[FFmpeg-cvslog] iac/imc: use planar sample format

Justin Ruggles git at videolan.org
Tue Oct 2 17:32:23 CEST 2012


ffmpeg | branch: master | Justin Ruggles <justin.ruggles at gmail.com> | Mon Aug 27 12:58:25 2012 -0400| [3fca0d72105a607926adea702065b1a00adb0b5d] | committer: Justin Ruggles

iac/imc: use planar sample format

> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=3fca0d72105a607926adea702065b1a00adb0b5d
---

 libavcodec/imc.c |   26 +++++++++++---------------
 1 file changed, 11 insertions(+), 15 deletions(-)

diff --git a/libavcodec/imc.c b/libavcodec/imc.c
index 8e6b5bf..1156e8a 100644
--- a/libavcodec/imc.c
+++ b/libavcodec/imc.c
@@ -241,7 +241,7 @@ static av_cold int imc_decode_init(AVCodecContext *avctx)
         return ret;
     }
     ff_dsputil_init(&q->dsp, avctx);
-    avctx->sample_fmt     = AV_SAMPLE_FMT_FLT;
+    avctx->sample_fmt     = AV_SAMPLE_FMT_FLTP;
     avctx->channel_layout = avctx->channels == 1 ? AV_CH_LAYOUT_MONO
                                                  : AV_CH_LAYOUT_STEREO;
 
@@ -661,7 +661,7 @@ static void imc_imdct256(IMCContext *q, IMCChannel *chctx, int channels)
     int i;
     float re, im;
     float *dst1 = q->out_samples;
-    float *dst2 = q->out_samples + (COEFFS - 1) * channels;
+    float *dst2 = q->out_samples + (COEFFS - 1);
 
     /* prerotation */
     for (i = 0; i < COEFFS / 2; i++) {
@@ -683,8 +683,8 @@ static void imc_imdct256(IMCContext *q, IMCChannel *chctx, int channels)
                + (q->mdct_sine_window[i * 2] * re);
         *dst2 =  (q->mdct_sine_window[i * 2] * chctx->last_fft_im[i])
                - (q->mdct_sine_window[COEFFS - 1 - i * 2] * re);
-        dst1 += channels * 2;
-        dst2 -= channels * 2;
+        dst1 += 2;
+        dst2 -= 2;
         chctx->last_fft_im[i] = im;
     }
 }
@@ -785,7 +785,6 @@ static int imc_decode_block(AVCodecContext *avctx, IMCContext *q, int ch)
         chctx->decoder_reset = 1;
 
     if (chctx->decoder_reset) {
-        memset(q->out_samples, 0, COEFFS * sizeof(*q->out_samples));
         for (i = 0; i < BANDS; i++)
             chctx->old_floor[i] = 1.0;
         for (i = 0; i < COEFFS; i++)
@@ -944,7 +943,7 @@ static int imc_decode_frame(AVCodecContext *avctx, void *data,
     }
 
     for (i = 0; i < avctx->channels; i++) {
-        q->out_samples = (float*)q->frame.data[0] + i;
+        q->out_samples = (float *)q->frame.extended_data[i];
 
         q->dsp.bswap16_buf(buf16, (const uint16_t*)buf, IMC_BLOCK_SIZE / 2);
 
@@ -957,15 +956,8 @@ static int imc_decode_frame(AVCodecContext *avctx, void *data,
     }
 
     if (avctx->channels == 2) {
-        float *src = (float*)q->frame.data[0], t1, t2;
-
-        for (i = 0; i < COEFFS; i++) {
-            t1     = src[0];
-            t2     = src[1];
-            src[0] = t1 + t2;
-            src[1] = t1 - t2;
-            src   += 2;
-        }
+        q->dsp.butterflies_float((float *)q->frame.extended_data[0],
+                                 (float *)q->frame.extended_data[1], COEFFS);
     }
 
     *got_frame_ptr   = 1;
@@ -995,6 +987,8 @@ AVCodec ff_imc_decoder = {
     .decode         = imc_decode_frame,
     .capabilities   = CODEC_CAP_DR1,
     .long_name      = NULL_IF_CONFIG_SMALL("IMC (Intel Music Coder)"),
+    .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
+                                                      AV_SAMPLE_FMT_NONE },
 };
 
 AVCodec ff_iac_decoder = {
@@ -1007,4 +1001,6 @@ AVCodec ff_iac_decoder = {
     .decode         = imc_decode_frame,
     .capabilities   = CODEC_CAP_DR1,
     .long_name      = NULL_IF_CONFIG_SMALL("IAC (Indeo Audio Coder)"),
+    .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
+                                                      AV_SAMPLE_FMT_NONE },
 };



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