Merge remote-tracking branch 'qatar/master'

* qatar/master: (35 commits)
  libopencore-amr: check output buffer size before decoding
  libopencore-amr: remove unneeded buf_size==0 check.
  libopencore-amr: remove unneeded frame_count field.
  aac_latm: remove unneeded check for zero-size packet.
  pcmdec: fix output buffer size check by calculating the actual output size prior to decoding.
  pcmdec: move codec-specific variable declarations to the corresponding codec blocks.
  pcmdec: return buf_size instead of src-buf.
  avcodec: remove the Zork PCM encoder.
  pcm_zork: use AV_SAMPLE_FMT_U8 instead of shifting all samples by 8.
  pcmenc: remove unneeded sample_fmt check.
  pcmdec: move number of channels check to pcm_decode_init()
  pcmdec: remove unnecessary check for sample_fmt change
  pcmdec: move DVD PCM bits_per_coded_sample check near to the code that sets the sample size.
  pcmdec: do not needlessly set *data_size to 0
  alacdec: remove unneeded NULL or zero-size packet checks.
  alacdec: simplify buffer allocation by using FF_ALLOC_OR_GOTO()
  alacdec: ask for a sample for unsupported sample depths.
  alacdec: cosmetics: use 'ch' instead of 'chan' to iterate channels
  alacdec: move some declarations to the top of the function
  alacdec: always use get_sbits_long() for uncompressed samples
  ...

Conflicts:
	libavcodec/pcm.c
	tests/ref/acodec/pcm

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

doc/general.texi

@@ -683,7 +683,7 @@ following image formats are supported:
 @item PCM unsigned 24-bit little-endian  @tab  X  @tab  X
 @item PCM unsigned 32-bit big-endian  @tab  X  @tab  X
 @item PCM unsigned 32-bit little-endian  @tab  X  @tab  X
-@item PCM Zork               @tab  X  @tab  X
+@item PCM Zork               @tab     @tab  X
 @item QCELP / PureVoice      @tab     @tab  X
 @item QDesign Music Codec 2  @tab     @tab  X
     @tab There are still some distortions.

libavcodec/Makefile

@@ -508,7 +508,6 @@ OBJS-$(CONFIG_PCM_U32BE_ENCODER)          += pcm.o
 OBJS-$(CONFIG_PCM_U32LE_DECODER)          += pcm.o
 OBJS-$(CONFIG_PCM_U32LE_ENCODER)          += pcm.o
 OBJS-$(CONFIG_PCM_ZORK_DECODER)           += pcm.o
-OBJS-$(CONFIG_PCM_ZORK_ENCODER)           += pcm.o
 
 OBJS-$(CONFIG_ADPCM_4XM_DECODER)          += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_ADX_DECODER)          += adxdec.o

libavcodec/aacdec.c

@@ -2501,9 +2501,6 @@ static int latm_decode_frame(AVCodecContext *avctx, void *out, int *out_size,
     int                 muxlength, err;
     GetBitContext       gb;
 
-    if (avpkt->size == 0)
-        return 0;
-
     init_get_bits(&gb, avpkt->data, avpkt->size * 8);
 
     // check for LOAS sync word

libavcodec/alac.c

@@ -72,7 +72,7 @@ typedef struct {
 
     int32_t *outputsamples_buffer[MAX_CHANNELS];
 
-    int32_t *wasted_bits_buffer[MAX_CHANNELS];
+    int32_t *extra_bits_buffer[MAX_CHANNELS];
 
     /* stuff from setinfo */
     uint32_t setinfo_max_samples_per_frame; /* 0x1000 = 4096 */    /* max samples per frame? */
@@ -82,58 +82,9 @@ typedef struct {
     uint8_t setinfo_rice_kmodifier; /* 0x0e */
     /* end setinfo stuff */
 
-    int wasted_bits;
+    int extra_bits;                         /**< number of extra bits beyond 16-bit */
 } ALACContext;
 
-static void allocate_buffers(ALACContext *alac)
-{
-    int chan;
-    for (chan = 0; chan < MAX_CHANNELS; chan++) {
-        alac->predicterror_buffer[chan] =
-            av_malloc(alac->setinfo_max_samples_per_frame * 4);
-
-        alac->outputsamples_buffer[chan] =
-            av_malloc(alac->setinfo_max_samples_per_frame * 4);
-
-        alac->wasted_bits_buffer[chan] = av_malloc(alac->setinfo_max_samples_per_frame * 4);
-    }
-}
-
-static int alac_set_info(ALACContext *alac)
-{
-    const unsigned char *ptr = alac->avctx->extradata;
-
-    ptr += 4; /* size */
-    ptr += 4; /* alac */
-    ptr += 4; /* 0 ? */
-
-    if(AV_RB32(ptr) >= UINT_MAX/4){
-        av_log(alac->avctx, AV_LOG_ERROR, "setinfo_max_samples_per_frame too large\n");
-        return -1;
-    }
-
-    /* buffer size / 2 ? */
-    alac->setinfo_max_samples_per_frame = bytestream_get_be32(&ptr);
-    ptr++;                          /* ??? */
-    alac->setinfo_sample_size           = *ptr++;
-    if (alac->setinfo_sample_size > 32) {
-        av_log(alac->avctx, AV_LOG_ERROR, "setinfo_sample_size too large\n");
-        return -1;
-    }
-    alac->setinfo_rice_historymult      = *ptr++;
-    alac->setinfo_rice_initialhistory   = *ptr++;
-    alac->setinfo_rice_kmodifier        = *ptr++;
-    ptr++;                         /* channels? */
-    bytestream_get_be16(&ptr);      /* ??? */
-    bytestream_get_be32(&ptr);      /* max coded frame size */
-    bytestream_get_be32(&ptr);      /* bitrate ? */
-    bytestream_get_be32(&ptr);      /* samplerate */
-
-    allocate_buffers(alac);
-
-    return 0;
-}
-
 static inline int decode_scalar(GetBitContext *gb, int k, int limit, int readsamplesize){
     /* read x - number of 1s before 0 represent the rice */
     int x = get_unary_0_9(gb);
@@ -347,93 +298,56 @@ static void predictor_decompress_fir_adapt(int32_t *error_buffer,
     }
 }
 
-static void reconstruct_stereo_16(int32_t *buffer[MAX_CHANNELS],
+static void decorrelate_stereo(int32_t *buffer[MAX_CHANNELS],
-                                  int16_t *buffer_out,
+                               int numsamples, uint8_t interlacing_shift,
-                                  int numchannels, int numsamples,
+                               uint8_t interlacing_leftweight)
-                                  uint8_t interlacing_shift,
-                                  uint8_t interlacing_leftweight)
 {
     int i;
-    if (numsamples <= 0)
-        return;
 
-    /* weighted interlacing */
-    if (interlacing_leftweight) {
-        for (i = 0; i < numsamples; i++) {
-            int32_t a, b;
-
-            a = buffer[0][i];
-            b = buffer[1][i];
-
-            a -= (b * interlacing_leftweight) >> interlacing_shift;
-            b += a;
-
-            buffer_out[i*numchannels] = b;
-            buffer_out[i*numchannels + 1] = a;
-        }
-
-        return;
-    }
-
-    /* otherwise basic interlacing took place */
     for (i = 0; i < numsamples; i++) {
-        int16_t left, right;
+        int32_t a, b;
 
-        left = buffer[0][i];
+        a = buffer[0][i];
-        right = buffer[1][i];
+        b = buffer[1][i];
 
-        buffer_out[i*numchannels] = left;
+        a -= (b * interlacing_leftweight) >> interlacing_shift;
-        buffer_out[i*numchannels + 1] = right;
+        b += a;
+
+        buffer[0][i] = b;
+        buffer[1][i] = a;
     }
 }
 
-static void decorrelate_stereo_24(int32_t *buffer[MAX_CHANNELS],
+static void append_extra_bits(int32_t *buffer[MAX_CHANNELS],
-                                  int32_t *buffer_out,
+                              int32_t *extra_bits_buffer[MAX_CHANNELS],
-                                  int32_t *wasted_bits_buffer[MAX_CHANNELS],
+                              int extra_bits, int numchannels, int numsamples)
-                                  int wasted_bits,
+{
-                                  int numchannels, int numsamples,
+    int i, ch;
-                                  uint8_t interlacing_shift,
+
-                                  uint8_t interlacing_leftweight)
+    for (ch = 0; ch < numchannels; ch++)
+        for (i = 0; i < numsamples; i++)
+            buffer[ch][i] = (buffer[ch][i] << extra_bits) | extra_bits_buffer[ch][i];
+}
+
+static void interleave_stereo_16(int32_t *buffer[MAX_CHANNELS],
+                                 int16_t *buffer_out, int numsamples)
 {
     int i;
 
-    if (numsamples <= 0)
+    for (i = 0; i < numsamples; i++) {
-        return;
+        *buffer_out++ = buffer[0][i];
+        *buffer_out++ = buffer[1][i];
+    }
+}
 
-    /* weighted interlacing */
+static void interleave_stereo_24(int32_t *buffer[MAX_CHANNELS],
-    if (interlacing_leftweight) {
+                                 int32_t *buffer_out, int numsamples)
-        for (i = 0; i < numsamples; i++) {
+{
-            int32_t a, b;
+    int i;
 
-            a = buffer[0][i];
+    for (i = 0; i < numsamples; i++) {
-            b = buffer[1][i];
+        *buffer_out++ = buffer[0][i] << 8;
-
+        *buffer_out++ = buffer[1][i] << 8;
-            a -= (b * interlacing_leftweight) >> interlacing_shift;
-            b += a;
-
-            if (wasted_bits) {
-                b  = (b  << wasted_bits) | wasted_bits_buffer[0][i];
-                a  = (a  << wasted_bits) | wasted_bits_buffer[1][i];
-            }
-
-            buffer_out[i * numchannels]     = b << 8;
-            buffer_out[i * numchannels + 1] = a << 8;
-        }
-    } else {
-        for (i = 0; i < numsamples; i++) {
-            int32_t left, right;
-
-            left  = buffer[0][i];
-            right = buffer[1][i];
-
-            if (wasted_bits) {
-                left   = (left   << wasted_bits) | wasted_bits_buffer[0][i];
-                right  = (right  << wasted_bits) | wasted_bits_buffer[1][i];
-            }
-
-            buffer_out[i * numchannels]     = left  << 8;
-            buffer_out[i * numchannels + 1] = right << 8;
-        }
     }
 }
 
@@ -452,18 +366,14 @@ static int alac_decode_frame(AVCodecContext *avctx,
     int isnotcompressed;
     uint8_t interlacing_shift;
     uint8_t interlacing_leftweight;
-
+    int i, ch;
-    /* short-circuit null buffers */
-    if (!inbuffer || !input_buffer_size)
-        return -1;
 
     init_get_bits(&alac->gb, inbuffer, input_buffer_size * 8);
 
     channels = get_bits(&alac->gb, 3) + 1;
-    if (channels > MAX_CHANNELS) {
+    if (channels != avctx->channels) {
-        av_log(avctx, AV_LOG_ERROR, "channels > %d not supported\n",
+        av_log(avctx, AV_LOG_ERROR, "frame header channel count mismatch\n");
-               MAX_CHANNELS);
+        return AVERROR_INVALIDDATA;
-        return -1;
     }
 
     /* 2^result = something to do with output waiting.
@@ -476,7 +386,7 @@ static int alac_decode_frame(AVCodecContext *avctx,
     /* the output sample size is stored soon */
     hassize = get_bits1(&alac->gb);
 
-    alac->wasted_bits = get_bits(&alac->gb, 2) << 3;
+    alac->extra_bits = get_bits(&alac->gb, 2) << 3;
 
     /* whether the frame is compressed */
     isnotcompressed = get_bits1(&alac->gb);
@@ -491,17 +401,7 @@ static int alac_decode_frame(AVCodecContext *avctx,
     } else
         outputsamples = alac->setinfo_max_samples_per_frame;
 
-    switch (alac->setinfo_sample_size) {
+    alac->bytespersample = channels * av_get_bytes_per_sample(avctx->sample_fmt);
-    case 16: avctx->sample_fmt    = AV_SAMPLE_FMT_S16;
-             alac->bytespersample = channels << 1;
-             break;
-    case 24: avctx->sample_fmt    = AV_SAMPLE_FMT_S32;
-             alac->bytespersample = channels << 2;
-             break;
-    default: av_log(avctx, AV_LOG_ERROR, "Sample depth %d is not supported.\n",
-                    alac->setinfo_sample_size);
-             return -1;
-    }
 
     if(outputsamples > *outputsize / alac->bytespersample){
         av_log(avctx, AV_LOG_ERROR, "sample buffer too small\n");
@@ -509,7 +409,7 @@ static int alac_decode_frame(AVCodecContext *avctx,
     }
 
     *outputsize = outputsamples * alac->bytespersample;
-    readsamplesize = alac->setinfo_sample_size - (alac->wasted_bits) + channels - 1;
+    readsamplesize = alac->setinfo_sample_size - alac->extra_bits + channels - 1;
     if (readsamplesize > MIN_CACHE_BITS) {
         av_log(avctx, AV_LOG_ERROR, "readsamplesize too big (%d)\n", readsamplesize);
         return -1;
@@ -522,51 +422,49 @@ static int alac_decode_frame(AVCodecContext *avctx,
         int prediction_type[MAX_CHANNELS];
         int prediction_quantitization[MAX_CHANNELS];
         int ricemodifier[MAX_CHANNELS];
-        int i, chan;
 
         interlacing_shift = get_bits(&alac->gb, 8);
         interlacing_leftweight = get_bits(&alac->gb, 8);
 
-        for (chan = 0; chan < channels; chan++) {
+        for (ch = 0; ch < channels; ch++) {
-            prediction_type[chan] = get_bits(&alac->gb, 4);
+            prediction_type[ch] = get_bits(&alac->gb, 4);
-            prediction_quantitization[chan] = get_bits(&alac->gb, 4);
+            prediction_quantitization[ch] = get_bits(&alac->gb, 4);
 
-            ricemodifier[chan] = get_bits(&alac->gb, 3);
+            ricemodifier[ch] = get_bits(&alac->gb, 3);
-            predictor_coef_num[chan] = get_bits(&alac->gb, 5);
+            predictor_coef_num[ch] = get_bits(&alac->gb, 5);
 
             /* read the predictor table */
-            for (i = 0; i < predictor_coef_num[chan]; i++)
+            for (i = 0; i < predictor_coef_num[ch]; i++)
-                predictor_coef_table[chan][i] = (int16_t)get_bits(&alac->gb, 16);
+                predictor_coef_table[ch][i] = (int16_t)get_bits(&alac->gb, 16);
         }
 
-        if (alac->wasted_bits) {
+        if (alac->extra_bits) {
-            int i, ch;
             for (i = 0; i < outputsamples; i++) {
                 for (ch = 0; ch < channels; ch++)
-                    alac->wasted_bits_buffer[ch][i] = get_bits(&alac->gb, alac->wasted_bits);
+                    alac->extra_bits_buffer[ch][i] = get_bits(&alac->gb, alac->extra_bits);
             }
         }
-        for (chan = 0; chan < channels; chan++) {
+        for (ch = 0; ch < channels; ch++) {
             bastardized_rice_decompress(alac,
-                                        alac->predicterror_buffer[chan],
+                                        alac->predicterror_buffer[ch],
                                         outputsamples,
                                         readsamplesize,
                                         alac->setinfo_rice_initialhistory,
                                         alac->setinfo_rice_kmodifier,
-                                        ricemodifier[chan] * alac->setinfo_rice_historymult / 4,
+                                        ricemodifier[ch] * alac->setinfo_rice_historymult / 4,
                                         (1 << alac->setinfo_rice_kmodifier) - 1);
 
-            if (prediction_type[chan] == 0) {
+            if (prediction_type[ch] == 0) {
                 /* adaptive fir */
-                predictor_decompress_fir_adapt(alac->predicterror_buffer[chan],
+                predictor_decompress_fir_adapt(alac->predicterror_buffer[ch],
-                                               alac->outputsamples_buffer[chan],
+                                               alac->outputsamples_buffer[ch],
                                                outputsamples,
                                                readsamplesize,
-                                               predictor_coef_table[chan],
+                                               predictor_coef_table[ch],
-                                               predictor_coef_num[chan],
+                                               predictor_coef_num[ch],
-                                               prediction_quantitization[chan]);
+                                               prediction_quantitization[ch]);
             } else {
-                av_log(avctx, AV_LOG_ERROR, "FIXME: unhandled prediction type: %i\n", prediction_type[chan]);
+                av_log(avctx, AV_LOG_ERROR, "FIXME: unhandled prediction type: %i\n", prediction_type[ch]);
                 /* I think the only other prediction type (or perhaps this is
                  * just a boolean?) runs adaptive fir twice.. like:
                  * predictor_decompress_fir_adapt(predictor_error, tempout, ...)
@@ -577,44 +475,35 @@ static int alac_decode_frame(AVCodecContext *avctx,
         }
     } else {
         /* not compressed, easy case */
-        int i, chan;
+        for (i = 0; i < outputsamples; i++) {
-        if (alac->setinfo_sample_size <= 16) {
+            for (ch = 0; ch < channels; ch++) {
-        for (i = 0; i < outputsamples; i++)
+                alac->outputsamples_buffer[ch][i] = get_sbits_long(&alac->gb,
-            for (chan = 0; chan < channels; chan++) {
+                                                                   alac->setinfo_sample_size);
-                int32_t audiobits;
-
-                audiobits = get_sbits_long(&alac->gb, alac->setinfo_sample_size);
-
-                alac->outputsamples_buffer[chan][i] = audiobits;
-            }
-        } else {
-            for (i = 0; i < outputsamples; i++) {
-                for (chan = 0; chan < channels; chan++) {
-                    alac->outputsamples_buffer[chan][i] = get_bits(&alac->gb,
-                                                          alac->setinfo_sample_size);
-                    alac->outputsamples_buffer[chan][i] = sign_extend(alac->outputsamples_buffer[chan][i],
-                                                                      alac->setinfo_sample_size);
-                }
             }
         }
-        alac->wasted_bits = 0;
+        alac->extra_bits = 0;
         interlacing_shift = 0;
         interlacing_leftweight = 0;
     }
     if (get_bits(&alac->gb, 3) != 7)
         av_log(avctx, AV_LOG_ERROR, "Error : Wrong End Of Frame\n");
 
+    if (channels == 2 && interlacing_leftweight) {
+        decorrelate_stereo(alac->outputsamples_buffer, outputsamples,
+                           interlacing_shift, interlacing_leftweight);
+    }
+
+    if (alac->extra_bits) {
+        append_extra_bits(alac->outputsamples_buffer, alac->extra_bits_buffer,
+                          alac->extra_bits, alac->numchannels, outputsamples);
+    }
+
     switch(alac->setinfo_sample_size) {
     case 16:
         if (channels == 2) {
-            reconstruct_stereo_16(alac->outputsamples_buffer,
+            interleave_stereo_16(alac->outputsamples_buffer, outbuffer,
-                                  (int16_t*)outbuffer,
+                                 outputsamples);
-                                  alac->numchannels,
-                                  outputsamples,
-                                  interlacing_shift,
-                                  interlacing_leftweight);
         } else {
-            int i;
             for (i = 0; i < outputsamples; i++) {
                 ((int16_t*)outbuffer)[i] = alac->outputsamples_buffer[0][i];
             }
@@ -622,16 +511,9 @@ static int alac_decode_frame(AVCodecContext *avctx,
         break;
     case 24:
         if (channels == 2) {
-            decorrelate_stereo_24(alac->outputsamples_buffer,
+            interleave_stereo_24(alac->outputsamples_buffer, outbuffer,
-                                  outbuffer,
+                                 outputsamples);
-                                  alac->wasted_bits_buffer,
-                                  alac->wasted_bits,
-                                  alac->numchannels,
-                                  outputsamples,
-                                  interlacing_shift,
-                                  interlacing_leftweight);
         } else {
-            int i;
             for (i = 0; i < outputsamples; i++)
                 ((int32_t *)outbuffer)[i] = alac->outputsamples_buffer[0][i] << 8;
         }
@@ -644,11 +526,75 @@ static int alac_decode_frame(AVCodecContext *avctx,
     return input_buffer_size;
 }
 
-static av_cold int alac_decode_init(AVCodecContext * avctx)
+static av_cold int alac_decode_close(AVCodecContext *avctx)
 {
     ALACContext *alac = avctx->priv_data;
+
+    int ch;
+    for (ch = 0; ch < alac->numchannels; ch++) {
+        av_freep(&alac->predicterror_buffer[ch]);
+        av_freep(&alac->outputsamples_buffer[ch]);
+        av_freep(&alac->extra_bits_buffer[ch]);
+    }
+
+    return 0;
+}
+
+static int allocate_buffers(ALACContext *alac)
+{
+    int ch;
+    for (ch = 0; ch < alac->numchannels; ch++) {
+        int buf_size = alac->setinfo_max_samples_per_frame * sizeof(int32_t);
+
+        FF_ALLOC_OR_GOTO(alac->avctx, alac->predicterror_buffer[ch],
+                         buf_size, buf_alloc_fail);
+
+        FF_ALLOC_OR_GOTO(alac->avctx, alac->outputsamples_buffer[ch],
+                         buf_size, buf_alloc_fail);
+
+        FF_ALLOC_OR_GOTO(alac->avctx, alac->extra_bits_buffer[ch],
+                         buf_size, buf_alloc_fail);
+    }
+    return 0;
+buf_alloc_fail:
+    alac_decode_close(alac->avctx);
+    return AVERROR(ENOMEM);
+}
+
+static int alac_set_info(ALACContext *alac)
+{
+    const unsigned char *ptr = alac->avctx->extradata;
+
+    ptr += 4; /* size */
+    ptr += 4; /* alac */
+    ptr += 4; /* 0 ? */
+
+    if(AV_RB32(ptr) >= UINT_MAX/4){
+        av_log(alac->avctx, AV_LOG_ERROR, "setinfo_max_samples_per_frame too large\n");
+        return -1;
+    }
+
+    /* buffer size / 2 ? */
+    alac->setinfo_max_samples_per_frame = bytestream_get_be32(&ptr);
+    ptr++;                          /* ??? */
+    alac->setinfo_sample_size           = *ptr++;
+    alac->setinfo_rice_historymult      = *ptr++;
+    alac->setinfo_rice_initialhistory   = *ptr++;
+    alac->setinfo_rice_kmodifier        = *ptr++;
+    alac->numchannels                   = *ptr++;
+    bytestream_get_be16(&ptr);      /* ??? */
+    bytestream_get_be32(&ptr);      /* max coded frame size */
+    bytestream_get_be32(&ptr);      /* bitrate ? */
+    bytestream_get_be32(&ptr);      /* samplerate */
+
+    return 0;
+}
+
+static av_cold int alac_decode_init(AVCodecContext * avctx)
+{
+    int ret;
+    ALACContext *alac = avctx->priv_data;
     alac->avctx = avctx;
-    alac->numchannels = alac->avctx->channels;
 
     /* initialize from the extradata */
     if (alac->avctx->extradata_size != ALAC_EXTRADATA_SIZE) {
@@ -661,18 +607,34 @@ static av_cold int alac_decode_init(AVCodecContext * avctx)
         return -1;
     }
 
-    return 0;
+    switch (alac->setinfo_sample_size) {
-}
+    case 16: avctx->sample_fmt    = AV_SAMPLE_FMT_S16;
+             break;
+    case 24: avctx->sample_fmt    = AV_SAMPLE_FMT_S32;
+             break;
+    default: av_log_ask_for_sample(avctx, "Sample depth %d is not supported.\n",
+                                   alac->setinfo_sample_size);
+             return AVERROR_PATCHWELCOME;
+    }
 
-static av_cold int alac_decode_close(AVCodecContext *avctx)
+    if (alac->numchannels < 1) {
-{
+        av_log(avctx, AV_LOG_WARNING, "Invalid channel count\n");
-    ALACContext *alac = avctx->priv_data;
+        alac->numchannels = avctx->channels;
+    } else {
+        if (alac->numchannels > MAX_CHANNELS)
+            alac->numchannels = avctx->channels;
+        else
+            avctx->channels = alac->numchannels;
+    }
+    if (avctx->channels > MAX_CHANNELS) {
+        av_log(avctx, AV_LOG_ERROR, "Unsupported channel count: %d\n",
+               avctx->channels);
+        return AVERROR_PATCHWELCOME;
+    }
 
-    int chan;
+    if ((ret = allocate_buffers(alac)) < 0) {
-    for (chan = 0; chan < MAX_CHANNELS; chan++) {
+        av_log(avctx, AV_LOG_ERROR, "Error allocating buffers\n");
-        av_freep(&alac->predicterror_buffer[chan]);
+        return ret;
-        av_freep(&alac->outputsamples_buffer[chan]);
-        av_freep(&alac->wasted_bits_buffer[chan]);
     }
 
     return 0;

libavcodec/allcodecs.c

@@ -329,7 +329,7 @@ void avcodec_register_all(void)
     REGISTER_ENCDEC  (PCM_U24LE, pcm_u24le);
     REGISTER_ENCDEC  (PCM_U32BE, pcm_u32be);
     REGISTER_ENCDEC  (PCM_U32LE, pcm_u32le);
-    REGISTER_ENCDEC  (PCM_ZORK , pcm_zork);
+    REGISTER_DECODER (PCM_ZORK , pcm_zork);
 
     /* DPCM codecs */
     REGISTER_DECODER (INTERPLAY_DPCM, interplay_dpcm);

libavcodec/libopencore-amr.c

@@ -79,7 +79,6 @@ static int get_bitrate_mode(int bitrate, void *log_ctx)
 
 typedef struct AMRContext {
     AVClass *av_class;
-    int   frame_count;
     void *dec_state;
     void *enc_state;
     int   enc_bitrate;
@@ -100,7 +99,6 @@ static av_cold int amr_nb_decode_init(AVCodecContext *avctx)
 {
     AMRContext *s  = avctx->priv_data;
 
-    s->frame_count = 0;
     s->dec_state   = Decoder_Interface_init();
     if (!s->dec_state) {
         av_log(avctx, AV_LOG_ERROR, "Decoder_Interface_init error\n");
@@ -133,10 +131,16 @@ static int amr_nb_decode_frame(AVCodecContext *avctx, void *data,
     AMRContext *s      = avctx->priv_data;
     static const uint8_t block_size[16] = { 12, 13, 15, 17, 19, 20, 26, 31, 5, 0, 0, 0, 0, 0, 0, 0 };
     enum Mode dec_mode;
-    int packet_size;
+    int packet_size, out_size;
 
     av_dlog(avctx, "amr_decode_frame buf=%p buf_size=%d frame_count=%d!!\n",
-            buf, buf_size, s->frame_count);
+            buf, buf_size, avctx->frame_number);
+
+    out_size = 160 * 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);
+    }
 
     dec_mode    = (buf[0] >> 3) & 0x000F;
     packet_size = block_size[dec_mode] + 1;
@@ -147,12 +151,11 @@ static int amr_nb_decode_frame(AVCodecContext *avctx, void *data,
         return AVERROR_INVALIDDATA;
     }
 
-    s->frame_count++;
     av_dlog(avctx, "packet_size=%d buf= 0x%X %X %X %X\n",
               packet_size, buf[0], buf[1], buf[2], buf[3]);
     /* call decoder */
     Decoder_Interface_Decode(s->dec_state, buf, data, 0);
-    *data_size = 160 * 2;
+    *data_size = out_size;
 
     return packet_size;
 }
@@ -172,8 +175,6 @@ static av_cold int amr_nb_encode_init(AVCodecContext *avctx)
 {
     AMRContext *s = avctx->priv_data;
 
-    s->frame_count = 0;
-
     if (avctx->sample_rate != 8000) {
         av_log(avctx, AV_LOG_ERROR, "Only 8000Hz sample rate supported\n");
         return AVERROR(ENOSYS);
@@ -276,12 +277,14 @@ static int amr_wb_decode_frame(AVCodecContext *avctx, void *data,
     int buf_size       = avpkt->size;
     AMRWBContext *s    = avctx->priv_data;
     int mode;
-    int packet_size;
+    int packet_size, out_size;
     static const uint8_t block_size[16] = {18, 24, 33, 37, 41, 47, 51, 59, 61, 6, 6, 0, 0, 0, 1, 1};
 
-    if (!buf_size)
+    out_size = 320 * av_get_bytes_per_sample(avctx->sample_fmt);
-        /* nothing to do */
+    if (*data_size < out_size) {
-        return 0;
+        av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n");
+        return AVERROR(EINVAL);
+    }
 
     mode        = (buf[0] >> 3) & 0x000F;
     packet_size = block_size[mode];
@@ -293,7 +296,7 @@ static int amr_wb_decode_frame(AVCodecContext *avctx, void *data,
     }
 
     D_IF_decode(s->state, buf, data, _good_frame);
-    *data_size = 320 * 2;
+    *data_size = out_size;
     return packet_size;
 }
 

libavcodec/pcm.c

@@ -95,11 +95,6 @@ static int pcm_encode_frame(AVCodecContext *avctx,
     samples = data;
     dst = frame;
 
-    if (avctx->sample_fmt!=avctx->codec->sample_fmts[0]) {
-        av_log(avctx, AV_LOG_ERROR, "invalid sample_fmt\n");
-        return -1;
-    }
-
     switch(avctx->codec->id) {
     case CODEC_ID_PCM_U32LE:
         ENCODE(uint32_t, le32, samples, dst, n, 0, 0x80000000)
@@ -176,14 +171,6 @@ static int pcm_encode_frame(AVCodecContext *avctx,
         memcpy(dst, samples, n*sample_size);
         dst += n*sample_size;
         break;
-    case CODEC_ID_PCM_ZORK:
-        for(;n>0;n--) {
-            v= *samples++ >> 8;
-            if(v<0)   v = -v;
-            else      v+= 128;
-            *dst++ = v;
-        }
-        break;
     case CODEC_ID_PCM_ALAW:
         for(;n>0;n--) {
             v = *samples++;
@@ -213,6 +200,11 @@ static av_cold int pcm_decode_init(AVCodecContext * avctx)
     PCMDecode *s = avctx->priv_data;
     int i;
 
+    if (avctx->channels <= 0 || avctx->channels > MAX_CHANNELS) {
+        av_log(avctx, AV_LOG_ERROR, "PCM channels out of bounds\n");
+        return AVERROR(EINVAL);
+    }
+
     switch(avctx->codec->id) {
     case CODEC_ID_PCM_ALAW:
         for(i=0;i<256;i++)
@@ -255,34 +247,29 @@ static int pcm_decode_frame(AVCodecContext *avctx,
                             void *data, int *data_size,
                             AVPacket *avpkt)
 {
-    const uint8_t *buf = avpkt->data;
+    const uint8_t *src = avpkt->data;
     int buf_size = avpkt->size;
     PCMDecode *s = avctx->priv_data;
-    int sample_size, c, n, i;
+    int sample_size, c, n, out_size;
     uint8_t *samples;
-    const uint8_t *src, *src8, *src2[MAX_CHANNELS];
     int32_t *dst_int32_t;
 
     samples = data;
-    src = buf;
-
-    if (avctx->sample_fmt!=avctx->codec->sample_fmts[0]) {
-        av_log(avctx, AV_LOG_ERROR, "invalid sample_fmt\n");
-        return -1;
-    }
-
-    if(avctx->channels <= 0 || avctx->channels > MAX_CHANNELS){
-        av_log(avctx, AV_LOG_ERROR, "PCM channels out of bounds\n");
-        return -1;
-    }
 
     sample_size = av_get_bits_per_sample(avctx->codec_id)/8;
 
     /* av_get_bits_per_sample returns 0 for CODEC_ID_PCM_DVD */
-    if (CODEC_ID_PCM_DVD == avctx->codec_id)
+    if (CODEC_ID_PCM_DVD == avctx->codec_id) {
+        if (avctx->bits_per_coded_sample != 20 &&
+            avctx->bits_per_coded_sample != 24) {
+            av_log(avctx, AV_LOG_ERROR,
+                   "PCM DVD unsupported sample depth %i\n",
+                   avctx->bits_per_coded_sample);
+            return AVERROR(EINVAL);
+        }
         /* 2 samples are interleaved per block in PCM_DVD */
         sample_size = avctx->bits_per_coded_sample * 2 / 8;
-    else if (avctx->codec_id == CODEC_ID_PCM_LXF)
+    } else if (avctx->codec_id == CODEC_ID_PCM_LXF)
         /* we process 40-bit blocks per channel for LXF */
         sample_size = 5;
 
@@ -301,11 +288,17 @@ static int pcm_decode_frame(AVCodecContext *avctx,
             buf_size -= buf_size % n;
     }
 
-    buf_size= FFMIN(buf_size, *data_size/2);
-    *data_size=0;
-
     n = buf_size/sample_size;
 
+    out_size = n * av_get_bytes_per_sample(avctx->sample_fmt);
+    if (avctx->codec_id == CODEC_ID_PCM_DVD ||
+        avctx->codec_id == CODEC_ID_PCM_LXF)
+        out_size *= 2;
+    if (*data_size < out_size) {
+        av_log(avctx, AV_LOG_ERROR, "output buffer too small\n");
+        return AVERROR(EINVAL);
+    }
+
     switch(avctx->codec->id) {
     case CODEC_ID_PCM_U32LE:
         DECODE(32, le32, src, samples, n, 0, 0x80000000)
@@ -335,6 +328,8 @@ static int pcm_decode_frame(AVCodecContext *avctx,
         }
         break;
     case CODEC_ID_PCM_S16LE_PLANAR:
+    {
+        const uint8_t *src2[MAX_CHANNELS];
         n /= avctx->channels;
         for(c=0;c<avctx->channels;c++)
             src2[c] = &src[c*n*2];
@@ -343,8 +338,8 @@ static int pcm_decode_frame(AVCodecContext *avctx,
                 AV_WN16A(samples, bytestream_get_le16(&src2[c]));
                 samples += 2;
             }
-        src = src2[avctx->channels-1];
         break;
+    }
     case CODEC_ID_PCM_U16LE:
         DECODE(16, le16, src, samples, n, 0, 0x8000)
         break;
@@ -389,16 +384,14 @@ static int pcm_decode_frame(AVCodecContext *avctx,
 #endif /* HAVE_BIGENDIAN */
     case CODEC_ID_PCM_U8:
         memcpy(samples, src, n*sample_size);
-        src += n*sample_size;
         samples += n * sample_size;
         break;
     case CODEC_ID_PCM_ZORK:
-        for(;n>0;n--) {
+        for (; n > 0; n--) {
-            int x= *src++;
+            int v = *src++;
-            if(x&128) x-= 128;
+            if (v < 128)
-            else      x = -x;
+                v = 128 - v;
-            AV_WN16A(samples, x << 8);
+            *samples++ = v;
-            samples += 2;
         }
         break;
     case CODEC_ID_PCM_ALAW:
@@ -409,6 +402,8 @@ static int pcm_decode_frame(AVCodecContext *avctx,
         }
         break;
     case CODEC_ID_PCM_DVD:
+    {
+        const uint8_t *src8;
         dst_int32_t = data;
         n /= avctx->channels;
         switch (avctx->bits_per_coded_sample) {
@@ -434,15 +429,14 @@ static int pcm_decode_frame(AVCodecContext *avctx,
                 src = src8;
             }
             break;
-        default:
-            av_log(avctx, AV_LOG_ERROR,
-                   "PCM DVD unsupported sample depth %i\n",
-                   avctx->bits_per_coded_sample);
-            return -1;
         }
         samples = (uint8_t *) dst_int32_t;
         break;
+    }
     case CODEC_ID_PCM_LXF:
+    {
+        int i;
+        const uint8_t *src8;
         dst_int32_t = data;
         n /= avctx->channels;
         //unpack and de-planerize
@@ -459,14 +453,14 @@ static int pcm_decode_frame(AVCodecContext *avctx,
                                  ((src8[2] & 0xF0) << 8) | (src8[4] << 4) | (src8[3] >> 4);
             }
         }
-        src += n * avctx->channels * 5;
         samples = (uint8_t *) dst_int32_t;
         break;
+    }
     default:
         return -1;
     }
-    *data_size = samples - (uint8_t *)data;
+    *data_size = out_size;
-    return src - buf;
+    return buf_size;
 }
 
 #if CONFIG_ENCODERS
@@ -529,4 +523,4 @@ PCM_CODEC  (CODEC_ID_PCM_U24BE, AV_SAMPLE_FMT_S32, pcm_u24be, "PCM unsigned 24-b
 PCM_CODEC  (CODEC_ID_PCM_U24LE, AV_SAMPLE_FMT_S32, pcm_u24le, "PCM unsigned 24-bit little-endian");
 PCM_CODEC  (CODEC_ID_PCM_U32BE, AV_SAMPLE_FMT_S32, pcm_u32be, "PCM unsigned 32-bit big-endian");
 PCM_CODEC  (CODEC_ID_PCM_U32LE, AV_SAMPLE_FMT_S32, pcm_u32le, "PCM unsigned 32-bit little-endian");
-PCM_CODEC  (CODEC_ID_PCM_ZORK,  AV_SAMPLE_FMT_S16, pcm_zork, "PCM Zork");
+PCM_DECODER(CODEC_ID_PCM_ZORK,  AV_SAMPLE_FMT_U8,  pcm_zork,  "PCM Zork");

libavcodec/pthread.c

@@ -686,6 +686,7 @@ static void frame_thread_free(AVCodecContext *avctx, int thread_count)
     av_freep(&fctx->threads);
     pthread_mutex_destroy(&fctx->buffer_mutex);
     av_freep(&avctx->thread_opaque);
+    avctx->has_b_frames -= avctx->thread_count - 1;
 }
 
 static int frame_thread_init(AVCodecContext *avctx)

libavcodec/x86/dsputil_mmx.c

@@ -2666,18 +2666,18 @@ void dsputil_init_mmx(DSPContext* c, AVCodecContext *avctx)
         }
         if(mm_flags & AV_CPU_FLAG_SSE2){
             if (!high_bit_depth) {
-            H264_QPEL_FUNCS(0, 1, sse2);
+                H264_QPEL_FUNCS(0, 1, sse2);
-            H264_QPEL_FUNCS(0, 2, sse2);
+                H264_QPEL_FUNCS(0, 2, sse2);
-            H264_QPEL_FUNCS(0, 3, sse2);
+                H264_QPEL_FUNCS(0, 3, sse2);
-            H264_QPEL_FUNCS(1, 1, sse2);
+                H264_QPEL_FUNCS(1, 1, sse2);
-            H264_QPEL_FUNCS(1, 2, sse2);
+                H264_QPEL_FUNCS(1, 2, sse2);
-            H264_QPEL_FUNCS(1, 3, sse2);
+                H264_QPEL_FUNCS(1, 3, sse2);
-            H264_QPEL_FUNCS(2, 1, sse2);
+                H264_QPEL_FUNCS(2, 1, sse2);
-            H264_QPEL_FUNCS(2, 2, sse2);
+                H264_QPEL_FUNCS(2, 2, sse2);
-            H264_QPEL_FUNCS(2, 3, sse2);
+                H264_QPEL_FUNCS(2, 3, sse2);
-            H264_QPEL_FUNCS(3, 1, sse2);
+                H264_QPEL_FUNCS(3, 1, sse2);
-            H264_QPEL_FUNCS(3, 2, sse2);
+                H264_QPEL_FUNCS(3, 2, sse2);
-            H264_QPEL_FUNCS(3, 3, sse2);
+                H264_QPEL_FUNCS(3, 3, sse2);
             }
 #if HAVE_YASM
 #define H264_QPEL_FUNCS_10(x, y, CPU)\

tests/codec-regression.sh

@@ -393,6 +393,5 @@ do_audio_enc_dec au  flt pcm_f32be
 do_audio_enc_dec wav flt pcm_f32le
 do_audio_enc_dec au  dbl pcm_f64be
 do_audio_enc_dec wav dbl pcm_f64le
-do_audio_enc_dec wav s16 pcm_zork
 do_audio_enc_dec 302 s16 pcm_s24daud "-ac 6 -ar 96000"
 fi

tests/ref/acodec/pcm

@@ -62,10 +62,6 @@ ba17c6d1a270e1333e981f239bf7eb45 *./tests/data/acodec/pcm_f64le.wav
 4233680 ./tests/data/acodec/pcm_f64le.wav
 64151e4bcc2b717aa5a8454d424d6a1f *./tests/data/pcm.acodec.out.wav
 stddev:    0.00 PSNR:999.99 MAXDIFF:    0 bytes:  1058400/  1058400
-ebd38ed390ebdefe0bdf00d21bf12c6b *./tests/data/acodec/pcm_zork.wav
-529258 ./tests/data/acodec/pcm_zork.wav
-7b02646acdd063650bb3ebc444543ace *./tests/data/pcm.acodec.out.wav
-stddev:  633.11 PSNR: 40.30 MAXDIFF:32768 bytes:  1058400/  1058400
 1b75d5198ae789ab3c48f7024e08f4a9 *./tests/data/acodec/pcm_s24daud.302
 10368730 ./tests/data/acodec/pcm_s24daud.302
 4708f86529c594e29404603c64bb208c *./tests/data/pcm.acodec.out.wav

tests/ref/seek/pcm_zork_wav

@@ -1,53 +0,0 @@
-ret: 0         st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos:     58 size:  4096
-ret: 0         st:-1 flags:0  ts:-1.000000
-ret: 0         st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos:     58 size:  4096
-ret: 0         st:-1 flags:1  ts: 1.894167
-ret: 0         st: 0 flags:1 dts: 1.894127 pts: 1.894127 pos:  30364 size:  4096
-ret: 0         st: 0 flags:0  ts: 0.788345
-ret: 0         st: 0 flags:1 dts: 0.788367 pts: 0.788367 pos:  12672 size:  4096
-ret: 0         st: 0 flags:1  ts:-0.317506
-ret: 0         st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos:     58 size:  4096
-ret: 0         st:-1 flags:0  ts: 2.576668
-ret: 0         st: 0 flags:1 dts: 2.576757 pts: 2.576757 pos:  41286 size:  4096
-ret: 0         st:-1 flags:1  ts: 1.470835
-ret: 0         st: 0 flags:1 dts: 1.470748 pts: 1.470748 pos:  23590 size:  4096
-ret: 0         st: 0 flags:0  ts: 0.365011
-ret: 0         st: 0 flags:1 dts: 0.365125 pts: 0.365125 pos:   5900 size:  4096
-ret: 0         st: 0 flags:1  ts:-0.740839
-ret: 0         st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos:     58 size:  4096
-ret: 0         st:-1 flags:0  ts: 2.153336
-ret: 0         st: 0 flags:1 dts: 2.153379 pts: 2.153379 pos:  34512 size:  4096
-ret: 0         st:-1 flags:1  ts: 1.047503
-ret: 0         st: 0 flags:1 dts: 1.047506 pts: 1.047506 pos:  16818 size:  4096
-ret: 0         st: 0 flags:0  ts:-0.058322
-ret: 0         st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos:     58 size:  4096
-ret: 0         st: 0 flags:1  ts: 2.835828
-ret: 0         st: 0 flags:1 dts: 2.835760 pts: 2.835760 pos:  45430 size:  4096
-ret: 0         st:-1 flags:0  ts: 1.730004
-ret: 0         st: 0 flags:1 dts: 1.730000 pts: 1.730000 pos:  27738 size:  4096
-ret: 0         st:-1 flags:1  ts: 0.624171
-ret: 0         st: 0 flags:1 dts: 0.624127 pts: 0.624127 pos:  10044 size:  4096
-ret: 0         st: 0 flags:0  ts:-0.481655
-ret: 0         st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos:     58 size:  4096
-ret: 0         st: 0 flags:1  ts: 2.412494
-ret: 0         st: 0 flags:1 dts: 2.412381 pts: 2.412381 pos:  38656 size:  4096
-ret: 0         st:-1 flags:0  ts: 1.306672
-ret: 0         st: 0 flags:1 dts: 1.306757 pts: 1.306757 pos:  20966 size:  4096
-ret: 0         st:-1 flags:1  ts: 0.200839
-ret: 0         st: 0 flags:1 dts: 0.200748 pts: 0.200748 pos:   3270 size:  4096
-ret: 0         st: 0 flags:0  ts:-0.904989
-ret: 0         st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos:     58 size:  4096
-ret: 0         st: 0 flags:1  ts: 1.989184
-ret: 0         st: 0 flags:1 dts: 1.989116 pts: 1.989116 pos:  31884 size:  4096
-ret: 0         st:-1 flags:0  ts: 0.883340
-ret: 0         st: 0 flags:1 dts: 0.883379 pts: 0.883379 pos:  14192 size:  4096
-ret: 0         st:-1 flags:1  ts:-0.222493
-ret: 0         st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos:     58 size:  4096
-ret: 0         st: 0 flags:0  ts: 2.671678
-ret: 0         st: 0 flags:1 dts: 2.671746 pts: 2.671746 pos:  42806 size:  4096
-ret: 0         st: 0 flags:1  ts: 1.565850
-ret: 0         st: 0 flags:1 dts: 1.565760 pts: 1.565760 pos:  25110 size:  4096
-ret: 0         st:-1 flags:0  ts: 0.460008
-ret: 0         st: 0 flags:1 dts: 0.460000 pts: 0.460000 pos:   7418 size:  4096
-ret: 0         st:-1 flags:1  ts:-0.645825
-ret: 0         st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos:     58 size:  4096