malloc(avccontext->extradata_size + FF_INPUT_BUFFER_PADDING_SIZE);
p[0] = 2;
offset = 1;
av_malloc(avccontext->extradata_size + FF_INPUT_BUFFER_PADDING_SIZE);
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);
......@@ -140,56 +146,57 @@ static av_cold int oggvorbis_encode_init(AVCodecContext *avccontext) {
offset += header_code.bytes;
assert(offset == avccontext->extradata_size);
/* vorbis_block_clear(&context->vb);
#if 0
vorbis_block_clear(&context->vb);
vorbis_dsp_clear(&context->vd);
vorbis_info_clear(&context->vi);*/
vorbis_info_clear(&context->vi);
#endif
vorbis_comment_clear(&context->vc);
avccontext->frame_size = OGGVORBIS_FRAME_SIZE ;
avccontext->frame_size = OGGVORBIS_FRAME_SIZE;
avccontext->coded_frame= avcodec_alloc_frame();
avccontext->coded_frame->key_frame= 1;
avccontext->coded_frame = avcodec_alloc_frame();
avccontext->coded_frame->key_frame = 1;
return 0 ;
return 0;
}
static int oggvorbis_encode_frame(AVCodecContext *avccontext,
unsigned char *packets,
int buf_size, void *data)
int buf_size, void *data)
{
OggVorbisContext *context = avccontext->priv_data ;
ogg_packet op ;
signed short *audio = data ;
OggVorbisContext *context = avccontext->priv_data;
ogg_packet op;
signed short *audio = data;
int l;
if(data) {
if (data) {
const int samples = avccontext->frame_size;
float **buffer ;
float **buffer;
int c, channels = context->vi.channels;
buffer = vorbis_analysis_buffer(&context->vd, samples) ;
buffer = vorbis_analysis_buffer(&context->vd, samples);
for (c = 0; c < channels; c++) {
int co = (channels > 8) ? c :
ff_vorbis_encoding_channel_layout_offsets[channels-1][c];
for(l = 0 ; l < samples ; l++)
buffer[c][l]=audio[l*channels+co]/32768.f;
ff_vorbis_encoding_channel_layout_offsets[channels - 1][c];
for (l = 0; l < samples; l++)
buffer[c][l] = audio[l * channels + co] / 32768.f;
}
vorbis_analysis_wrote(&context->vd, samples) ;
vorbis_analysis_wrote(&context->vd, samples);
} else {
if(!context->eof)
vorbis_analysis_wrote(&context->vd, 0) ;
if (!context->eof)
vorbis_analysis_wrote(&context->vd, 0);
context->eof = 1;
}
while(vorbis_analysis_blockout(&context->vd, &context->vb) == 1) {
while (vorbis_analysis_blockout(&context->vd, &context->vb) == 1) {
vorbis_analysis(&context->vb, NULL);
vorbis_bitrate_addblock(&context->vb) ;
vorbis_bitrate_addblock(&context->vb);
while(vorbis_bitrate_flushpacket(&context->vd, &op)) {
while (vorbis_bitrate_flushpacket(&context->vd, &op)) {
/* i'd love to say the following line is a hack, but sadly it's
* not, apparently the end of stream decision is in libogg. */
if(op.bytes==1 && op.e_o_s)
if (op.bytes == 1 && op.e_o_s)
continue;
if (context->buffer_index + sizeof(ogg_packet) + op.bytes > BUFFER_SIZE) {
av_log(avccontext, AV_LOG_ERROR, "libvorbis: buffer overflow.");
......@@ -203,13 +210,13 @@ static int oggvorbis_encode_frame(AVCodecContext *avccontext,
}
}
l=0;
if(context->buffer_index){
ogg_packet *op2= (ogg_packet*)context->buffer;
l = 0;
if (context->buffer_index) {
ogg_packet *op2 = (ogg_packet *)context->buffer;
op2->packet = context->buffer + sizeof(ogg_packet);
l= op2->bytes;
avccontext->coded_frame->pts= av_rescale_q(op2->granulepos, (AVRational){1, avccontext->sample_rate}, avccontext->time_base);
l = op2->bytes;
avccontext->coded_frame->pts = av_rescale_q(op2->granulepos, (AVRational) { 1, avccontext->sample_rate }, avccontext->time_base);
//FIXME we should reorder the user supplied pts and not assume that they are spaced by 1/sample_rate
if (l > buf_size) {
......@@ -226,12 +233,12 @@ static int oggvorbis_encode_frame(AVCodecContext *avccontext,
return l;
}
static av_cold int oggvorbis_encode_close(AVCodecContext *avccontext) {
OggVorbisContext *context = avccontext->priv_data ;
static av_cold int oggvorbis_encode_close(AVCodecContext *avccontext)
{
OggVorbisContext *context = avccontext->priv_data;
/* ogg_packet op ; */
vorbis_analysis_wrote(&context->vd, 0) ; /* notify vorbisenc this is EOF */
vorbis_analysis_wrote(&context->vd, 0); /* notify vorbisenc this is EOF */
vorbis_block_clear(&context->vb);
vorbis_dsp_clear(&context->vd);
......@@ -240,10 +247,9 @@ static av_cold int oggvorbis_encode_close(AVCodecContext *avccontext) {
av_freep(&avccontext->coded_frame);
av_freep(&avccontext->extradata);
return 0 ;
return 0;
}
AVCodec ff_libvorbis_encoder = {
.name = "libvorbis",
.type = AVMEDIA_TYPE_AUDIO,
......@@ -253,7 +259,7 @@ AVCodec ff_libvorbis_encoder = {
.encode = oggvorbis_encode_frame,
.close = oggvorbis_encode_close,
.capabilities = CODEC_CAP_DELAY,
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE },
.long_name = NULL_IF_CONFIG_SMALL("libvorbis Vorbis"),
.priv_class = &class,
};
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