libvorbis: use VBR by default, with default quality of 3

[ffmpeg] / libavcodec / atrac1.c

diff --git a/libavcodec/atrac1.c b/libavcodec/atrac1.c
index ef5156c650790f8d4bc1ca45aa0a6a3099dc61a7..1afd62680a58b7e76b4b8cdef52f8ef881ab7b30 100644 (file)
--- 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
 
@@ -261,39 +262,40 @@ static void at1_subband_synthesis(AT1Ctx *q, AT1SUCtx* su, float *pOut)
     float iqmf_temp[512 + 46];
 
     /* combine low and middle bands */
-    atrac_iqmf(q->bands[0], q->bands[1], 128, temp, su->fst_qmf_delay, iqmf_temp);
+    ff_atrac_iqmf(q->bands[0], q->bands[1], 128, temp, su->fst_qmf_delay, iqmf_temp);
 
     /* delay the signal of the high band by 23 samples */
     memcpy( su->last_qmf_delay,    &su->last_qmf_delay[256], sizeof(float) *  23);
     memcpy(&su->last_qmf_delay[23], q->bands[2],             sizeof(float) * 256);
 
     /* combine (low + middle) and high bands */
-    atrac_iqmf(temp, su->last_qmf_delay, 256, pOut, su->snd_qmf_delay, iqmf_temp);
+    ff_atrac_iqmf(temp, su->last_qmf_delay, 256, pOut, su->snd_qmf_delay, iqmf_temp);
 }
 
 
 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");
+        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];
@@ -321,7 +323,9 @@ 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;
 }
 
@@ -374,9 +378,9 @@ static av_cold int atrac1_decode_init(AVCodecContext *avctx)
 
     ff_init_ff_sine_windows(5);
 
-    atrac_generate_tables();
+    ff_atrac_generate_tables();
 
-    dsputil_init(&q->dsp, avctx);
+    ff_dsputil_init(&q->dsp, avctx);
     ff_fmt_convert_init(&q->fmt_conv, avctx);
 
     q->bands[0] = q->low;
@@ -389,6 +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;
 
+    avcodec_get_frame_defaults(&q->frame);
+    avctx->coded_frame = &q->frame;
+
     return 0;
 }
 
@@ -401,5 +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)"),
 };