X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fatrac1.c;h=9ead80d5c83e06425d45dae1a10c891bd94f599a;hb=4424fe9c029ccefc5868823b4967c792e51487f3;hp=c646ba9fc1d7916e359c282d32c11dc40926e9d0;hpb=21dcecc310e701d035b6a877139b9dd8e2a82a1a;p=ffmpeg

diff --git a/libavcodec/atrac1.c b/libavcodec/atrac1.c
index c646ba9fc1d..9ead80d5c83 100644
--- a/libavcodec/atrac1.c
+++ b/libavcodec/atrac1.c
@@ -72,6 +72,7 @@ typedef struct {
  * The atrac1 context, holds all needed parameters for decoding
  */
 typedef struct {
+    AVFrame frame;
     AT1SUCtx            SUs[AT1_MAX_CHANNELS];              ///< channel sound unit
     DECLARE_ALIGNED(32, float, spec)[AT1_SU_SAMPLES];      ///< the mdct spectrum buffer
 
@@ -131,7 +132,7 @@ 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;
@@ -175,14 +176,14 @@ 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);
@@ -231,7 +232,7 @@ 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];
@@ -273,27 +274,28 @@ static void at1_subband_synthesis(AT1Ctx *q, AT1SUCtx* su, float *pOut)
 
 
 static int atrac1_decode_frame(AVCodecContext *avctx, void *data,
-                               int *data_size, AVPacket *avpkt)
+                               int *got_frame_ptr, AVPacket *avpkt)
 {
     const uint8_t *buf = avpkt->data;
     int buf_size       = avpkt->size;
     AT1Ctx *q          = avctx->priv_data;
-    int ch, ret, out_size;
+    int ch, ret;
     GetBitContext gb;
-    float* samples = data;
+    float *samples;
 
 
     if (buf_size < 212 * q->channels) {
-        av_log(q,AV_LOG_ERROR,"Not enough 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);
+    /* get output buffer */
+    q->frame.nb_samples = AT1_SU_SAMPLES;
+    if ((ret = avctx->get_buffer(avctx, &q->frame)) < 0) {
+        av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+        return ret;
     }
+    samples = (float *)q->frame.data[0];
 
     for (ch = 0; ch < q->channels; ch++) {
         AT1SUCtx* su = &q->SUs[ch];
@@ -321,14 +323,31 @@ static int atrac1_decode_frame(AVCodecContext *avctx, void *data,
                                      AT1_SU_SAMPLES, 2);
     }
 
-    *data_size = out_size;
+    *got_frame_ptr   = 1;
+    *(AVFrame *)data = q->frame;
+
     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;
 
@@ -349,9 +368,13 @@ static av_cold int atrac1_decode_init(AVCodecContext *avctx)
     }
 
     /* 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);
 
@@ -370,18 +393,9 @@ static av_cold int atrac1_decode_init(AVCodecContext *avctx)
     q->SUs[1].spectrum[0] = q->SUs[1].spec1;
     q->SUs[1].spectrum[1] = q->SUs[1].spec2;
 
-    return 0;
-}
-
+    avcodec_get_frame_defaults(&q->frame);
+    avctx->coded_frame = &q->frame;
 
-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;
 }
 
@@ -394,5 +408,6 @@ AVCodec ff_atrac1_decoder = {
     .init = atrac1_decode_init,
     .close = atrac1_decode_end,
     .decode = atrac1_decode_frame,
+    .capabilities = CODEC_CAP_DR1,
     .long_name = NULL_IF_CONFIG_SMALL("Atrac 1 (Adaptive TRansform Acoustic Coding)"),
 };