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>

1 files changed, 429 insertions, 0 deletions

diff --git a/libavcodec/opus.h b/libavcodec/opus.h
new file mode 100644
index 0000000000..a6eea02803
--- /dev/null
+++ b/libavcodec/opus.h
@@ -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 */