AC3 encoding patch ba (Ross Martin <ffmpeg at ross dot interwrx dot com>)
authorMichael Niedermayer <michaelni@gmx.at>
Wed, 20 Aug 2003 07:57:00 +0000 (07:57 +0000)
committerMichael Niedermayer <michaelni@gmx.at>
Wed, 20 Aug 2003 07:57:00 +0000 (07:57 +0000)
Originally committed as revision 2129 to svn://svn.ffmpeg.org/ffmpeg/trunk

ffmpeg.c
libavcodec/ac3enc.c
libavcodec/resample.c
tests/ffmpeg.regression.ref
tests/rotozoom.regression.ref

index 94abd74948d9ab2339e5efb1d7bacbbeaf64e270..92f009421c257b9e60871d0f9f9de58d0652db68 100644 (file)
--- a/ffmpeg.c
+++ b/ffmpeg.c
@@ -936,6 +936,11 @@ static int av_encode(AVFormatContext **output_files,
                             ost->resample = audio_resample_init(codec->channels, icodec->channels,
                                                         codec->sample_rate, 
                                                         icodec->sample_rate);
                             ost->resample = audio_resample_init(codec->channels, icodec->channels,
                                                         codec->sample_rate, 
                                                         icodec->sample_rate);
+                           if(!ost->resample)
+                             {
+                               printf("Can't resample.  Aborting.\n");
+                               av_abort();
+                             }
                         }
                         /* Request specific number of channels */
                         icodec->channels = codec->channels;
                         }
                         /* Request specific number of channels */
                         icodec->channels = codec->channels;
@@ -944,6 +949,11 @@ static int av_encode(AVFormatContext **output_files,
                         ost->resample = audio_resample_init(codec->channels, icodec->channels,
                                                         codec->sample_rate, 
                                                         icodec->sample_rate);
                         ost->resample = audio_resample_init(codec->channels, icodec->channels,
                                                         codec->sample_rate, 
                                                         icodec->sample_rate);
+                       if(!ost->resample)
+                         {
+                           printf("Can't resample.  Aborting.\n");
+                           av_abort();
+                         }
                     }
                 }
                 ist->decoding_needed = 1;
                     }
                 }
                 ist->decoding_needed = 1;
index 3e83fb1fcf3965d265af3d6ffb2a83a13a173023..4093fb75d85ced6778ff98987e94acb4091b5f0a 100644 (file)
@@ -978,7 +978,7 @@ static void output_audio_block(AC3EncodeContext *s,
                                int8_t global_exp[AC3_MAX_CHANNELS],
                                int block_num)
 {
                                int8_t global_exp[AC3_MAX_CHANNELS],
                                int block_num)
 {
-    int ch, nb_groups, group_size, i, baie;
+    int ch, nb_groups, group_size, i, baie, rbnd;
     uint8_t *p;
     uint16_t qmant[AC3_MAX_CHANNELS][N/2];
     int exp0, exp1;
     uint8_t *p;
     uint16_t qmant[AC3_MAX_CHANNELS][N/2];
     int exp0, exp1;
@@ -1000,14 +1000,28 @@ static void output_audio_block(AC3EncodeContext *s,
         put_bits(&s->pb, 1, 0); /* no new coupling strategy */
     }
 
         put_bits(&s->pb, 1, 0); /* no new coupling strategy */
     }
 
-    if (s->acmod == 2) {
-        put_bits(&s->pb, 1, 0); /* no matrixing (but should be used in the future) */
-    }
+    if (s->acmod == 2)
+      {
+       if(block_num==0)
+         {
+           /* first block must define rematrixing (rematstr)  */
+           put_bits(&s->pb, 1, 1); 
+           
+           /* dummy rematrixing rematflg(1:4)=0 */
+           for (rbnd=0;rbnd<4;rbnd++)
+             put_bits(&s->pb, 1, 0); 
+         }
+       else 
+         {
+           /* no matrixing (but should be used in the future) */
+           put_bits(&s->pb, 1, 0);
+         } 
+      }
 
 #if defined(DEBUG) 
     {
 
 #if defined(DEBUG) 
     {
-        static int count = 0;
-        printf("Block #%d (%d)\n", block_num, count++);
+      static int count = 0;
+      printf("Block #%d (%d)\n", block_num, count++);
     }
 #endif
     /* exponent strategy */
     }
 #endif
     /* exponent strategy */
@@ -1329,7 +1343,8 @@ static int output_frame_end(AC3EncodeContext *s)
     frame = s->pb.buf;
     n = 2 * s->frame_size - (pbBufPtr(&s->pb) - frame) - 2;
     assert(n >= 0);
     frame = s->pb.buf;
     n = 2 * s->frame_size - (pbBufPtr(&s->pb) - frame) - 2;
     assert(n >= 0);
-    memset(pbBufPtr(&s->pb), 0, n);
+    if(n>0)
+      memset(pbBufPtr(&s->pb), 0, n);
     
     /* Now we must compute both crcs : this is not so easy for crc1
        because it is at the beginning of the data... */
     
     /* Now we must compute both crcs : this is not so easy for crc1
        because it is at the beginning of the data... */
index 86bed847c4b6ec78a7a9d4b43467deff2fac1609..c4ae085f936afa4ff92319a5643493151af44a2c 100644 (file)
@@ -194,6 +194,23 @@ static void stereo_mux(short *output, short *input1, short *input2, int n)
     }
 }
 
     }
 }
 
+static void ac3_5p1_mux(short *output, short *input1, short *input2, int n)
+{
+    int i;
+    short l,r;
+
+    for(i=0;i<n;i++) {
+      l=*input1++;
+      r=*input2++;
+      *output++ = l;           /* left */
+      *output++ = (l/2)+(r/2); /* center */
+      *output++ = r;           /* right */
+      *output++ = 0;           /* left surround */
+      *output++ = 0;           /* right surroud */
+      *output++ = 0;           /* low freq */
+    }
+}
+
 static int mono_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
 {
     short *buf1;
 static int mono_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
 {
     short *buf1;
@@ -225,12 +242,18 @@ ReSampleContext *audio_resample_init(int output_channels, int input_channels,
     ReSampleContext *s;
     int i;
     
     ReSampleContext *s;
     int i;
     
-    if (output_channels > 2 || input_channels > 2)
-        return NULL;
+    if ( input_channels > 2)
+      {
+       printf("Resampling with input channels greater than 2 unsupported.");
+       return NULL;
+      }
 
     s = av_mallocz(sizeof(ReSampleContext));
     if (!s)
 
     s = av_mallocz(sizeof(ReSampleContext));
     if (!s)
-        return NULL;
+      {
+       printf("Can't allocate memory for resample context.");
+       return NULL;
+      }
 
     s->ratio = (float)output_rate / (float)input_rate;
     
 
     s->ratio = (float)output_rate / (float)input_rate;
     
@@ -241,6 +264,14 @@ ReSampleContext *audio_resample_init(int output_channels, int input_channels,
     if (s->output_channels < s->filter_channels)
         s->filter_channels = s->output_channels;
 
     if (s->output_channels < s->filter_channels)
         s->filter_channels = s->output_channels;
 
+/*
+ * ac3 output is the only case where filter_channels could be greater than 2.
+ * input channels can't be greater than 2, so resample the 2 channels and then
+ * expand to 6 channels after the resampling.
+ */
+    if(s->filter_channels>2)
+      s->filter_channels = 2;
+
     for(i=0;i<s->filter_channels;i++) {
         init_mono_resample(&s->channel_ctx[i], s->ratio);
     }
     for(i=0;i<s->filter_channels;i++) {
         init_mono_resample(&s->channel_ctx[i], s->ratio);
     }
@@ -279,10 +310,10 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl
         buftmp2[0] = bufin[0];
         buftmp3[0] = output;
         stereo_to_mono(buftmp2[0], input, nb_samples);
         buftmp2[0] = bufin[0];
         buftmp3[0] = output;
         stereo_to_mono(buftmp2[0], input, nb_samples);
-    } else if (s->output_channels == 2 && s->input_channels == 1) {
+    } else if (s->output_channels >= 2 && s->input_channels == 1) {
         buftmp2[0] = input;
         buftmp3[0] = bufout[0];
         buftmp2[0] = input;
         buftmp3[0] = bufout[0];
-    } else if (s->output_channels == 2) {
+    } else if (s->output_channels >= 2) {
         buftmp2[0] = bufin[0];
         buftmp2[1] = bufin[1];
         buftmp3[0] = bufout[0];
         buftmp2[0] = bufin[0];
         buftmp2[1] = bufin[1];
         buftmp3[0] = bufout[0];
@@ -303,6 +334,8 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl
         mono_to_stereo(output, buftmp3[0], nb_samples1);
     } else if (s->output_channels == 2) {
         stereo_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
         mono_to_stereo(output, buftmp3[0], nb_samples1);
     } else if (s->output_channels == 2) {
         stereo_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
+    } else if (s->output_channels == 6) {
+        ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
     }
 
     av_free(bufin[0]);
     }
 
     av_free(bufin[0]);
index de33c3a38cc8cc11e6e1f8d219d40f238b8e18f9..a20dcf3727920dc1f24e08a7262aa574fe671991 100644 (file)
@@ -52,4 +52,4 @@ stddev: 19.19 bytes:7602176
 stddev:  8.19 bytes:7602176
 21f8ff9f1daacd9133683bb4ea0f50a4 *./data/a-mp2.mp2
 116d1290ba1b4eb98fdee52e423417b1 *./data/out.wav
 stddev:  8.19 bytes:7602176
 21f8ff9f1daacd9133683bb4ea0f50a4 *./data/a-mp2.mp2
 116d1290ba1b4eb98fdee52e423417b1 *./data/out.wav
-048b9c3444c788bac6ce5cc3a8f4db00 *./data/a-ac3.rm
+d056da679e6d6682812fffb28a7f0db6 *./data/a-ac3.rm
index af1ab41b5f1054cc04bed7a3b8eaf662701ef3f0..e1e264537f5cab5b0f11596cb0ace170786cdf86 100644 (file)
@@ -52,4 +52,4 @@ bee27a404ab6a1b7ab1d3551eb4f1877 *./data/a-flv.flv
 stddev:  5.29 bytes:7602176
 21f8ff9f1daacd9133683bb4ea0f50a4 *./data/a-mp2.mp2
 116d1290ba1b4eb98fdee52e423417b1 *./data/out.wav
 stddev:  5.29 bytes:7602176
 21f8ff9f1daacd9133683bb4ea0f50a4 *./data/a-mp2.mp2
 116d1290ba1b4eb98fdee52e423417b1 *./data/out.wav
-048b9c3444c788bac6ce5cc3a8f4db00 *./data/a-ac3.rm
+d056da679e6d6682812fffb28a7f0db6 *./data/a-ac3.rm