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Merge remote-tracking branch 'qatar/master'

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* qatar/master:
  libvorbis: Give consistent names to all functions, structs, and defines

Conflicts:
	libavcodec/libvorbisenc.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer 2014-02-21 13:53:24 +01:00
commit 6e63867771
1 changed files with 19 additions and 19 deletions
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38
libavcodec/libvorbisenc.c
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@ -35,11 +35,11 @@
* an output packet will always start at the same point as one of the input
* packets.
*/
#define OGGVORBIS_FRAME_SIZE 64
#define LIBVORBIS_FRAME_SIZE 64
#define BUFFER_SIZE (1024 * 64)
typedef struct OggVorbisEncContext {
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 */
@ -51,10 +51,10 @@ typedef struct OggVorbisEncContext {
double iblock; /**< impulse block bias option */
VorbisParseContext vp; /**< parse context to get durations */
AudioFrameQueue afq; /**< frame queue for timestamps */
} OggVorbisEncContext;
} LibvorbisEncContext;
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 },
{ "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 }
};
@ -80,9 +80,9 @@ static int vorbis_error_to_averror(int ov_err)
}
}
static av_cold int oggvorbis_encode_init_internal(vorbis_info *vi, AVCodecContext *avctx)
static av_cold int libvorbis_setup(vorbis_info *vi, AVCodecContext *avctx)
{
OggVorbisEncContext *s = avctx->priv_data;
LibvorbisEncContext *s = avctx->priv_data;
double cfreq;
int ret;
@ -171,9 +171,9 @@ static int xiph_len(int l)
return 1 + l / 255 + l;
}
static av_cold int oggvorbis_encode_close(AVCodecContext *avctx)
static av_cold int libvorbis_encode_close(AVCodecContext *avctx)
{
OggVorbisEncContext *s = avctx->priv_data;
LibvorbisEncContext *s = avctx->priv_data;
/* notify vorbisenc this is EOF */
if (s->dsp_initialized)
@ -190,16 +190,16 @@ static av_cold int oggvorbis_encode_close(AVCodecContext *avctx)
return 0;
}
static av_cold int oggvorbis_encode_init(AVCodecContext *avctx)
static av_cold int libvorbis_encode_init(AVCodecContext *avctx)
{
OggVorbisEncContext *s = avctx->priv_data;
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 = oggvorbis_encode_init_internal(&s->vi, avctx))) {
if ((ret = libvorbis_setup(&s->vi, avctx))) {
av_log(avctx, AV_LOG_ERROR, "encoder setup failed\n");
goto error;
}
@ -253,7 +253,7 @@ static av_cold int oggvorbis_encode_init(AVCodecContext *avctx)
vorbis_comment_clear(&s->vc);
avctx->frame_size = OGGVORBIS_FRAME_SIZE;
avctx->frame_size = LIBVORBIS_FRAME_SIZE;
ff_af_queue_init(avctx, &s->afq);
s->pkt_fifo = av_fifo_alloc(BUFFER_SIZE);
@ -264,14 +264,14 @@ static av_cold int oggvorbis_encode_init(AVCodecContext *avctx)
return 0;
error:
oggvorbis_encode_close(avctx);
libvorbis_encode_close(avctx);
return ret;
}
static int oggvorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
static int libvorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
OggVorbisEncContext *s = avctx->priv_data;
LibvorbisEncContext *s = avctx->priv_data;
ogg_packet op;
int ret, duration;
@ -364,10 +364,10 @@ AVCodec ff_libvorbis_encoder = {
.long_name = NULL_IF_CONFIG_SMALL("libvorbis"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_VORBIS,
.priv_data_size = sizeof(OggVorbisEncContext),
.init = oggvorbis_encode_init,
.encode2 = oggvorbis_encode_frame,
.close = oggvorbis_encode_close,
.priv_data_size = sizeof(LibvorbisEncContext),
.init = libvorbis_encode_init,
.encode2 = libvorbis_encode_frame,
.close = libvorbis_encode_close,
.capabilities = CODEC_CAP_DELAY,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE },