Merge remote-tracking branch 'qatar/master'
[ffmpeg.git] / libavcodec / alac.c
index b7071d2..83e0d81 100644 (file)
@@ -2,20 +2,20 @@
  * ALAC (Apple Lossless Audio Codec) decoder
  * Copyright (c) 2005 David Hammerton
  *
- * This file is part of Libav.
+ * This file is part of FFmpeg.
  *
- * Libav is free software; you can redistribute it and/or
+ * FFmpeg is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Lesser General Public
  * License as published by the Free Software Foundation; either
  * version 2.1 of the License, or (at your option) any later version.
  *
- * Libav is distributed in the hope that it will be useful,
+ * FFmpeg is distributed in the hope that it will be useful,
  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  * Lesser General Public License for more details.
  *
  * You should have received a copy of the GNU Lesser General Public
- * License along with Libav; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
  * @author 2005 David Hammerton
  * @see http://crazney.net/programs/itunes/alac.html
  *
- * Note: This decoder expects a 36- (0x24-)byte QuickTime atom to be
+ * Note: This decoder expects a 36-byte QuickTime atom to be
  * passed through the extradata[_size] fields. This atom is tacked onto
  * the end of an 'alac' stsd atom and has the following format:
- *  bytes 0-3   atom size (0x24), big-endian
- *  bytes 4-7   atom type ('alac', not the 'alac' tag from start of stsd)
- *  bytes 8-35  data bytes needed by decoder
  *
- * Extradata:
- * 32bit  size
- * 32bit  tag (=alac)
- * 32bit  zero?
- * 32bit  max sample per frame
- *  8bit  ?? (zero?)
+ * 32bit  atom size
+ * 32bit  tag                  ("alac")
+ * 32bit  tag version          (0)
+ * 32bit  samples per frame    (used when not set explicitly in the frames)
+ *  8bit  compatible version   (0)
  *  8bit  sample size
- *  8bit  history mult
- *  8bit  initial history
- *  8bit  kmodifier
- *  8bit  channels?
- * 16bit  ??
- * 32bit  max coded frame size
- * 32bit  bitrate?
+ *  8bit  history mult         (40)
+ *  8bit  initial history      (14)
+ *  8bit  kmodifier            (10)
+ *  8bit  channels
+ * 16bit  maxRun               (255)
+ * 32bit  max coded frame size (0 means unknown)
+ * 32bit  average bitrate      (0 means unknown)
  * 32bit  samplerate
  */
 
 typedef struct {
 
     AVCodecContext *avctx;
+    AVFrame frame;
     GetBitContext gb;
 
     int numchannels;
-    int bytespersample;
 
     /* buffers */
     int32_t *predicterror_buffer[MAX_CHANNELS];
@@ -112,7 +108,7 @@ static inline int decode_scalar(GetBitContext *gb, int k, int limit, int readsam
     return x;
 }
 
-static void bastardized_rice_decompress(ALACContext *alac,
+static int bastardized_rice_decompress(ALACContext *alac,
                                  int32_t *output_buffer,
                                  int output_size,
                                  int readsamplesize, /* arg_10 */
@@ -134,6 +130,9 @@ static void bastardized_rice_decompress(ALACContext *alac,
         /* standard rice encoding */
         int k; /* size of extra bits */
 
+        if(get_bits_left(&alac->gb) <= 0)
+            return -1;
+
         /* read k, that is bits as is */
         k = av_log2((history >> 9) + 3);
         x= decode_scalar(&alac->gb, k, rice_kmodifier, readsamplesize);
@@ -179,6 +178,7 @@ static void bastardized_rice_decompress(ALACContext *alac,
             history = 0;
         }
     }
+    return 0;
 }
 
 static inline int sign_only(int v)
@@ -299,8 +299,7 @@ static void predictor_decompress_fir_adapt(int32_t *error_buffer,
 }
 
 static void decorrelate_stereo(int32_t *buffer[MAX_CHANNELS],
-                               int numchannels, int numsamples,
-                               uint8_t interlacing_shift,
+                               int numsamples, uint8_t interlacing_shift,
                                uint8_t interlacing_leftweight)
 {
     int i;
@@ -330,43 +329,41 @@ static void append_extra_bits(int32_t *buffer[MAX_CHANNELS],
             buffer[ch][i] = (buffer[ch][i] << extra_bits) | extra_bits_buffer[ch][i];
 }
 
-static void reconstruct_stereo_16(int32_t *buffer[MAX_CHANNELS],
-                                  int16_t *buffer_out,
-                                  int numchannels, int numsamples)
+static void interleave_stereo_16(int32_t *buffer[MAX_CHANNELS],
+                                 int16_t *buffer_out, int numsamples)
 {
     int i;
 
     for (i = 0; i < numsamples; i++) {
-        int16_t left, right;
-
-        left = buffer[0][i];
-        right = buffer[1][i];
-
-        buffer_out[i*numchannels] = left;
-        buffer_out[i*numchannels + 1] = right;
+        *buffer_out++ = buffer[0][i];
+        *buffer_out++ = buffer[1][i];
     }
 }
 
-static void decorrelate_stereo_24(int32_t *buffer[MAX_CHANNELS],
-                                  int32_t *buffer_out,
-                                  int numchannels, int numsamples)
+static void interleave_stereo_24(int32_t *buffer[MAX_CHANNELS],
+                                 int32_t *buffer_out, int numsamples)
 {
     int i;
 
-        for (i = 0; i < numsamples; i++) {
-            int32_t left, right;
+    for (i = 0; i < numsamples; i++) {
+        *buffer_out++ = buffer[0][i] << 8;
+        *buffer_out++ = buffer[1][i] << 8;
+    }
+}
 
-            left  = buffer[0][i];
-            right = buffer[1][i];
+static void interleave_stereo_32(int32_t *buffer[MAX_CHANNELS],
+                                 int32_t *buffer_out, int numsamples)
+{
+    int i;
 
-            buffer_out[i * numchannels]     = left  << 8;
-            buffer_out[i * numchannels + 1] = right << 8;
-        }
+    for (i = 0; i < numsamples; i++) {
+        *buffer_out++ = buffer[0][i];
+        *buffer_out++ = buffer[1][i];
+    }
 }
 
-static int alac_decode_frame(AVCodecContext *avctx,
-                             void *outbuffer, int *outputsize,
-                             AVPacket *avpkt)
+static int alac_decode_frame(AVCodecContext *avctx, void *data,
+                             int *got_frame_ptr, AVPacket *avpkt)
 {
     const uint8_t *inbuffer = avpkt->data;
     int input_buffer_size = avpkt->size;
@@ -379,10 +376,7 @@ static int alac_decode_frame(AVCodecContext *avctx,
     int isnotcompressed;
     uint8_t interlacing_shift;
     uint8_t interlacing_leftweight;
-
-    /* short-circuit null buffers */
-    if (!inbuffer || !input_buffer_size)
-        return -1;
+    int i, ch, ret;
 
     init_get_bits(&alac->gb, inbuffer, input_buffer_size * 8);
 
@@ -417,14 +411,17 @@ static int alac_decode_frame(AVCodecContext *avctx,
     } else
         outputsamples = alac->setinfo_max_samples_per_frame;
 
-    alac->bytespersample = channels * av_get_bytes_per_sample(avctx->sample_fmt);
-
-    if(outputsamples > *outputsize / alac->bytespersample){
-        av_log(avctx, AV_LOG_ERROR, "sample buffer too small\n");
-        return -1;
+    /* get output buffer */
+    if (outputsamples > INT32_MAX) {
+        av_log(avctx, AV_LOG_ERROR, "unsupported block size: %u\n", outputsamples);
+        return AVERROR_INVALIDDATA;
+    }
+    alac->frame.nb_samples = outputsamples;
+    if ((ret = avctx->get_buffer(avctx, &alac->frame)) < 0) {
+        av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+        return ret;
     }
 
-    *outputsize = outputsamples * alac->bytespersample;
     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);
@@ -438,79 +435,74 @@ 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++) {
-            prediction_type[chan] = get_bits(&alac->gb, 4);
-            prediction_quantitization[chan] = get_bits(&alac->gb, 4);
+        for (ch = 0; ch < channels; ch++) {
+            prediction_type[ch] = get_bits(&alac->gb, 4);
+            prediction_quantitization[ch] = get_bits(&alac->gb, 4);
 
-            ricemodifier[chan] = get_bits(&alac->gb, 3);
-            predictor_coef_num[chan] = get_bits(&alac->gb, 5);
+            ricemodifier[ch] = get_bits(&alac->gb, 3);
+            predictor_coef_num[ch] = get_bits(&alac->gb, 5);
 
             /* read the predictor table */
-            for (i = 0; i < predictor_coef_num[chan]; i++)
-                predictor_coef_table[chan][i] = (int16_t)get_bits(&alac->gb, 16);
+            for (i = 0; i < predictor_coef_num[ch]; i++)
+                predictor_coef_table[ch][i] = (int16_t)get_bits(&alac->gb, 16);
         }
 
         if (alac->extra_bits) {
-            int i, ch;
             for (i = 0; i < outputsamples; i++) {
+                if(get_bits_left(&alac->gb) <= 0)
+                    return -1;
                 for (ch = 0; ch < channels; ch++)
                     alac->extra_bits_buffer[ch][i] = get_bits(&alac->gb, alac->extra_bits);
             }
         }
-        for (chan = 0; chan < channels; chan++) {
-            bastardized_rice_decompress(alac,
-                                        alac->predicterror_buffer[chan],
+        for (ch = 0; ch < channels; ch++) {
+            int ret = bastardized_rice_decompress(alac,
+                                        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) {
-                /* adaptive fir */
-                predictor_decompress_fir_adapt(alac->predicterror_buffer[chan],
-                                               alac->outputsamples_buffer[chan],
-                                               outputsamples,
-                                               readsamplesize,
-                                               predictor_coef_table[chan],
-                                               predictor_coef_num[chan],
-                                               prediction_quantitization[chan]);
-            } else {
-                av_log(avctx, AV_LOG_ERROR, "FIXME: unhandled prediction type: %i\n", prediction_type[chan]);
-                /* 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, ...)
-                 * predictor_decompress_fir_adapt(predictor_error, outputsamples ...)
-                 * little strange..
+            if(ret<0)
+                return ret;
+
+            /* adaptive FIR filter */
+            if (prediction_type[ch] == 15) {
+                /* Prediction type 15 runs the adaptive FIR twice.
+                 * The first pass uses the special-case coef_num = 31, while
+                 * the second pass uses the coefs from the bitstream.
+                 *
+                 * However, this prediction type is not currently used by the
+                 * reference encoder.
                  */
+                predictor_decompress_fir_adapt(alac->predicterror_buffer[ch],
+                                               alac->predicterror_buffer[ch],
+                                               outputsamples, readsamplesize,
+                                               NULL, 31, 0);
+            } else if (prediction_type[ch] > 0) {
+                av_log(avctx, AV_LOG_WARNING, "unknown prediction type: %i\n",
+                       prediction_type[ch]);
             }
+            predictor_decompress_fir_adapt(alac->predicterror_buffer[ch],
+                                           alac->outputsamples_buffer[ch],
+                                           outputsamples, readsamplesize,
+                                           predictor_coef_table[ch],
+                                           predictor_coef_num[ch],
+                                           prediction_quantitization[ch]);
         }
     } else {
         /* not compressed, easy case */
-        int i, chan;
-        if (alac->setinfo_sample_size <= 16) {
-        for (i = 0; i < outputsamples; i++)
-            for (chan = 0; chan < channels; chan++) {
-                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);
-                }
+        for (i = 0; i < outputsamples; i++) {
+            if(get_bits_left(&alac->gb) <= 0)
+                return -1;
+            for (ch = 0; ch < channels; ch++) {
+                alac->outputsamples_buffer[ch][i] = get_sbits_long(&alac->gb,
+                                                                   alac->setinfo_sample_size);
             }
         }
         alac->extra_bits = 0;
@@ -521,9 +513,8 @@ static int alac_decode_frame(AVCodecContext *avctx,
         av_log(avctx, AV_LOG_ERROR, "Error : Wrong End Of Frame\n");
 
     if (channels == 2 && interlacing_leftweight) {
-        decorrelate_stereo(alac->outputsamples_buffer, alac->numchannels,
-                           outputsamples, interlacing_shift,
-                           interlacing_leftweight);
+        decorrelate_stereo(alac->outputsamples_buffer, outputsamples,
+                           interlacing_shift, interlacing_leftweight);
     }
 
     if (alac->extra_bits) {
@@ -534,27 +525,33 @@ static int alac_decode_frame(AVCodecContext *avctx,
     switch(alac->setinfo_sample_size) {
     case 16:
         if (channels == 2) {
-            reconstruct_stereo_16(alac->outputsamples_buffer,
-                                  (int16_t*)outbuffer,
-                                  alac->numchannels,
-                                  outputsamples);
+            interleave_stereo_16(alac->outputsamples_buffer,
+                                 (int16_t *)alac->frame.data[0], outputsamples);
         } else {
-            int i;
+            int16_t *outbuffer = (int16_t *)alac->frame.data[0];
             for (i = 0; i < outputsamples; i++) {
-                ((int16_t*)outbuffer)[i] = alac->outputsamples_buffer[0][i];
+                outbuffer[i] = alac->outputsamples_buffer[0][i];
             }
         }
         break;
     case 24:
         if (channels == 2) {
-            decorrelate_stereo_24(alac->outputsamples_buffer,
-                                  outbuffer,
-                                  alac->numchannels,
-                                  outputsamples);
+            interleave_stereo_24(alac->outputsamples_buffer,
+                                 (int32_t *)alac->frame.data[0], outputsamples);
         } else {
-            int i;
+            int32_t *outbuffer = (int32_t *)alac->frame.data[0];
             for (i = 0; i < outputsamples; i++)
-                ((int32_t *)outbuffer)[i] = alac->outputsamples_buffer[0][i] << 8;
+                outbuffer[i] = alac->outputsamples_buffer[0][i] << 8;
+        }
+        break;
+    case 32:
+        if (channels == 2) {
+            interleave_stereo_32(alac->outputsamples_buffer,
+                                 (int32_t *)alac->frame.data[0], outputsamples);
+        } else {
+            int32_t *outbuffer = (int32_t *)alac->frame.data[0];
+            for (i = 0; i < outputsamples; i++)
+                outbuffer[i] = alac->outputsamples_buffer[0][i];
         }
         break;
     }
@@ -562,6 +559,9 @@ static int alac_decode_frame(AVCodecContext *avctx,
     if (input_buffer_size * 8 - get_bits_count(&alac->gb) > 8)
         av_log(avctx, AV_LOG_ERROR, "Error : %d bits left\n", input_buffer_size * 8 - get_bits_count(&alac->gb));
 
+    *got_frame_ptr   = 1;
+    *(AVFrame *)data = alac->frame;
+
     return input_buffer_size;
 }
 
@@ -569,11 +569,11 @@ static av_cold int alac_decode_close(AVCodecContext *avctx)
 {
     ALACContext *alac = avctx->priv_data;
 
-    int chan;
-    for (chan = 0; chan < alac->numchannels; chan++) {
-        av_freep(&alac->predicterror_buffer[chan]);
-        av_freep(&alac->outputsamples_buffer[chan]);
-        av_freep(&alac->extra_bits_buffer[chan]);
+    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;
@@ -581,24 +581,23 @@ static av_cold int alac_decode_close(AVCodecContext *avctx)
 
 static int allocate_buffers(ALACContext *alac)
 {
-    int chan;
-    for (chan = 0; chan < alac->numchannels; chan++) {
-        alac->predicterror_buffer[chan] =
-            av_malloc(alac->setinfo_max_samples_per_frame * 4);
+    int ch;
+    for (ch = 0; ch < alac->numchannels; ch++) {
+        int buf_size = alac->setinfo_max_samples_per_frame * sizeof(int32_t);
 
-        alac->outputsamples_buffer[chan] =
-            av_malloc(alac->setinfo_max_samples_per_frame * 4);
+        FF_ALLOC_OR_GOTO(alac->avctx, alac->predicterror_buffer[ch],
+                         buf_size, buf_alloc_fail);
 
-        alac->extra_bits_buffer[chan] = av_malloc(alac->setinfo_max_samples_per_frame * 4);
+        FF_ALLOC_OR_GOTO(alac->avctx, alac->outputsamples_buffer[ch],
+                         buf_size, buf_alloc_fail);
 
-        if (!alac->predicterror_buffer[chan]  ||
-            !alac->outputsamples_buffer[chan] ||
-            !alac->extra_bits_buffer[chan]) {
-            alac_decode_close(alac->avctx);
-            return AVERROR(ENOMEM);
-        }
+        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)
@@ -607,7 +606,7 @@ static int alac_set_info(ALACContext *alac)
 
     ptr += 4; /* size */
     ptr += 4; /* alac */
-    ptr += 4; /* 0 ? */
+    ptr += 4; /* version */
 
     if(AV_RB32(ptr) >= UINT_MAX/4){
         av_log(alac->avctx, AV_LOG_ERROR, "setinfo_max_samples_per_frame too large\n");
@@ -616,15 +615,15 @@ static int alac_set_info(ALACContext *alac)
 
     /* buffer size / 2 ? */
     alac->setinfo_max_samples_per_frame = bytestream_get_be32(&ptr);
-    ptr++;                          /* ??? */
+    ptr++;                          /* compatible version */
     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_be16(&ptr);      /* maxRun */
     bytestream_get_be32(&ptr);      /* max coded frame size */
-    bytestream_get_be32(&ptr);      /* bitrate ? */
+    bytestream_get_be32(&ptr);      /* average bitrate */
     bytestream_get_be32(&ptr);      /* samplerate */
 
     return 0;
@@ -650,11 +649,12 @@ static av_cold int alac_decode_init(AVCodecContext * avctx)
     switch (alac->setinfo_sample_size) {
     case 16: avctx->sample_fmt    = AV_SAMPLE_FMT_S16;
              break;
+    case 32:
     case 24: avctx->sample_fmt    = AV_SAMPLE_FMT_S32;
              break;
-    default: av_log(avctx, AV_LOG_ERROR, "Sample depth %d is not supported.\n",
-                    alac->setinfo_sample_size);
-             return -1;
+    default: av_log_ask_for_sample(avctx, "Sample depth %d is not supported.\n",
+                                   alac->setinfo_sample_size);
+             return AVERROR_PATCHWELCOME;
     }
 
     if (alac->numchannels < 1) {
@@ -677,6 +677,9 @@ static av_cold int alac_decode_init(AVCodecContext * avctx)
         return ret;
     }
 
+    avcodec_get_frame_defaults(&alac->frame);
+    avctx->coded_frame = &alac->frame;
+
     return 0;
 }
 
@@ -688,5 +691,6 @@ AVCodec ff_alac_decoder = {
     .init           = alac_decode_init,
     .close          = alac_decode_close,
     .decode         = alac_decode_frame,
+    .capabilities   = CODEC_CAP_DR1,
     .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
 };