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Allow resampling with no channel count change for up to 8 channels.

This commit is contained in:
Alex Converse 2011-05-10 14:24:05 -07:00 committed by Alex Converse
parent 918a540953
commit 3e00ababc4
1 changed files with 41 additions and 43 deletions
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84
libavcodec/resample.c
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@ -29,6 +29,8 @@
#include "libavutil/opt.h"
#include "libavutil/samplefmt.h"
#define MAX_CHANNELS 8
struct AVResampleContext;
static const char *context_to_name(void *ptr)
@ -41,7 +43,7 @@ static const AVClass audioresample_context_class = { "ReSampleContext", context_
struct ReSampleContext {
struct AVResampleContext *resample_context;
short *temp[2];
short *temp[MAX_CHANNELS];
int temp_len;
float ratio;
/* channel convert */
@ -104,24 +106,25 @@ static void mono_to_stereo(short *output, short *input, int n1)
}
}
/* XXX: should use more abstract 'N' channels system */
static void stereo_split(short *output1, short *output2, short *input, int n)
static void deinterleave(short **output, short *input, int channels, int samples)
{
int i;
int i, j;
for(i=0;i<n;i++) {
*output1++ = *input++;
*output2++ = *input++;
for (i = 0; i < samples; i++) {
for (j = 0; j < channels; j++) {
*output[j]++ = *input++;
}
}
}
static void stereo_mux(short *output, short *input1, short *input2, int n)
static void interleave(short *output, short **input, int channels, int samples)
{
int i;
int i, j;
for(i=0;i<n;i++) {
*output++ = *input1++;
*output++ = *input2++;
for (i = 0; i < samples; i++) {
for (j = 0; j < channels; j++) {
*output++ = *input[j]++;
}
}
}
@ -151,14 +154,18 @@ ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
{
ReSampleContext *s;
if ( input_channels > 2)
if (input_channels > MAX_CHANNELS)
{
av_log(NULL, AV_LOG_ERROR, "Resampling with input channels greater than 2 unsupported.\n");
av_log(NULL, AV_LOG_ERROR,
"Resampling with input channels greater than %d is unsupported.\n",
MAX_CHANNELS);
return NULL;
}
if (output_channels > 2 && !(output_channels == 6 && input_channels == 2)) {
if ( output_channels > 2 &&
!(output_channels == 6 && input_channels == 2) &&
output_channels != input_channels) {
av_log(NULL, AV_LOG_ERROR,
"Resampling output channel count must be 1 or 2 for mono input and 1, 2 or 6 for stereo input.\n");
"Resampling output channel count must be 1 or 2 for mono input; 1, 2 or 6 for stereo input; or N for N channel input.\n");
return NULL;
}
@ -206,14 +213,6 @@ ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
}
}
/*
* AC-3 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;
#define TAPS 16
s->resample_context= av_resample_init(output_rate, input_rate,
filter_length, log2_phase_count, linear, cutoff);
@ -228,9 +227,9 @@ ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples)
{
int i, nb_samples1;
short *bufin[2];
short *bufout[2];
short *buftmp2[2], *buftmp3[2];
short *bufin[MAX_CHANNELS];
short *bufout[MAX_CHANNELS];
short *buftmp2[MAX_CHANNELS], *buftmp3[MAX_CHANNELS];
short *output_bak = NULL;
int lenout;
@ -291,12 +290,9 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl
bufin[i]= av_malloc( (nb_samples + s->temp_len) * sizeof(short) );
memcpy(bufin[i], s->temp[i], s->temp_len * sizeof(short));
buftmp2[i] = bufin[i] + s->temp_len;
bufout[i] = av_malloc(lenout * sizeof(short));
}
/* make some zoom to avoid round pb */
bufout[0]= av_malloc( lenout * sizeof(short) );
bufout[1]= av_malloc( lenout * sizeof(short) );
if (s->input_channels == 2 &&
s->output_channels == 1) {
buftmp3[0] = output;
@ -304,10 +300,11 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl
} else if (s->output_channels >= 2 && s->input_channels == 1) {
buftmp3[0] = bufout[0];
memcpy(buftmp2[0], input, nb_samples*sizeof(short));
} else if (s->output_channels >= 2) {
buftmp3[0] = bufout[0];
buftmp3[1] = bufout[1];
stereo_split(buftmp2[0], buftmp2[1], input, nb_samples);
} else if (s->output_channels >= s->input_channels && s->input_channels >= 2) {
for (i = 0; i < s->input_channels; i++) {
buftmp3[i] = bufout[i];
}
deinterleave(buftmp2, input, s->input_channels, nb_samples);
} else {
buftmp3[0] = output;
memcpy(buftmp2[0], input, nb_samples*sizeof(short));
@ -329,10 +326,10 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl
if (s->output_channels == 2 && s->input_channels == 1) {
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) {
} else if (s->output_channels == 6 && s->input_channels == 2) {
ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
} else if (s->output_channels == s->input_channels && s->input_channels >= 2) {
interleave(output, buftmp3, s->output_channels, nb_samples1);
}
if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
@ -348,19 +345,20 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl
}
}
for(i=0; i<s->filter_channels; i++)
for (i = 0; i < s->filter_channels; i++) {
av_free(bufin[i]);
av_free(bufout[i]);
}
av_free(bufout[0]);
av_free(bufout[1]);
return nb_samples1;
}
void audio_resample_close(ReSampleContext *s)
{
int i;
av_resample_close(s->resample_context);
av_freep(&s->temp[0]);
av_freep(&s->temp[1]);
for (i = 0; i < s->filter_channels; i++)
av_freep(&s->temp[i]);
av_freep(&s->buffer[0]);
av_freep(&s->buffer[1]);
av_audio_convert_free(s->convert_ctx[0]);