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>

1 files changed, 60 insertions, 33 deletions

diff --git a/libavcodec/atrac1.c b/libavcodec/atrac1.c
index 2ad99bf..d4d5986 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,