swresample: Add swr_get_out_samples()
Previous version reviewed-by: Pavel Koshevoy <pkoshevoy@gmail.com> Previous version reviewed-by: wm4 <nfxjfg@googlemail.com> Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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@ -15,6 +15,9 @@ libavutil: 2014-08-09
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API changes, most recent first:
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2015-06-04 - xxxxxxx - lswr 1.2.100
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Add swr_get_out_samples()
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2015-05-27 - xxxxxxx - lavu 54.26.100 - cpu.h
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Add AV_CPU_FLAG_AVXSLOW.
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@ -345,6 +345,25 @@ static int64_t get_delay(struct SwrContext *s, int64_t base){
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return av_rescale(num, base, s->in_sample_rate*(int64_t)c->src_incr << c->phase_shift);
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}
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static int64_t get_out_samples(struct SwrContext *s, int in_samples) {
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ResampleContext *c = s->resample;
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// The + 2 are added to allow implementations to be slightly inaccuarte, they should not be needed currently
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// They also make it easier to proof that changes and optimizations do not
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// break the upper bound
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int64_t num = s->in_buffer_count + 2LL + in_samples;
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num *= 1 << c->phase_shift;
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num -= c->index;
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num = av_rescale_rnd(num, s->out_sample_rate, ((int64_t)s->in_sample_rate) << c->phase_shift, AV_ROUND_UP) + 2;
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if (c->compensation_distance) {
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if (num > INT_MAX)
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return AVERROR(EINVAL);
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num = FFMAX(num, (num * c->ideal_dst_incr - 1) / c->dst_incr + 1);
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}
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return num;
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}
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static int resample_flush(struct SwrContext *s) {
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AudioData *a= &s->in_buffer;
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int i, j, ret;
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@ -414,4 +433,5 @@ struct Resampler const swri_resampler={
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set_compensation,
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get_delay,
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invert_initial_buffer,
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get_out_samples,
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};
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@ -673,11 +673,15 @@ int attribute_align_arg swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_C
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const uint8_t *in_arg [SWR_CH_MAX], int in_count){
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AudioData * in= &s->in;
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AudioData *out= &s->out;
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int av_unused max_output;
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if (!swr_is_initialized(s)) {
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av_log(s, AV_LOG_ERROR, "Context has not been initialized\n");
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return AVERROR(EINVAL);
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}
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#if ASSERT_LEVEL >1
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max_output = swr_get_out_samples(s, in_count);
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#endif
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while(s->drop_output > 0){
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int ret;
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@ -720,6 +724,9 @@ int attribute_align_arg swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_C
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int ret = swr_convert_internal(s, out, out_count, in, in_count);
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if(ret>0 && !s->drop_output)
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s->outpts += ret * (int64_t)s->in_sample_rate;
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av_assert2(max_output < 0 || ret < 0 || ret <= max_output);
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return ret;
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}else{
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AudioData tmp= *in;
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@ -771,6 +778,7 @@ int attribute_align_arg swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_C
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}
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if(ret2>0 && !s->drop_output)
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s->outpts += ret2 * (int64_t)s->in_sample_rate;
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av_assert2(max_output < 0 || ret2 < 0 || ret2 <= max_output);
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return ret2;
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}
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}
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@ -822,6 +830,28 @@ int64_t swr_get_delay(struct SwrContext *s, int64_t base){
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}
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}
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int swr_get_out_samples(struct SwrContext *s, int in_samples)
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{
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int64_t out_samples;
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if (in_samples < 0)
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return AVERROR(EINVAL);
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if (s->resampler && s->resample) {
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if (!s->resampler->get_out_samples)
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return AVERROR(ENOSYS);
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out_samples = s->resampler->get_out_samples(s, in_samples);
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} else {
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out_samples = s->in_buffer_count + in_samples;
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av_assert0(s->out_sample_rate == s->in_sample_rate);
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}
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if (out_samples > INT_MAX)
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return AVERROR(EINVAL);
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return out_samples;
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}
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int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance){
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int ret;
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@ -294,9 +294,10 @@ void swr_close(struct SwrContext *s);
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* in and in_count can be set to 0 to flush the last few samples out at the
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* end.
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*
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* If more input is provided than output space then the input will be buffered.
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* You can avoid this buffering by providing more output space than input.
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* Conversion will run directly without copying whenever possible.
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* If more input is provided than output space, then the input will be buffered.
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* You can avoid this buffering by using swr_get_out_samples() to retrieve an
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* upper bound on the required number of output samples for the given number of
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* input samples. Conversion will run directly without copying whenever possible.
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*
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* @param s allocated Swr context, with parameters set
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* @param out output buffers, only the first one need be set in case of packed audio
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@ -435,6 +436,24 @@ int swr_inject_silence(struct SwrContext *s, int count);
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*/
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int64_t swr_get_delay(struct SwrContext *s, int64_t base);
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/**
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* Find an upper bound on the number of samples that the next swr_convert
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* call will output, if called with in_samples of input samples. This
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* depends on the internal state, and anything changing the internal state
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* (like further swr_convert() calls) will may change the number of samples
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* swr_get_out_samples() returns for the same number of input samples.
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*
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* @param in_samples number of input samples.
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* @note any call to swr_inject_silence(), swr_convert(), swr_next_pts()
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* or swr_set_compensation() invalidates this limit
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* @note it is recommended to pass the correct available buffer size
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* to all functions like swr_convert() even if swr_get_out_samples()
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* indicates that less would be used.
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* @returns an upper bound on the number of samples that the next swr_convert
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* will output or a negative value to indicate an error
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*/
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int swr_get_out_samples(struct SwrContext *s, int in_samples);
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/**
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* @}
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*
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@ -76,6 +76,7 @@ typedef int (* resample_flush_func)(struct SwrContext *c);
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typedef int (* set_compensation_func)(struct ResampleContext *c, int sample_delta, int compensation_distance);
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typedef int64_t (* get_delay_func)(struct SwrContext *s, int64_t base);
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typedef int (* invert_initial_buffer_func)(struct ResampleContext *c, AudioData *dst, const AudioData *src, int src_size, int *dst_idx, int *dst_count);
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typedef int64_t (* get_out_samples_func)(struct SwrContext *s, int in_samples);
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struct Resampler {
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resample_init_func init;
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@ -85,6 +86,7 @@ struct Resampler {
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set_compensation_func set_compensation;
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get_delay_func get_delay;
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invert_initial_buffer_func invert_initial_buffer;
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get_out_samples_func get_out_samples;
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};
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extern struct Resampler const swri_resampler;
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@ -29,7 +29,7 @@
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#include "libavutil/avutil.h"
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#define LIBSWRESAMPLE_VERSION_MAJOR 1
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#define LIBSWRESAMPLE_VERSION_MINOR 1
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#define LIBSWRESAMPLE_VERSION_MINOR 2
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#define LIBSWRESAMPLE_VERSION_MICRO 100
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#define LIBSWRESAMPLE_VERSION_INT AV_VERSION_INT(LIBSWRESAMPLE_VERSION_MAJOR, \
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