X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fflacdec.c;h=9ca55cc1dedeb7ff2b4b99e3fd8a781928ba9f06;hb=4fbb62a21bd04bf261da2382d5ba6c249c702af8;hp=4563329e5aba1afbddeaf1472c700c45e35ef364;hpb=5756bc7b025d305a2b5461b9865ff69fd913d546;p=ffmpeg

diff --git a/libavcodec/flacdec.c b/libavcodec/flacdec.c
index 4563329e5ab..9ca55cc1ded 100644
--- a/libavcodec/flacdec.c
+++ b/libavcodec/flacdec.c
@@ -2,30 +2,28 @@
  * FLAC (Free Lossless Audio Codec) decoder
  * Copyright (c) 2003 Alex Beregszaszi
  *
- * This file is part of FFmpeg.
+ * This file is part of Libav.
  *
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav 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,
+ * Libav 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
+ * License along with Libav; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
 /**
- * @file libavcodec/flacdec.c
+ * @file
  * FLAC (Free Lossless Audio Codec) decoder
  * @author Alex Beregszaszi
- *
- * For more information on the FLAC format, visit:
- *  http://flac.sourceforge.net/
+ * @see http://flac.sourceforge.net/
  *
  * This decoder can be used in 1 of 2 ways: Either raw FLAC data can be fed
  * through, starting from the initial 'fLaC' signature; or by passing the
@@ -35,26 +33,16 @@
 
 #include <limits.h>
 
+#include "libavutil/channel_layout.h"
 #include "libavutil/crc.h"
 #include "avcodec.h"
 #include "internal.h"
-#include "bitstream.h"
+#include "get_bits.h"
 #include "bytestream.h"
 #include "golomb.h"
 #include "flac.h"
-
-#undef NDEBUG
-#include <assert.h>
-
-#define MAX_CHANNELS 8
-#define MAX_BLOCKSIZE 65535
-
-enum decorrelation_type {
-    INDEPENDENT,
-    LEFT_SIDE,
-    RIGHT_SIDE,
-    MID_SIDE,
-};
+#include "flacdata.h"
+#include "flacdsp.h"
 
 typedef struct FLACContext {
     FLACSTREAMINFO
@@ -63,89 +51,64 @@ typedef struct FLACContext {
     GetBitContext gb;                       ///< GetBitContext initialized to start at the current frame
 
     int blocksize;                          ///< number of samples in the current frame
-    int curr_bps;                           ///< bps for current subframe, adjusted for channel correlation and wasted bits
     int sample_shift;                       ///< shift required to make output samples 16-bit or 32-bit
-    int is32;                               ///< flag to indicate if output should be 32-bit instead of 16-bit
-    enum decorrelation_type decorrelation;  ///< channel decorrelation type in the current frame
+    int ch_mode;                            ///< channel decorrelation type in the current frame
     int got_streaminfo;                     ///< indicates if the STREAMINFO has been read
 
-    int32_t *decoded[MAX_CHANNELS];         ///< decoded samples
-    uint8_t *bitstream;
-    unsigned int bitstream_size;
-    unsigned int bitstream_index;
-    unsigned int allocated_bitstream_size;
-} FLACContext;
-
-static const int sample_rate_table[] =
-{ 0,
-  88200, 176400, 192000,
-  8000, 16000, 22050, 24000, 32000, 44100, 48000, 96000,
-  0, 0, 0, 0 };
-
-static const int sample_size_table[] =
-{ 0, 8, 12, 0, 16, 20, 24, 0 };
-
-static const int blocksize_table[] = {
-     0,    192, 576<<0, 576<<1, 576<<2, 576<<3,      0,      0,
-256<<0, 256<<1, 256<<2, 256<<3, 256<<4, 256<<5, 256<<6, 256<<7
-};
+    int32_t *decoded[FLAC_MAX_CHANNELS];    ///< decoded samples
+    uint8_t *decoded_buffer;
+    unsigned int decoded_buffer_size;
 
-static int64_t get_utf8(GetBitContext *gb)
-{
-    int64_t val;
-    GET_UTF8(val, get_bits(gb, 8), return -1;)
-    return val;
-}
+    FLACDSPContext dsp;
+} FLACContext;
 
-static void allocate_buffers(FLACContext *s);
+static int allocate_buffers(FLACContext *s);
 
-int ff_flac_is_extradata_valid(AVCodecContext *avctx,
-                               enum FLACExtradataFormat *format,
-                               uint8_t **streaminfo_start)
+static void flac_set_bps(FLACContext *s)
 {
-    if (!avctx->extradata || avctx->extradata_size < FLAC_STREAMINFO_SIZE) {
-        av_log(avctx, AV_LOG_ERROR, "extradata NULL or too small.\n");
-        return 0;
-    }
-    if (AV_RL32(avctx->extradata) != MKTAG('f','L','a','C')) {
-        /* extradata contains STREAMINFO only */
-        if (avctx->extradata_size != FLAC_STREAMINFO_SIZE) {
-            av_log(avctx, AV_LOG_WARNING, "extradata contains %d bytes too many.\n",
-                   FLAC_STREAMINFO_SIZE-avctx->extradata_size);
-        }
-        *format = FLAC_EXTRADATA_FORMAT_STREAMINFO;
-        *streaminfo_start = avctx->extradata;
+    enum AVSampleFormat req = s->avctx->request_sample_fmt;
+    int need32 = s->bps > 16;
+    int want32 = av_get_bytes_per_sample(req) > 2;
+    int planar = av_sample_fmt_is_planar(req);
+
+    if (need32 || want32) {
+        if (planar)
+            s->avctx->sample_fmt = AV_SAMPLE_FMT_S32P;
+        else
+            s->avctx->sample_fmt = AV_SAMPLE_FMT_S32;
+        s->sample_shift = 32 - s->bps;
     } else {
-        if (avctx->extradata_size < 8+FLAC_STREAMINFO_SIZE) {
-            av_log(avctx, AV_LOG_ERROR, "extradata too small.\n");
-            return 0;
-        }
-        *format = FLAC_EXTRADATA_FORMAT_FULL_HEADER;
-        *streaminfo_start = &avctx->extradata[8];
+        if (planar)
+            s->avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
+        else
+            s->avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+        s->sample_shift = 16 - s->bps;
     }
-    return 1;
 }
 
 static av_cold int flac_decode_init(AVCodecContext *avctx)
 {
     enum FLACExtradataFormat format;
     uint8_t *streaminfo;
+    int ret;
     FLACContext *s = avctx->priv_data;
     s->avctx = avctx;
 
-    avctx->sample_fmt = SAMPLE_FMT_S16;
-
     /* for now, the raw FLAC header is allowed to be passed to the decoder as
        frame data instead of extradata. */
     if (!avctx->extradata)
         return 0;
 
-    if (!ff_flac_is_extradata_valid(avctx, &format, &streaminfo))
-        return -1;
+    if (!avpriv_flac_is_extradata_valid(avctx, &format, &streaminfo))
+        return AVERROR_INVALIDDATA;
 
     /* initialize based on the demuxer-supplied streamdata header */
-    ff_flac_parse_streaminfo(avctx, (FLACStreaminfo *)s, streaminfo);
-    allocate_buffers(s);
+    avpriv_flac_parse_streaminfo(avctx, (FLACStreaminfo *)s, streaminfo);
+    ret = allocate_buffers(s);
+    if (ret < 0)
+        return ret;
+    flac_set_bps(s);
+    ff_flacdsp_init(&s->dsp, avctx->sample_fmt, s->bps);
     s->got_streaminfo = 1;
 
     return 0;
@@ -160,64 +123,22 @@ static void dump_headers(AVCodecContext *avctx, FLACStreaminfo *s)
     av_log(avctx, AV_LOG_DEBUG, "  Bits: %d\n", s->bps);
 }
 
-static void allocate_buffers(FLACContext *s)
+static int allocate_buffers(FLACContext *s)
 {
-    int i;
+    int buf_size;
 
-    assert(s->max_blocksize);
+    buf_size = av_samples_get_buffer_size(NULL, s->channels, s->max_blocksize,
+                                          AV_SAMPLE_FMT_S32P, 0);
+    if (buf_size < 0)
+        return buf_size;
 
-    if (s->max_framesize == 0 && s->max_blocksize) {
-        // FIXME header overhead
-        s->max_framesize= (s->channels * s->bps * s->max_blocksize + 7)/ 8;
-    }
+    av_fast_malloc(&s->decoded_buffer, &s->decoded_buffer_size, buf_size);
+    if (!s->decoded_buffer)
+        return AVERROR(ENOMEM);
 
-    for (i = 0; i < s->channels; i++) {
-        s->decoded[i] = av_realloc(s->decoded[i],
-                                   sizeof(int32_t)*s->max_blocksize);
-    }
-
-    if (s->allocated_bitstream_size < s->max_framesize)
-        s->bitstream= av_fast_realloc(s->bitstream,
-                                      &s->allocated_bitstream_size,
-                                      s->max_framesize);
-}
-
-void ff_flac_parse_streaminfo(AVCodecContext *avctx, struct FLACStreaminfo *s,
-                              const uint8_t *buffer)
-{
-    GetBitContext gb;
-    init_get_bits(&gb, buffer, FLAC_STREAMINFO_SIZE*8);
-
-    skip_bits(&gb, 16); /* skip min blocksize */
-    s->max_blocksize = get_bits(&gb, 16);
-    if (s->max_blocksize < 16) {
-        av_log(avctx, AV_LOG_WARNING, "invalid max blocksize: %d\n",
-               s->max_blocksize);
-        s->max_blocksize = 16;
-    }
-
-    skip_bits(&gb, 24); /* skip min frame size */
-    s->max_framesize = get_bits_long(&gb, 24);
-
-    s->samplerate = get_bits_long(&gb, 20);
-    s->channels = get_bits(&gb, 3) + 1;
-    s->bps = get_bits(&gb, 5) + 1;
-
-    avctx->channels = s->channels;
-    avctx->sample_rate = s->samplerate;
-    avctx->bits_per_raw_sample = s->bps;
-    if (s->bps > 16)
-        avctx->sample_fmt = SAMPLE_FMT_S32;
-    else
-        avctx->sample_fmt = SAMPLE_FMT_S16;
-
-    s->samples  = get_bits_long(&gb, 32) << 4;
-    s->samples |= get_bits(&gb, 4);
-
-    skip_bits_long(&gb, 64); /* md5 sum */
-    skip_bits_long(&gb, 64); /* md5 sum */
-
-    dump_headers(avctx, s);
+    return av_samples_fill_arrays((uint8_t **)s->decoded, NULL,
+                                  s->decoded_buffer, s->channels,
+                                  s->max_blocksize, AV_SAMPLE_FMT_S32P, 0);
 }
 
 /**
@@ -229,21 +150,23 @@ void ff_flac_parse_streaminfo(AVCodecContext *avctx, struct FLACStreaminfo *s,
  */
 static int parse_streaminfo(FLACContext *s, const uint8_t *buf, int buf_size)
 {
-    int metadata_type, metadata_size;
+    int metadata_type, metadata_size, ret;
 
     if (buf_size < FLAC_STREAMINFO_SIZE+8) {
         /* need more data */
         return 0;
     }
-    buf += 4;
-    metadata_type = bytestream_get_byte(&buf) & 0x7F;
-    metadata_size = bytestream_get_be24(&buf);
+    flac_parse_block_header(&buf[4], NULL, &metadata_type, &metadata_size);
     if (metadata_type != FLAC_METADATA_TYPE_STREAMINFO ||
         metadata_size != FLAC_STREAMINFO_SIZE) {
         return AVERROR_INVALIDDATA;
     }
-    ff_flac_parse_streaminfo(s->avctx, (FLACStreaminfo *)s, buf);
-    allocate_buffers(s);
+    avpriv_flac_parse_streaminfo(s->avctx, (FLACStreaminfo *)s, &buf[8]);
+    ret = allocate_buffers(s);
+    if (ret < 0)
+        return ret;
+    flac_set_bps(s);
+    ff_flacdsp_init(&s->dsp, s->avctx->sample_fmt, s->bps);
     s->got_streaminfo = 1;
 
     return 0;
@@ -262,9 +185,11 @@ static int get_metadata_size(const uint8_t *buf, int buf_size)
 
     buf += 4;
     do {
-        metadata_last = bytestream_get_byte(&buf) & 0x80;
-        metadata_size = bytestream_get_be24(&buf);
-        if (buf + metadata_size > buf_end) {
+        if (buf_end - buf < 4)
+            return 0;
+        flac_parse_block_header(buf, &metadata_last, NULL, &metadata_size);
+        buf += 4;
+        if (buf_end - buf < metadata_size) {
             /* need more data in order to read the complete header */
             return 0;
         }
@@ -274,16 +199,17 @@ static int get_metadata_size(const uint8_t *buf, int buf_size)
     return buf_size - (buf_end - buf);
 }
 
-static int decode_residuals(FLACContext *s, int channel, int pred_order)
+static int decode_residuals(FLACContext *s, int32_t *decoded, int pred_order)
 {
     int i, tmp, partition, method_type, rice_order;
-    int sample = 0, samples;
+    int rice_bits, rice_esc;
+    int samples;
 
     method_type = get_bits(&s->gb, 2);
     if (method_type > 1) {
         av_log(s->avctx, AV_LOG_ERROR, "illegal residual coding method %d\n",
                method_type);
-        return -1;
+        return AVERROR_INVALIDDATA;
     }
 
     rice_order = get_bits(&s->gb, 4);
@@ -292,20 +218,23 @@ static int decode_residuals(FLACContext *s, int channel, int pred_order)
     if (pred_order > samples) {
         av_log(s->avctx, AV_LOG_ERROR, "invalid predictor order: %i > %i\n",
                pred_order, samples);
-        return -1;
+        return AVERROR_INVALIDDATA;
     }
 
-    sample=
+    rice_bits = 4 + method_type;
+    rice_esc  = (1 << rice_bits) - 1;
+
+    decoded += pred_order;
     i= pred_order;
     for (partition = 0; partition < (1 << rice_order); partition++) {
-        tmp = get_bits(&s->gb, method_type == 0 ? 4 : 5);
-        if (tmp == (method_type == 0 ? 15 : 31)) {
+        tmp = get_bits(&s->gb, rice_bits);
+        if (tmp == rice_esc) {
             tmp = get_bits(&s->gb, 5);
-            for (; i < samples; i++, sample++)
-                s->decoded[channel][sample] = get_sbits_long(&s->gb, tmp);
+            for (; i < samples; i++)
+                *decoded++ = get_sbits_long(&s->gb, tmp);
         } else {
-            for (; i < samples; i++, sample++) {
-                s->decoded[channel][sample] = get_sr_golomb_flac(&s->gb, tmp, INT_MAX, 0);
+            for (; i < samples; i++) {
+                *decoded++ = get_sr_golomb_flac(&s->gb, tmp, INT_MAX, 0);
             }
         }
         i= 0;
@@ -314,19 +243,19 @@ static int decode_residuals(FLACContext *s, int channel, int pred_order)
     return 0;
 }
 
-static int decode_subframe_fixed(FLACContext *s, int channel, int pred_order)
+static int decode_subframe_fixed(FLACContext *s, int32_t *decoded,
+                                 int pred_order, int bps)
 {
     const int blocksize = s->blocksize;
-    int32_t *decoded = s->decoded[channel];
-    int av_uninit(a), av_uninit(b), av_uninit(c), av_uninit(d), i;
+    int a, b, c, d, i, ret;
 
     /* warm up samples */
     for (i = 0; i < pred_order; i++) {
-        decoded[i] = get_sbits_long(&s->gb, s->curr_bps);
+        decoded[i] = get_sbits_long(&s->gb, bps);
     }
 
-    if (decode_residuals(s, channel, pred_order) < 0)
-        return -1;
+    if ((ret = decode_residuals(s, decoded, pred_order)) < 0)
+        return ret;
 
     if (pred_order > 0)
         a = decoded[pred_order-1];
@@ -358,415 +287,287 @@ static int decode_subframe_fixed(FLACContext *s, int channel, int pred_order)
         break;
     default:
         av_log(s->avctx, AV_LOG_ERROR, "illegal pred order %d\n", pred_order);
-        return -1;
+        return AVERROR_INVALIDDATA;
     }
 
     return 0;
 }
 
-static int decode_subframe_lpc(FLACContext *s, int channel, int pred_order)
+static int decode_subframe_lpc(FLACContext *s, int32_t *decoded, int pred_order,
+                               int bps)
 {
-    int i, j;
+    int i, ret;
     int coeff_prec, qlevel;
-    int coeffs[pred_order];
-    int32_t *decoded = s->decoded[channel];
+    int coeffs[32];
 
     /* warm up samples */
     for (i = 0; i < pred_order; i++) {
-        decoded[i] = get_sbits_long(&s->gb, s->curr_bps);
+        decoded[i] = get_sbits_long(&s->gb, bps);
     }
 
     coeff_prec = get_bits(&s->gb, 4) + 1;
     if (coeff_prec == 16) {
         av_log(s->avctx, AV_LOG_ERROR, "invalid coeff precision\n");
-        return -1;
+        return AVERROR_INVALIDDATA;
     }
     qlevel = get_sbits(&s->gb, 5);
     if (qlevel < 0) {
         av_log(s->avctx, AV_LOG_ERROR, "qlevel %d not supported, maybe buggy stream\n",
                qlevel);
-        return -1;
+        return AVERROR_INVALIDDATA;
     }
 
     for (i = 0; i < pred_order; i++) {
-        coeffs[i] = get_sbits(&s->gb, coeff_prec);
+        coeffs[pred_order - i - 1] = get_sbits(&s->gb, coeff_prec);
     }
 
-    if (decode_residuals(s, channel, pred_order) < 0)
-        return -1;
+    if ((ret = decode_residuals(s, decoded, pred_order)) < 0)
+        return ret;
 
-    if (s->bps > 16) {
-        int64_t sum;
-        for (i = pred_order; i < s->blocksize; i++) {
-            sum = 0;
-            for (j = 0; j < pred_order; j++)
-                sum += (int64_t)coeffs[j] * decoded[i-j-1];
-            decoded[i] += sum >> qlevel;
-        }
-    } else {
-        for (i = pred_order; i < s->blocksize-1; i += 2) {
-            int c;
-            int d = decoded[i-pred_order];
-            int s0 = 0, s1 = 0;
-            for (j = pred_order-1; j > 0; j--) {
-                c = coeffs[j];
-                s0 += c*d;
-                d = decoded[i-j];
-                s1 += c*d;
-            }
-            c = coeffs[0];
-            s0 += c*d;
-            d = decoded[i] += s0 >> qlevel;
-            s1 += c*d;
-            decoded[i+1] += s1 >> qlevel;
-        }
-        if (i < s->blocksize) {
-            int sum = 0;
-            for (j = 0; j < pred_order; j++)
-                sum += coeffs[j] * decoded[i-j-1];
-            decoded[i] += sum >> qlevel;
-        }
-    }
+    s->dsp.lpc(decoded, coeffs, pred_order, qlevel, s->blocksize);
 
     return 0;
 }
 
 static inline int decode_subframe(FLACContext *s, int channel)
 {
+    int32_t *decoded = s->decoded[channel];
     int type, wasted = 0;
-    int i, tmp;
+    int bps = s->bps;
+    int i, tmp, ret;
 
-    s->curr_bps = s->bps;
     if (channel == 0) {
-        if (s->decorrelation == RIGHT_SIDE)
-            s->curr_bps++;
+        if (s->ch_mode == FLAC_CHMODE_RIGHT_SIDE)
+            bps++;
     } else {
-        if (s->decorrelation == LEFT_SIDE || s->decorrelation == MID_SIDE)
-            s->curr_bps++;
+        if (s->ch_mode == FLAC_CHMODE_LEFT_SIDE || s->ch_mode == FLAC_CHMODE_MID_SIDE)
+            bps++;
     }
 
     if (get_bits1(&s->gb)) {
         av_log(s->avctx, AV_LOG_ERROR, "invalid subframe padding\n");
-        return -1;
+        return AVERROR_INVALIDDATA;
     }
     type = get_bits(&s->gb, 6);
 
     if (get_bits1(&s->gb)) {
+        int left = get_bits_left(&s->gb);
         wasted = 1;
+        if ( left < 0 ||
+            (left < bps && !show_bits_long(&s->gb, left)) ||
+                           !show_bits_long(&s->gb, bps)) {
+            av_log(s->avctx, AV_LOG_ERROR,
+                   "Invalid number of wasted bits > available bits (%d) - left=%d\n",
+                   bps, left);
+            return AVERROR_INVALIDDATA;
+        }
         while (!get_bits1(&s->gb))
             wasted++;
-        s->curr_bps -= wasted;
+        bps -= wasted;
     }
-    if (s->curr_bps > 32) {
-        ff_log_missing_feature(s->avctx, "decorrelated bit depth > 32", 0);
-        return -1;
+    if (bps > 32) {
+        avpriv_report_missing_feature(s->avctx, "Decorrelated bit depth > 32");
+        return AVERROR_PATCHWELCOME;
     }
 
 //FIXME use av_log2 for types
     if (type == 0) {
-        tmp = get_sbits_long(&s->gb, s->curr_bps);
+        tmp = get_sbits_long(&s->gb, bps);
         for (i = 0; i < s->blocksize; i++)
-            s->decoded[channel][i] = tmp;
+            decoded[i] = tmp;
     } else if (type == 1) {
         for (i = 0; i < s->blocksize; i++)
-            s->decoded[channel][i] = get_sbits_long(&s->gb, s->curr_bps);
+            decoded[i] = get_sbits_long(&s->gb, bps);
     } else if ((type >= 8) && (type <= 12)) {
-        if (decode_subframe_fixed(s, channel, type & ~0x8) < 0)
-            return -1;
+        if ((ret = decode_subframe_fixed(s, decoded, type & ~0x8, bps)) < 0)
+            return ret;
     } else if (type >= 32) {
-        if (decode_subframe_lpc(s, channel, (type & ~0x20)+1) < 0)
-            return -1;
+        if ((ret = decode_subframe_lpc(s, decoded, (type & ~0x20)+1, bps)) < 0)
+            return ret;
     } else {
         av_log(s->avctx, AV_LOG_ERROR, "invalid coding type\n");
-        return -1;
+        return AVERROR_INVALIDDATA;
     }
 
     if (wasted) {
         int i;
         for (i = 0; i < s->blocksize; i++)
-            s->decoded[channel][i] <<= wasted;
+            decoded[i] <<= wasted;
     }
 
     return 0;
 }
 
-static int decode_frame(FLACContext *s, int alloc_data_size)
+static int decode_frame(FLACContext *s)
 {
-    int blocksize_code, sample_rate_code, sample_size_code, assignment, i, crc8;
-    int decorrelation, bps, blocksize, samplerate;
-
-    blocksize_code = get_bits(&s->gb, 4);
-
-    sample_rate_code = get_bits(&s->gb, 4);
-
-    assignment = get_bits(&s->gb, 4); /* channel assignment */
-    if (assignment < 8 && s->channels == assignment+1)
-        decorrelation = INDEPENDENT;
-    else if (assignment >=8 && assignment < 11 && s->channels == 2)
-        decorrelation = LEFT_SIDE + assignment - 8;
-    else {
-        av_log(s->avctx, AV_LOG_ERROR, "unsupported channel assignment %d (channels=%d)\n",
-               assignment, s->channels);
-        return -1;
-    }
-
-    sample_size_code = get_bits(&s->gb, 3);
-    if (sample_size_code == 0)
-        bps= s->bps;
-    else if ((sample_size_code != 3) && (sample_size_code != 7))
-        bps = sample_size_table[sample_size_code];
-    else {
-        av_log(s->avctx, AV_LOG_ERROR, "invalid sample size code (%d)\n",
-               sample_size_code);
-        return -1;
-    }
-    if (bps > 16) {
-        s->avctx->sample_fmt = SAMPLE_FMT_S32;
-        s->sample_shift = 32 - bps;
-        s->is32 = 1;
-    } else {
-        s->avctx->sample_fmt = SAMPLE_FMT_S16;
-        s->sample_shift = 16 - bps;
-        s->is32 = 0;
+    int i, ret;
+    GetBitContext *gb = &s->gb;
+    FLACFrameInfo fi;
+
+    if ((ret = ff_flac_decode_frame_header(s->avctx, gb, &fi, 0)) < 0) {
+        av_log(s->avctx, AV_LOG_ERROR, "invalid frame header\n");
+        return ret;
     }
-    s->bps = s->avctx->bits_per_raw_sample = bps;
 
-    if (get_bits1(&s->gb)) {
-        av_log(s->avctx, AV_LOG_ERROR, "broken stream, invalid padding\n");
-        return -1;
+    if (s->channels && fi.channels != s->channels && s->got_streaminfo) {
+        s->channels = s->avctx->channels = fi.channels;
+        ff_flac_set_channel_layout(s->avctx);
+        ret = allocate_buffers(s);
+        if (ret < 0)
+            return ret;
     }
+    s->channels = s->avctx->channels = fi.channels;
+    if (!s->avctx->channel_layout)
+        ff_flac_set_channel_layout(s->avctx);
+    s->ch_mode = fi.ch_mode;
 
-    if (get_utf8(&s->gb) < 0) {
-        av_log(s->avctx, AV_LOG_ERROR, "utf8 fscked\n");
-        return -1;
+    if (!s->bps && !fi.bps) {
+        av_log(s->avctx, AV_LOG_ERROR, "bps not found in STREAMINFO or frame header\n");
+        return AVERROR_INVALIDDATA;
+    }
+    if (!fi.bps) {
+        fi.bps = s->bps;
+    } else if (s->bps && fi.bps != s->bps) {
+        av_log(s->avctx, AV_LOG_ERROR, "switching bps mid-stream is not "
+                                       "supported\n");
+        return AVERROR_INVALIDDATA;
     }
 
-    if (blocksize_code == 0) {
-        av_log(s->avctx, AV_LOG_ERROR, "reserved blocksize code: 0\n");
-        return -1;
-    } else if (blocksize_code == 6)
-        blocksize = get_bits(&s->gb, 8)+1;
-    else if (blocksize_code == 7)
-        blocksize = get_bits(&s->gb, 16)+1;
-    else
-        blocksize = blocksize_table[blocksize_code];
+    if (!s->bps) {
+        s->bps = s->avctx->bits_per_raw_sample = fi.bps;
+        flac_set_bps(s);
+    }
 
-    if (blocksize > s->max_blocksize) {
-        av_log(s->avctx, AV_LOG_ERROR, "blocksize %d > %d\n", blocksize,
+    if (!s->max_blocksize)
+        s->max_blocksize = FLAC_MAX_BLOCKSIZE;
+    if (fi.blocksize > s->max_blocksize) {
+        av_log(s->avctx, AV_LOG_ERROR, "blocksize %d > %d\n", fi.blocksize,
                s->max_blocksize);
-        return -1;
+        return AVERROR_INVALIDDATA;
     }
+    s->blocksize = fi.blocksize;
 
-    if (blocksize * s->channels * (s->is32 ? 4 : 2) > alloc_data_size)
-        return -1;
-
-    if (sample_rate_code == 0)
-        samplerate= s->samplerate;
-    else if (sample_rate_code < 12)
-        samplerate = sample_rate_table[sample_rate_code];
-    else if (sample_rate_code == 12)
-        samplerate = get_bits(&s->gb, 8) * 1000;
-    else if (sample_rate_code == 13)
-        samplerate = get_bits(&s->gb, 16);
-    else if (sample_rate_code == 14)
-        samplerate = get_bits(&s->gb, 16) * 10;
-    else {
-        av_log(s->avctx, AV_LOG_ERROR, "illegal sample rate code %d\n",
-               sample_rate_code);
-        return -1;
+    if (!s->samplerate && !fi.samplerate) {
+        av_log(s->avctx, AV_LOG_ERROR, "sample rate not found in STREAMINFO"
+                                        " or frame header\n");
+        return AVERROR_INVALIDDATA;
     }
+    if (fi.samplerate == 0)
+        fi.samplerate = s->samplerate;
+    s->samplerate = s->avctx->sample_rate = fi.samplerate;
 
-    skip_bits(&s->gb, 8);
-    crc8 = av_crc(av_crc_get_table(AV_CRC_8_ATM), 0,
-                  s->gb.buffer, get_bits_count(&s->gb)/8);
-    if (crc8) {
-        av_log(s->avctx, AV_LOG_ERROR, "header crc mismatch crc=%2X\n", crc8);
-        return -1;
+    if (!s->got_streaminfo) {
+        ret = allocate_buffers(s);
+        if (ret < 0)
+            return ret;
+        ff_flacdsp_init(&s->dsp, s->avctx->sample_fmt, s->bps);
+        s->got_streaminfo = 1;
+        dump_headers(s->avctx, (FLACStreaminfo *)s);
     }
 
-    s->blocksize    = blocksize;
-    s->samplerate   = samplerate;
-    s->bps          = bps;
-    s->decorrelation= decorrelation;
-
 //    dump_headers(s->avctx, (FLACStreaminfo *)s);
 
     /* subframes */
     for (i = 0; i < s->channels; i++) {
-        if (decode_subframe(s, i) < 0)
-            return -1;
+        if ((ret = decode_subframe(s, i)) < 0)
+            return ret;
     }
 
-    align_get_bits(&s->gb);
+    align_get_bits(gb);
 
     /* frame footer */
-    skip_bits(&s->gb, 16); /* data crc */
+    skip_bits(gb, 16); /* data crc */
 
     return 0;
 }
 
-static int flac_decode_frame(AVCodecContext *avctx,
-                            void *data, int *data_size,
-                            const uint8_t *buf, int buf_size)
+static int flac_decode_frame(AVCodecContext *avctx, void *data,
+                             int *got_frame_ptr, AVPacket *avpkt)
 {
+    AVFrame *frame     = data;
+    const uint8_t *buf = avpkt->data;
+    int buf_size = avpkt->size;
     FLACContext *s = avctx->priv_data;
-    int i, j = 0, input_buf_size = 0, bytes_read = 0;
-    int16_t *samples_16 = data;
-    int32_t *samples_32 = data;
-    int alloc_data_size= *data_size;
+    int bytes_read = 0;
+    int ret;
 
-    *data_size=0;
+    *got_frame_ptr = 0;
 
     if (s->max_framesize == 0) {
-        s->max_framesize= FFMAX(4, buf_size); // should hopefully be enough for the first header
-        s->bitstream= av_fast_realloc(s->bitstream, &s->allocated_bitstream_size, s->max_framesize);
-    }
-
-    if (1 && s->max_framesize) { //FIXME truncated
-        if (s->bitstream_size < 4 || AV_RL32(s->bitstream) != MKTAG('f','L','a','C'))
-            buf_size= FFMIN(buf_size, s->max_framesize - FFMIN(s->bitstream_size, s->max_framesize));
-        input_buf_size= buf_size;
-
-        if (s->bitstream_size + buf_size < buf_size || s->bitstream_index + s->bitstream_size + buf_size < s->bitstream_index)
-            return -1;
-
-        if (s->allocated_bitstream_size < s->bitstream_size + buf_size)
-            s->bitstream= av_fast_realloc(s->bitstream, &s->allocated_bitstream_size, s->bitstream_size + buf_size);
-
-        if (s->bitstream_index + s->bitstream_size + buf_size > s->allocated_bitstream_size) {
-            memmove(s->bitstream, &s->bitstream[s->bitstream_index],
-                    s->bitstream_size);
-            s->bitstream_index=0;
-        }
-        memcpy(&s->bitstream[s->bitstream_index + s->bitstream_size],
-               buf, buf_size);
-        buf= &s->bitstream[s->bitstream_index];
-        buf_size += s->bitstream_size;
-        s->bitstream_size= buf_size;
-
-        if (buf_size < s->max_framesize && input_buf_size) {
-            return input_buf_size;
-        }
+        s->max_framesize =
+            ff_flac_get_max_frame_size(s->max_blocksize ? s->max_blocksize : FLAC_MAX_BLOCKSIZE,
+                                       FLAC_MAX_CHANNELS, 32);
     }
 
     /* check that there is at least the smallest decodable amount of data.
-       this amount corresponds to the smallest valid FLAC frame possible. */
-    if (buf_size < 16)
-        goto end;
+       this amount corresponds to the smallest valid FLAC frame possible.
+       FF F8 69 02 00 00 9A 00 00 34 46 */
+    if (buf_size < FLAC_MIN_FRAME_SIZE)
+        return buf_size;
 
     /* check for inline header */
     if (AV_RB32(buf) == MKBETAG('f','L','a','C')) {
-        if (!s->got_streaminfo && parse_streaminfo(s, buf, buf_size)) {
+        if (!s->got_streaminfo && (ret = parse_streaminfo(s, buf, buf_size))) {
             av_log(s->avctx, AV_LOG_ERROR, "invalid header\n");
-            return -1;
+            return ret;
         }
-        bytes_read = get_metadata_size(buf, buf_size);
-        goto end;
-    }
-
-    /* check for frame sync code and resync stream if necessary */
-    if ((AV_RB16(buf) & 0xFFFE) != 0xFFF8) {
-        const uint8_t *buf_end = buf + buf_size;
-        av_log(s->avctx, AV_LOG_ERROR, "FRAME HEADER not here\n");
-        while (buf+2 < buf_end && (AV_RB16(buf) & 0xFFFE) != 0xFFF8)
-            buf++;
-        bytes_read = buf_size - (buf_end - buf);
-        goto end; // we may not have enough bits left to decode a frame, so try next time
+        return get_metadata_size(buf, buf_size);
     }
 
     /* decode frame */
     init_get_bits(&s->gb, buf, buf_size*8);
-    skip_bits(&s->gb, 16);
-    if (decode_frame(s, alloc_data_size) < 0) {
+    if ((ret = decode_frame(s)) < 0) {
         av_log(s->avctx, AV_LOG_ERROR, "decode_frame() failed\n");
-        s->bitstream_size=0;
-        s->bitstream_index=0;
-        return -1;
+        return ret;
     }
-    *data_size = s->blocksize * s->channels * (s->is32 ? 4 : 2);
     bytes_read = (get_bits_count(&s->gb)+7)/8;
 
-#define DECORRELATE(left, right)\
-            assert(s->channels == 2);\
-            for (i = 0; i < s->blocksize; i++) {\
-                int a= s->decoded[0][i];\
-                int b= s->decoded[1][i];\
-                if (s->is32) {\
-                    *samples_32++ = (left)  << s->sample_shift;\
-                    *samples_32++ = (right) << s->sample_shift;\
-                } else {\
-                    *samples_16++ = (left)  << s->sample_shift;\
-                    *samples_16++ = (right) << s->sample_shift;\
-                }\
-            }\
-            break;
-
-    switch (s->decorrelation) {
-    case INDEPENDENT:
-        for (j = 0; j < s->blocksize; j++) {
-            for (i = 0; i < s->channels; i++) {
-                if (s->is32)
-                    *samples_32++ = s->decoded[i][j] << s->sample_shift;
-                else
-                    *samples_16++ = s->decoded[i][j] << s->sample_shift;
-            }
-        }
-        break;
-    case LEFT_SIDE:
-        DECORRELATE(a,a-b)
-    case RIGHT_SIDE:
-        DECORRELATE(a+b,b)
-    case MID_SIDE:
-        DECORRELATE( (a-=b>>1) + b, a)
+    /* get output buffer */
+    frame->nb_samples = s->blocksize;
+    if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
+        av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+        return ret;
     }
 
-end:
+    s->dsp.decorrelate[s->ch_mode](frame->data, s->decoded, s->channels,
+                                   s->blocksize, s->sample_shift);
+
     if (bytes_read > buf_size) {
         av_log(s->avctx, AV_LOG_ERROR, "overread: %d\n", bytes_read - buf_size);
-        s->bitstream_size=0;
-        s->bitstream_index=0;
-        return -1;
+        return AVERROR_INVALIDDATA;
+    }
+    if (bytes_read < buf_size) {
+        av_log(s->avctx, AV_LOG_DEBUG, "underread: %d orig size: %d\n",
+               buf_size - bytes_read, buf_size);
     }
 
-    if (s->bitstream_size) {
-        s->bitstream_index += bytes_read;
-        s->bitstream_size  -= bytes_read;
-        return input_buf_size;
-    } else
-        return bytes_read;
+    *got_frame_ptr = 1;
+
+    return bytes_read;
 }
 
 static av_cold int flac_decode_close(AVCodecContext *avctx)
 {
     FLACContext *s = avctx->priv_data;
-    int i;
 
-    for (i = 0; i < s->channels; i++) {
-        av_freep(&s->decoded[i]);
-    }
-    av_freep(&s->bitstream);
+    av_freep(&s->decoded_buffer);
 
     return 0;
 }
 
-static void flac_flush(AVCodecContext *avctx)
-{
-    FLACContext *s = avctx->priv_data;
-
-    s->bitstream_size=
-    s->bitstream_index= 0;
-}
-
-AVCodec flac_decoder = {
-    "flac",
-    CODEC_TYPE_AUDIO,
-    CODEC_ID_FLAC,
-    sizeof(FLACContext),
-    flac_decode_init,
-    NULL,
-    flac_decode_close,
-    flac_decode_frame,
-    CODEC_CAP_DELAY,
-    .flush= flac_flush,
-    .long_name= NULL_IF_CONFIG_SMALL("FLAC (Free Lossless Audio Codec)"),
+AVCodec ff_flac_decoder = {
+    .name           = "flac",
+    .long_name      = NULL_IF_CONFIG_SMALL("FLAC (Free Lossless Audio Codec)"),
+    .type           = AVMEDIA_TYPE_AUDIO,
+    .id             = AV_CODEC_ID_FLAC,
+    .priv_data_size = sizeof(FLACContext),
+    .init           = flac_decode_init,
+    .close          = flac_decode_close,
+    .decode         = flac_decode_frame,
+    .capabilities   = CODEC_CAP_DR1,
+    .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16,
+                                                      AV_SAMPLE_FMT_S16P,
+                                                      AV_SAMPLE_FMT_S32,
+                                                      AV_SAMPLE_FMT_S32P,
+                                                      -1 },
 };