#include "libavutil/common.h"
#include "libavutil/dict.h"
// #include "libavutil/error.h"
+#include "libavutil/frame.h"
#include "libavutil/log.h"
#include "libavutil/mem.h"
#include "libavutil/opt.h"
{
int ret;
+ if (avresample_is_open(avr)) {
+ av_log(avr, AV_LOG_ERROR, "The resampling context is already open.\n");
+ return AVERROR(EINVAL);
+ }
+
/* set channel mixing parameters */
avr->in_channels = av_get_channel_layout_nb_channels(avr->in_channel_layout);
if (avr->in_channels <= 0 || avr->in_channels > AVRESAMPLE_MAX_CHANNELS) {
av_get_sample_fmt_name(avr->internal_sample_fmt));
}
- /* treat all mono as planar for easier comparison */
- if (avr->in_channels == 1)
- avr->in_sample_fmt = av_get_planar_sample_fmt(avr->in_sample_fmt);
- if (avr->out_channels == 1)
- avr->out_sample_fmt = av_get_planar_sample_fmt(avr->out_sample_fmt);
-
/* we may need to add an extra conversion in order to remap channels if
the output format is not planar */
if (avr->use_channel_map && !avr->mixing_needed && !avr->resample_needed &&
- !av_sample_fmt_is_planar(avr->out_sample_fmt)) {
+ !ff_sample_fmt_is_planar(avr->out_sample_fmt, avr->out_channels)) {
avr->internal_sample_fmt = av_get_planar_sample_fmt(avr->out_sample_fmt);
}
avr->in_convert_needed = avr->in_sample_fmt != avr->internal_sample_fmt;
else
avr->in_convert_needed = avr->use_channel_map &&
- !av_sample_fmt_is_planar(avr->out_sample_fmt);
+ !ff_sample_fmt_is_planar(avr->out_sample_fmt, avr->out_channels);
if (avr->resample_needed || avr->mixing_needed || avr->in_convert_needed)
avr->out_convert_needed = avr->internal_sample_fmt != avr->out_sample_fmt;
}
if (avr->resample_needed) {
avr->resample_out_buffer = ff_audio_data_alloc(avr->out_channels,
- 0, avr->internal_sample_fmt,
+ 1024, avr->internal_sample_fmt,
"resample_out_buffer");
if (!avr->resample_out_buffer) {
ret = AVERROR(EINVAL);
return ret;
}
+int avresample_is_open(AVAudioResampleContext *avr)
+{
+ return !!avr->out_fifo;
+}
+
void avresample_close(AVAudioResampleContext *avr)
{
ff_audio_data_free(&avr->in_buffer);
current_buffer);
}
+int avresample_config(AVAudioResampleContext *avr, AVFrame *out, AVFrame *in)
+{
+ if (avresample_is_open(avr)) {
+ avresample_close(avr);
+ }
+
+ if (in) {
+ avr->in_channel_layout = in->channel_layout;
+ avr->in_sample_rate = in->sample_rate;
+ avr->in_sample_fmt = in->format;
+ }
+
+ if (out) {
+ avr->out_channel_layout = out->channel_layout;
+ avr->out_sample_rate = out->sample_rate;
+ avr->out_sample_fmt = out->format;
+ }
+
+ return 0;
+}
+
+static int config_changed(AVAudioResampleContext *avr,
+ AVFrame *out, AVFrame *in)
+{
+ int ret = 0;
+
+ if (in) {
+ if (avr->in_channel_layout != in->channel_layout ||
+ avr->in_sample_rate != in->sample_rate ||
+ avr->in_sample_fmt != in->format) {
+ ret |= AVERROR_INPUT_CHANGED;
+ }
+ }
+
+ if (out) {
+ if (avr->out_channel_layout != out->channel_layout ||
+ avr->out_sample_rate != out->sample_rate ||
+ avr->out_sample_fmt != out->format) {
+ ret |= AVERROR_OUTPUT_CHANGED;
+ }
+ }
+
+ return ret;
+}
+
+static inline int convert_frame(AVAudioResampleContext *avr,
+ AVFrame *out, AVFrame *in)
+{
+ int ret;
+ uint8_t **out_data = NULL, **in_data = NULL;
+ int out_linesize = 0, in_linesize = 0;
+ int out_nb_samples = 0, in_nb_samples = 0;
+
+ if (out) {
+ out_data = out->extended_data;
+ out_linesize = out->linesize[0];
+ out_nb_samples = out->nb_samples;
+ }
+
+ if (in) {
+ in_data = in->extended_data;
+ in_linesize = in->linesize[0];
+ in_nb_samples = in->nb_samples;
+ }
+
+ ret = avresample_convert(avr, out_data, out_linesize,
+ out_nb_samples,
+ in_data, in_linesize,
+ in_nb_samples);
+
+ if (ret < 0) {
+ if (out)
+ out->nb_samples = 0;
+ return ret;
+ }
+
+ if (out)
+ out->nb_samples = ret;
+
+ return 0;
+}
+
+static inline int available_samples(AVFrame *out)
+{
+ int bytes_per_sample = av_get_bytes_per_sample(out->format);
+ int samples = out->linesize[0] / bytes_per_sample;
+
+ if (av_sample_fmt_is_planar(out->format)) {
+ return samples;
+ } else {
+ int channels = av_get_channel_layout_nb_channels(out->channel_layout);
+ return samples / channels;
+ }
+}
+
+int avresample_convert_frame(AVAudioResampleContext *avr,
+ AVFrame *out, AVFrame *in)
+{
+ int ret, setup = 0;
+
+ if (!avresample_is_open(avr)) {
+ if ((ret = avresample_config(avr, out, in)) < 0)
+ return ret;
+ if ((ret = avresample_open(avr)) < 0)
+ return ret;
+ setup = 1;
+ } else {
+ // return as is or reconfigure for input changes?
+ if ((ret = config_changed(avr, out, in)))
+ return ret;
+ }
+
+ if (out) {
+ if (!out->linesize[0]) {
+ out->nb_samples = avresample_get_out_samples(avr, in->nb_samples);
+ if ((ret = av_frame_get_buffer(out, 0)) < 0) {
+ if (setup)
+ avresample_close(avr);
+ return ret;
+ }
+ } else {
+ if (!out->nb_samples)
+ out->nb_samples = available_samples(out);
+ }
+ }
+
+ return convert_frame(avr, out, in);
+}
+
int avresample_get_matrix(AVAudioResampleContext *avr, double *matrix,
int stride)
{
return av_audio_fifo_size(avr->out_fifo);
}
+int avresample_get_out_samples(AVAudioResampleContext *avr, int in_nb_samples)
+{
+ int64_t samples = avresample_get_delay(avr) + (int64_t)in_nb_samples;
+
+ if (avr->resample_needed) {
+ samples = av_rescale_rnd(samples,
+ avr->out_sample_rate,
+ avr->in_sample_rate,
+ AV_ROUND_UP);
+ }
+
+ samples += avresample_available(avr);
+
+ if (samples > INT_MAX)
+ return AVERROR(EINVAL);
+
+ return samples;
+}
+
int avresample_read(AVAudioResampleContext *avr, uint8_t **output, int nb_samples)
{
if (!output)
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