X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fdsddec.c;h=19fb75ee850a94c3d4bb8153396ff01e75a0a03b;hb=83b6471dcb762859f20b2c414decc755fcceb5e8;hp=2c5c357acc51fa86740e37d142b95d77077d8e85;hpb=0084eed5bffebd7f3915bc0f9eba7350e8bc0ef7;p=ffmpeg

diff --git a/libavcodec/dsddec.c b/libavcodec/dsddec.c
index 2c5c357acc5..19fb75ee850 100644
--- a/libavcodec/dsddec.c
+++ b/libavcodec/dsddec.c
@@ -27,11 +27,11 @@
  */
 
 #include "libavcodec/internal.h"
-#include "libavcodec/mathops.h"
 #include "avcodec.h"
 #include "dsd.h"
 
 #define DSD_SILENCE 0x69
+#define DSD_SILENCE_REVERSED 0x96
 /* 0x69 = 01101001
  * This pattern "on repeat" makes a low energy 352.8 kHz tone
  * and a high energy 1.0584 MHz tone which should be filtered
@@ -44,13 +44,16 @@ static av_cold int decode_init(AVCodecContext *avctx)
     int i;
     uint8_t silence;
 
+    if (!avctx->channels)
+        return AVERROR_INVALIDDATA;
+
     ff_init_dsd_data();
 
     s = av_malloc_array(sizeof(DSDContext), avctx->channels);
     if (!s)
         return AVERROR(ENOMEM);
 
-    silence = avctx->codec_id == AV_CODEC_ID_DSD_LSBF || avctx->codec_id == AV_CODEC_ID_DSD_LSBF_PLANAR ? ff_reverse[DSD_SILENCE] : DSD_SILENCE;
+    silence = avctx->codec_id == AV_CODEC_ID_DSD_LSBF || avctx->codec_id == AV_CODEC_ID_DSD_LSBF_PLANAR ? DSD_SILENCE_REVERSED : DSD_SILENCE;
     for (i = 0; i < avctx->channels; i++) {
         s[i].pos = 0;
         memset(s[i].buf, silence, sizeof(s[i].buf));
@@ -61,17 +64,20 @@ static av_cold int decode_init(AVCodecContext *avctx)
     return 0;
 }
 
-static int decode_frame(AVCodecContext *avctx, void *data,
-                        int *got_frame_ptr, AVPacket *avpkt)
+typedef struct ThreadData {
+    AVFrame *frame;
+    const AVPacket *avpkt;
+} ThreadData;
+
+static int dsd_channel(AVCodecContext *avctx, void *tdata, int j, int threadnr)
 {
-    DSDContext * s = avctx->priv_data;
-    AVFrame *frame = data;
-    int ret, i;
     int lsbf = avctx->codec_id == AV_CODEC_ID_DSD_LSBF || avctx->codec_id == AV_CODEC_ID_DSD_LSBF_PLANAR;
-    int src_next;
-    int src_stride;
-
-    frame->nb_samples = avpkt->size / avctx->channels;
+    DSDContext *s = avctx->priv_data;
+    ThreadData *td = tdata;
+    AVFrame *frame = td->frame;
+    const AVPacket *avpkt = td->avpkt;
+    int src_next, src_stride;
+    float *dst = ((float **)frame->extended_data)[j];
 
     if (avctx->codec_id == AV_CODEC_ID_DSD_LSBF_PLANAR || avctx->codec_id == AV_CODEC_ID_DSD_MSBF_PLANAR) {
         src_next   = frame->nb_samples;
@@ -81,30 +87,45 @@ static int decode_frame(AVCodecContext *avctx, void *data,
         src_stride = avctx->channels;
     }
 
+    ff_dsd2pcm_translate(&s[j], frame->nb_samples, lsbf,
+                         avpkt->data + j * src_next, src_stride,
+                         dst, 1);
+
+    return 0;
+}
+
+static int decode_frame(AVCodecContext *avctx, void *data,
+                        int *got_frame_ptr, AVPacket *avpkt)
+{
+    ThreadData td;
+    AVFrame *frame = data;
+    int ret;
+
+    frame->nb_samples = avpkt->size / avctx->channels;
+
     if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
         return ret;
 
-    for (i = 0; i < avctx->channels; i++) {
-        float * dst = ((float **)frame->extended_data)[i];
-        ff_dsd2pcm_translate(&s[i], frame->nb_samples, lsbf,
-            avpkt->data + i * src_next, src_stride,
-            dst, 1);
-    }
+    td.frame = frame;
+    td.avpkt = avpkt;
+    avctx->execute2(avctx, dsd_channel, &td, NULL, avctx->channels);
 
     *got_frame_ptr = 1;
     return frame->nb_samples * avctx->channels;
 }
 
 #define DSD_DECODER(id_, name_, long_name_) \
-AVCodec ff_##name_##_decoder = { \
+const AVCodec ff_ ## name_ ## _decoder = { \
     .name         = #name_, \
     .long_name    = NULL_IF_CONFIG_SMALL(long_name_), \
     .type         = AVMEDIA_TYPE_AUDIO, \
     .id           = AV_CODEC_ID_##id_, \
     .init         = decode_init, \
     .decode       = decode_frame, \
+    .capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_SLICE_THREADS, \
     .sample_fmts  = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP, \
                                                    AV_SAMPLE_FMT_NONE }, \
+    .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE, \
 };
 
 DSD_DECODER(DSD_LSBF, dsd_lsbf, "DSD (Direct Stream Digital), least significant bit first")