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AC3 encoding patch ba (Ross Martin <ffmpeg at ross dot interwrx dot com>)

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Originally committed as revision 2129 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Michael Niedermayer 2003-08-20 07:57:00 +00:00
parent b928ec649c
commit 743739d2c5
5 changed files with 72 additions and 14 deletions
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10
ffmpeg.c
View File

@ -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);
if(!ost->resample)
{
printf("Can't resample. Aborting.\n");
av_abort();
}
}
/* 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);
if(!ost->resample)
{
printf("Can't resample. Aborting.\n");
av_abort();
}
}
}
ist->decoding_needed = 1;

29
libavcodec/ac3enc.c
View File

@ -978,7 +978,7 @@ static void output_audio_block(AC3EncodeContext *s,
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;
@ -1000,14 +1000,28 @@ static void output_audio_block(AC3EncodeContext *s,
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)
{
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 */
@ -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);
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... */

43
libavcodec/resample.c
View 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;
@ -225,12 +242,18 @@ ReSampleContext *audio_resample_init(int output_channels, int input_channels,
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)
return NULL;
{
printf("Can't allocate memory for resample context.");
return NULL;
}
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;
/*
* 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);
}
@ -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);
} 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];
} else if (s->output_channels == 2) {
} else if (s->output_channels >= 2) {
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);
} else if (s->output_channels == 6) {
ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
}
av_free(bufin[0]);

2
tests/ffmpeg.regression.ref
View File

@ -52,4 +52,4 @@ stddev: 19.19 bytes:7602176
stddev: 8.19 bytes:7602176
21f8ff9f1daacd9133683bb4ea0f50a4 *./data/a-mp2.mp2
116d1290ba1b4eb98fdee52e423417b1 *./data/out.wav
048b9c3444c788bac6ce5cc3a8f4db00 *./data/a-ac3.rm
d056da679e6d6682812fffb28a7f0db6 *./data/a-ac3.rm

2
tests/rotozoom.regression.ref
View File

@ -52,4 +52,4 @@ bee27a404ab6a1b7ab1d3551eb4f1877 *./data/a-flv.flv
stddev: 5.29 bytes:7602176
21f8ff9f1daacd9133683bb4ea0f50a4 *./data/a-mp2.mp2
116d1290ba1b4eb98fdee52e423417b1 *./data/out.wav
048b9c3444c788bac6ce5cc3a8f4db00 *./data/a-ac3.rm
d056da679e6d6682812fffb28a7f0db6 *./data/a-ac3.rm