parser: Move Doxygen documentation to the header files

[ffmpeg] / libavcodec / aacenc.c

diff --git a/libavcodec/aacenc.c b/libavcodec/aacenc.c
index e4fbf0304b1d567d3672ab2514cc88f598e6d058..e9f6e2ffbfeb93b6d76437e1930943dc6e505a2b 100644 (file)
--- a/libavcodec/aacenc.c
+++ b/libavcodec/aacenc.c
@@ -302,7 +302,7 @@ static void encode_ms_info(PutBitContext *pb, ChannelElement *cpe)
 /**
  * Produce integer coefficients from scalefactors provided by the model.
  */
-static void adjust_frame_information(AACEncContext *apc, ChannelElement *cpe, int chans)
+static void adjust_frame_information(ChannelElement *cpe, int chans)
 {
     int i, w, w2, g, ch;
     int start, maxsfb, cmaxsfb;
@@ -460,8 +460,7 @@ static int encode_individual_channel(AVCodecContext *avctx, AACEncContext *s,
 /**
  * Write some auxiliary information about the created AAC file.
  */
-static void put_bitstream_info(AVCodecContext *avctx, AACEncContext *s,
-                               const char *name)
+static void put_bitstream_info(AACEncContext *s, const char *name)
 {
     int i, namelen, padbits;
 
@@ -479,31 +478,28 @@ static void put_bitstream_info(AVCodecContext *avctx, AACEncContext *s,
 }
 
 /*
- * Deinterleave input samples.
+ * Copy input samples.
  * Channels are reordered from Libav's default order to AAC order.
  */
-static void deinterleave_input_samples(AACEncContext *s, const AVFrame *frame)
+static void copy_input_samples(AACEncContext *s, const AVFrame *frame)
 {
-    int ch, i;
-    const int sinc = s->channels;
-    const uint8_t *channel_map = aac_chan_maps[sinc - 1];
+    int ch;
+    int end = 2048 + (frame ? frame->nb_samples : 0);
+    const uint8_t *channel_map = aac_chan_maps[s->channels - 1];
 
-    /* deinterleave and remap input samples */
-    for (ch = 0; ch < sinc; ch++) {
+    /* copy and remap input samples */
+    for (ch = 0; ch < s->channels; ch++) {
         /* copy last 1024 samples of previous frame to the start of the current frame */
         memcpy(&s->planar_samples[ch][1024], &s->planar_samples[ch][2048], 1024 * sizeof(s->planar_samples[0][0]));
 
-        /* deinterleave */
-        i = 2048;
+        /* copy new samples and zero any remaining samples */
         if (frame) {
-            const float *sptr = ((const float *)frame->data[0]) + channel_map[ch];
-            for (; i < 2048 + frame->nb_samples; i++) {
-                s->planar_samples[ch][i] = *sptr;
-                sptr += sinc;
-            }
+            memcpy(&s->planar_samples[ch][2048],
+                   frame->extended_data[channel_map[ch]],
+                   frame->nb_samples * sizeof(s->planar_samples[0][0]));
         }
-        memset(&s->planar_samples[ch][i], 0,
-               (3072 - i) * sizeof(s->planar_samples[0][0]));
+        memset(&s->planar_samples[ch][end], 0,
+               (3072 - end) * sizeof(s->planar_samples[0][0]));
     }
 }
 
@@ -526,7 +522,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
             return ret;
     }
 
-    deinterleave_input_samples(s, frame);
+    copy_input_samples(s, frame);
     if (s->psypp)
         ff_psy_preprocess(s->psypp, s->planar_samples, s->channels);
 
@@ -587,7 +583,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
         init_put_bits(&s->pb, avpkt->data, avpkt->size);
 
         if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & CODEC_FLAG_BITEXACT))
-            put_bitstream_info(avctx, s, LIBAVCODEC_IDENT);
+            put_bitstream_info(s, LIBAVCODEC_IDENT);
         start_ch = 0;
         memset(chan_el_counter, 0, sizeof(chan_el_counter));
         for (i = 0; i < s->chan_map[0]; i++) {
@@ -629,7 +625,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
                     s->coder->search_for_ms(s, cpe, s->lambda);
                 }
             }
-            adjust_frame_information(s, cpe, chans);
+            adjust_frame_information(cpe, chans);
             if (chans == 2) {
                 put_bits(&s->pb, 1, cpe->common_window);
                 if (cpe->common_window) {
@@ -802,7 +798,7 @@ fail:
 
 #define AACENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
 static const AVOption aacenc_options[] = {
-    {"stereo_mode", "Stereo coding method", offsetof(AACEncContext, options.stereo_mode), AV_OPT_TYPE_INT, {.dbl = 0}, -1, 1, AACENC_FLAGS, "stereo_mode"},
+    {"stereo_mode", "Stereo coding method", offsetof(AACEncContext, options.stereo_mode), AV_OPT_TYPE_INT, {.i64 = 0}, -1, 1, AACENC_FLAGS, "stereo_mode"},
         {"auto",     "Selected by the Encoder", 0, AV_OPT_TYPE_CONST, {.i64 = -1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
         {"ms_off",   "Disable Mid/Side coding", 0, AV_OPT_TYPE_CONST, {.i64 =  0 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
         {"ms_force", "Force Mid/Side for the whole frame if possible", 0, AV_OPT_TYPE_CONST, {.i64 =  1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
@@ -826,7 +822,7 @@ AVCodec ff_aac_encoder = {
     .close          = aac_encode_end,
     .capabilities   = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY |
                       CODEC_CAP_EXPERIMENTAL,
-    .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT,
+    .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,
                                                      AV_SAMPLE_FMT_NONE },
     .long_name      = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
     .priv_class     = &aacenc_class,