Merge remote-tracking branch 'qatar/master'

* qatar/master: (51 commits)
  cin audio: use sign_extend() instead of casting to int16_t
  cin audio: restructure decoding loop to avoid a separate counter variable
  cin audio: use local variable for delta value
  cin audio: remove unneeded cast from void*
  cin audio: validate the channel count
  cin audio: remove unneeded AVCodecContext pointer from CinAudioContext
  dsicin: fix several audio-related fields in the CIN demuxer
  flacdec: use av_get_bytes_per_sample() to get sample size
  dca: handle errors from dca_decode_block()
  dca: return error if the frame header is invalid
  dca: return proper error codes instead of -1
  utvideo: handle empty Huffman trees
  binkaudio: change short to int16_t
  binkaudio: only decode one block at a time.
  binkaudio: store interleaved overlap samples in BinkAudioContext.
  binkaudio: pre-calculate quantization factors
  binkaudio: add some buffer overread checks.
  atrac3: support float or int16 output using request_sample_fmt
  atrac3: add CODEC_CAP_SUBFRAMES capability
  atrac3: return appropriate error codes instead of -1
  ...

Conflicts:
	libavcodec/atrac1.c
	libavcodec/dca.c
	libavformat/mov.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>

14 files changed, 493 insertions, 277 deletions

diff --git a/libavcodec/atrac1.c b/libavcodec/atrac1.c
index 2ad99bf473..d4d5986821 100644
--- a/libavcodec/atrac1.c
+++ b/libavcodec/atrac1.c
@@ -36,6 +36,7 @@
 #include "get_bits.h"
 #include "dsputil.h"
 #include "fft.h"
+#include "fmtconvert.h"
 #include "sinewin.h"
 
 #include "atrac.h"
@@ -78,10 +79,11 @@ typedef struct {
     DECLARE_ALIGNED(32, float,  mid)[256];
     DECLARE_ALIGNED(32, float, high)[512];
     float*              bands[3];
-    DECLARE_ALIGNED(32, float, out_samples)[AT1_MAX_CHANNELS][AT1_SU_SAMPLES];
+    float              *out_samples[AT1_MAX_CHANNELS];
     FFTContext          mdct_ctx[3];
     int                 channels;
     DSPContext          dsp;
+    FmtConvertContext   fmt_conv;
 } AT1Ctx;
 
 /** size of the transform in samples in the long mode for each QMF band */
@@ -129,7 +131,7 @@ static int at1_imdct_block(AT1SUCtx* su, AT1Ctx *q)
             nbits = mdct_long_nbits[band_num] - log2_block_count;
 
             if (nbits != 5 && nbits != 7 && nbits != 8)
-                return -1;
+                return AVERROR_INVALIDDATA;
         } else {
             block_size = 32;
             nbits = 5;
@@ -173,14 +175,14 @@ static int at1_parse_bsm(GetBitContext* gb, int log2_block_cnt[AT1_QMF_BANDS])
         /* low and mid band */
         log2_block_count_tmp = get_bits(gb, 2);
         if (log2_block_count_tmp & 1)
-            return -1;
+            return AVERROR_INVALIDDATA;
         log2_block_cnt[i] = 2 - log2_block_count_tmp;
     }
 
     /* high band */
     log2_block_count_tmp = get_bits(gb, 2);
     if (log2_block_count_tmp != 0 && log2_block_count_tmp != 3)
-        return -1;
+        return AVERROR_INVALIDDATA;
     log2_block_cnt[IDX_HIGH_BAND] = 3 - log2_block_count_tmp;
 
     skip_bits(gb, 2);
@@ -229,7 +231,7 @@ static int at1_unpack_dequant(GetBitContext* gb, AT1SUCtx* su,
 
             /* check for bitstream overflow */
             if (bits_used > AT1_SU_MAX_BITS)
-                return -1;
+                return AVERROR_INVALIDDATA;
 
             /* get the position of the 1st spec according to the block size mode */
             pos = su->log2_block_count[band_num] ? bfu_start_short[bfu_num] : bfu_start_long[bfu_num];
@@ -276,14 +278,21 @@ static int atrac1_decode_frame(AVCodecContext *avctx, void *data,
     const uint8_t *buf = avpkt->data;
     int buf_size       = avpkt->size;
     AT1Ctx *q          = avctx->priv_data;
-    int ch, ret, i;
+    int ch, ret, out_size;
     GetBitContext gb;
     float* samples = data;
 
 
     if (buf_size < 212 * q->channels) {
-        av_log(avctx, AV_LOG_ERROR,"Not enought data to decode!\n");
-        return -1;
+        av_log(avctx,AV_LOG_ERROR,"Not enough data to decode!\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    out_size = q->channels * AT1_SU_SAMPLES *
+               av_get_bytes_per_sample(avctx->sample_fmt);
+    if (*data_size < out_size) {
+        av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
+        return AVERROR(EINVAL);
     }
 
     for (ch = 0; ch < q->channels; ch++) {
@@ -303,44 +312,72 @@ static int atrac1_decode_frame(AVCodecContext *avctx, void *data,
         ret = at1_imdct_block(su, q);
         if (ret < 0)
             return ret;
-        at1_subband_synthesis(q, su, q->out_samples[ch]);
+        at1_subband_synthesis(q, su, q->channels == 1 ? samples : q->out_samples[ch]);
     }
 
-    /* interleave; FIXME, should create/use a DSP function */
-    if (q->channels == 1) {
-        /* mono */
-        memcpy(samples, q->out_samples[0], AT1_SU_SAMPLES * 4);
-    } else {
-        /* stereo */
-        for (i = 0; i < AT1_SU_SAMPLES; i++) {
-            samples[i * 2]     = q->out_samples[0][i];
-            samples[i * 2 + 1] = q->out_samples[1][i];
-        }
+    /* interleave */
+    if (q->channels == 2) {
+        q->fmt_conv.float_interleave(samples, (const float **)q->out_samples,
+                                     AT1_SU_SAMPLES, 2);
     }
 
-    *data_size = q->channels * AT1_SU_SAMPLES * sizeof(*samples);
+    *data_size = out_size;
     return avctx->block_align;
 }
 
 
+static av_cold int atrac1_decode_end(AVCodecContext * avctx)
+{
+    AT1Ctx *q = avctx->priv_data;
+
+    av_freep(&q->out_samples[0]);
+
+    ff_mdct_end(&q->mdct_ctx[0]);
+    ff_mdct_end(&q->mdct_ctx[1]);
+    ff_mdct_end(&q->mdct_ctx[2]);
+
+    return 0;
+}
+
+
 static av_cold int atrac1_decode_init(AVCodecContext *avctx)
 {
     AT1Ctx *q = avctx->priv_data;
+    int ret;
 
     avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
 
+    if (avctx->channels < 1 || avctx->channels > AT1_MAX_CHANNELS) {
+        av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %d\n",
+               avctx->channels);
+        return AVERROR(EINVAL);
+    }
     q->channels = avctx->channels;
 
+    if (avctx->channels == 2) {
+        q->out_samples[0] = av_malloc(2 * AT1_SU_SAMPLES * sizeof(*q->out_samples[0]));
+        q->out_samples[1] = q->out_samples[0] + AT1_SU_SAMPLES;
+        if (!q->out_samples[0]) {
+            av_freep(&q->out_samples[0]);
+            return AVERROR(ENOMEM);
+        }
+    }
+
     /* Init the mdct transforms */
-    ff_mdct_init(&q->mdct_ctx[0], 6, 1, -1.0/ (1 << 15));
-    ff_mdct_init(&q->mdct_ctx[1], 8, 1, -1.0/ (1 << 15));
-    ff_mdct_init(&q->mdct_ctx[2], 9, 1, -1.0/ (1 << 15));
+    if ((ret = ff_mdct_init(&q->mdct_ctx[0], 6, 1, -1.0/ (1 << 15))) ||
+        (ret = ff_mdct_init(&q->mdct_ctx[1], 8, 1, -1.0/ (1 << 15))) ||
+        (ret = ff_mdct_init(&q->mdct_ctx[2], 9, 1, -1.0/ (1 << 15)))) {
+        av_log(avctx, AV_LOG_ERROR, "Error initializing MDCT\n");
+        atrac1_decode_end(avctx);
+        return ret;
+    }
 
     ff_init_ff_sine_windows(5);
 
     atrac_generate_tables();
 
     dsputil_init(&q->dsp, avctx);
+    ff_fmt_convert_init(&q->fmt_conv, avctx);
 
     q->bands[0] = q->low;
     q->bands[1] = q->mid;
@@ -356,16 +393,6 @@ static av_cold int atrac1_decode_init(AVCodecContext *avctx)
 }
 
 
-static av_cold int atrac1_decode_end(AVCodecContext * avctx) {
-    AT1Ctx *q = avctx->priv_data;
-
-    ff_mdct_end(&q->mdct_ctx[0]);
-    ff_mdct_end(&q->mdct_ctx[1]);
-    ff_mdct_end(&q->mdct_ctx[2]);
-    return 0;
-}
-
-
 AVCodec ff_atrac1_decoder = {
     .name = "atrac1",
     .type = AVMEDIA_TYPE_AUDIO,
diff --git a/libavcodec/atrac3.c b/libavcodec/atrac3.c
index 20ab75dfd7..25beeeeb6c 100644
--- a/libavcodec/atrac3.c
+++ b/libavcodec/atrac3.c
@@ -41,6 +41,7 @@
 #include "dsputil.h"
 #include "bytestream.h"
 #include "fft.h"
+#include "fmtconvert.h"
 
 #include "atrac.h"
 #include "atrac3data.h"
@@ -48,6 +49,8 @@
 #define JOINT_STEREO    0x12
 #define STEREO          0x2
 
+#define SAMPLES_PER_FRAME 1024
+#define MDCT_SIZE          512
 
 /* These structures are needed to store the parsed gain control data. */
 typedef struct {
@@ -70,12 +73,12 @@ typedef struct {
     int               bandsCoded;
     int               numComponents;
     tonal_component   components[64];
-    float             prevFrame[1024];
+    float             prevFrame[SAMPLES_PER_FRAME];
     int               gcBlkSwitch;
     gain_block        gainBlock[2];
 
-    DECLARE_ALIGNED(32, float, spectrum)[1024];
-    DECLARE_ALIGNED(32, float, IMDCT_buf)[1024];
+    DECLARE_ALIGNED(32, float, spectrum)[SAMPLES_PER_FRAME];
+    DECLARE_ALIGNED(32, float, IMDCT_buf)[SAMPLES_PER_FRAME];
 
     float             delayBuf1[46]; ///<qmf delay buffers
     float             delayBuf2[46];
@@ -107,7 +110,7 @@ typedef struct {
     //@}
     //@{
     /** data buffers */
-    float               outSamples[2048];
+    float              *outSamples[2];
     uint8_t*            decoded_bytes_buffer;
     float               tempBuf[1070];
     //@}
@@ -120,9 +123,10 @@ typedef struct {
     //@}
 
     FFTContext          mdct_ctx;
+    FmtConvertContext   fmt_conv;
 } ATRAC3Context;
 
-static DECLARE_ALIGNED(32, float, mdct_window)[512];
+static DECLARE_ALIGNED(32, float, mdct_window)[MDCT_SIZE];
 static VLC              spectral_coeff_tab[7];
 static float            gain_tab1[16];
 static float            gain_tab2[31];
@@ -159,7 +163,7 @@ static void IMLT(ATRAC3Context *q, float *pInput, float *pOutput, int odd_band)
     q->mdct_ctx.imdct_calc(&q->mdct_ctx,pOutput,pInput);
 
     /* Perform windowing on the output. */
-    dsp.vector_fmul(pOutput, pOutput, mdct_window, 512);
+    dsp.vector_fmul(pOutput, pOutput, mdct_window, MDCT_SIZE);
 
 }
 
@@ -192,7 +196,7 @@ static int decode_bytes(const uint8_t* inbuffer, uint8_t* out, int bytes){
 }
 
 
-static av_cold void init_atrac3_transforms(ATRAC3Context *q) {
+static av_cold int init_atrac3_transforms(ATRAC3Context *q, int is_float) {
     float enc_window[256];
     int i;
 
@@ -208,7 +212,7 @@ static av_cold void init_atrac3_transforms(ATRAC3Context *q) {
         }
 
     /* Initialize the MDCT transform. */
-    ff_mdct_init(&q->mdct_ctx, 9, 1, 1.0);
+    return ff_mdct_init(&q->mdct_ctx, 9, 1, is_float ? 1.0 / 32768 : 1.0);
 }
 
 /**
@@ -221,6 +225,8 @@ static av_cold int atrac3_decode_close(AVCodecContext *avctx)
 
     av_free(q->pUnits);
     av_free(q->decoded_bytes_buffer);
+    av_freep(&q->outSamples[0]);
+
     ff_mdct_end(&q->mdct_ctx);
 
     return 0;
@@ -340,7 +346,7 @@ static int decodeSpectrum (GetBitContext *gb, float *pOut)
 
     /* Clear the subbands that were not coded. */
     first = subbandTab[cnt];
-    memset(pOut+first, 0, (1024 - first) * sizeof(float));
+    memset(pOut+first, 0, (SAMPLES_PER_FRAME - first) * sizeof(float));
     return numSubbands;
 }
 
@@ -370,7 +376,7 @@ static int decodeTonalComponents (GetBitContext *gb, tonal_component *pComponent
 
     coding_mode_selector = get_bits(gb,2);
     if (coding_mode_selector == 2)
-        return -1;
+        return AVERROR_INVALIDDATA;
 
     coding_mode = coding_mode_selector & 1;
 
@@ -382,7 +388,7 @@ static int decodeTonalComponents (GetBitContext *gb, tonal_component *pComponent
 
         quant_step_index = get_bits(gb,3);
         if (quant_step_index <= 1)
-            return -1;
+            return AVERROR_INVALIDDATA;
 
         if (coding_mode_selector == 3)
             coding_mode = get_bits1(gb);
@@ -396,7 +402,7 @@ static int decodeTonalComponents (GetBitContext *gb, tonal_component *pComponent
             for (k=0; k<coded_components; k++) {
                 sfIndx = get_bits(gb,6);
                 pComponent[component_count].pos = j * 64 + (get_bits(gb,6));
-                max_coded_values = 1024 - pComponent[component_count].pos;
+                max_coded_values = SAMPLES_PER_FRAME - pComponent[component_count].pos;
                 coded_values = coded_values_per_component + 1;
                 coded_values = FFMIN(max_coded_values,coded_values);
 
@@ -445,7 +451,7 @@ static int decodeGainControl (GetBitContext *gb, gain_block *pGb, int numBands)
             pLevel[cf]= get_bits(gb,4);
             pLoc  [cf]= get_bits(gb,5);
             if(cf && pLoc[cf] <= pLoc[cf-1])
-                return -1;
+                return AVERROR_INVALIDDATA;
         }
     }
 
@@ -662,12 +668,12 @@ static int decodeChannelSoundUnit (ATRAC3Context *q, GetBitContext *gb, channel_
     if (codingMode == JOINT_STEREO && channelNum == 1) {
         if (get_bits(gb,2) != 3) {
             av_log(NULL,AV_LOG_ERROR,"JS mono Sound Unit id != 3.\n");
-            return -1;
+            return AVERROR_INVALIDDATA;
         }
     } else {
         if (get_bits(gb,6) != 0x28) {
             av_log(NULL,AV_LOG_ERROR,"Sound Unit id != 0x28.\n");
-            return -1;
+            return AVERROR_INVALIDDATA;
         }
     }
 
@@ -719,7 +725,8 @@ static int decodeChannelSoundUnit (ATRAC3Context *q, GetBitContext *gb, channel_
  * @param databuf       the input data
  */
 
-static int decodeFrame(ATRAC3Context *q, const uint8_t* databuf)
+static int decodeFrame(ATRAC3Context *q, const uint8_t* databuf,
+                       float **out_samples)
 {
     int   result, i;
     float   *p1, *p2, *p3, *p4;
@@ -731,7 +738,7 @@ static int decodeFrame(ATRAC3Context *q, const uint8_t* databuf)
         /* decode Sound Unit 1 */
         init_get_bits(&q->gb,databuf,q->bits_per_frame);
 
-        result = decodeChannelSoundUnit(q,&q->gb, q->pUnits, q->outSamples, 0, JOINT_STEREO);
+        result = decodeChannelSoundUnit(q,&q->gb, q->pUnits, out_samples[0], 0, JOINT_STEREO);
         if (result != 0)
             return (result);
 
@@ -753,7 +760,7 @@ static int decodeFrame(ATRAC3Context *q, const uint8_t* databuf)
         ptr1 = q->decoded_bytes_buffer;
         for (i = 4; *ptr1 == 0xF8; i++, ptr1++) {
             if (i >= q->bytes_per_frame)
-                return -1;
+                return AVERROR_INVALIDDATA;
         }
 
 
@@ -772,14 +779,14 @@ static int decodeFrame(ATRAC3Context *q, const uint8_t* databuf)
         }
 
         /* Decode Sound Unit 2. */
-        result = decodeChannelSoundUnit(q,&q->gb, &q->pUnits[1], &q->outSamples[1024], 1, JOINT_STEREO);
+        result = decodeChannelSoundUnit(q,&q->gb, &q->pUnits[1], out_samples[1], 1, JOINT_STEREO);
         if (result != 0)
             return (result);
 
         /* Reconstruct the channel coefficients. */
-        reverseMatrixing(q->outSamples, &q->outSamples[1024], q->matrix_coeff_index_prev, q->matrix_coeff_index_now);
+        reverseMatrixing(out_samples[0], out_samples[1], q->matrix_coeff_index_prev, q->matrix_coeff_index_now);
 
-        channelWeighting(q->outSamples, &q->outSamples[1024], q->weighting_delay);
+        channelWeighting(out_samples[0], out_samples[1], q->weighting_delay);
 
     } else {
         /* normal stereo mode or mono */
@@ -789,22 +796,21 @@ static int decodeFrame(ATRAC3Context *q, const uint8_t* databuf)
             /* Set the bitstream reader at the start of a channel sound unit. */
             init_get_bits(&q->gb, databuf+((i*q->bytes_per_frame)/q->channels), (q->bits_per_frame)/q->channels);
 
-            result = decodeChannelSoundUnit(q,&q->gb, &q->pUnits[i], &q->outSamples[i*1024], i, q->codingMode);
+            result = decodeChannelSoundUnit(q,&q->gb, &q->pUnits[i], out_samples[i], i, q->codingMode);
             if (result != 0)
                 return (result);
         }
     }
 
     /* Apply the iQMF synthesis filter. */
-    p1= q->outSamples;
     for (i=0 ; i<q->channels ; i++) {
+        p1 = out_samples[i];
         p2= p1+256;
         p3= p2+256;
         p4= p3+256;
         atrac_iqmf (p1, p2, 256, p1, q->pUnits[i].delayBuf1, q->tempBuf);
         atrac_iqmf (p4, p3, 256, p3, q->pUnits[i].delayBuf2, q->tempBuf);
         atrac_iqmf (p1, p3, 512, p1, q->pUnits[i].delayBuf3, q->tempBuf);
-        p1 +=1024;
     }
 
     return 0;
@@ -823,15 +829,22 @@ static int atrac3_decode_frame(AVCodecContext *avctx,
     const uint8_t *buf = avpkt->data;
     int buf_size = avpkt->size;
     ATRAC3Context *q = avctx->priv_data;
-    int result = 0, i;
+    int result = 0, out_size;
     const uint8_t* databuf;
-    int16_t* samples = data;
+    float   *samples_flt = data;
+    int16_t *samples_s16 = data;
 
     if (buf_size < avctx->block_align) {
         av_log(avctx, AV_LOG_ERROR,
                "Frame too small (%d bytes). Truncated file?\n", buf_size);
-        *data_size = 0;
-        return buf_size;
+        return AVERROR_INVALIDDATA;
+    }
+
+    out_size = SAMPLES_PER_FRAME * q->channels *
+               av_get_bytes_per_sample(avctx->sample_fmt);
+    if (*data_size < out_size) {
+        av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
+        return AVERROR(EINVAL);
     }
 
     /* Check if we need to descramble and what buffer to pass on. */
@@ -842,26 +855,27 @@ static int atrac3_decode_frame(AVCodecContext *avctx,
         databuf = buf;
     }
 
-    result = decodeFrame(q, databuf);
+    if (q->channels == 1 && avctx->sample_fmt == AV_SAMPLE_FMT_FLT)
+        result = decodeFrame(q, databuf, &samples_flt);
+    else
+        result = decodeFrame(q, databuf, q->outSamples);
 
     if (result != 0) {
         av_log(NULL,AV_LOG_ERROR,"Frame decoding error!\n");
-        return -1;
+        return result;
     }
 
-    if (q->channels == 1) {
-        /* mono */
-        for (i = 0; i<1024; i++)
-            samples[i] = av_clip_int16(round(q->outSamples[i]));
-        *data_size = 1024 * sizeof(int16_t);
-    } else {
-        /* stereo */
-        for (i = 0; i < 1024; i++) {
-            samples[i*2] = av_clip_int16(round(q->outSamples[i]));
-            samples[i*2+1] = av_clip_int16(round(q->outSamples[1024+i]));
-        }
-        *data_size = 2048 * sizeof(int16_t);
+    /* interleave */
+    if (q->channels == 2 && avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {
+        q->fmt_conv.float_interleave(samples_flt,
+                                     (const float **)q->outSamples,
+                                     SAMPLES_PER_FRAME, 2);
+    } else if (avctx->sample_fmt == AV_SAMPLE_FMT_S16) {
+        q->fmt_conv.float_to_int16_interleave(samples_s16,
+                                              (const float **)q->outSamples,
+                                              SAMPLES_PER_FRAME, q->channels);
     }
+    *data_size = out_size;
 
     return avctx->block_align;
 }
@@ -875,7 +889,7 @@ static int atrac3_decode_frame(AVCodecContext *avctx,
 
 static av_cold int atrac3_decode_init(AVCodecContext *avctx)
 {
-    int i;
+    int i, ret;
     const uint8_t *edata_ptr = avctx->extradata;
     ATRAC3Context *q = avctx->priv_data;
     static VLC_TYPE atrac3_vlc_table[4096][2];
@@ -899,7 +913,7 @@ static av_cold int atrac3_decode_init(AVCodecContext *avctx)
         av_log(avctx,AV_LOG_DEBUG,"[12-13] %d\n",bytestream_get_le16(&edata_ptr));  //Unknown always 0
 
         /* setup */
-        q->samples_per_frame = 1024 * q->channels;
+        q->samples_per_frame = SAMPLES_PER_FRAME * q->channels;
         q->atrac3version = 4;
         q->delay = 0x88E;
         if (q->codingMode)
@@ -912,7 +926,7 @@ static av_cold int atrac3_decode_init(AVCodecContext *avctx)
         if ((q->bytes_per_frame == 96*q->channels*q->frame_factor) || (q->bytes_per_frame == 152*q->channels*q->frame_factor) || (q->bytes_per_frame == 192*q->channels*q->frame_factor)) {
         } else {
             av_log(avctx,AV_LOG_ERROR,"Unknown frame/channel/frame_factor configuration %d/%d/%d\n", q->bytes_per_frame, q->channels, q->frame_factor);
-            return -1;
+            return AVERROR_INVALIDDATA;
         }
 
     } else if (avctx->extradata_size == 10) {
@@ -932,17 +946,17 @@ static av_cold int atrac3_decode_init(AVCodecContext *avctx)
 
     if (q->atrac3version != 4) {
         av_log(avctx,AV_LOG_ERROR,"Version %d != 4.\n",q->atrac3version);
-        return -1;
+        return AVERROR_INVALIDDATA;
     }
 
-    if (q->samples_per_frame != 1024 && q->samples_per_frame != 2048) {
+    if (q->samples_per_frame != SAMPLES_PER_FRAME && q->samples_per_frame != SAMPLES_PER_FRAME*2) {
         av_log(avctx,AV_LOG_ERROR,"Unknown amount of samples per frame %d.\n",q->samples_per_frame);
-        return -1;
+        return AVERROR_INVALIDDATA;
     }
 
     if (q->delay != 0x88E) {
         av_log(avctx,AV_LOG_ERROR,"Unknown amount of delay %x != 0x88E.\n",q->delay);
-        return -1;
+        return AVERROR_INVALIDDATA;
     }
 
     if (q->codingMode == STEREO) {
@@ -951,17 +965,17 @@ static av_cold int atrac3_decode_init(AVCodecContext *avctx)
         av_log(avctx,AV_LOG_DEBUG,"Joint stereo detected.\n");
     } else {
         av_log(avctx,AV_LOG_ERROR,"Unknown channel coding mode %x!\n",q->codingMode);
-        return -1;
+        return AVERROR_INVALIDDATA;
     }
 
     if (avctx->channels <= 0 || avctx->channels > 2 /*|| ((avctx->channels * 1024) != q->samples_per_frame)*/) {
         av_log(avctx,AV_LOG_ERROR,"Channel configuration error!\n");
-        return -1;
+        return AVERROR(EINVAL);
     }
 
 
     if(avctx->block_align >= UINT_MAX/2)
-        return -1;
+        return AVERROR(EINVAL);
 
     /* Pad the data buffer with FF_INPUT_BUFFER_PADDING_SIZE,
      * this is for the bitstream reader. */
@@ -981,7 +995,16 @@ static av_cold int atrac3_decode_init(AVCodecContext *avctx)
         vlcs_initialized = 1;
     }
 
-    init_atrac3_transforms(q);
+    if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT)
+        avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
+    else
+        avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+
+    if ((ret = init_atrac3_transforms(q, avctx->sample_fmt == AV_SAMPLE_FMT_FLT))) {
+        av_log(avctx, AV_LOG_ERROR, "Error initializing MDCT\n");
+        av_freep(&q->decoded_bytes_buffer);
+        return ret;
+    }
 
     atrac_generate_tables();
 
@@ -1007,14 +1030,23 @@ static av_cold int atrac3_decode_init(AVCodecContext *avctx)
     }
 
     dsputil_init(&dsp, avctx);
+    ff_fmt_convert_init(&q->fmt_conv, avctx);
 
     q->pUnits = av_mallocz(sizeof(channel_unit)*q->channels);
     if (!q->pUnits) {
-        av_free(q->decoded_bytes_buffer);
+        atrac3_decode_close(avctx);
         return AVERROR(ENOMEM);
     }
 
-    avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+    if (avctx->channels > 1 || avctx->sample_fmt == AV_SAMPLE_FMT_S16) {
+        q->outSamples[0] = av_mallocz(SAMPLES_PER_FRAME * avctx->channels * sizeof(*q->outSamples[0]));
+        q->outSamples[1] = q->outSamples[0] + SAMPLES_PER_FRAME;
+        if (!q->outSamples[0]) {
+            atrac3_decode_close(avctx);
+            return AVERROR(ENOMEM);
+        }
+    }
+
     return 0;
 }
 
@@ -1028,5 +1060,6 @@ AVCodec ff_atrac3_decoder =
     .init = atrac3_decode_init,
     .close = atrac3_decode_close,
     .decode = atrac3_decode_frame,
+    .capabilities = CODEC_CAP_SUBFRAMES,
     .long_name = NULL_IF_CONFIG_SMALL("Atrac 3 (Adaptive TRansform Acoustic Coding 3)"),
 };
diff --git a/libavcodec/binkaudio.c b/libavcodec/binkaudio.c
index 2d06aaa9e9..b1e4de2711 100644
--- a/libavcodec/binkaudio.c
+++ b/libavcodec/binkaudio.c
@@ -39,6 +39,8 @@
 
 extern const uint16_t ff_wma_critical_freqs[25];
 
+static float quant_table[95];
+
 #define MAX_CHANNELS 2
 #define BINK_BLOCK_MAX_SIZE (MAX_CHANNELS << 11)
 
@@ -56,8 +58,11 @@ typedef struct {
     unsigned int *bands;
     float root;
     DECLARE_ALIGNED(32, FFTSample, coeffs)[BINK_BLOCK_MAX_SIZE];
-    DECLARE_ALIGNED(16, short, previous)[BINK_BLOCK_MAX_SIZE / 16];  ///< coeffs from previous audio block
+    DECLARE_ALIGNED(16, int16_t, previous)[BINK_BLOCK_MAX_SIZE / 16];  ///< coeffs from previous audio block
+    DECLARE_ALIGNED(16, int16_t, current)[BINK_BLOCK_MAX_SIZE / 16];
     float *coeffs_ptr[MAX_CHANNELS]; ///< pointers to the coeffs arrays for float_to_int16_interleave
+    float *prev_ptr[MAX_CHANNELS];   ///< pointers to the overlap points in the coeffs array
+    uint8_t *packet_buffer;
     union {
         RDFTContext rdft;
         DCTContext dct;
@@ -107,6 +112,10 @@ static av_cold int decode_init(AVCodecContext *avctx)
     s->block_size    = (s->frame_len - s->overlap_len) * s->channels;
     sample_rate_half = (sample_rate + 1) / 2;
     s->root          = 2.0 / sqrt(s->frame_len);
+    for (i = 0; i < 95; i++) {
+        /* constant is result of 0.066399999/log10(M_E) */
+        quant_table[i] = expf(i * 0.15289164787221953823f) * s->root;
+    }
 
     /* calculate number of bands */
     for (s->num_bands = 1; s->num_bands < 25; s->num_bands++)
@@ -126,8 +135,10 @@ static av_cold int decode_init(AVCodecContext *avctx)
     s->first = 1;
     avctx->sample_fmt = AV_SAMPLE_FMT_S16;
 
-    for (i = 0; i < s->channels; i++)
+    for (i = 0; i < s->channels; i++) {
         s->coeffs_ptr[i] = s->coeffs + i * s->frame_len;
+        s->prev_ptr[i]   = s->coeffs_ptr[i] + s->frame_len - s->overlap_len;
+    }
 
     if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == CODEC_ID_BINKAUDIO_RDFT)
         ff_rdft_init(&s->trans.rdft, frame_len_bits, DFT_C2R);
@@ -152,11 +163,18 @@ static const uint8_t rle_length_tab[16] = {
     2, 3, 4, 5, 6, 8, 9, 10, 11, 12, 13, 14, 15, 16, 32, 64
 };
 
+#define GET_BITS_SAFE(out, nbits) do {  \
+    if (get_bits_left(gb) < nbits)      \
+        return AVERROR_INVALIDDATA;     \
+    out = get_bits(gb, nbits);          \
+} while (0)
+
 /**
  * Decode Bink Audio block
  * @param[out] out Output buffer (must contain s->block_size elements)
+ * @return 0 on success, negative error code on failure
  */
-static void decode_block(BinkAudioContext *s, short *out, int use_dct)
+static int decode_block(BinkAudioContext *s, int16_t *out, int use_dct)
 {
     int ch, i, j, k;
     float q, quant[25];
@@ -169,17 +187,22 @@ static void decode_block(BinkAudioContext *s, short *out, int use_dct)
     for (ch = 0; ch < s->channels; ch++) {
         FFTSample *coeffs = s->coeffs_ptr[ch];
         if (s->version_b) {
+            if (get_bits_left(gb) < 64)
+                return AVERROR_INVALIDDATA;
             coeffs[0] = av_int2flt(get_bits(gb, 32)) * s->root;
             coeffs[1] = av_int2flt(get_bits(gb, 32)) * s->root;
         } else {
+            if (get_bits_left(gb) < 58)
+                return AVERROR_INVALIDDATA;
             coeffs[0] = get_float(gb) * s->root;
             coeffs[1] = get_float(gb) * s->root;
         }
 
+        if (get_bits_left(gb) < s->num_bands * 8)
+            return AVERROR_INVALIDDATA;
         for (i = 0; i < s->num_bands; i++) {
-            /* constant is result of 0.066399999/log10(M_E) */
             int value = get_bits(gb, 8);
-            quant[i] = expf(FFMIN(value, 95) * 0.15289164787221953823f) * s->root;
+            quant[i]  = quant_table[FFMIN(value, 95)];
         }
 
         k = 0;
@@ -190,15 +213,20 @@ static void decode_block(BinkAudioContext *s, short *out, int use_dct)
         while (i < s->frame_len) {
             if (s->version_b) {
                 j = i + 16;
-            } else if (get_bits1(gb)) {
-                j = i + rle_length_tab[get_bits(gb, 4)] * 8;
             } else {
-                j = i + 8;
+                int v;
+                GET_BITS_SAFE(v, 1);
+                if (v) {
+                    GET_BITS_SAFE(v, 4);
+                    j = i + rle_length_tab[v] * 8;
+                } else {
+                    j = i + 8;
+                }
             }
 
             j = FFMIN(j, s->frame_len);
 
-            width = get_bits(gb, 4);
+            GET_BITS_SAFE(width, 4);
             if (width == 0) {
                 memset(coeffs + i, 0, (j - i) * sizeof(*coeffs));
                 i = j;
@@ -208,9 +236,11 @@ static void decode_block(BinkAudioContext *s, short *out, int use_dct)
                 while (i < j) {
                     if (s->bands[k] == i)
                         q = quant[k++];
-                    coeff = get_bits(gb, width);
+                    GET_BITS_SAFE(coeff, width);
                     if (coeff) {
-                        if (get_bits1(gb))
+                        int v;
+                        GET_BITS_SAFE(v, 1);
+                        if (v)
                             coeffs[i] = -q * coeff;
                         else
                             coeffs[i] =  q * coeff;
@@ -231,8 +261,12 @@ static void decode_block(BinkAudioContext *s, short *out, int use_dct)
             s->trans.rdft.rdft_calc(&s->trans.rdft, coeffs);
     }
 
+    s->fmt_conv.float_to_int16_interleave(s->current,
+                                          (const float **)s->prev_ptr,
+                                          s->overlap_len, s->channels);
     s->fmt_conv.float_to_int16_interleave(out, (const float **)s->coeffs_ptr,
-                                          s->frame_len, s->channels);
+                                          s->frame_len - s->overlap_len,
+                                          s->channels);
 
     if (!s->first) {
         int count = s->overlap_len * s->channels;
@@ -242,16 +276,19 @@ static void decode_block(BinkAudioContext *s, short *out, int use_dct)
         }
     }
 
-    memcpy(s->previous, out + s->block_size,
-           s->overlap_len * s->channels * sizeof(*out));
+    memcpy(s->previous, s->current,
+           s->overlap_len * s->channels * sizeof(*s->previous));
 
     s->first = 0;
+
+    return 0;
 }
 
 static av_cold int decode_end(AVCodecContext *avctx)
 {
     BinkAudioContext * s = avctx->priv_data;
     av_freep(&s->bands);
+    av_freep(&s->packet_buffer);
     if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == CODEC_ID_BINKAUDIO_RDFT)
         ff_rdft_end(&s->trans.rdft);
     else if (CONFIG_BINKAUDIO_DCT_DECODER)
@@ -270,25 +307,47 @@ static int decode_frame(AVCodecContext *avctx,
                         AVPacket *avpkt)
 {
     BinkAudioContext *s = avctx->priv_data;
-    const uint8_t *buf  = avpkt->data;
-    int buf_size        = avpkt->size;
-    short *samples      = data;
-    short *samples_end  = (short*)((uint8_t*)data + *data_size);
-    int reported_size;
+    int16_t *samples      = data;
     GetBitContext *gb = &s->gb;
+    int out_size, consumed = 0;
+
+    if (!get_bits_left(gb)) {
+        uint8_t *buf;
+        /* handle end-of-stream */
+        if (!avpkt->size) {
+            *data_size = 0;
+            return 0;
+        }
+        if (avpkt->size < 4) {
+            av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
+            return AVERROR_INVALIDDATA;
+        }
+        buf = av_realloc(s->packet_buffer, avpkt->size + FF_INPUT_BUFFER_PADDING_SIZE);
+        if (!buf)
+            return AVERROR(ENOMEM);
+        s->packet_buffer = buf;
+        memcpy(s->packet_buffer, avpkt->data, avpkt->size);
+        init_get_bits(gb, s->packet_buffer, avpkt->size * 8);
+        consumed = avpkt->size;
+
+        /* skip reported size */
+        skip_bits_long(gb, 32);
+    }
 
-    init_get_bits(gb, buf, buf_size * 8);
+    out_size = s->block_size * av_get_bytes_per_sample(avctx->sample_fmt);
+    if (*data_size < out_size) {
+        av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
+        return AVERROR(EINVAL);
+    }
 
-    reported_size = get_bits_long(gb, 32);
-    while (get_bits_count(gb) / 8 < buf_size &&
-           samples + s->block_size <= samples_end) {
-        decode_block(s, samples, avctx->codec->id == CODEC_ID_BINKAUDIO_DCT);
-        samples += s->block_size;
-        get_bits_align32(gb);
+    if (decode_block(s, samples, avctx->codec->id == CODEC_ID_BINKAUDIO_DCT)) {
+        av_log(avctx, AV_LOG_ERROR, "Incomplete packet\n");
+        return AVERROR_INVALIDDATA;
     }
+    get_bits_align32(gb);
 
-    *data_size = FFMIN(reported_size, (uint8_t*)samples - (uint8_t*)data);
-    return buf_size;
+    *data_size = out_size;
+    return consumed;
 }
 
 AVCodec ff_binkaudio_rdft_decoder = {
@@ -299,6 +358,7 @@ AVCodec ff_binkaudio_rdft_decoder = {
     .init           = decode_init,
     .close          = decode_end,
     .decode         = decode_frame,
+    .capabilities   = CODEC_CAP_DELAY,
     .long_name = NULL_IF_CONFIG_SMALL("Bink Audio (RDFT)")
 };
 
@@ -310,5 +370,6 @@ AVCodec ff_binkaudio_dct_decoder = {
     .init           = decode_init,
     .close          = decode_end,
     .decode         = decode_frame,
+    .capabilities   = CODEC_CAP_DELAY,
     .long_name = NULL_IF_CONFIG_SMALL("Bink Audio (DCT)")
 };
diff --git a/libavcodec/cook.c b/libavcodec/cook.c
index 0d09bb83fb..9cfd3960e0 100644
--- a/libavcodec/cook.c
+++ b/libavcodec/cook.c
@@ -42,12 +42,7 @@
  * available.
  */
 
-#include <math.h>
-#include <stddef.h>
-#include <stdio.h>
-
 #include "libavutil/lfg.h"
-#include "libavutil/random_seed.h"
 #include "avcodec.h"
 #include "get_bits.h"
 #include "dsputil.h"
@@ -124,7 +119,7 @@ typedef struct cook {
     void (* interpolate) (struct cook *q, float* buffer,
                           int gain_index, int gain_index_next);
 
-    void (* saturate_output) (struct cook *q, int chan, int16_t *out);
+    void (* saturate_output) (struct cook *q, int chan, float *out);
 
     AVCodecContext*     avctx;
     GetBitContext       gb;
@@ -217,11 +212,11 @@ static av_cold int init_cook_vlc_tables(COOKContext *q) {
 }
 
 static av_cold int init_cook_mlt(COOKContext *q) {
-    int j;
+    int j, ret;
     int mlt_size = q->samples_per_channel;
 
-    if ((q->mlt_window = av_malloc(sizeof(float)*mlt_size)) == 0)
-      return -1;
+    if ((q->mlt_window = av_malloc(mlt_size * sizeof(*q->mlt_window))) == 0)
+        return AVERROR(ENOMEM);
 
     /* Initialize the MLT window: simple sine window. */
     ff_sine_window_init(q->mlt_window, mlt_size);
@@ -229,9 +224,9 @@ static av_cold int init_cook_mlt(COOKContext *q) {
         q->mlt_window[j] *= sqrt(2.0 / q->samples_per_channel);
 
     /* Initialize the MDCT. */
-    if (ff_mdct_init(&q->mdct_ctx, av_log2(mlt_size)+1, 1, 1.0)) {
-      av_free(q->mlt_window);
-      return -1;
+    if ((ret = ff_mdct_init(&q->mdct_ctx, av_log2(mlt_size)+1, 1, 1.0/32768.0))) {
+        av_free(q->mlt_window);
+        return ret;
     }
     av_log(q->avctx,AV_LOG_DEBUG,"MDCT initialized, order = %d.\n",
            av_log2(mlt_size)+1);
@@ -410,9 +405,9 @@ static void categorize(COOKContext *q, COOKSubpacket *p, int* quant_index_table,
         //av_log(q->avctx, AV_LOG_ERROR, "bits_left = %d\n",bits_left);
     }
 
-    memset(&exp_index1,0,102*sizeof(int));
-    memset(&exp_index2,0,102*sizeof(int));
-    memset(&tmp_categorize_array,0,128*2*sizeof(int));
+    memset(&exp_index1,           0, sizeof(exp_index1));
+    memset(&exp_index2,           0, sizeof(exp_index2));
+    memset(&tmp_categorize_array, 0, sizeof(tmp_categorize_array));
 
     bias=-32;
 
@@ -633,8 +628,8 @@ static void mono_decode(COOKContext *q, COOKSubpacket *p, float* mlt_buffer) {
     int quant_index_table[102];
     int category[128];
 
-    memset(&category, 0, 128*sizeof(int));
-    memset(&category_index, 0, 128*sizeof(int));
+    memset(&category,       0, sizeof(category));
+    memset(&category_index, 0, sizeof(category_index));
 
     decode_envelope(q, p, quant_index_table);
     q->num_vectors = get_bits(&q->gb,p->log2_numvector_size);
@@ -663,14 +658,12 @@ static void interpolate_float(COOKContext *q, float* buffer,
         for(i=0 ; i<q->gain_size_factor ; i++){
             buffer[i]*=fc1;
         }
-        return;
     } else {                                        //smooth gain
         fc2 = q->gain_table[11 + (gain_index_next-gain_index)];
         for(i=0 ; i<q->gain_size_factor ; i++){
             buffer[i]*=fc1;
             fc1*=fc2;
         }
-        return;
     }
 }
 
@@ -733,7 +726,8 @@ static void imlt_gain(COOKContext *q, float *inbuffer,
     }
 
     /* Save away the current to be previous block. */
-    memcpy(previous_buffer, buffer0, sizeof(float)*q->samples_per_channel);
+    memcpy(previous_buffer, buffer0,
+           q->samples_per_channel * sizeof(*previous_buffer));
 }
 
 
@@ -744,27 +738,24 @@ static void imlt_gain(COOKContext *q, float *inbuffer,
  * @param decouple_tab      decoupling array
  *
  */
+static void decouple_info(COOKContext *q, COOKSubpacket *p, int *decouple_tab)
+{
+    int i;
+    int vlc    = get_bits1(&q->gb);
+    int start  = cplband[p->js_subband_start];
+    int end    = cplband[p->subbands-1];
+    int length = end - start + 1;
 
-static void decouple_info(COOKContext *q, COOKSubpacket *p, int* decouple_tab){
-    int length, i;
-
-    if(get_bits1(&q->gb)) {
-        if(cplband[p->js_subband_start] > cplband[p->subbands-1]) return;
-
-        length = cplband[p->subbands-1] - cplband[p->js_subband_start] + 1;
-        for (i=0 ; i<length ; i++) {
-            decouple_tab[cplband[p->js_subband_start] + i] = get_vlc2(&q->gb, p->ccpl.table, p->ccpl.bits, 2);
-        }
+    if (start > end)
         return;
-    }
-
-    if(cplband[p->js_subband_start] > cplband[p->subbands-1]) return;
 
-    length = cplband[p->subbands-1] - cplband[p->js_subband_start] + 1;
-    for (i=0 ; i<length ; i++) {
-       decouple_tab[cplband[p->js_subband_start] + i] = get_bits(&q->gb, p->js_vlc_bits);
+    if (vlc) {
+        for (i = 0; i < length; i++)
+            decouple_tab[start + i] = get_vlc2(&q->gb, p->ccpl.table, p->ccpl.bits, 2);
+    } else {
+        for (i = 0; i < length; i++)
+            decouple_tab[start + i] = get_bits(&q->gb, p->js_vlc_bits);
     }
-    return;
 }
 
 /*
@@ -811,11 +802,11 @@ static void joint_decode(COOKContext *q, COOKSubpacket *p, float* mlt_buffer1,
     const float* cplscale;
 
     memset(decouple_tab, 0, sizeof(decouple_tab));
-    memset(decode_buffer, 0, sizeof(decode_buffer));
+    memset(decode_buffer, 0, sizeof(q->decode_buffer_0));
 
     /* Make sure the buffers are zeroed out. */
-    memset(mlt_buffer1,0, 1024*sizeof(float));
-    memset(mlt_buffer2,0, 1024*sizeof(float));
+    memset(mlt_buffer1, 0, 1024 * sizeof(*mlt_buffer1));
+    memset(mlt_buffer2, 0, 1024 * sizeof(*mlt_buffer2));
     decouple_info(q, p, decouple_tab);
     mono_decode(q, p, decode_buffer);
 
@@ -867,22 +858,18 @@ decode_bytes_and_gain(COOKContext *q, COOKSubpacket *p, const uint8_t *inbuffer,
 }
 
  /**
- * Saturate the output signal to signed 16bit integers.
+ * Saturate the output signal and interleave.
  *
  * @param q                 pointer to the COOKContext
  * @param chan              channel to saturate
  * @param out               pointer to the output vector
  */
-static void
-saturate_output_float (COOKContext *q, int chan, int16_t *out)
+static void saturate_output_float(COOKContext *q, int chan, float *out)
 {
     int j;
     float *output = q->mono_mdct_output + q->samples_per_channel;
-    /* Clip and convert floats to 16 bits.
-     */
     for (j = 0; j < q->samples_per_channel; j++) {
-        out[chan + q->nb_channels * j] =
-          av_clip_int16(lrintf(output[j]));
+        out[chan + q->nb_channels * j] = av_clipf(output[j], -1.0, 1.0);
     }
 }
 
@@ -902,7 +889,7 @@ saturate_output_float (COOKContext *q, int chan, int16_t *out)
 static inline void
 mlt_compensate_output(COOKContext *q, float *decode_buffer,
                       cook_gains *gains_ptr, float *previous_buffer,
-                      int16_t *out, int chan)
+                      float *out, int chan)
 {
     imlt_gain(q, decode_buffer, gains_ptr, previous_buffer);
     q->saturate_output (q, chan, out);
@@ -917,7 +904,9 @@ mlt_compensate_output(COOKContext *q, float *decode_buffer,
  * @param inbuffer          pointer to the inbuffer
  * @param outbuffer         pointer to the outbuffer
  */
-static void decode_subpacket(COOKContext *q, COOKSubpacket* p, const uint8_t *inbuffer, int16_t *outbuffer) {
+static void decode_subpacket(COOKContext *q, COOKSubpacket *p,
+                             const uint8_t *inbuffer, float *outbuffer)
+{
     int sub_packet_size = p->size;
     /* packet dump */
 //    for (i=0 ; i<sub_packet_size ; i++) {
@@ -966,13 +955,20 @@ static int cook_decode_frame(AVCodecContext *avctx,
     const uint8_t *buf = avpkt->data;
     int buf_size = avpkt->size;
     COOKContext *q = avctx->priv_data;
-    int i;
+    int i, out_size;
     int offset = 0;
     int chidx = 0;
 
     if (buf_size < avctx->block_align)
         return buf_size;
 
+    out_size = q->nb_channels * q->samples_per_channel *
+               av_get_bytes_per_sample(avctx->sample_fmt);
+    if (*data_size < out_size) {
+        av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
+        return AVERROR(EINVAL);
+    }
+
     /* estimate subpacket sizes */
     q->subpacket[0].size = avctx->block_align;
 
@@ -981,22 +977,21 @@ static int cook_decode_frame(AVCodecContext *avctx,
         q->subpacket[0].size -= q->subpacket[i].size + 1;
         if (q->subpacket[0].size < 0) {
             av_log(avctx,AV_LOG_DEBUG,"frame subpacket size total > avctx->block_align!\n");
-            return -1;
+            return AVERROR_INVALIDDATA;
         }
     }
 
     /* decode supbackets */
-    *data_size = 0;
     for(i=0;i<q->num_subpackets;i++){
         q->subpacket[i].bits_per_subpacket = (q->subpacket[i].size*8)>>q->subpacket[i].bits_per_subpdiv;
         q->subpacket[i].ch_idx = chidx;
         av_log(avctx,AV_LOG_DEBUG,"subpacket[%i] size %i js %i %i block_align %i\n",i,q->subpacket[i].size,q->subpacket[i].joint_stereo,offset,avctx->block_align);
-        decode_subpacket(q, &q->subpacket[i], buf + offset, (int16_t*)data);
+        decode_subpacket(q, &q->subpacket[i], buf + offset, data);
         offset += q->subpacket[i].size;
         chidx += q->subpacket[i].num_channels;
         av_log(avctx,AV_LOG_DEBUG,"subpacket[%i] %i %i\n",i,q->subpacket[i].size * 8,get_bits_count(&q->gb));
     }
-    *data_size = sizeof(int16_t) * q->nb_channels * q->samples_per_channel;
+    *data_size = out_size;
 
     /* Discard the first two frames: no valid audio. */
     if (avctx->frame_number < 2) *data_size = 0;
@@ -1053,12 +1048,13 @@ static av_cold int cook_decode_init(AVCodecContext *avctx)
     int extradata_size = avctx->extradata_size;
     int s = 0;
     unsigned int channel_mask = 0;
+    int ret;
     q->avctx = avctx;
 
     /* Take care of the codec specific extradata. */
     if (extradata_size <= 0) {
         av_log(avctx,AV_LOG_ERROR,"Necessary extradata missing!\n");
-        return -1;
+        return AVERROR_INVALIDDATA;
     }
     av_log(avctx,AV_LOG_DEBUG,"codecdata_length=%d\n",avctx->extradata_size);
 
@@ -1103,7 +1099,7 @@ static av_cold int cook_decode_init(AVCodecContext *avctx)
             case MONO:
                 if (q->nb_channels != 1) {
                     av_log_ask_for_sample(avctx, "Container channels != 1.\n");
-                    return -1;
+                    return AVERROR_PATCHWELCOME;
                 }
                 av_log(avctx,AV_LOG_DEBUG,"MONO\n");
                 break;
@@ -1117,7 +1113,7 @@ static av_cold int cook_decode_init(AVCodecContext *avctx)
             case JOINT_STEREO:
                 if (q->nb_channels != 2) {
                     av_log_ask_for_sample(avctx, "Container channels != 2.\n");
-                    return -1;
+                    return AVERROR_PATCHWELCOME;
                 }
                 av_log(avctx,AV_LOG_DEBUG,"JOINT_STEREO\n");
                 if (avctx->extradata_size >= 16){
@@ -1155,12 +1151,12 @@ static av_cold int cook_decode_init(AVCodecContext *avctx)
                 break;
             default:
                 av_log_ask_for_sample(avctx, "Unknown Cook version.\n");
-                return -1;
+                return AVERROR_PATCHWELCOME;
         }
 
         if(s > 1 && q->subpacket[s].samples_per_channel != q->samples_per_channel) {
             av_log(avctx,AV_LOG_ERROR,"different number of samples per channel!\n");
-            return -1;
+            return AVERROR_INVALIDDATA;
         } else
             q->samples_per_channel = q->subpacket[0].samples_per_channel;
 
@@ -1171,18 +1167,18 @@ static av_cold int cook_decode_init(AVCodecContext *avctx)
         /* Try to catch some obviously faulty streams, othervise it might be exploitable */
         if (q->subpacket[s].total_subbands > 53) {
             av_log_ask_for_sample(avctx, "total_subbands > 53\n");
-            return -1;
+            return AVERROR_PATCHWELCOME;
         }
 
         if ((q->subpacket[s].js_vlc_bits > 6) || (q->subpacket[s].js_vlc_bits < 2*q->subpacket[s].joint_stereo)) {
             av_log(avctx,AV_LOG_ERROR,"js_vlc_bits = %d, only >= %d and <= 6 allowed!\n",
                    q->subpacket[s].js_vlc_bits, 2*q->subpacket[s].joint_stereo);
-            return -1;
+            return AVERROR_INVALIDDATA;
         }
 
         if (q->subpacket[s].subbands > 50) {
             av_log_ask_for_sample(avctx, "subbands > 50\n");
-            return -1;
+            return AVERROR_PATCHWELCOME;
         }
         q->subpacket[s].gains1.now      = q->subpacket[s].gain_1;
         q->subpacket[s].gains1.previous = q->subpacket[s].gain_2;
@@ -1193,7 +1189,7 @@ static av_cold int cook_decode_init(AVCodecContext *avctx)
         s++;
         if (s > MAX_SUBPACKETS) {
             av_log_ask_for_sample(avctx, "Too many subpackets > 5\n");
-            return -1;
+            return AVERROR_PATCHWELCOME;
         }
     }
     /* Generate tables */
@@ -1201,12 +1197,12 @@ static av_cold int cook_decode_init(AVCodecContext *avctx)
     init_gain_table(q);
     init_cplscales_table(q);
 
-    if (init_cook_vlc_tables(q) != 0)
-        return -1;
+    if ((ret = init_cook_vlc_tables(q)))
+        return ret;
 
 
     if(avctx->block_align >= UINT_MAX/2)
-        return -1;
+        return AVERROR(EINVAL);
 
     /* Pad the databuffer with:
        DECODE_BYTES_PAD1 or DECODE_BYTES_PAD2 for decode_bytes(),
@@ -1216,11 +1212,11 @@ static av_cold int cook_decode_init(AVCodecContext *avctx)
                      + DECODE_BYTES_PAD1(avctx->block_align)
                      + FF_INPUT_BUFFER_PADDING_SIZE);
     if (q->decoded_bytes_buffer == NULL)
-        return -1;
+        return AVERROR(ENOMEM);
 
     /* Initialize transform. */
-    if ( init_cook_mlt(q) != 0 )
-        return -1;
+    if ((ret = init_cook_mlt(q)))
+        return ret;
 
     /* Initialize COOK signal arithmetic handling */
     if (1) {
@@ -1237,10 +1233,10 @@ static av_cold int cook_decode_init(AVCodecContext *avctx)
         av_log_ask_for_sample(avctx,
                               "unknown amount of samples_per_channel = %d\n",
                               q->samples_per_channel);
-        return -1;
+        return AVERROR_PATCHWELCOME;
     }
 
-    avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+    avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
     if (channel_mask)
         avctx->channel_layout = channel_mask;
     else
diff --git a/libavcodec/dca.c b/libavcodec/dca.c
index 5cf5b2629a..37977e5c55 100644
--- a/libavcodec/dca.c
+++ b/libavcodec/dca.c
@@ -528,15 +528,15 @@ static int dca_parse_frame_header(DCAContext * s)
     s->sample_blocks     = get_bits(&s->gb, 7) + 1;
     s->frame_size        = get_bits(&s->gb, 14) + 1;
     if (s->frame_size < 95)
-        return -1;
+        return AVERROR_INVALIDDATA;
     s->amode             = get_bits(&s->gb, 6);
     s->sample_rate       = dca_sample_rates[get_bits(&s->gb, 4)];
     if (!s->sample_rate)
-        return -1;
+        return AVERROR_INVALIDDATA;
     s->bit_rate_index    = get_bits(&s->gb, 5);
     s->bit_rate          = dca_bit_rates[s->bit_rate_index];
     if (!s->bit_rate)
-        return -1;
+        return AVERROR_INVALIDDATA;
 
     s->downmix           = get_bits(&s->gb, 1);
     s->dynrange          = get_bits(&s->gb, 1);
@@ -626,7 +626,7 @@ static int dca_subframe_header(DCAContext * s, int base_channel, int block_index
     int j, k;
 
     if (get_bits_left(&s->gb) < 0)
-        return -1;
+        return AVERROR_INVALIDDATA;
 
     if (!base_channel) {
         s->subsubframes[s->current_subframe] = get_bits(&s->gb, 2) + 1;
@@ -658,7 +658,7 @@ static int dca_subframe_header(DCAContext * s, int base_channel, int block_index
             else if (s->bitalloc_huffman[j] == 7) {
                 av_log(s->avctx, AV_LOG_ERROR,
                        "Invalid bit allocation index\n");
-                return -1;
+                return AVERROR_INVALIDDATA;
             } else {
                 s->bitalloc[j][k] =
                     get_bitalloc(&s->gb, &dca_bitalloc_index, s->bitalloc_huffman[j]);
@@ -667,7 +667,7 @@ static int dca_subframe_header(DCAContext * s, int base_channel, int block_index
             if (s->bitalloc[j][k] > 26) {
 //                 av_log(s->avctx,AV_LOG_DEBUG,"bitalloc index [%i][%i] too big (%i)\n",
 //                          j, k, s->bitalloc[j][k]);
-                return -1;
+                return AVERROR_INVALIDDATA;
             }
         }
     }
@@ -685,7 +685,7 @@ static int dca_subframe_header(DCAContext * s, int base_channel, int block_index
     }
 
     if (get_bits_left(&s->gb) < 0)
-        return -1;
+        return AVERROR_INVALIDDATA;
 
     for (j = base_channel; j < s->prim_channels; j++) {
         const uint32_t *scale_table;
@@ -723,7 +723,7 @@ static int dca_subframe_header(DCAContext * s, int base_channel, int block_index
     }
 
     if (get_bits_left(&s->gb) < 0)
-        return -1;
+        return AVERROR_INVALIDDATA;
 
     /* Scale factors for joint subband coding */
     for (j = base_channel; j < s->prim_channels; j++) {
@@ -1055,7 +1055,7 @@ static int decode_blockcode(int code, int levels, int *values)
         return 0;
     else {
         av_log(NULL, AV_LOG_ERROR, "ERROR: block code look-up failed\n");
-        return -1;
+        return AVERROR_INVALIDDATA;
     }
 }
 #endif
@@ -1096,7 +1096,7 @@ static int dca_subsubframe(DCAContext * s, int base_channel, int block_index)
 
     for (k = base_channel; k < s->prim_channels; k++) {
         if (get_bits_left(&s->gb) < 0)
-            return -1;
+            return AVERROR_INVALIDDATA;
 
         for (l = 0; l < s->vq_start_subband[k]; l++) {
             int m;
@@ -1275,12 +1275,13 @@ static int dca_subframe_footer(DCAContext * s, int base_channel)
 
 static int dca_decode_block(DCAContext * s, int base_channel, int block_index)
 {
+    int ret;
 
     /* Sanity check */
     if (s->current_subframe >= s->subframes) {
         av_log(s->avctx, AV_LOG_DEBUG, "check failed: %i>%i",
                s->current_subframe, s->subframes);
-        return -1;
+        return AVERROR_INVALIDDATA;
     }
 
     if (!s->current_subsubframe) {
@@ -1288,16 +1289,16 @@ static int dca_decode_block(DCAContext * s, int base_channel, int block_index)
         av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_header\n");
 #endif
         /* Read subframe header */
-        if (dca_subframe_header(s, base_channel, block_index))
-            return -1;
+        if ((ret = dca_subframe_header(s, base_channel, block_index)))
+            return ret;
     }
 
     /* Read subsubframe */
 #ifdef TRACE
     av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subsubframe\n");
 #endif
-    if (dca_subsubframe(s, base_channel, block_index))
-        return -1;
+    if ((ret = dca_subsubframe(s, base_channel, block_index)))
+        return ret;
 
     /* Update state */
     s->current_subsubframe++;
@@ -1310,8 +1311,8 @@ static int dca_decode_block(DCAContext * s, int base_channel, int block_index)
         av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_footer\n");
 #endif
         /* Read subframe footer */
-        if (dca_subframe_footer(s, base_channel))
-            return -1;
+        if ((ret = dca_subframe_footer(s, base_channel)))
+            return ret;
     }
 
     return 0;
@@ -1354,7 +1355,7 @@ static int dca_convert_bitstream(const uint8_t * src, int src_size, uint8_t * ds
         flush_put_bits(&pb);
         return (put_bits_count(&pb) + 7) >> 3;
     default:
-        return -1;
+        return AVERROR_INVALIDDATA;
     }
 }
 
@@ -1637,7 +1638,7 @@ static int dca_decode_frame(AVCodecContext * avctx,
 
     int lfe_samples;
     int num_core_channels = 0;
-    int i;
+    int i, ret;
     float   *samples_flt = data;
     int16_t *samples_s16 = data;
     int out_size;
@@ -1650,16 +1651,15 @@ static int dca_decode_frame(AVCodecContext * avctx,
 
     s->dca_buffer_size = dca_convert_bitstream(buf, buf_size, s->dca_buffer,
                                                DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE);
-    if (s->dca_buffer_size == -1) {
+    if (s->dca_buffer_size == AVERROR_INVALIDDATA) {
         av_log(avctx, AV_LOG_ERROR, "Not a valid DCA frame\n");
-        return -1;
+        return AVERROR_INVALIDDATA;
     }
 
     init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);
-    if (dca_parse_frame_header(s) < 0) {
+    if ((ret = dca_parse_frame_header(s)) < 0) {
         //seems like the frame is corrupt, try with the next one
-        *data_size=0;
-        return buf_size;
+        return ret;
     }
     //set AVCodec values with parsed data
     avctx->sample_rate = s->sample_rate;
@@ -1669,7 +1669,10 @@ static int dca_decode_frame(AVCodecContext * avctx,
     s->profile = FF_PROFILE_DTS;
 
     for (i = 0; i < (s->sample_blocks / 8); i++) {
-        dca_decode_block(s, 0, i);
+        if ((ret = dca_decode_block(s, 0, i))) {
+            av_log(avctx, AV_LOG_ERROR, "error decoding block\n");
+            return ret;
+        }
     }
 
     /* record number of core channels incase less than max channels are requested */
@@ -1725,7 +1728,10 @@ static int dca_decode_frame(AVCodecContext * avctx,
             dca_parse_audio_coding_header(s, s->xch_base_channel);
 
             for (i = 0; i < (s->sample_blocks / 8); i++) {
-                dca_decode_block(s, s->xch_base_channel, i);
+                if ((ret = dca_decode_block(s, s->xch_base_channel, i))) {
+                    av_log(avctx, AV_LOG_ERROR, "error decoding XCh extension\n");
+                    continue;
+                }
             }
 
             s->xch_present = 1;
@@ -1799,7 +1805,7 @@ static int dca_decode_frame(AVCodecContext * avctx,
 
         if (channels > !!s->lfe &&
             s->channel_order_tab[channels - 1 - !!s->lfe] < 0)
-            return -1;
+            return AVERROR_INVALIDDATA;
 
         if (avctx->request_channels == 2 && s->prim_channels > 2) {
             channels = 2;
@@ -1812,7 +1818,7 @@ static int dca_decode_frame(AVCodecContext * avctx,
         }
     } else {
         av_log(avctx, AV_LOG_ERROR, "Non standard configuration %d !\n",s->amode);
-        return -1;
+        return AVERROR_INVALIDDATA;
     }
 
     if (avctx->channels != channels) {
@@ -1824,7 +1830,7 @@ static int dca_decode_frame(AVCodecContext * avctx,
     out_size = 256 / 8 * s->sample_blocks * channels *
                av_get_bytes_per_sample(avctx->sample_fmt);
     if (*data_size < out_size)
-        return -1;
+        return AVERROR(EINVAL);
     *data_size = out_size;
 
     /* filter to get final output */
diff --git a/libavcodec/dsicinav.c b/libavcodec/dsicinav.c
index 05d9e4cc14..53d4f90d2e 100644
--- a/libavcodec/dsicinav.c
+++ b/libavcodec/dsicinav.c
@@ -26,6 +26,7 @@
 
 #include "avcodec.h"
 #include "bytestream.h"
+#include "mathops.h"
 
 
 typedef enum CinVideoBitmapIndex {
@@ -43,7 +44,6 @@ typedef struct CinVideoContext {
 } CinVideoContext;
 
 typedef struct CinAudioContext {
-    AVCodecContext *avctx;
     int initial_decode_frame;
     int delta;
 } CinAudioContext;
@@ -309,7 +309,11 @@ static av_cold int cinaudio_decode_init(AVCodecContext *avctx)
 {
     CinAudioContext *cin = avctx->priv_data;
 
-    cin->avctx = avctx;
+    if (avctx->channels != 1) {
+        av_log_ask_for_sample(avctx, "Number of channels is not supported\n");
+        return AVERROR_PATCHWELCOME;
+    }
+
     cin->initial_decode_frame = 1;
     cin->delta = 0;
     avctx->sample_fmt = AV_SAMPLE_FMT_S16;
@@ -322,29 +326,35 @@ static int cinaudio_decode_frame(AVCodecContext *avctx,
                                  AVPacket *avpkt)
 {
     const uint8_t *buf = avpkt->data;
-    int buf_size = avpkt->size;
     CinAudioContext *cin = avctx->priv_data;
-    const uint8_t *src = buf;
-    int16_t *samples = (int16_t *)data;
-
-    buf_size = FFMIN(buf_size, *data_size/2);
+    const uint8_t *buf_end = buf + avpkt->size;
+    int16_t *samples = data;
+    int delta, out_size;
+
+    out_size = (avpkt->size - cin->initial_decode_frame) *
+               av_get_bytes_per_sample(avctx->sample_fmt);
+    if (*data_size < out_size) {
+        av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
+        return AVERROR(EINVAL);
+    }
 
+    delta = cin->delta;
     if (cin->initial_decode_frame) {
         cin->initial_decode_frame = 0;
-        cin->delta = (int16_t)AV_RL16(src); src += 2;
-        *samples++ = cin->delta;
-        buf_size -= 2;
+        delta = sign_extend(AV_RL16(buf), 16);
+        buf += 2;
+        *samples++ = delta;
     }
-    while (buf_size > 0) {
-        cin->delta += cinaudio_delta16_table[*src++];
-        cin->delta = av_clip_int16(cin->delta);
-        *samples++ = cin->delta;
-        --buf_size;
+    while (buf < buf_end) {
+        delta += cinaudio_delta16_table[*buf++];
+        delta = av_clip_int16(delta);
+        *samples++ = delta;
     }
+    cin->delta = delta;
 
-    *data_size = (uint8_t *)samples - (uint8_t *)data;
+    *data_size = out_size;
 
-    return src - buf;
+    return avpkt->size;
 }
 
 
diff --git a/libavcodec/flacdec.c b/libavcodec/flacdec.c
index cebe3e3e62..c140440436 100644
--- a/libavcodec/flacdec.c
+++ b/libavcodec/flacdec.c
@@ -587,7 +587,8 @@ static int flac_decode_frame(AVCodecContext *avctx,
     bytes_read = (get_bits_count(&s->gb)+7)/8;
 
     /* check if allocated data size is large enough for output */
-    output_size = s->blocksize * s->channels * (s->is32 ? 4 : 2);
+    output_size = s->blocksize * s->channels *
+                  av_get_bytes_per_sample(avctx->sample_fmt);
     if (output_size > alloc_data_size) {
         av_log(s->avctx, AV_LOG_ERROR, "output data size is larger than "
                                        "allocated data size\n");
diff --git a/libavcodec/h264.c b/libavcodec/h264.c
index 7792d03ce7..3302f71993 100644
--- a/libavcodec/h264.c
+++ b/libavcodec/h264.c
@@ -1815,7 +1815,7 @@ static av_always_inline void hl_decode_mb_predict_luma(H264Context *h, int mb_ty
                     static const uint8_t dc_mapping[16] = { 0*16, 1*16, 4*16, 5*16, 2*16, 3*16, 6*16, 7*16,
                                                             8*16, 9*16,12*16,13*16,10*16,11*16,14*16,15*16};
                     for(i = 0; i < 16; i++)
-                        dctcoef_set(h->mb+p*256, pixel_shift, dc_mapping[i], dctcoef_get(h->mb_luma_dc[p], pixel_shift, i));
+                        dctcoef_set(h->mb+(p*256 << pixel_shift), pixel_shift, dc_mapping[i], dctcoef_get(h->mb_luma_dc[p], pixel_shift, i));
                 }
             }
         }else
@@ -2033,7 +2033,7 @@ static av_always_inline void hl_decode_mb_internal(H264Context *h, int simple, i
                         }
                         if (chroma422) {
                             for(i=j*16+4; i<j*16+8; i++){
-                                if(h->non_zero_count_cache[ scan8[i] ] || dctcoef_get(h->mb, pixel_shift, i*16))
+                                if(h->non_zero_count_cache[ scan8[i+4] ] || dctcoef_get(h->mb, pixel_shift, i*16))
                                     idct_add   (dest[j-1] + block_offset[i+4], h->mb + (i*16 << pixel_shift), uvlinesize);
                             }
                         }
diff --git a/libavcodec/utvideo.c b/libavcodec/utvideo.c
index aac3969b15..4c3b2a1621 100644
--- a/libavcodec/utvideo.c
+++ b/libavcodec/utvideo.c
@@ -66,7 +66,7 @@ static int huff_cmp(const void *a, const void *b)
     return (aa->len - bb->len)*256 + aa->sym - bb->sym;
 }
 
-static int build_huff(const uint8_t *src, VLC *vlc)
+static int build_huff(const uint8_t *src, VLC *vlc, int *fsym)
 {
     int i;
     HuffEntry he[256];
@@ -76,13 +76,18 @@ static int build_huff(const uint8_t *src, VLC *vlc)
     uint8_t syms[256];
     uint32_t code;
 
+    *fsym = -1;
     for (i = 0; i < 256; i++) {
         he[i].sym = i;
         he[i].len = *src++;
     }
     qsort(he, 256, sizeof(*he), huff_cmp);
 
-    if (!he[0].len || he[0].len > 32)
+    if (!he[0].len) {
+        *fsym = he[0].sym;
+        return 0;
+    }
+    if (he[0].len > 32)
         return -1;
 
     last = 255;
@@ -112,12 +117,37 @@ static int decode_plane(UtvideoContext *c, int plane_no,
     int sstart, send;
     VLC vlc;
     GetBitContext gb;
-    int prev;
+    int prev, fsym;
+    const int cmask = ~(!plane_no && c->avctx->pix_fmt == PIX_FMT_YUV420P);
 
-    if (build_huff(src, &vlc)) {
+    if (build_huff(src, &vlc, &fsym)) {
         av_log(c->avctx, AV_LOG_ERROR, "Cannot build Huffman codes\n");
         return AVERROR_INVALIDDATA;
     }
+    if (fsym >= 0) { // build_huff reported a symbol to fill slices with
+        send = 0;
+        for (slice = 0; slice < c->slices; slice++) {
+            uint8_t *dest;
+
+            sstart = send;
+            send   = (height * (slice + 1) / c->slices) & cmask;
+            dest   = dst + sstart * stride;
+
+            prev = 0x80;
+            for (j = sstart; j < send; j++) {
+                for (i = 0; i < width * step; i += step) {
+                    pix = fsym;
+                    if (use_pred) {
+                        prev += pix;
+                        pix   = prev;
+                    }
+                    dest[i] = pix;
+                }
+                dest += stride;
+            }
+        }
+        return 0;
+    }
 
     src      += 256;
     src_size -= 256;
@@ -128,7 +158,7 @@ static int decode_plane(UtvideoContext *c, int plane_no,
         int slice_data_start, slice_data_end, slice_size;
 
         sstart = send;
-        send   = height * (slice + 1) / c->slices;
+        send   = (height * (slice + 1) / c->slices) & cmask;
         dest   = dst + sstart * stride;
 
         // slice offset and size validation was done earlier
@@ -204,16 +234,17 @@ static void restore_rgb_planes(uint8_t *src, int step, int stride, int width, in
 }
 
 static void restore_median(uint8_t *src, int step, int stride,
-                           int width, int height, int slices)
+                           int width, int height, int slices, int rmode)
 {
     int i, j, slice;
     int A, B, C;
     uint8_t *bsrc;
     int slice_start, slice_height;
+    const int cmask = ~rmode;
 
     for (slice = 0; slice < slices; slice++) {
-        slice_start = (slice * height) / slices;
-        slice_height = ((slice + 1) * height) / slices - slice_start;
+        slice_start = ((slice * height) / slices) & cmask;
+        slice_height = ((((slice + 1) * height) / slices) & cmask) - slice_start;
 
         bsrc = src + slice_start * stride;
 
@@ -337,7 +368,7 @@ static int decode_frame(AVCodecContext *avctx, void *data, int *data_size, AVPac
             if (c->frame_pred == PRED_MEDIAN)
                 restore_median(c->pic.data[0] + rgb_order[i], c->planes,
                                c->pic.linesize[0], avctx->width, avctx->height,
-                               c->slices);
+                               c->slices, 0);
         }
         restore_rgb_planes(c->pic.data[0], c->planes, c->pic.linesize[0],
                            avctx->width, avctx->height);
@@ -353,7 +384,7 @@ static int decode_frame(AVCodecContext *avctx, void *data, int *data_size, AVPac
             if (c->frame_pred == PRED_MEDIAN)
                 restore_median(c->pic.data[i], 1, c->pic.linesize[i],
                                avctx->width >> !!i, avctx->height >> !!i,
-                               c->slices);
+                               c->slices, !i);
         }
         break;
     case PIX_FMT_YUV422P:
@@ -366,7 +397,7 @@ static int decode_frame(AVCodecContext *avctx, void *data, int *data_size, AVPac
                 return ret;
             if (c->frame_pred == PRED_MEDIAN)
                 restore_median(c->pic.data[i], 1, c->pic.linesize[i],
-                               avctx->width >> !!i, avctx->height, c->slices);
+                               avctx->width >> !!i, avctx->height, c->slices, 0);
         }
         break;
     }
diff --git a/libavcodec/vp3.c b/libavcodec/vp3.c
index 77a3151b45..1f8841acb7 100644
--- a/libavcodec/vp3.c
+++ b/libavcodec/vp3.c
@@ -45,6 +45,7 @@
 #define FRAGMENT_PIXELS 8
 
 static av_cold int vp3_decode_end(AVCodecContext *avctx);
+static void vp3_decode_flush(AVCodecContext *avctx);
 
 //FIXME split things out into their own arrays
 typedef struct Vp3Fragment {
@@ -890,7 +891,7 @@ static int unpack_vlcs(Vp3DecodeContext *s, GetBitContext *gb,
             /* decode a VLC into a token */
             token = get_vlc2(gb, vlc_table, 11, 3);
             /* use the token to get a zero run, a coefficient, and an eob run */
-            if (token <= 6) {
+            if ((unsigned) token <= 6U) {
                 eob_run = eob_run_base[token];
                 if (eob_run_get_bits[token])
                     eob_run += get_bits(gb, eob_run_get_bits[token]);
@@ -908,7 +909,7 @@ static int unpack_vlcs(Vp3DecodeContext *s, GetBitContext *gb,
                     coeff_i        += eob_run;
                     eob_run = 0;
                 }
-            } else {
+            } else if (token >= 0) {
                 bits_to_get = coeff_get_bits[token];
                 if (bits_to_get)
                     bits_to_get = get_bits(gb, bits_to_get);
@@ -942,6 +943,10 @@ static int unpack_vlcs(Vp3DecodeContext *s, GetBitContext *gb,
                 for (i = coeff_index+1; i <= coeff_index+zero_run; i++)
                     s->num_coded_frags[plane][i]--;
                 coeff_i++;
+            } else {
+                av_log(s->avctx, AV_LOG_ERROR,
+                       "Invalid token %d\n", token);
+                return -1;
             }
     }
 
@@ -991,6 +996,8 @@ static int unpack_dct_coeffs(Vp3DecodeContext *s, GetBitContext *gb)
     /* unpack the Y plane DC coefficients */
     residual_eob_run = unpack_vlcs(s, gb, &s->dc_vlc[dc_y_table], 0,
         0, residual_eob_run);
+    if (residual_eob_run < 0)
+        return residual_eob_run;
 
     /* reverse prediction of the Y-plane DC coefficients */
     reverse_dc_prediction(s, 0, s->fragment_width[0], s->fragment_height[0]);
@@ -998,8 +1005,12 @@ static int unpack_dct_coeffs(Vp3DecodeContext *s, GetBitContext *gb)
     /* unpack the C plane DC coefficients */
     residual_eob_run = unpack_vlcs(s, gb, &s->dc_vlc[dc_c_table], 0,
         1, residual_eob_run);
+    if (residual_eob_run < 0)
+        return residual_eob_run;
     residual_eob_run = unpack_vlcs(s, gb, &s->dc_vlc[dc_c_table], 0,
         2, residual_eob_run);
+    if (residual_eob_run < 0)
+        return residual_eob_run;
 
     /* reverse prediction of the C-plane DC coefficients */
     if (!(s->avctx->flags & CODEC_FLAG_GRAY))
@@ -1036,11 +1047,17 @@ static int unpack_dct_coeffs(Vp3DecodeContext *s, GetBitContext *gb)
     for (i = 1; i <= 63; i++) {
             residual_eob_run = unpack_vlcs(s, gb, y_tables[i], i,
                 0, residual_eob_run);
+            if (residual_eob_run < 0)
+                return residual_eob_run;
 
             residual_eob_run = unpack_vlcs(s, gb, c_tables[i], i,
                 1, residual_eob_run);
+            if (residual_eob_run < 0)
+                return residual_eob_run;
             residual_eob_run = unpack_vlcs(s, gb, c_tables[i], i,
                 2, residual_eob_run);
+            if (residual_eob_run < 0)
+                return residual_eob_run;
     }
 
     return 0;
@@ -1777,10 +1794,15 @@ static int vp3_update_thread_context(AVCodecContext *dst, const AVCodecContext *
     Vp3DecodeContext *s = dst->priv_data, *s1 = src->priv_data;
     int qps_changed = 0, i, err;
 
+#define copy_fields(to, from, start_field, end_field) memcpy(&to->start_field, &from->start_field, (char*)&to->end_field - (char*)&to->start_field)
+
     if (!s1->current_frame.data[0]
         ||s->width != s1->width
-        ||s->height!= s1->height)
+        ||s->height!= s1->height) {
+        if (s != s1)
+            copy_fields(s, s1, golden_frame, current_frame);
         return -1;
+    }
 
     if (s != s1) {
         // init tables if the first frame hasn't been decoded
@@ -1796,8 +1818,6 @@ static int vp3_update_thread_context(AVCodecContext *dst, const AVCodecContext *
             memcpy(s->motion_val[1], s1->motion_val[1], c_fragment_count * sizeof(*s->motion_val[1]));
         }
 
-#define copy_fields(to, from, start_field, end_field) memcpy(&to->start_field, &from->start_field, (char*)&to->end_field - (char*)&to->start_field)
-
         // copy previous frame data
         copy_fields(s, s1, golden_frame, dsp);
 
@@ -1990,9 +2010,6 @@ static av_cold int vp3_decode_end(AVCodecContext *avctx)
     Vp3DecodeContext *s = avctx->priv_data;
     int i;
 
-    if (avctx->is_copy && !s->current_frame.data[0])
-        return 0;
-
     av_free(s->superblock_coding);
     av_free(s->all_fragments);
     av_free(s->coded_fragment_list[0]);
@@ -2339,6 +2356,23 @@ static void vp3_decode_flush(AVCodecContext *avctx)
         ff_thread_release_buffer(avctx, &s->current_frame);
 }
 
+static int vp3_init_thread_copy(AVCodecContext *avctx)
+{
+    Vp3DecodeContext *s = avctx->priv_data;
+
+    s->superblock_coding      = NULL;
+    s->all_fragments          = NULL;
+    s->coded_fragment_list[0] = NULL;
+    s->dct_tokens_base        = NULL;
+    s->superblock_fragments   = NULL;
+    s->macroblock_coding      = NULL;
+    s->motion_val[0]          = NULL;
+    s->motion_val[1]          = NULL;
+    s->edge_emu_buffer        = NULL;
+
+    return 0;
+}
+
 AVCodec ff_theora_decoder = {
     .name           = "theora",
     .type           = AVMEDIA_TYPE_VIDEO,
@@ -2350,6 +2384,7 @@ AVCodec ff_theora_decoder = {
     .capabilities   = CODEC_CAP_DR1 | CODEC_CAP_DRAW_HORIZ_BAND | CODEC_CAP_FRAME_THREADS,
     .flush = vp3_decode_flush,
     .long_name = NULL_IF_CONFIG_SMALL("Theora"),
+    .init_thread_copy      = ONLY_IF_THREADS_ENABLED(vp3_init_thread_copy),
     .update_thread_context = ONLY_IF_THREADS_ENABLED(vp3_update_thread_context)
 };
 #endif
@@ -2365,5 +2400,6 @@ AVCodec ff_vp3_decoder = {
     .capabilities   = CODEC_CAP_DR1 | CODEC_CAP_DRAW_HORIZ_BAND | CODEC_CAP_FRAME_THREADS,
     .flush = vp3_decode_flush,
     .long_name = NULL_IF_CONFIG_SMALL("On2 VP3"),
+    .init_thread_copy      = ONLY_IF_THREADS_ENABLED(vp3_init_thread_copy),
     .update_thread_context = ONLY_IF_THREADS_ENABLED(vp3_update_thread_context)
 };
diff --git a/libavcodec/vp8.c b/libavcodec/vp8.c
index 9b07608078..37bdcf7525 100644
--- a/libavcodec/vp8.c
+++ b/libavcodec/vp8.c
@@ -50,7 +50,8 @@ static int vp8_alloc_frame(VP8Context *s, AVFrame *f)
     int ret;
     if ((ret = ff_thread_get_buffer(s->avctx, f)) < 0)
         return ret;
-    if (!s->maps_are_invalid && s->num_maps_to_be_freed) {
+    if (s->num_maps_to_be_freed) {
+        assert(!s->maps_are_invalid);
         f->ref_index[0] = s->segmentation_maps[--s->num_maps_to_be_freed];
     } else if (!(f->ref_index[0] = av_mallocz(s->mb_width * s->mb_height))) {
         ff_thread_release_buffer(s->avctx, f);
@@ -59,39 +60,50 @@ static int vp8_alloc_frame(VP8Context *s, AVFrame *f)
     return 0;
 }
 
-static void vp8_release_frame(VP8Context *s, AVFrame *f, int is_close)
+static void vp8_release_frame(VP8Context *s, AVFrame *f, int prefer_delayed_free, int can_direct_free)
 {
-    if (!is_close) {
-        if (f->ref_index[0]) {
-            assert(s->num_maps_to_be_freed < FF_ARRAY_ELEMS(s->segmentation_maps));
-            s->segmentation_maps[s->num_maps_to_be_freed++] = f->ref_index[0];
+    if (f->ref_index[0]) {
+        if (prefer_delayed_free) {
+            /* Upon a size change, we want to free the maps but other threads may still
+             * be using them, so queue them. Upon a seek, all threads are inactive so
+             * we want to cache one to prevent re-allocation in the next decoding
+             * iteration, but the rest we can free directly. */
+            int max_queued_maps = can_direct_free ? 1 : FF_ARRAY_ELEMS(s->segmentation_maps);
+            if (s->num_maps_to_be_freed < max_queued_maps) {
+                s->segmentation_maps[s->num_maps_to_be_freed++] = f->ref_index[0];
+            } else if (can_direct_free) /* vp8_decode_flush(), but our queue is full */ {
+                av_free(f->ref_index[0]);
+            } /* else: MEMLEAK (should never happen, but better that than crash) */
             f->ref_index[0] = NULL;
+        } else /* vp8_decode_free() */ {
+            av_free(f->ref_index[0]);
         }
-    } else {
-        av_freep(&f->ref_index[0]);
     }
     ff_thread_release_buffer(s->avctx, f);
 }
 
-static void vp8_decode_flush_impl(AVCodecContext *avctx, int force, int is_close)
+static void vp8_decode_flush_impl(AVCodecContext *avctx,
+                                  int prefer_delayed_free, int can_direct_free, int free_mem)
 {
     VP8Context *s = avctx->priv_data;
     int i;
 
-    if (!avctx->is_copy || force) {
+    if (!avctx->is_copy) {
         for (i = 0; i < 5; i++)
             if (s->frames[i].data[0])
-                vp8_release_frame(s, &s->frames[i], is_close);
+                vp8_release_frame(s, &s->frames[i], prefer_delayed_free, can_direct_free);
     }
     memset(s->framep, 0, sizeof(s->framep));
 
-    free_buffers(s);
-    s->maps_are_invalid = 1;
+    if (free_mem) {
+        free_buffers(s);
+        s->maps_are_invalid = 1;
+    }
 }
 
 static void vp8_decode_flush(AVCodecContext *avctx)
 {
-    vp8_decode_flush_impl(avctx, 0, 0);
+    vp8_decode_flush_impl(avctx, 1, 1, 0);
 }
 
 static int update_dimensions(VP8Context *s, int width, int height)
@@ -101,7 +113,7 @@ static int update_dimensions(VP8Context *s, int width, int height)
         if (av_image_check_size(width, height, 0, s->avctx))
             return AVERROR_INVALIDDATA;
 
-        vp8_decode_flush_impl(s->avctx, 1, 0);
+        vp8_decode_flush_impl(s->avctx, 1, 0, 1);
 
         avcodec_set_dimensions(s->avctx, width, height);
     }
@@ -1581,7 +1593,7 @@ static int vp8_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
             &s->frames[i] != s->framep[VP56_FRAME_PREVIOUS] &&
             &s->frames[i] != s->framep[VP56_FRAME_GOLDEN] &&
             &s->frames[i] != s->framep[VP56_FRAME_GOLDEN2])
-            vp8_release_frame(s, &s->frames[i], 0);
+            vp8_release_frame(s, &s->frames[i], 1, 0);
 
     // find a free buffer
     for (i = 0; i < 5; i++)
@@ -1597,7 +1609,7 @@ static int vp8_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
         abort();
     }
     if (curframe->data[0])
-        ff_thread_release_buffer(avctx, curframe);
+        vp8_release_frame(s, curframe, 1, 0);
 
     curframe->key_frame = s->keyframe;
     curframe->pict_type = s->keyframe ? AV_PICTURE_TYPE_I : AV_PICTURE_TYPE_P;
@@ -1778,7 +1790,7 @@ static av_cold int vp8_decode_init(AVCodecContext *avctx)
 
 static av_cold int vp8_decode_free(AVCodecContext *avctx)
 {
-    vp8_decode_flush_impl(avctx, 0, 1);
+    vp8_decode_flush_impl(avctx, 0, 1, 1);
     release_queued_segmaps(avctx->priv_data, 1);
     return 0;
 }
diff --git a/libavformat/avformat.h b/libavformat/avformat.h
index 60ff6dcf03..90cf3ad44d 100644
--- a/libavformat/avformat.h
+++ b/libavformat/avformat.h
@@ -676,6 +676,7 @@ typedef struct AVStream {
         int duration_count;
         double duration_error[2][2][MAX_STD_TIMEBASES];
         int64_t codec_info_duration;
+        int nb_decoded_frames;
     } *info;
 
     /**
diff --git a/libavformat/dsicin.c b/libavformat/dsicin.c
index 09e80e944f..ab098dc017 100644
--- a/libavformat/dsicin.c
+++ b/libavformat/dsicin.c
@@ -131,9 +131,8 @@ static int cin_read_header(AVFormatContext *s, AVFormatParameters *ap)
     st->codec->codec_tag = 0;  /* no tag */
     st->codec->channels = 1;
     st->codec->sample_rate = 22050;
-    st->codec->bits_per_coded_sample = 16;
+    st->codec->bits_per_coded_sample = 8;
     st->codec->bit_rate = st->codec->sample_rate * st->codec->bits_per_coded_sample * st->codec->channels;
-    st->codec->block_align = st->codec->channels * st->codec->bits_per_coded_sample;
 
     return 0;
 }
@@ -211,7 +210,8 @@ static int cin_read_packet(AVFormatContext *s, AVPacket *pkt)
 
     pkt->stream_index = cin->audio_stream_index;
     pkt->pts = cin->audio_stream_pts;
-    cin->audio_stream_pts += cin->audio_buffer_size * 2 / cin->file_header.audio_frame_size;
+    pkt->duration = cin->audio_buffer_size - (pkt->pts == 0);
+    cin->audio_stream_pts += pkt->duration;
     cin->audio_buffer_size = 0;
     return 0;
 }
diff --git a/libavformat/utils.c b/libavformat/utils.c
index 8a78308447..2cf42d3c43 100644
--- a/libavformat/utils.c
+++ b/libavformat/utils.c
@@ -2200,7 +2200,7 @@ static int has_codec_parameters(AVCodecContext *avctx)
 static int has_decode_delay_been_guessed(AVStream *st)
 {
     return st->codec->codec_id != CODEC_ID_H264 ||
-        st->codec_info_nb_frames >= 6 + st->codec->has_b_frames;
+        st->info->nb_decoded_frames >= 6;
 }
 
 static int try_decode_frame(AVStream *st, AVPacket *avpkt, AVDictionary **options)
@@ -2226,6 +2226,8 @@ static int try_decode_frame(AVStream *st, AVPacket *avpkt, AVDictionary **option
             avcodec_get_frame_defaults(&picture);
             ret = avcodec_decode_video2(st->codec, &picture,
                                         &got_picture, avpkt);
+            if (got_picture)
+                st->info->nb_decoded_frames++;
             break;
         case AVMEDIA_TYPE_AUDIO:
             data_size = FFMAX(avpkt->size, AVCODEC_MAX_AUDIO_FRAME_SIZE);