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