mirror of https://git.ffmpeg.org/ffmpeg.git
Merge commit 'b70d7a4ac72d23f3448f3b08b770fdf5f57de222'
* commit 'b70d7a4ac72d23f3448f3b08b770fdf5f57de222': lavc: add a native Opus decoder. Conflicts: Changelog configure libavcodec/version.h Fate tests pass with both avresample as well as swresample based opus decoder, but are disabled (reference files are very large so i want to think a day or 2 about if theres an alternative or if they could be avoided, they also dont match the official samples) Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer
commit
2c7d3ecfc9
|
|
@ -23,6 +23,7 @@ version <next>:
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|||
- support for decoding through DXVA2 in ffmpeg
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- libbs2b-based stereo-to-binaural audio filter
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- libx264 reference frames count limiting depending on level
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- native Opus decoder
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version 2.2:
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|
|
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|||
|
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@ -2110,6 +2110,7 @@ nellymoser_decoder_select="mdct sinewin"
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nellymoser_encoder_select="audio_frame_queue mdct sinewin"
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nuv_decoder_select="dsputil lzo"
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on2avc_decoder_select="mdct"
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opus_decoder_deps="avresample"
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png_decoder_select="zlib"
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png_encoder_select="dsputil zlib"
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prores_decoder_select="dsputil"
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@ -5139,6 +5140,8 @@ enabled subtitles_filter && prepend avfilter_deps "avformat avcodec"
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enabled lavfi_indev && prepend avdevice_deps "avfilter"
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enabled opus_decoder && prepend avcodec_deps "avresample"
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expand_deps(){
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lib_deps=${1}_deps
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eval "deps=\$$lib_deps"
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|
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@ -332,6 +332,9 @@ OBJS-$(CONFIG_NELLYMOSER_DECODER) += nellymoserdec.o nellymoser.o
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OBJS-$(CONFIG_NELLYMOSER_ENCODER) += nellymoserenc.o nellymoser.o
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OBJS-$(CONFIG_NUV_DECODER) += nuv.o rtjpeg.o
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OBJS-$(CONFIG_ON2AVC_DECODER) += on2avc.o on2avcdata.o
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OBJS-$(CONFIG_OPUS_DECODER) += opusdec.o opus.o opus_celt.o \
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opus_imdct.o opus_silk.o \
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vorbis_data.o
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OBJS-$(CONFIG_PAF_VIDEO_DECODER) += paf.o
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OBJS-$(CONFIG_PAF_AUDIO_DECODER) += paf.o
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OBJS-$(CONFIG_PAM_DECODER) += pnmdec.o pnm.o
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@ -779,6 +782,7 @@ OBJS-$(CONFIG_MPEGAUDIO_PARSER) += mpegaudio_parser.o \
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mpegaudiodecheader.o mpegaudiodata.o
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OBJS-$(CONFIG_MPEGVIDEO_PARSER) += mpegvideo_parser.o \
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mpeg12.o mpeg12data.o
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OBJS-$(CONFIG_OPUS_PARSER) += opus_parser.o opus.o vorbis_data.o
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OBJS-$(CONFIG_PNG_PARSER) += png_parser.o
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OBJS-$(CONFIG_PNM_PARSER) += pnm_parser.o pnm.o
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OBJS-$(CONFIG_RV30_PARSER) += rv34_parser.o
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|
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@ -372,6 +372,7 @@ void avcodec_register_all(void)
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REGISTER_DECODER(MPC8, mpc8);
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REGISTER_ENCDEC (NELLYMOSER, nellymoser);
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REGISTER_DECODER(ON2AVC, on2avc);
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REGISTER_DECODER(OPUS, opus);
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REGISTER_DECODER(PAF_AUDIO, paf_audio);
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REGISTER_DECODER(QCELP, qcelp);
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REGISTER_DECODER(QDM2, qdm2);
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|
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@ -562,6 +563,7 @@ void avcodec_register_all(void)
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REGISTER_PARSER(MPEG4VIDEO, mpeg4video);
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REGISTER_PARSER(MPEGAUDIO, mpegaudio);
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REGISTER_PARSER(MPEGVIDEO, mpegvideo);
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REGISTER_PARSER(OPUS, opus);
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REGISTER_PARSER(PNG, png);
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REGISTER_PARSER(PNM, pnm);
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REGISTER_PARSER(RV30, rv30);
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|
|
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@ -0,0 +1,428 @@
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/*
|
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* Copyright (c) 2012 Andrew D'Addesio
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* Copyright (c) 2013-2014 Mozilla Corporation
|
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*
|
||||
* This file is part of FFmpeg.
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*
|
||||
* 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
|
||||
*/
|
||||
|
||||
/**
|
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* @file
|
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* Opus decoder/parser shared code
|
||||
*/
|
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|
||||
#include <stdint.h>
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|
||||
#include "libavutil/error.h"
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|
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#include "opus.h"
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#include "vorbis.h"
|
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|
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static const uint16_t opus_frame_duration[32] = {
|
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480, 960, 1920, 2880,
|
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480, 960, 1920, 2880,
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480, 960, 1920, 2880,
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480, 960,
|
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480, 960,
|
||||
120, 240, 480, 960,
|
||||
120, 240, 480, 960,
|
||||
120, 240, 480, 960,
|
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120, 240, 480, 960,
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||||
};
|
||||
|
||||
/**
|
||||
* Read a 1- or 2-byte frame length
|
||||
*/
|
||||
static inline int xiph_lacing_16bit(const uint8_t **ptr, const uint8_t *end)
|
||||
{
|
||||
int val;
|
||||
|
||||
if (*ptr >= end)
|
||||
return AVERROR_INVALIDDATA;
|
||||
val = *(*ptr)++;
|
||||
if (val >= 252) {
|
||||
if (*ptr >= end)
|
||||
return AVERROR_INVALIDDATA;
|
||||
val += 4 * *(*ptr)++;
|
||||
}
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return val;
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||||
}
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||||
|
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/**
|
||||
* Read a multi-byte length (used for code 3 packet padding size)
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*/
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static inline int xiph_lacing_full(const uint8_t **ptr, const uint8_t *end)
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{
|
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int val = 0;
|
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int next;
|
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|
||||
while (1) {
|
||||
if (*ptr >= end || val > INT_MAX - 254)
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||||
return AVERROR_INVALIDDATA;
|
||||
next = *(*ptr)++;
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||||
val += next;
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if (next < 255)
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||||
break;
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||||
else
|
||||
val--;
|
||||
}
|
||||
return val;
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||||
}
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||||
|
||||
/**
|
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* Parse Opus packet info from raw packet data
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||||
*/
|
||||
int ff_opus_parse_packet(OpusPacket *pkt, const uint8_t *buf, int buf_size,
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int self_delimiting)
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{
|
||||
const uint8_t *ptr = buf;
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||||
const uint8_t *end = buf + buf_size;
|
||||
int padding = 0;
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||||
int frame_bytes, i;
|
||||
|
||||
if (buf_size < 1)
|
||||
goto fail;
|
||||
|
||||
/* TOC byte */
|
||||
i = *ptr++;
|
||||
pkt->code = (i ) & 0x3;
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||||
pkt->stereo = (i >> 2) & 0x1;
|
||||
pkt->config = (i >> 3) & 0x1F;
|
||||
|
||||
/* code 2 and code 3 packets have at least 1 byte after the TOC */
|
||||
if (pkt->code >= 2 && buf_size < 2)
|
||||
goto fail;
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||||
|
||||
switch (pkt->code) {
|
||||
case 0:
|
||||
/* 1 frame */
|
||||
pkt->frame_count = 1;
|
||||
pkt->vbr = 0;
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||||
|
||||
if (self_delimiting) {
|
||||
int len = xiph_lacing_16bit(&ptr, end);
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if (len < 0 || len > end - ptr)
|
||||
goto fail;
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||||
end = ptr + len;
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||||
buf_size = end - buf;
|
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}
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|
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frame_bytes = end - ptr;
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||||
if (frame_bytes > MAX_FRAME_SIZE)
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goto fail;
|
||||
pkt->frame_offset[0] = ptr - buf;
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||||
pkt->frame_size[0] = frame_bytes;
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||||
break;
|
||||
case 1:
|
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/* 2 frames, equal size */
|
||||
pkt->frame_count = 2;
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||||
pkt->vbr = 0;
|
||||
|
||||
if (self_delimiting) {
|
||||
int len = xiph_lacing_16bit(&ptr, end);
|
||||
if (len < 0 || 2 * len > end - ptr)
|
||||
goto fail;
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||||
end = ptr + 2 * len;
|
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buf_size = end - buf;
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||||
}
|
||||
|
||||
frame_bytes = end - ptr;
|
||||
if (frame_bytes & 1 || frame_bytes >> 1 > MAX_FRAME_SIZE)
|
||||
goto fail;
|
||||
pkt->frame_offset[0] = ptr - buf;
|
||||
pkt->frame_size[0] = frame_bytes >> 1;
|
||||
pkt->frame_offset[1] = pkt->frame_offset[0] + pkt->frame_size[0];
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||||
pkt->frame_size[1] = frame_bytes >> 1;
|
||||
break;
|
||||
case 2:
|
||||
/* 2 frames, different sizes */
|
||||
pkt->frame_count = 2;
|
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pkt->vbr = 1;
|
||||
|
||||
/* read 1st frame size */
|
||||
frame_bytes = xiph_lacing_16bit(&ptr, end);
|
||||
if (frame_bytes < 0)
|
||||
goto fail;
|
||||
|
||||
if (self_delimiting) {
|
||||
int len = xiph_lacing_16bit(&ptr, end);
|
||||
if (len < 0 || len + frame_bytes > end - ptr)
|
||||
goto fail;
|
||||
end = ptr + frame_bytes + len;
|
||||
buf_size = end - buf;
|
||||
}
|
||||
|
||||
pkt->frame_offset[0] = ptr - buf;
|
||||
pkt->frame_size[0] = frame_bytes;
|
||||
|
||||
/* calculate 2nd frame size */
|
||||
frame_bytes = end - ptr - pkt->frame_size[0];
|
||||
if (frame_bytes < 0 || frame_bytes > MAX_FRAME_SIZE)
|
||||
goto fail;
|
||||
pkt->frame_offset[1] = pkt->frame_offset[0] + pkt->frame_size[0];
|
||||
pkt->frame_size[1] = frame_bytes;
|
||||
break;
|
||||
case 3:
|
||||
/* 1 to 48 frames, can be different sizes */
|
||||
i = *ptr++;
|
||||
pkt->frame_count = (i ) & 0x3F;
|
||||
padding = (i >> 6) & 0x01;
|
||||
pkt->vbr = (i >> 7) & 0x01;
|
||||
|
||||
if (pkt->frame_count == 0 || pkt->frame_count > MAX_FRAMES)
|
||||
goto fail;
|
||||
|
||||
/* read padding size */
|
||||
if (padding) {
|
||||
padding = xiph_lacing_full(&ptr, end);
|
||||
if (padding < 0)
|
||||
goto fail;
|
||||
}
|
||||
|
||||
/* read frame sizes */
|
||||
if (pkt->vbr) {
|
||||
/* for VBR, all frames except the final one have their size coded
|
||||
in the bitstream. the last frame size is implicit. */
|
||||
int total_bytes = 0;
|
||||
for (i = 0; i < pkt->frame_count - 1; i++) {
|
||||
frame_bytes = xiph_lacing_16bit(&ptr, end);
|
||||
if (frame_bytes < 0)
|
||||
goto fail;
|
||||
pkt->frame_size[i] = frame_bytes;
|
||||
total_bytes += frame_bytes;
|
||||
}
|
||||
|
||||
if (self_delimiting) {
|
||||
int len = xiph_lacing_16bit(&ptr, end);
|
||||
if (len < 0 || len + total_bytes + padding > end - ptr)
|
||||
goto fail;
|
||||
end = ptr + total_bytes + len + padding;
|
||||
buf_size = end - buf;
|
||||
}
|
||||
|
||||
frame_bytes = end - ptr - padding;
|
||||
if (total_bytes > frame_bytes)
|
||||
goto fail;
|
||||
pkt->frame_offset[0] = ptr - buf;
|
||||
for (i = 1; i < pkt->frame_count; i++)
|
||||
pkt->frame_offset[i] = pkt->frame_offset[i-1] + pkt->frame_size[i-1];
|
||||
pkt->frame_size[pkt->frame_count-1] = frame_bytes - total_bytes;
|
||||
} else {
|
||||
/* for CBR, the remaining packet bytes are divided evenly between
|
||||
the frames */
|
||||
if (self_delimiting) {
|
||||
frame_bytes = xiph_lacing_16bit(&ptr, end);
|
||||
if (frame_bytes < 0 || pkt->frame_count * frame_bytes + padding > end - ptr)
|
||||
goto fail;
|
||||
end = ptr + pkt->frame_count * frame_bytes + padding;
|
||||
buf_size = end - buf;
|
||||
} else {
|
||||
frame_bytes = end - ptr - padding;
|
||||
if (frame_bytes % pkt->frame_count ||
|
||||
frame_bytes / pkt->frame_count > MAX_FRAME_SIZE)
|
||||
goto fail;
|
||||
frame_bytes /= pkt->frame_count;
|
||||
}
|
||||
|
||||
pkt->frame_offset[0] = ptr - buf;
|
||||
pkt->frame_size[0] = frame_bytes;
|
||||
for (i = 1; i < pkt->frame_count; i++) {
|
||||
pkt->frame_offset[i] = pkt->frame_offset[i-1] + pkt->frame_size[i-1];
|
||||
pkt->frame_size[i] = frame_bytes;
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
pkt->packet_size = buf_size;
|
||||
pkt->data_size = pkt->packet_size - padding;
|
||||
|
||||
/* total packet duration cannot be larger than 120ms */
|
||||
pkt->frame_duration = opus_frame_duration[pkt->config];
|
||||
if (pkt->frame_duration * pkt->frame_count > MAX_PACKET_DUR)
|
||||
goto fail;
|
||||
|
||||
/* set mode and bandwidth */
|
||||
if (pkt->config < 12) {
|
||||
pkt->mode = OPUS_MODE_SILK;
|
||||
pkt->bandwidth = pkt->config >> 2;
|
||||
} else if (pkt->config < 16) {
|
||||
pkt->mode = OPUS_MODE_HYBRID;
|
||||
pkt->bandwidth = OPUS_BANDWIDTH_SUPERWIDEBAND + (pkt->config >= 14);
|
||||
} else {
|
||||
pkt->mode = OPUS_MODE_CELT;
|
||||
pkt->bandwidth = (pkt->config - 16) >> 2;
|
||||
/* skip mediumband */
|
||||
if (pkt->bandwidth)
|
||||
pkt->bandwidth++;
|
||||
}
|
||||
|
||||
return 0;
|
||||
|
||||
fail:
|
||||
memset(pkt, 0, sizeof(*pkt));
|
||||
return AVERROR_INVALIDDATA;
|
||||
}
|
||||
|
||||
static int channel_reorder_vorbis(int nb_channels, int channel_idx)
|
||||
{
|
||||
return ff_vorbis_channel_layout_offsets[nb_channels - 1][channel_idx];
|
||||
}
|
||||
|
||||
static int channel_reorder_unknown(int nb_channels, int channel_idx)
|
||||
{
|
||||
return channel_idx;
|
||||
}
|
||||
|
||||
av_cold int ff_opus_parse_extradata(AVCodecContext *avctx,
|
||||
OpusContext *s)
|
||||
{
|
||||
static const uint8_t default_channel_map[2] = { 0, 1 };
|
||||
uint8_t default_extradata[19] = {
|
||||
'O', 'p', 'u', 's', 'H', 'e', 'a', 'd',
|
||||
1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
|
||||
};
|
||||
|
||||
int (*channel_reorder)(int, int) = channel_reorder_unknown;
|
||||
|
||||
const uint8_t *extradata, *channel_map;
|
||||
int extradata_size;
|
||||
int version, channels, map_type, streams, stereo_streams, i, j;
|
||||
uint64_t layout;
|
||||
|
||||
if (!avctx->extradata) {
|
||||
if (avctx->channels > 2) {
|
||||
av_log(avctx, AV_LOG_ERROR,
|
||||
"Multichannel configuration without extradata.\n");
|
||||
return AVERROR(EINVAL);
|
||||
}
|
||||
default_extradata[9] = (avctx->channels == 1) ? 1 : 2;
|
||||
extradata = default_extradata;
|
||||
extradata_size = sizeof(default_extradata);
|
||||
} else {
|
||||
extradata = avctx->extradata;
|
||||
extradata_size = avctx->extradata_size;
|
||||
}
|
||||
|
||||
if (extradata_size < 19) {
|
||||
av_log(avctx, AV_LOG_ERROR, "Invalid extradata size: %d\n",
|
||||
extradata_size);
|
||||
return AVERROR_INVALIDDATA;
|
||||
}
|
||||
|
||||
version = extradata[8];
|
||||
if (version > 15) {
|
||||
avpriv_request_sample(avctx, "Extradata version %d", version);
|
||||
return AVERROR_PATCHWELCOME;
|
||||
}
|
||||
|
||||
avctx->delay = AV_RL16(extradata + 10);
|
||||
|
||||
channels = extradata[9];
|
||||
if (!channels) {
|
||||
av_log(avctx, AV_LOG_ERROR, "Zero channel count specified in the extadata\n");
|
||||
return AVERROR_INVALIDDATA;
|
||||
}
|
||||
|
||||
s->gain_i = AV_RL16(extradata + 16);
|
||||
if (s->gain_i)
|
||||
s->gain = pow(10, s->gain_i / (20.0 * 256));
|
||||
|
||||
map_type = extradata[18];
|
||||
if (!map_type) {
|
||||
if (channels > 2) {
|
||||
av_log(avctx, AV_LOG_ERROR,
|
||||
"Channel mapping 0 is only specified for up to 2 channels\n");
|
||||
return AVERROR_INVALIDDATA;
|
||||
}
|
||||
layout = (channels == 1) ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;
|
||||
streams = 1;
|
||||
stereo_streams = channels - 1;
|
||||
channel_map = default_channel_map;
|
||||
} else if (map_type == 1 || map_type == 255) {
|
||||
if (extradata_size < 21 + channels) {
|
||||
av_log(avctx, AV_LOG_ERROR, "Invalid extradata size: %d\n",
|
||||
extradata_size);
|
||||
return AVERROR_INVALIDDATA;
|
||||
}
|
||||
|
||||
streams = extradata[19];
|
||||
stereo_streams = extradata[20];
|
||||
if (!streams || stereo_streams > streams ||
|
||||
streams + stereo_streams > 255) {
|
||||
av_log(avctx, AV_LOG_ERROR,
|
||||
"Invalid stream/stereo stream count: %d/%d\n", streams, stereo_streams);
|
||||
return AVERROR_INVALIDDATA;
|
||||
}
|
||||
|
||||
if (map_type == 1) {
|
||||
if (channels > 8) {
|
||||
av_log(avctx, AV_LOG_ERROR,
|
||||
"Channel mapping 1 is only specified for up to 8 channels\n");
|
||||
return AVERROR_INVALIDDATA;
|
||||
}
|
||||
layout = ff_vorbis_channel_layouts[channels - 1];
|
||||
channel_reorder = channel_reorder_vorbis;
|
||||
} else
|
||||
layout = 0;
|
||||
|
||||
channel_map = extradata + 21;
|
||||
} else {
|
||||
avpriv_request_sample(avctx, "Mapping type %d", map_type);
|
||||
return AVERROR_PATCHWELCOME;
|
||||
}
|
||||
|
||||
s->channel_maps = av_mallocz_array(channels, sizeof(*s->channel_maps));
|
||||
if (!s->channel_maps)
|
||||
return AVERROR(ENOMEM);
|
||||
|
||||
for (i = 0; i < channels; i++) {
|
||||
ChannelMap *map = &s->channel_maps[i];
|
||||
uint8_t idx = channel_map[channel_reorder(channels, i)];
|
||||
|
||||
if (idx == 255) {
|
||||
map->silence = 1;
|
||||
continue;
|
||||
} else if (idx >= streams + stereo_streams) {
|
||||
av_log(avctx, AV_LOG_ERROR,
|
||||
"Invalid channel map for output channel %d: %d\n", i, idx);
|
||||
return AVERROR_INVALIDDATA;
|
||||
}
|
||||
|
||||
/* check that we din't see this index yet */
|
||||
map->copy = 0;
|
||||
for (j = 0; j < i; j++)
|
||||
if (channel_map[channel_reorder(channels, j)] == idx) {
|
||||
map->copy = 1;
|
||||
map->copy_idx = j;
|
||||
break;
|
||||
}
|
||||
|
||||
if (idx < 2 * stereo_streams) {
|
||||
map->stream_idx = idx / 2;
|
||||
map->channel_idx = idx & 1;
|
||||
} else {
|
||||
map->stream_idx = idx - stereo_streams;
|
||||
map->channel_idx = 0;
|
||||
}
|
||||
}
|
||||
|
||||
avctx->channels = channels;
|
||||
avctx->channel_layout = layout;
|
||||
s->nb_streams = streams;
|
||||
s->nb_stereo_streams = stereo_streams;
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
|
@ -0,0 +1,429 @@
|
|||
/*
|
||||
* Opus decoder/demuxer common functions
|
||||
* Copyright (c) 2012 Andrew D'Addesio
|
||||
* Copyright (c) 2013-2014 Mozilla Corporation
|
||||
*
|
||||
* 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
|
||||
*/
|
||||
|
||||
#ifndef AVCODEC_OPUS_H
|
||||
#define AVCODEC_OPUS_H
|
||||
|
||||
#include <stdint.h>
|
||||
|
||||
#include "libavutil/audio_fifo.h"
|
||||
#include "libavutil/float_dsp.h"
|
||||
#include "libavutil/frame.h"
|
||||
|
||||
#include "libavresample/avresample.h"
|
||||
|
||||
#include "avcodec.h"
|
||||
#include "get_bits.h"
|
||||
|
||||
#define MAX_FRAME_SIZE 1275
|
||||
#define MAX_FRAMES 48
|
||||
#define MAX_PACKET_DUR 5760
|
||||
|
||||
#define CELT_SHORT_BLOCKSIZE 120
|
||||
#define CELT_OVERLAP CELT_SHORT_BLOCKSIZE
|
||||
#define CELT_MAX_LOG_BLOCKS 3
|
||||
#define CELT_MAX_FRAME_SIZE (CELT_SHORT_BLOCKSIZE * (1 << CELT_MAX_LOG_BLOCKS))
|
||||
#define CELT_MAX_BANDS 21
|
||||
#define CELT_VECTORS 11
|
||||
#define CELT_ALLOC_STEPS 6
|
||||
#define CELT_FINE_OFFSET 21
|
||||
#define CELT_MAX_FINE_BITS 8
|
||||
#define CELT_NORM_SCALE 16384
|
||||
#define CELT_QTHETA_OFFSET 4
|
||||
#define CELT_QTHETA_OFFSET_TWOPHASE 16
|
||||
#define CELT_DEEMPH_COEFF 0.85000610f
|
||||
#define CELT_POSTFILTER_MINPERIOD 15
|
||||
#define CELT_ENERGY_SILENCE (-28.0f)
|
||||
|
||||
#define SILK_HISTORY 322
|
||||
#define SILK_MAX_LPC 16
|
||||
|
||||
#define ROUND_MULL(a,b,s) (((MUL64(a, b) >> (s - 1)) + 1) >> 1)
|
||||
#define ROUND_MUL16(a,b) ((MUL16(a, b) + 16384) >> 15)
|
||||
#define opus_ilog(i) (av_log2(i) + !!(i))
|
||||
|
||||
enum OpusMode {
|
||||
OPUS_MODE_SILK,
|
||||
OPUS_MODE_HYBRID,
|
||||
OPUS_MODE_CELT
|
||||
};
|
||||
|
||||
enum OpusBandwidth {
|
||||
OPUS_BANDWIDTH_NARROWBAND,
|
||||
OPUS_BANDWIDTH_MEDIUMBAND,
|
||||
OPUS_BANDWIDTH_WIDEBAND,
|
||||
OPUS_BANDWIDTH_SUPERWIDEBAND,
|
||||
OPUS_BANDWIDTH_FULLBAND
|
||||
};
|
||||
|
||||
typedef struct RawBitsContext {
|
||||
const uint8_t *position;
|
||||
unsigned int bytes;
|
||||
unsigned int cachelen;
|
||||
unsigned int cacheval;
|
||||
} RawBitsContext;
|
||||
|
||||
typedef struct OpusRangeCoder {
|
||||
GetBitContext gb;
|
||||
RawBitsContext rb;
|
||||
unsigned int range;
|
||||
unsigned int value;
|
||||
unsigned int total_read_bits;
|
||||
} OpusRangeCoder;
|
||||
|
||||
typedef struct SilkContext SilkContext;
|
||||
|
||||
typedef struct CeltIMDCTContext CeltIMDCTContext;
|
||||
|
||||
typedef struct CeltContext CeltContext;
|
||||
|
||||
typedef struct OpusPacket {
|
||||
int packet_size; /** packet size */
|
||||
int data_size; /** size of the useful data -- packet size - padding */
|
||||
int code; /** packet code: specifies the frame layout */
|
||||
int stereo; /** whether this packet is mono or stereo */
|
||||
int vbr; /** vbr flag */
|
||||
int config; /** configuration: tells the audio mode,
|
||||
** bandwidth, and frame duration */
|
||||
int frame_count; /** frame count */
|
||||
int frame_offset[MAX_FRAMES]; /** frame offsets */
|
||||
int frame_size[MAX_FRAMES]; /** frame sizes */
|
||||
int frame_duration; /** frame duration, in samples @ 48kHz */
|
||||
enum OpusMode mode; /** mode */
|
||||
enum OpusBandwidth bandwidth; /** bandwidth */
|
||||
} OpusPacket;
|
||||
|
||||
typedef struct OpusStreamContext {
|
||||
AVCodecContext *avctx;
|
||||
int output_channels;
|
||||
|
||||
OpusRangeCoder rc;
|
||||
OpusRangeCoder redundancy_rc;
|
||||
SilkContext *silk;
|
||||
CeltContext *celt;
|
||||
AVFloatDSPContext *fdsp;
|
||||
|
||||
float silk_buf[2][960];
|
||||
float *silk_output[2];
|
||||
DECLARE_ALIGNED(32, float, celt_buf)[2][960];
|
||||
float *celt_output[2];
|
||||
|
||||
float redundancy_buf[2][960];
|
||||
float *redundancy_output[2];
|
||||
|
||||
/* data buffers for the final output data */
|
||||
float *out[2];
|
||||
int out_size;
|
||||
|
||||
float *out_dummy;
|
||||
int out_dummy_allocated_size;
|
||||
|
||||
AVAudioResampleContext *avr;
|
||||
AVAudioFifo *celt_delay;
|
||||
int silk_samplerate;
|
||||
/* number of samples we still want to get from the resampler */
|
||||
int delayed_samples;
|
||||
|
||||
OpusPacket packet;
|
||||
|
||||
int redundancy_idx;
|
||||
} OpusStreamContext;
|
||||
|
||||
// a mapping between an opus stream and an output channel
|
||||
typedef struct ChannelMap {
|
||||
int stream_idx;
|
||||
int channel_idx;
|
||||
|
||||
// when a single decoded channel is mapped to multiple output channels, we
|
||||
// write to the first output directly and copy from it to the others
|
||||
// this field is set to 1 for those copied output channels
|
||||
int copy;
|
||||
// this is the index of the output channel to copy from
|
||||
int copy_idx;
|
||||
|
||||
// this channel is silent
|
||||
int silence;
|
||||
} ChannelMap;
|
||||
|
||||
typedef struct OpusContext {
|
||||
OpusStreamContext *streams;
|
||||
int nb_streams;
|
||||
int nb_stereo_streams;
|
||||
|
||||
AVFloatDSPContext fdsp;
|
||||
int16_t gain_i;
|
||||
float gain;
|
||||
|
||||
ChannelMap *channel_maps;
|
||||
} OpusContext;
|
||||
|
||||
static av_always_inline void opus_rc_normalize(OpusRangeCoder *rc)
|
||||
{
|
||||
while (rc->range <= 1<<23) {
|
||||
rc->value = ((rc->value << 8) | (get_bits(&rc->gb, 8) ^ 0xFF)) & ((1u << 31) - 1);
|
||||
rc->range <<= 8;
|
||||
rc->total_read_bits += 8;
|
||||
}
|
||||
}
|
||||
|
||||
static av_always_inline void opus_rc_update(OpusRangeCoder *rc, unsigned int scale,
|
||||
unsigned int low, unsigned int high,
|
||||
unsigned int total)
|
||||
{
|
||||
rc->value -= scale * (total - high);
|
||||
rc->range = low ? scale * (high - low)
|
||||
: rc->range - scale * (total - high);
|
||||
opus_rc_normalize(rc);
|
||||
}
|
||||
|
||||
static av_always_inline unsigned int opus_rc_getsymbol(OpusRangeCoder *rc, const uint16_t *cdf)
|
||||
{
|
||||
unsigned int k, scale, total, symbol, low, high;
|
||||
|
||||
total = *cdf++;
|
||||
|
||||
scale = rc->range / total;
|
||||
symbol = rc->value / scale + 1;
|
||||
symbol = total - FFMIN(symbol, total);
|
||||
|
||||
for (k = 0; cdf[k] <= symbol; k++);
|
||||
high = cdf[k];
|
||||
low = k ? cdf[k-1] : 0;
|
||||
|
||||
opus_rc_update(rc, scale, low, high, total);
|
||||
|
||||
return k;
|
||||
}
|
||||
|
||||
static av_always_inline unsigned int opus_rc_p2model(OpusRangeCoder *rc, unsigned int bits)
|
||||
{
|
||||
unsigned int k, scale;
|
||||
scale = rc->range >> bits; // in this case, scale = symbol
|
||||
|
||||
if (rc->value >= scale) {
|
||||
rc->value -= scale;
|
||||
rc->range -= scale;
|
||||
k = 0;
|
||||
} else {
|
||||
rc->range = scale;
|
||||
k = 1;
|
||||
}
|
||||
opus_rc_normalize(rc);
|
||||
return k;
|
||||
}
|
||||
|
||||
/**
|
||||
* CELT: estimate bits of entropy that have thus far been consumed for the
|
||||
* current CELT frame, to integer and fractional (1/8th bit) precision
|
||||
*/
|
||||
static av_always_inline unsigned int opus_rc_tell(const OpusRangeCoder *rc)
|
||||
{
|
||||
return rc->total_read_bits - av_log2(rc->range) - 1;
|
||||
}
|
||||
|
||||
static av_always_inline unsigned int opus_rc_tell_frac(const OpusRangeCoder *rc)
|
||||
{
|
||||
unsigned int i, total_bits, rcbuffer, range;
|
||||
|
||||
total_bits = rc->total_read_bits << 3;
|
||||
rcbuffer = av_log2(rc->range) + 1;
|
||||
range = rc->range >> (rcbuffer-16);
|
||||
|
||||
for (i = 0; i < 3; i++) {
|
||||
int bit;
|
||||
range = range * range >> 15;
|
||||
bit = range >> 16;
|
||||
rcbuffer = rcbuffer << 1 | bit;
|
||||
range >>= bit;
|
||||
}
|
||||
|
||||
return total_bits - rcbuffer;
|
||||
}
|
||||
|
||||
/**
|
||||
* CELT: read 1-25 raw bits at the end of the frame, backwards byte-wise
|
||||
*/
|
||||
static av_always_inline unsigned int opus_getrawbits(OpusRangeCoder *rc, unsigned int count)
|
||||
{
|
||||
unsigned int value = 0;
|
||||
|
||||
while (rc->rb.bytes && rc->rb.cachelen < count) {
|
||||
rc->rb.cacheval |= *--rc->rb.position << rc->rb.cachelen;
|
||||
rc->rb.cachelen += 8;
|
||||
rc->rb.bytes--;
|
||||
}
|
||||
|
||||
value = rc->rb.cacheval & ((1<<count)-1);
|
||||
rc->rb.cacheval >>= count;
|
||||
rc->rb.cachelen -= count;
|
||||
rc->total_read_bits += count;
|
||||
|
||||
return value;
|
||||
}
|
||||
|
||||
/**
|
||||
* CELT: read a uniform distribution
|
||||
*/
|
||||
static av_always_inline unsigned int opus_rc_unimodel(OpusRangeCoder *rc, unsigned int size)
|
||||
{
|
||||
unsigned int bits, k, scale, total;
|
||||
|
||||
bits = opus_ilog(size - 1);
|
||||
total = (bits > 8) ? ((size - 1) >> (bits - 8)) + 1 : size;
|
||||
|
||||
scale = rc->range / total;
|
||||
k = rc->value / scale + 1;
|
||||
k = total - FFMIN(k, total);
|
||||
opus_rc_update(rc, scale, k, k + 1, total);
|
||||
|
||||
if (bits > 8) {
|
||||
k = k << (bits - 8) | opus_getrawbits(rc, bits - 8);
|
||||
return FFMIN(k, size - 1);
|
||||
} else
|
||||
return k;
|
||||
}
|
||||
|
||||
static av_always_inline int opus_rc_laplace(OpusRangeCoder *rc, unsigned int symbol, int decay)
|
||||
{
|
||||
/* extends the range coder to model a Laplace distribution */
|
||||
int value = 0;
|
||||
unsigned int scale, low = 0, center;
|
||||
|
||||
scale = rc->range >> 15;
|
||||
center = rc->value / scale + 1;
|
||||
center = (1 << 15) - FFMIN(center, 1 << 15);
|
||||
|
||||
if (center >= symbol) {
|
||||
value++;
|
||||
low = symbol;
|
||||
symbol = 1 + ((32768 - 32 - symbol) * (16384-decay) >> 15);
|
||||
|
||||
while (symbol > 1 && center >= low + 2 * symbol) {
|
||||
value++;
|
||||
symbol *= 2;
|
||||
low += symbol;
|
||||
symbol = (((symbol - 2) * decay) >> 15) + 1;
|
||||
}
|
||||
|
||||
if (symbol <= 1) {
|
||||
int distance = (center - low) >> 1;
|
||||
value += distance;
|
||||
low += 2 * distance;
|
||||
}
|
||||
|
||||
if (center < low + symbol)
|
||||
value *= -1;
|
||||
else
|
||||
low += symbol;
|
||||
}
|
||||
|
||||
opus_rc_update(rc, scale, low, FFMIN(low + symbol, 32768), 32768);
|
||||
|
||||
return value;
|
||||
}
|
||||
|
||||
static av_always_inline unsigned int opus_rc_stepmodel(OpusRangeCoder *rc, int k0)
|
||||
{
|
||||
/* Use a probability of 3 up to itheta=8192 and then use 1 after */
|
||||
unsigned int k, scale, symbol, total = (k0+1)*3 + k0;
|
||||
scale = rc->range / total;
|
||||
symbol = rc->value / scale + 1;
|
||||
symbol = total - FFMIN(symbol, total);
|
||||
|
||||
k = (symbol < (k0+1)*3) ? symbol/3 : symbol - (k0+1)*2;
|
||||
|
||||
opus_rc_update(rc, scale, (k <= k0) ? 3*(k+0) : (k-1-k0) + 3*(k0+1),
|
||||
(k <= k0) ? 3*(k+1) : (k-0-k0) + 3*(k0+1), total);
|
||||
return k;
|
||||
}
|
||||
|
||||
static av_always_inline unsigned int opus_rc_trimodel(OpusRangeCoder *rc, int qn)
|
||||
{
|
||||
unsigned int k, scale, symbol, total, low, center;
|
||||
|
||||
total = ((qn>>1) + 1) * ((qn>>1) + 1);
|
||||
scale = rc->range / total;
|
||||
center = rc->value / scale + 1;
|
||||
center = total - FFMIN(center, total);
|
||||
|
||||
if (center < total >> 1) {
|
||||
k = (ff_sqrt(8 * center + 1) - 1) >> 1;
|
||||
low = k * (k + 1) >> 1;
|
||||
symbol = k + 1;
|
||||
} else {
|
||||
k = (2*(qn + 1) - ff_sqrt(8*(total - center - 1) + 1)) >> 1;
|
||||
low = total - ((qn + 1 - k) * (qn + 2 - k) >> 1);
|
||||
symbol = qn + 1 - k;
|
||||
}
|
||||
|
||||
opus_rc_update(rc, scale, low, low + symbol, total);
|
||||
|
||||
return k;
|
||||
}
|
||||
|
||||
int ff_opus_parse_packet(OpusPacket *pkt, const uint8_t *buf, int buf_size,
|
||||
int self_delimited);
|
||||
|
||||
int ff_opus_parse_extradata(AVCodecContext *avctx, OpusContext *s);
|
||||
|
||||
int ff_silk_init(AVCodecContext *avctx, SilkContext **ps, int output_channels);
|
||||
void ff_silk_free(SilkContext **ps);
|
||||
void ff_silk_flush(SilkContext *s);
|
||||
|
||||
/**
|
||||
* Decode the LP layer of one Opus frame (which may correspond to several SILK
|
||||
* frames).
|
||||
*/
|
||||
int ff_silk_decode_superframe(SilkContext *s, OpusRangeCoder *rc,
|
||||
float *output[2],
|
||||
enum OpusBandwidth bandwidth, int coded_channels,
|
||||
int duration_ms);
|
||||
|
||||
/**
|
||||
* Init an iMDCT of the length 2 * 15 * (2^N)
|
||||
*/
|
||||
int ff_celt_imdct_init(CeltIMDCTContext **s, int N);
|
||||
|
||||
/**
|
||||
* Free an iMDCT.
|
||||
*/
|
||||
void ff_celt_imdct_uninit(CeltIMDCTContext **s);
|
||||
|
||||
/**
|
||||
* Calculate the middle half of the iMDCT
|
||||
*/
|
||||
void ff_celt_imdct_half(CeltIMDCTContext *s, float *dst, const float *src,
|
||||
int src_stride, float scale);
|
||||
|
||||
int ff_celt_init(AVCodecContext *avctx, CeltContext **s, int output_channels);
|
||||
|
||||
void ff_celt_free(CeltContext **s);
|
||||
|
||||
void ff_celt_flush(CeltContext *s);
|
||||
|
||||
int ff_celt_decode_frame(CeltContext *s, OpusRangeCoder *rc,
|
||||
float **output, int coded_channels, int frame_size,
|
||||
int startband, int endband);
|
||||
|
||||
extern const float ff_celt_window2[120];
|
||||
|
||||
#endif /* AVCODEC_OPUS_H */
|
||||
File diff suppressed because it is too large
Load Diff
|
|
@ -0,0 +1,268 @@
|
|||
/*
|
||||
* Copyright (c) 2013-2014 Mozilla Corporation
|
||||
*
|
||||
* 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
|
||||
*/
|
||||
|
||||
/**
|
||||
* @file
|
||||
* Celt non-power of 2 iMDCT
|
||||
*/
|
||||
|
||||
#include <float.h>
|
||||
#include <math.h>
|
||||
|
||||
#include "libavutil/attributes.h"
|
||||
#include "libavutil/common.h"
|
||||
|
||||
#include "fft.h"
|
||||
#include "opus.h"
|
||||
|
||||
// complex c = a * b
|
||||
#define CMUL3(cre, cim, are, aim, bre, bim) \
|
||||
do { \
|
||||
cre = are * bre - aim * bim; \
|
||||
cim = are * bim + aim * bre; \
|
||||
} while (0)
|
||||
|
||||
#define CMUL(c, a, b) CMUL3((c).re, (c).im, (a).re, (a).im, (b).re, (b).im)
|
||||
|
||||
// complex c = a * b
|
||||
// d = a * conjugate(b)
|
||||
#define CMUL2(c, d, a, b) \
|
||||
do { \
|
||||
float are = (a).re; \
|
||||
float aim = (a).im; \
|
||||
float bre = (b).re; \
|
||||
float bim = (b).im; \
|
||||
float rr = are * bre; \
|
||||
float ri = are * bim; \
|
||||
float ir = aim * bre; \
|
||||
float ii = aim * bim; \
|
||||
(c).re = rr - ii; \
|
||||
(c).im = ri + ir; \
|
||||
(d).re = rr + ii; \
|
||||
(d).im = -ri + ir; \
|
||||
} while (0)
|
||||
|
||||
struct CeltIMDCTContext {
|
||||
int fft_n;
|
||||
int len2;
|
||||
int len4;
|
||||
|
||||
FFTComplex *tmp;
|
||||
|
||||
FFTComplex *twiddle_exptab;
|
||||
|
||||
FFTComplex *exptab[6];
|
||||
};
|
||||
|
||||
av_cold void ff_celt_imdct_uninit(CeltIMDCTContext **ps)
|
||||
{
|
||||
CeltIMDCTContext *s = *ps;
|
||||
int i;
|
||||
|
||||
if (!s)
|
||||
return;
|
||||
|
||||
for (i = 0; i < FF_ARRAY_ELEMS(s->exptab); i++)
|
||||
av_freep(&s->exptab[i]);
|
||||
|
||||
av_freep(&s->twiddle_exptab);
|
||||
|
||||
av_freep(&s->tmp);
|
||||
|
||||
av_freep(ps);
|
||||
}
|
||||
|
||||
av_cold int ff_celt_imdct_init(CeltIMDCTContext **ps, int N)
|
||||
{
|
||||
CeltIMDCTContext *s;
|
||||
int len2 = 15 * (1 << N);
|
||||
int len = 2 * len2;
|
||||
int i, j;
|
||||
|
||||
if (len2 > CELT_MAX_FRAME_SIZE)
|
||||
return AVERROR(EINVAL);
|
||||
|
||||
s = av_mallocz(sizeof(*s));
|
||||
if (!s)
|
||||
return AVERROR(ENOMEM);
|
||||
|
||||
s->fft_n = N - 1;
|
||||
s->len4 = len2 / 2;
|
||||
s->len2 = len2;
|
||||
|
||||
s->tmp = av_malloc(len * 2 * sizeof(*s->tmp));
|
||||
if (!s->tmp)
|
||||
goto fail;
|
||||
|
||||
s->twiddle_exptab = av_malloc(s->len4 * sizeof(*s->twiddle_exptab));
|
||||
if (!s->twiddle_exptab)
|
||||
goto fail;
|
||||
|
||||
for (i = 0; i < s->len4; i++) {
|
||||
s->twiddle_exptab[i].re = cos(2 * M_PI * (i + 0.125 + s->len4) / len);
|
||||
s->twiddle_exptab[i].im = sin(2 * M_PI * (i + 0.125 + s->len4) / len);
|
||||
}
|
||||
|
||||
for (i = 0; i < FF_ARRAY_ELEMS(s->exptab); i++) {
|
||||
int N = 15 * (1 << i);
|
||||
s->exptab[i] = av_malloc(sizeof(*s->exptab[i]) * FFMAX(N, 19));
|
||||
if (!s->exptab[i])
|
||||
goto fail;
|
||||
|
||||
for (j = 0; j < N; j++) {
|
||||
s->exptab[i][j].re = cos(2 * M_PI * j / N);
|
||||
s->exptab[i][j].im = sin(2 * M_PI * j / N);
|
||||
}
|
||||
}
|
||||
|
||||
// wrap around to simplify fft15
|
||||
for (j = 15; j < 19; j++)
|
||||
s->exptab[0][j] = s->exptab[0][j - 15];
|
||||
|
||||
*ps = s;
|
||||
|
||||
return 0;
|
||||
fail:
|
||||
ff_celt_imdct_uninit(&s);
|
||||
return AVERROR(ENOMEM);
|
||||
}
|
||||
|
||||
static void fft5(FFTComplex *out, const FFTComplex *in, int stride)
|
||||
{
|
||||
// [0] = exp(2 * i * pi / 5), [1] = exp(2 * i * pi * 2 / 5)
|
||||
static const FFTComplex fact[] = { { 0.30901699437494745, 0.95105651629515353 },
|
||||
{ -0.80901699437494734, 0.58778525229247325 } };
|
||||
|
||||
FFTComplex z[4][4];
|
||||
|
||||
CMUL2(z[0][0], z[0][3], in[1 * stride], fact[0]);
|
||||
CMUL2(z[0][1], z[0][2], in[1 * stride], fact[1]);
|
||||
CMUL2(z[1][0], z[1][3], in[2 * stride], fact[0]);
|
||||
CMUL2(z[1][1], z[1][2], in[2 * stride], fact[1]);
|
||||
CMUL2(z[2][0], z[2][3], in[3 * stride], fact[0]);
|
||||
CMUL2(z[2][1], z[2][2], in[3 * stride], fact[1]);
|
||||
CMUL2(z[3][0], z[3][3], in[4 * stride], fact[0]);
|
||||
CMUL2(z[3][1], z[3][2], in[4 * stride], fact[1]);
|
||||
|
||||
out[0].re = in[0].re + in[stride].re + in[2 * stride].re + in[3 * stride].re + in[4 * stride].re;
|
||||
out[0].im = in[0].im + in[stride].im + in[2 * stride].im + in[3 * stride].im + in[4 * stride].im;
|
||||
|
||||
out[1].re = in[0].re + z[0][0].re + z[1][1].re + z[2][2].re + z[3][3].re;
|
||||
out[1].im = in[0].im + z[0][0].im + z[1][1].im + z[2][2].im + z[3][3].im;
|
||||
|
||||
out[2].re = in[0].re + z[0][1].re + z[1][3].re + z[2][0].re + z[3][2].re;
|
||||
out[2].im = in[0].im + z[0][1].im + z[1][3].im + z[2][0].im + z[3][2].im;
|
||||
|
||||
out[3].re = in[0].re + z[0][2].re + z[1][0].re + z[2][3].re + z[3][1].re;
|
||||
out[3].im = in[0].im + z[0][2].im + z[1][0].im + z[2][3].im + z[3][1].im;
|
||||
|
||||
out[4].re = in[0].re + z[0][3].re + z[1][2].re + z[2][1].re + z[3][0].re;
|
||||
out[4].im = in[0].im + z[0][3].im + z[1][2].im + z[2][1].im + z[3][0].im;
|
||||
}
|
||||
|
||||
static void fft15(CeltIMDCTContext *s, FFTComplex *out, const FFTComplex *in, int stride)
|
||||
{
|
||||
const FFTComplex *exptab = s->exptab[0];
|
||||
FFTComplex tmp[5];
|
||||
FFTComplex tmp1[5];
|
||||
FFTComplex tmp2[5];
|
||||
int k;
|
||||
|
||||
fft5(tmp, in, stride * 3);
|
||||
fft5(tmp1, in + stride, stride * 3);
|
||||
fft5(tmp2, in + 2 * stride, stride * 3);
|
||||
|
||||
for (k = 0; k < 5; k++) {
|
||||
FFTComplex t1, t2;
|
||||
|
||||
CMUL(t1, tmp1[k], exptab[k]);
|
||||
CMUL(t2, tmp2[k], exptab[2 * k]);
|
||||
out[k].re = tmp[k].re + t1.re + t2.re;
|
||||
out[k].im = tmp[k].im + t1.im + t2.im;
|
||||
|
||||
CMUL(t1, tmp1[k], exptab[k + 5]);
|
||||
CMUL(t2, tmp2[k], exptab[2 * (k + 5)]);
|
||||
out[k + 5].re = tmp[k].re + t1.re + t2.re;
|
||||
out[k + 5].im = tmp[k].im + t1.im + t2.im;
|
||||
|
||||
CMUL(t1, tmp1[k], exptab[k + 10]);
|
||||
CMUL(t2, tmp2[k], exptab[2 * k + 5]);
|
||||
out[k + 10].re = tmp[k].re + t1.re + t2.re;
|
||||
out[k + 10].im = tmp[k].im + t1.im + t2.im;
|
||||
}
|
||||
}
|
||||
|
||||
/*
|
||||
* FFT of the length 15 * (2^N)
|
||||
*/
|
||||
static void fft_calc(CeltIMDCTContext *s, FFTComplex *out, const FFTComplex *in, int N, int stride)
|
||||
{
|
||||
if (N) {
|
||||
const FFTComplex *exptab = s->exptab[N];
|
||||
const int len2 = 15 * (1 << (N - 1));
|
||||
int k;
|
||||
|
||||
fft_calc(s, out, in, N - 1, stride * 2);
|
||||
fft_calc(s, out + len2, in + stride, N - 1, stride * 2);
|
||||
|
||||
for (k = 0; k < len2; k++) {
|
||||
FFTComplex t;
|
||||
|
||||
CMUL(t, out[len2 + k], exptab[k]);
|
||||
|
||||
out[len2 + k].re = out[k].re - t.re;
|
||||
out[len2 + k].im = out[k].im - t.im;
|
||||
|
||||
out[k].re += t.re;
|
||||
out[k].im += t.im;
|
||||
}
|
||||
} else
|
||||
fft15(s, out, in, stride);
|
||||
}
|
||||
|
||||
void ff_celt_imdct_half(CeltIMDCTContext *s, float *dst, const float *src,
|
||||
int stride, float scale)
|
||||
{
|
||||
FFTComplex *z = (FFTComplex *)dst;
|
||||
const int len8 = s->len4 / 2;
|
||||
const float *in1 = src;
|
||||
const float *in2 = src + (s->len2 - 1) * stride;
|
||||
int i;
|
||||
|
||||
for (i = 0; i < s->len4; i++) {
|
||||
FFTComplex tmp = { *in2, *in1 };
|
||||
CMUL(s->tmp[i], tmp, s->twiddle_exptab[i]);
|
||||
in1 += 2 * stride;
|
||||
in2 -= 2 * stride;
|
||||
}
|
||||
|
||||
fft_calc(s, z, s->tmp, s->fft_n, 1);
|
||||
|
||||
for (i = 0; i < len8; i++) {
|
||||
float r0, i0, r1, i1;
|
||||
|
||||
CMUL3(r0, i1, z[len8 - i - 1].im, z[len8 - i - 1].re, s->twiddle_exptab[len8 - i - 1].im, s->twiddle_exptab[len8 - i - 1].re);
|
||||
CMUL3(r1, i0, z[len8 + i].im, z[len8 + i].re, s->twiddle_exptab[len8 + i].im, s->twiddle_exptab[len8 + i].re);
|
||||
z[len8 - i - 1].re = scale * r0;
|
||||
z[len8 - i - 1].im = scale * i0;
|
||||
z[len8 + i].re = scale * r1;
|
||||
z[len8 + i].im = scale * i1;
|
||||
}
|
||||
}
|
||||
|
|
@ -0,0 +1,75 @@
|
|||
/*
|
||||
* Copyright (c) 2013-2014 Mozilla Corporation
|
||||
*
|
||||
* 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
|
||||
*/
|
||||
|
||||
/**
|
||||
* @file
|
||||
* Opus parser
|
||||
*
|
||||
* Determines the duration for each packet.
|
||||
*/
|
||||
|
||||
#include "avcodec.h"
|
||||
#include "opus.h"
|
||||
|
||||
typedef struct OpusParseContext {
|
||||
OpusContext ctx;
|
||||
OpusPacket pkt;
|
||||
int extradata_parsed;
|
||||
} OpusParseContext;
|
||||
|
||||
static int opus_parse(AVCodecParserContext *ctx, AVCodecContext *avctx,
|
||||
const uint8_t **poutbuf, int *poutbuf_size,
|
||||
const uint8_t *buf, int buf_size)
|
||||
{
|
||||
OpusParseContext *s = ctx->priv_data;
|
||||
int ret;
|
||||
|
||||
if (!buf_size)
|
||||
return 0;
|
||||
|
||||
if (avctx->extradata && !s->extradata_parsed) {
|
||||
ret = ff_opus_parse_extradata(avctx, &s->ctx);
|
||||
if (ret < 0) {
|
||||
av_log(avctx, AV_LOG_ERROR, "Error parsing Ogg extradata.\n");
|
||||
goto fail;
|
||||
}
|
||||
av_freep(&s->ctx.channel_maps);
|
||||
s->extradata_parsed = 1;
|
||||
}
|
||||
|
||||
ret = ff_opus_parse_packet(&s->pkt, buf, buf_size, s->ctx.nb_streams > 1);
|
||||
if (ret < 0) {
|
||||
av_log(avctx, AV_LOG_ERROR, "Error parsing Opus packet header.\n");
|
||||
goto fail;
|
||||
}
|
||||
|
||||
ctx->duration = s->pkt.frame_count * s->pkt.frame_duration;
|
||||
|
||||
fail:
|
||||
*poutbuf = buf;
|
||||
*poutbuf_size = buf_size;
|
||||
return buf_size;
|
||||
}
|
||||
|
||||
AVCodecParser ff_opus_parser = {
|
||||
.codec_ids = { AV_CODEC_ID_OPUS },
|
||||
.priv_data_size = sizeof(OpusParseContext),
|
||||
.parser_parse = opus_parse,
|
||||
};
|
||||
File diff suppressed because it is too large
Load Diff
|
|
@ -0,0 +1,674 @@
|
|||
/*
|
||||
* Opus decoder
|
||||
* Copyright (c) 2012 Andrew D'Addesio
|
||||
* Copyright (c) 2013-2014 Mozilla Corporation
|
||||
*
|
||||
* 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
|
||||
*/
|
||||
|
||||
/**
|
||||
* @file
|
||||
* Opus decoder
|
||||
* @author Andrew D'Addesio, Anton Khirnov
|
||||
*
|
||||
* Codec homepage: http://opus-codec.org/
|
||||
* Specification: http://tools.ietf.org/html/rfc6716
|
||||
* Ogg Opus specification: https://tools.ietf.org/html/draft-ietf-codec-oggopus-03
|
||||
*
|
||||
* Ogg-contained .opus files can be produced with opus-tools:
|
||||
* http://git.xiph.org/?p=opus-tools.git
|
||||
*/
|
||||
|
||||
#include <stdint.h>
|
||||
|
||||
#include "libavutil/attributes.h"
|
||||
#include "libavutil/audio_fifo.h"
|
||||
#include "libavutil/channel_layout.h"
|
||||
#include "libavutil/opt.h"
|
||||
|
||||
#include "libavresample/avresample.h"
|
||||
|
||||
#include "avcodec.h"
|
||||
#include "celp_filters.h"
|
||||
#include "fft.h"
|
||||
#include "get_bits.h"
|
||||
#include "internal.h"
|
||||
#include "mathops.h"
|
||||
#include "opus.h"
|
||||
|
||||
static const uint16_t silk_frame_duration_ms[16] = {
|
||||
10, 20, 40, 60,
|
||||
10, 20, 40, 60,
|
||||
10, 20, 40, 60,
|
||||
10, 20,
|
||||
10, 20,
|
||||
};
|
||||
|
||||
/* number of samples of silence to feed to the resampler
|
||||
* at the beginning */
|
||||
static const int silk_resample_delay[] = {
|
||||
4, 8, 11, 11, 11
|
||||
};
|
||||
|
||||
static const uint8_t celt_band_end[] = { 13, 17, 17, 19, 21 };
|
||||
|
||||
static int get_silk_samplerate(int config)
|
||||
{
|
||||
if (config < 4)
|
||||
return 8000;
|
||||
else if (config < 8)
|
||||
return 12000;
|
||||
return 16000;
|
||||
}
|
||||
|
||||
/**
|
||||
* Range decoder
|
||||
*/
|
||||
static int opus_rc_init(OpusRangeCoder *rc, const uint8_t *data, int size)
|
||||
{
|
||||
int ret = init_get_bits8(&rc->gb, data, size);
|
||||
if (ret < 0)
|
||||
return ret;
|
||||
|
||||
rc->range = 128;
|
||||
rc->value = 127 - get_bits(&rc->gb, 7);
|
||||
rc->total_read_bits = 9;
|
||||
opus_rc_normalize(rc);
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
static void opus_raw_init(OpusRangeCoder *rc, const uint8_t *rightend,
|
||||
unsigned int bytes)
|
||||
{
|
||||
rc->rb.position = rightend;
|
||||
rc->rb.bytes = bytes;
|
||||
rc->rb.cachelen = 0;
|
||||
rc->rb.cacheval = 0;
|
||||
}
|
||||
|
||||
static void opus_fade(float *out,
|
||||
const float *in1, const float *in2,
|
||||
const float *window, int len)
|
||||
{
|
||||
int i;
|
||||
for (i = 0; i < len; i++)
|
||||
out[i] = in2[i] * window[i] + in1[i] * (1.0 - window[i]);
|
||||
}
|
||||
|
||||
static int opus_flush_resample(OpusStreamContext *s, int nb_samples)
|
||||
{
|
||||
int celt_size = av_audio_fifo_size(s->celt_delay);
|
||||
int ret, i;
|
||||
|
||||
ret = avresample_convert(s->avr, (uint8_t**)s->out, s->out_size, nb_samples,
|
||||
NULL, 0, 0);
|
||||
if (ret < 0)
|
||||
return ret;
|
||||
else if (ret != nb_samples) {
|
||||
av_log(s->avctx, AV_LOG_ERROR, "Wrong number of flushed samples: %d\n",
|
||||
ret);
|
||||
return AVERROR_BUG;
|
||||
}
|
||||
|
||||
if (celt_size) {
|
||||
if (celt_size != nb_samples) {
|
||||
av_log(s->avctx, AV_LOG_ERROR, "Wrong number of CELT delay samples.\n");
|
||||
return AVERROR_BUG;
|
||||
}
|
||||
av_audio_fifo_read(s->celt_delay, (void**)s->celt_output, nb_samples);
|
||||
for (i = 0; i < s->output_channels; i++) {
|
||||
s->fdsp->vector_fmac_scalar(s->out[i],
|
||||
s->celt_output[i], 1.0,
|
||||
nb_samples);
|
||||
}
|
||||
}
|
||||
|
||||
if (s->redundancy_idx) {
|
||||
for (i = 0; i < s->output_channels; i++)
|
||||
opus_fade(s->out[i], s->out[i],
|
||||
s->redundancy_output[i] + 120 + s->redundancy_idx,
|
||||
ff_celt_window2 + s->redundancy_idx, 120 - s->redundancy_idx);
|
||||
s->redundancy_idx = 0;
|
||||
}
|
||||
|
||||
s->out[0] += nb_samples;
|
||||
s->out[1] += nb_samples;
|
||||
s->out_size -= nb_samples * sizeof(float);
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
static int opus_init_resample(OpusStreamContext *s)
|
||||
{
|
||||
float delay[16] = { 0.0 };
|
||||
uint8_t *delayptr[2] = { (uint8_t*)delay, (uint8_t*)delay };
|
||||
int ret;
|
||||
|
||||
av_opt_set_int(s->avr, "in_sample_rate", s->silk_samplerate, 0);
|
||||
ret = avresample_open(s->avr);
|
||||
if (ret < 0) {
|
||||
av_log(s->avctx, AV_LOG_ERROR, "Error opening the resampler.\n");
|
||||
return ret;
|
||||
}
|
||||
|
||||
ret = avresample_convert(s->avr, NULL, 0, 0, delayptr, sizeof(delay),
|
||||
silk_resample_delay[s->packet.bandwidth]);
|
||||
if (ret < 0) {
|
||||
av_log(s->avctx, AV_LOG_ERROR,
|
||||
"Error feeding initial silence to the resampler.\n");
|
||||
return ret;
|
||||
}
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
static int opus_decode_redundancy(OpusStreamContext *s, const uint8_t *data, int size)
|
||||
{
|
||||
int ret;
|
||||
enum OpusBandwidth bw = s->packet.bandwidth;
|
||||
|
||||
if (s->packet.mode == OPUS_MODE_SILK &&
|
||||
bw == OPUS_BANDWIDTH_MEDIUMBAND)
|
||||
bw = OPUS_BANDWIDTH_WIDEBAND;
|
||||
|
||||
ret = opus_rc_init(&s->redundancy_rc, data, size);
|
||||
if (ret < 0)
|
||||
goto fail;
|
||||
opus_raw_init(&s->redundancy_rc, data + size, size);
|
||||
|
||||
ret = ff_celt_decode_frame(s->celt, &s->redundancy_rc,
|
||||
s->redundancy_output,
|
||||
s->packet.stereo + 1, 240,
|
||||
0, celt_band_end[s->packet.bandwidth]);
|
||||
if (ret < 0)
|
||||
goto fail;
|
||||
|
||||
return 0;
|
||||
fail:
|
||||
av_log(s->avctx, AV_LOG_ERROR, "Error decoding the redundancy frame.\n");
|
||||
return ret;
|
||||
}
|
||||
|
||||
static int opus_decode_frame(OpusStreamContext *s, const uint8_t *data, int size)
|
||||
{
|
||||
int samples = s->packet.frame_duration;
|
||||
int redundancy = 0;
|
||||
int redundancy_size, redundancy_pos;
|
||||
int ret, i, consumed;
|
||||
int delayed_samples = s->delayed_samples;
|
||||
|
||||
ret = opus_rc_init(&s->rc, data, size);
|
||||
if (ret < 0)
|
||||
return ret;
|
||||
|
||||
/* decode the silk frame */
|
||||
if (s->packet.mode == OPUS_MODE_SILK || s->packet.mode == OPUS_MODE_HYBRID) {
|
||||
if (!avresample_is_open(s->avr)) {
|
||||
ret = opus_init_resample(s);
|
||||
if (ret < 0)
|
||||
return ret;
|
||||
}
|
||||
|
||||
samples = ff_silk_decode_superframe(s->silk, &s->rc, s->silk_output,
|
||||
FFMIN(s->packet.bandwidth, OPUS_BANDWIDTH_WIDEBAND),
|
||||
s->packet.stereo + 1,
|
||||
silk_frame_duration_ms[s->packet.config]);
|
||||
if (samples < 0) {
|
||||
av_log(s->avctx, AV_LOG_ERROR, "Error decoding a SILK frame.\n");
|
||||
return samples;
|
||||
}
|
||||
|
||||
samples = avresample_convert(s->avr, (uint8_t**)s->out, s->out_size,
|
||||
s->packet.frame_duration,
|
||||
(uint8_t**)s->silk_output,
|
||||
sizeof(s->silk_buf[0]),
|
||||
samples);
|
||||
if (samples < 0) {
|
||||
av_log(s->avctx, AV_LOG_ERROR, "Error resampling SILK data.\n");
|
||||
return samples;
|
||||
}
|
||||
s->delayed_samples += s->packet.frame_duration - samples;
|
||||
} else
|
||||
ff_silk_flush(s->silk);
|
||||
|
||||
// decode redundancy information
|
||||
consumed = opus_rc_tell(&s->rc);
|
||||
if (s->packet.mode == OPUS_MODE_HYBRID && consumed + 37 <= size * 8)
|
||||
redundancy = opus_rc_p2model(&s->rc, 12);
|
||||
else if (s->packet.mode == OPUS_MODE_SILK && consumed + 17 <= size * 8)
|
||||
redundancy = 1;
|
||||
|
||||
if (redundancy) {
|
||||
redundancy_pos = opus_rc_p2model(&s->rc, 1);
|
||||
|
||||
if (s->packet.mode == OPUS_MODE_HYBRID)
|
||||
redundancy_size = opus_rc_unimodel(&s->rc, 256) + 2;
|
||||
else
|
||||
redundancy_size = size - (consumed + 7) / 8;
|
||||
size -= redundancy_size;
|
||||
if (size < 0) {
|
||||
av_log(s->avctx, AV_LOG_ERROR, "Invalid redundancy frame size.\n");
|
||||
return AVERROR_INVALIDDATA;
|
||||
}
|
||||
|
||||
if (redundancy_pos) {
|
||||
ret = opus_decode_redundancy(s, data + size, redundancy_size);
|
||||
if (ret < 0)
|
||||
return ret;
|
||||
ff_celt_flush(s->celt);
|
||||
}
|
||||
}
|
||||
|
||||
/* decode the CELT frame */
|
||||
if (s->packet.mode == OPUS_MODE_CELT || s->packet.mode == OPUS_MODE_HYBRID) {
|
||||
float *out_tmp[2] = { s->out[0], s->out[1] };
|
||||
float **dst = (s->packet.mode == OPUS_MODE_CELT) ?
|
||||
out_tmp : s->celt_output;
|
||||
int celt_output_samples = samples;
|
||||
int delay_samples = av_audio_fifo_size(s->celt_delay);
|
||||
|
||||
if (delay_samples) {
|
||||
if (s->packet.mode == OPUS_MODE_HYBRID) {
|
||||
av_audio_fifo_read(s->celt_delay, (void**)s->celt_output, delay_samples);
|
||||
|
||||
for (i = 0; i < s->output_channels; i++) {
|
||||
s->fdsp->vector_fmac_scalar(out_tmp[i], s->celt_output[i], 1.0,
|
||||
delay_samples);
|
||||
out_tmp[i] += delay_samples;
|
||||
}
|
||||
celt_output_samples -= delay_samples;
|
||||
} else {
|
||||
av_log(s->avctx, AV_LOG_WARNING,
|
||||
"Spurious CELT delay samples present.\n");
|
||||
av_audio_fifo_drain(s->celt_delay, delay_samples);
|
||||
if (s->avctx->err_recognition & AV_EF_EXPLODE)
|
||||
return AVERROR_BUG;
|
||||
}
|
||||
}
|
||||
|
||||
opus_raw_init(&s->rc, data + size, size);
|
||||
|
||||
ret = ff_celt_decode_frame(s->celt, &s->rc, dst,
|
||||
s->packet.stereo + 1,
|
||||
s->packet.frame_duration,
|
||||
(s->packet.mode == OPUS_MODE_HYBRID) ? 17 : 0,
|
||||
celt_band_end[s->packet.bandwidth]);
|
||||
if (ret < 0)
|
||||
return ret;
|
||||
|
||||
if (s->packet.mode == OPUS_MODE_HYBRID) {
|
||||
int celt_delay = s->packet.frame_duration - celt_output_samples;
|
||||
void *delaybuf[2] = { s->celt_output[0] + celt_output_samples,
|
||||
s->celt_output[1] + celt_output_samples };
|
||||
|
||||
for (i = 0; i < s->output_channels; i++) {
|
||||
s->fdsp->vector_fmac_scalar(out_tmp[i],
|
||||
s->celt_output[i], 1.0,
|
||||
celt_output_samples);
|
||||
}
|
||||
|
||||
ret = av_audio_fifo_write(s->celt_delay, delaybuf, celt_delay);
|
||||
if (ret < 0)
|
||||
return ret;
|
||||
}
|
||||
} else
|
||||
ff_celt_flush(s->celt);
|
||||
|
||||
if (s->redundancy_idx) {
|
||||
for (i = 0; i < s->output_channels; i++)
|
||||
opus_fade(s->out[i], s->out[i],
|
||||
s->redundancy_output[i] + 120 + s->redundancy_idx,
|
||||
ff_celt_window2 + s->redundancy_idx, 120 - s->redundancy_idx);
|
||||
s->redundancy_idx = 0;
|
||||
}
|
||||
if (redundancy) {
|
||||
if (!redundancy_pos) {
|
||||
ff_celt_flush(s->celt);
|
||||
ret = opus_decode_redundancy(s, data + size, redundancy_size);
|
||||
if (ret < 0)
|
||||
return ret;
|
||||
|
||||
for (i = 0; i < s->output_channels; i++) {
|
||||
opus_fade(s->out[i] + samples - 120 + delayed_samples,
|
||||
s->out[i] + samples - 120 + delayed_samples,
|
||||
s->redundancy_output[i] + 120,
|
||||
ff_celt_window2, 120 - delayed_samples);
|
||||
if (delayed_samples)
|
||||
s->redundancy_idx = 120 - delayed_samples;
|
||||
}
|
||||
} else {
|
||||
for (i = 0; i < s->output_channels; i++) {
|
||||
memcpy(s->out[i] + delayed_samples, s->redundancy_output[i], 120 * sizeof(float));
|
||||
opus_fade(s->out[i] + 120 + delayed_samples,
|
||||
s->redundancy_output[i] + 120,
|
||||
s->out[i] + 120 + delayed_samples,
|
||||
ff_celt_window2, 120);
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
return samples;
|
||||
}
|
||||
|
||||
static int opus_decode_subpacket(OpusStreamContext *s,
|
||||
const uint8_t *buf, int buf_size,
|
||||
int nb_samples)
|
||||
{
|
||||
int output_samples = 0;
|
||||
int flush_needed = 0;
|
||||
int i, j, ret;
|
||||
|
||||
/* check if we need to flush the resampler */
|
||||
if (avresample_is_open(s->avr)) {
|
||||
if (buf) {
|
||||
int64_t cur_samplerate;
|
||||
av_opt_get_int(s->avr, "in_sample_rate", 0, &cur_samplerate);
|
||||
flush_needed = (s->packet.mode == OPUS_MODE_CELT) || (cur_samplerate != s->silk_samplerate);
|
||||
} else {
|
||||
flush_needed = !!s->delayed_samples;
|
||||
}
|
||||
}
|
||||
|
||||
if (!buf && !flush_needed)
|
||||
return 0;
|
||||
|
||||
/* use dummy output buffers if the channel is not mapped to anything */
|
||||
if (!s->out[0] ||
|
||||
(s->output_channels == 2 && !s->out[1])) {
|
||||
av_fast_malloc(&s->out_dummy, &s->out_dummy_allocated_size, s->out_size);
|
||||
if (!s->out_dummy)
|
||||
return AVERROR(ENOMEM);
|
||||
if (!s->out[0])
|
||||
s->out[0] = s->out_dummy;
|
||||
if (!s->out[1])
|
||||
s->out[1] = s->out_dummy;
|
||||
}
|
||||
|
||||
/* flush the resampler if necessary */
|
||||
if (flush_needed) {
|
||||
ret = opus_flush_resample(s, s->delayed_samples);
|
||||
if (ret < 0) {
|
||||
av_log(s->avctx, AV_LOG_ERROR, "Error flushing the resampler.\n");
|
||||
return ret;
|
||||
}
|
||||
avresample_close(s->avr);
|
||||
output_samples += s->delayed_samples;
|
||||
s->delayed_samples = 0;
|
||||
|
||||
if (!buf)
|
||||
goto finish;
|
||||
}
|
||||
|
||||
/* decode all the frames in the packet */
|
||||
for (i = 0; i < s->packet.frame_count; i++) {
|
||||
int size = s->packet.frame_size[i];
|
||||
int samples = opus_decode_frame(s, buf + s->packet.frame_offset[i], size);
|
||||
|
||||
if (samples < 0) {
|
||||
av_log(s->avctx, AV_LOG_ERROR, "Error decoding an Opus frame.\n");
|
||||
if (s->avctx->err_recognition & AV_EF_EXPLODE)
|
||||
return samples;
|
||||
|
||||
for (j = 0; j < s->output_channels; j++)
|
||||
memset(s->out[j], 0, s->packet.frame_duration * sizeof(float));
|
||||
samples = s->packet.frame_duration;
|
||||
}
|
||||
output_samples += samples;
|
||||
|
||||
for (j = 0; j < s->output_channels; j++)
|
||||
s->out[j] += samples;
|
||||
s->out_size -= samples * sizeof(float);
|
||||
}
|
||||
|
||||
finish:
|
||||
s->out[0] = s->out[1] = NULL;
|
||||
s->out_size = 0;
|
||||
|
||||
return output_samples;
|
||||
}
|
||||
|
||||
static int opus_decode_packet(AVCodecContext *avctx, void *data,
|
||||
int *got_frame_ptr, AVPacket *avpkt)
|
||||
{
|
||||
OpusContext *c = avctx->priv_data;
|
||||
AVFrame *frame = data;
|
||||
const uint8_t *buf = avpkt->data;
|
||||
int buf_size = avpkt->size;
|
||||
int coded_samples = 0;
|
||||
int decoded_samples = 0;
|
||||
int i, ret;
|
||||
|
||||
/* decode the header of the first sub-packet to find out the sample count */
|
||||
if (buf) {
|
||||
OpusPacket *pkt = &c->streams[0].packet;
|
||||
ret = ff_opus_parse_packet(pkt, buf, buf_size, c->nb_streams > 1);
|
||||
if (ret < 0) {
|
||||
av_log(avctx, AV_LOG_ERROR, "Error parsing the packet header.\n");
|
||||
return ret;
|
||||
}
|
||||
coded_samples += pkt->frame_count * pkt->frame_duration;
|
||||
c->streams[0].silk_samplerate = get_silk_samplerate(pkt->config);
|
||||
}
|
||||
|
||||
frame->nb_samples = coded_samples + c->streams[0].delayed_samples;
|
||||
|
||||
/* no input or buffered data => nothing to do */
|
||||
if (!frame->nb_samples) {
|
||||
*got_frame_ptr = 0;
|
||||
return 0;
|
||||
}
|
||||
|
||||
/* setup the data buffers */
|
||||
ret = ff_get_buffer(avctx, frame, 0);
|
||||
if (ret < 0) {
|
||||
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
|
||||
return ret;
|
||||
}
|
||||
frame->nb_samples = 0;
|
||||
|
||||
for (i = 0; i < avctx->channels; i++) {
|
||||
ChannelMap *map = &c->channel_maps[i];
|
||||
if (!map->copy)
|
||||
c->streams[map->stream_idx].out[map->channel_idx] = (float*)frame->extended_data[i];
|
||||
}
|
||||
|
||||
for (i = 0; i < c->nb_streams; i++)
|
||||
c->streams[i].out_size = frame->linesize[0];
|
||||
|
||||
/* decode each sub-packet */
|
||||
for (i = 0; i < c->nb_streams; i++) {
|
||||
OpusStreamContext *s = &c->streams[i];
|
||||
|
||||
if (i && buf) {
|
||||
ret = ff_opus_parse_packet(&s->packet, buf, buf_size, i != c->nb_streams - 1);
|
||||
if (ret < 0) {
|
||||
av_log(avctx, AV_LOG_ERROR, "Error parsing the packet header.\n");
|
||||
return ret;
|
||||
}
|
||||
s->silk_samplerate = get_silk_samplerate(s->packet.config);
|
||||
}
|
||||
|
||||
ret = opus_decode_subpacket(&c->streams[i], buf,
|
||||
s->packet.data_size, coded_samples);
|
||||
if (ret < 0)
|
||||
return ret;
|
||||
if (decoded_samples && ret != decoded_samples) {
|
||||
av_log(avctx, AV_LOG_ERROR, "Different numbers of decoded samples "
|
||||
"in a multi-channel stream\n");
|
||||
return AVERROR_INVALIDDATA;
|
||||
}
|
||||
decoded_samples = ret;
|
||||
buf += s->packet.packet_size;
|
||||
buf_size -= s->packet.packet_size;
|
||||
}
|
||||
|
||||
for (i = 0; i < avctx->channels; i++) {
|
||||
ChannelMap *map = &c->channel_maps[i];
|
||||
|
||||
/* handle copied channels */
|
||||
if (map->copy) {
|
||||
memcpy(frame->extended_data[i],
|
||||
frame->extended_data[map->copy_idx],
|
||||
frame->linesize[0]);
|
||||
} else if (map->silence) {
|
||||
memset(frame->extended_data[i], 0, frame->linesize[0]);
|
||||
}
|
||||
|
||||
if (c->gain_i) {
|
||||
c->fdsp.vector_fmul_scalar((float*)frame->extended_data[i],
|
||||
(float*)frame->extended_data[i],
|
||||
c->gain, FFALIGN(decoded_samples, 8));
|
||||
}
|
||||
}
|
||||
|
||||
frame->nb_samples = decoded_samples;
|
||||
*got_frame_ptr = !!decoded_samples;
|
||||
|
||||
return avpkt->size;
|
||||
}
|
||||
|
||||
static av_cold void opus_decode_flush(AVCodecContext *ctx)
|
||||
{
|
||||
OpusContext *c = ctx->priv_data;
|
||||
int i;
|
||||
|
||||
for (i = 0; i < c->nb_streams; i++) {
|
||||
OpusStreamContext *s = &c->streams[i];
|
||||
|
||||
memset(&s->packet, 0, sizeof(s->packet));
|
||||
s->delayed_samples = 0;
|
||||
|
||||
if (s->celt_delay)
|
||||
av_audio_fifo_drain(s->celt_delay, av_audio_fifo_size(s->celt_delay));
|
||||
avresample_close(s->avr);
|
||||
|
||||
ff_silk_flush(s->silk);
|
||||
ff_celt_flush(s->celt);
|
||||
}
|
||||
}
|
||||
|
||||
static av_cold int opus_decode_close(AVCodecContext *avctx)
|
||||
{
|
||||
OpusContext *c = avctx->priv_data;
|
||||
int i;
|
||||
|
||||
for (i = 0; i < c->nb_streams; i++) {
|
||||
OpusStreamContext *s = &c->streams[i];
|
||||
|
||||
ff_silk_free(&s->silk);
|
||||
ff_celt_free(&s->celt);
|
||||
|
||||
av_freep(&s->out_dummy);
|
||||
s->out_dummy_allocated_size = 0;
|
||||
|
||||
av_audio_fifo_free(s->celt_delay);
|
||||
avresample_free(&s->avr);
|
||||
}
|
||||
|
||||
av_freep(&c->streams);
|
||||
c->nb_streams = 0;
|
||||
|
||||
av_freep(&c->channel_maps);
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
static av_cold int opus_decode_init(AVCodecContext *avctx)
|
||||
{
|
||||
OpusContext *c = avctx->priv_data;
|
||||
int ret, i, j;
|
||||
|
||||
avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
|
||||
avctx->sample_rate = 48000;
|
||||
|
||||
avpriv_float_dsp_init(&c->fdsp, 0);
|
||||
|
||||
/* find out the channel configuration */
|
||||
ret = ff_opus_parse_extradata(avctx, c);
|
||||
if (ret < 0)
|
||||
return ret;
|
||||
|
||||
/* allocate and init each independent decoder */
|
||||
c->streams = av_mallocz_array(c->nb_streams, sizeof(*c->streams));
|
||||
if (!c->streams) {
|
||||
c->nb_streams = 0;
|
||||
ret = AVERROR(ENOMEM);
|
||||
goto fail;
|
||||
}
|
||||
|
||||
for (i = 0; i < c->nb_streams; i++) {
|
||||
OpusStreamContext *s = &c->streams[i];
|
||||
uint64_t layout;
|
||||
|
||||
s->output_channels = (i < c->nb_stereo_streams) ? 2 : 1;
|
||||
|
||||
s->avctx = avctx;
|
||||
|
||||
for (j = 0; j < s->output_channels; j++) {
|
||||
s->silk_output[j] = s->silk_buf[j];
|
||||
s->celt_output[j] = s->celt_buf[j];
|
||||
s->redundancy_output[j] = s->redundancy_buf[j];
|
||||
}
|
||||
|
||||
s->fdsp = &c->fdsp;
|
||||
|
||||
s->avr = avresample_alloc_context();
|
||||
if (!s->avr)
|
||||
goto fail;
|
||||
|
||||
layout = (s->output_channels == 1) ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;
|
||||
av_opt_set_int(s->avr, "in_sample_fmt", avctx->sample_fmt, 0);
|
||||
av_opt_set_int(s->avr, "out_sample_fmt", avctx->sample_fmt, 0);
|
||||
av_opt_set_int(s->avr, "in_channel_layout", layout, 0);
|
||||
av_opt_set_int(s->avr, "out_channel_layout", layout, 0);
|
||||
av_opt_set_int(s->avr, "out_sample_rate", avctx->sample_rate, 0);
|
||||
|
||||
ret = ff_silk_init(avctx, &s->silk, s->output_channels);
|
||||
if (ret < 0)
|
||||
goto fail;
|
||||
|
||||
ret = ff_celt_init(avctx, &s->celt, s->output_channels);
|
||||
if (ret < 0)
|
||||
goto fail;
|
||||
|
||||
s->celt_delay = av_audio_fifo_alloc(avctx->sample_fmt,
|
||||
s->output_channels, 1024);
|
||||
if (!s->celt_delay) {
|
||||
ret = AVERROR(ENOMEM);
|
||||
goto fail;
|
||||
}
|
||||
}
|
||||
|
||||
return 0;
|
||||
fail:
|
||||
opus_decode_close(avctx);
|
||||
return ret;
|
||||
}
|
||||
|
||||
AVCodec ff_opus_decoder = {
|
||||
.name = "opus",
|
||||
.long_name = NULL_IF_CONFIG_SMALL("Opus"),
|
||||
.type = AVMEDIA_TYPE_AUDIO,
|
||||
.id = AV_CODEC_ID_OPUS,
|
||||
.priv_data_size = sizeof(OpusContext),
|
||||
.init = opus_decode_init,
|
||||
.close = opus_decode_close,
|
||||
.decode = opus_decode_packet,
|
||||
.flush = opus_decode_flush,
|
||||
.capabilities = CODEC_CAP_DR1 | CODEC_CAP_DELAY,
|
||||
};
|
||||
|
|
@ -29,8 +29,8 @@
|
|||
#include "libavutil/version.h"
|
||||
|
||||
#define LIBAVCODEC_VERSION_MAJOR 55
|
||||
#define LIBAVCODEC_VERSION_MINOR 61
|
||||
#define LIBAVCODEC_VERSION_MICRO 101
|
||||
#define LIBAVCODEC_VERSION_MINOR 62
|
||||
#define LIBAVCODEC_VERSION_MICRO 100
|
||||
|
||||
#define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
|
||||
LIBAVCODEC_VERSION_MINOR, \
|
||||
|
|
|
|||
|
|
@ -119,6 +119,7 @@ include $(SRC_PATH)/tests/fate/microsoft.mak
|
|||
include $(SRC_PATH)/tests/fate/monkeysaudio.mak
|
||||
include $(SRC_PATH)/tests/fate/mp3.mak
|
||||
include $(SRC_PATH)/tests/fate/mpc.mak
|
||||
include $(SRC_PATH)/tests/fate/opus.mak
|
||||
include $(SRC_PATH)/tests/fate/pcm.mak
|
||||
include $(SRC_PATH)/tests/fate/probe.mak
|
||||
include $(SRC_PATH)/tests/fate/prores.mak
|
||||
|
|
|
|||
|
|
@ -0,0 +1,39 @@
|
|||
# The samples were produced by simply rewrapping the official test vectors from
|
||||
# their custom format into Matroska.
|
||||
# The reference files were created with our decoder and tested against the
|
||||
# libopus output with the official opus_compare tool. We cannot use libopus
|
||||
# output as reference directly, because the use of different resamplers would
|
||||
# require too high fuzz values, which can hide bugs.
|
||||
# Before adding new tests here, always make sure they pass opus_compare.
|
||||
|
||||
OPUS_CELT_SAMPLES = $(addprefix testvector, 01 07 11) tron.6ch.tinypkts
|
||||
OPUS_HYBRID_SAMPLES = $(addprefix testvector, 05 06)
|
||||
OPUS_SILK_SAMPLES = $(addprefix testvector, 02 03 04)
|
||||
OPUS_SAMPLES = $(addprefix testvector, 08 09 10 12)
|
||||
|
||||
define FATE_OPUS_TEST
|
||||
FATE_OPUS += fate-opus-$(1)
|
||||
FATE_OPUS$(2) += fate-opus-$(1)
|
||||
fate-opus-$(1): CMD = ffmpeg -i $(TARGET_SAMPLES)/opus/$(1).mka -f f32le -
|
||||
fate-opus-$(1): REF = $(TARGET_SAMPLES)/opus/$(1).f32
|
||||
endef
|
||||
|
||||
$(foreach N,$(OPUS_CELT_SAMPLES), $(eval $(call FATE_OPUS_TEST,$(N),_CELT)))
|
||||
$(foreach N,$(OPUS_HYBRID_SAMPLES),$(eval $(call FATE_OPUS_TEST,$(N),_HYBRID)))
|
||||
$(foreach N,$(OPUS_SILK_SAMPLES), $(eval $(call FATE_OPUS_TEST,$(N),_SILK)))
|
||||
$(foreach N,$(OPUS_SAMPLES), $(eval $(call FATE_OPUS_TEST,$(N),)))
|
||||
|
||||
FATE_OPUS := $(sort $(FATE_OPUS))
|
||||
|
||||
$(FATE_OPUS): CMP = stddev
|
||||
$(FATE_OPUS): CMP_UNIT = f32
|
||||
$(FATE_OPUS): FUZZ = 3
|
||||
|
||||
$(FATE_OPUS_CELT): CMP = oneoff
|
||||
$(FATE_OPUS_CELT): FUZZ = 5
|
||||
|
||||
#FATE_SAMPLES_AVCONV-$(call DEMDEC, MATROSKA, OPUS) += $(FATE_OPUS)
|
||||
fate-opus-celt: $(FATE_OPUS_CELT)
|
||||
fate-opus-hybrid: $(FATE_OPUS_HYBRID)
|
||||
fate-opus-silk: $(FATE_OPUS_SILK)
|
||||
fate-opus: $(FATE_OPUS)
|
||||
Loading…
Reference in New Issue