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avcodec/opusdec: switch to swresample 

This also fixes linking failures in doc/examples which where apparently
caused by the linking order between avcodec and avresample

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer 2014-05-15 18:28:45 +02:00
parent 96cb4c8718
commit ffa05e0802
3 changed files with 30 additions and 31 deletions
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4
configure vendored
View File

@ -2110,7 +2110,7 @@ nellymoser_decoder_select="mdct sinewin"
nellymoser_encoder_select="audio_frame_queue mdct sinewin"
nuv_decoder_select="dsputil lzo"
on2avc_decoder_select="mdct"
opus_decoder_deps="avresample"
opus_decoder_deps="swresample"
png_decoder_select="zlib"
png_encoder_select="dsputil zlib"
prores_decoder_select="dsputil"
@ -5140,7 +5140,7 @@ enabled subtitles_filter && prepend avfilter_deps "avformat avcodec"
enabled lavfi_indev && prepend avdevice_deps "avfilter"
enabled opus_decoder && prepend avcodec_deps "avresample"
enabled opus_decoder && prepend avcodec_deps "swresample"
expand_deps(){
lib_deps=${1}_deps

4
libavcodec/opus.h
View File

@ -29,7 +29,7 @@
#include "libavutil/float_dsp.h"
#include "libavutil/frame.h"
#include "libavresample/avresample.h"
#include "libswresample/swresample.h"
#include "avcodec.h"
#include "get_bits.h"
@ -137,7 +137,7 @@ typedef struct OpusStreamContext {
float *out_dummy;
int out_dummy_allocated_size;
AVAudioResampleContext *avr;
SwrContext *swr;
AVAudioFifo *celt_delay;
int silk_samplerate;
/* number of samples we still want to get from the resampler */

53
libavcodec/opusdec.c
View File

@ -40,7 +40,7 @@
#include "libavutil/channel_layout.h"
#include "libavutil/opt.h"
#include "libavresample/avresample.h"
#include "libswresample/swresample.h"
#include "avcodec.h"
#include "celp_filters.h"
@ -114,9 +114,9 @@ static int opus_flush_resample(OpusStreamContext *s, int nb_samples)
{
int celt_size = av_audio_fifo_size(s->celt_delay);
int ret, i;
ret = avresample_convert(s->avr, (uint8_t**)s->out, s->out_size, nb_samples,
NULL, 0, 0);
ret = swr_convert(s->swr,
(uint8_t**)s->out, nb_samples,
NULL, 0);
if (ret < 0)
return ret;
else if (ret != nb_samples) {
@ -159,15 +159,16 @@ static int opus_init_resample(OpusStreamContext *s)
uint8_t *delayptr[2] = { (uint8_t*)delay, (uint8_t*)delay };
int ret;
av_opt_set_int(s->avr, "in_sample_rate", s->silk_samplerate, 0);
ret = avresample_open(s->avr);
av_opt_set_int(s->swr, "in_sample_rate", s->silk_samplerate, 0);
ret = swr_init(s->swr);
if (ret < 0) {
av_log(s->avctx, AV_LOG_ERROR, "Error opening the resampler.\n");
return ret;
}
ret = avresample_convert(s->avr, NULL, 0, 0, delayptr, sizeof(delay),
silk_resample_delay[s->packet.bandwidth]);
ret = swr_convert(s->swr,
NULL, 0,
delayptr, silk_resample_delay[s->packet.bandwidth]);
if (ret < 0) {
av_log(s->avctx, AV_LOG_ERROR,
"Error feeding initial silence to the resampler.\n");
@ -218,7 +219,7 @@ static int opus_decode_frame(OpusStreamContext *s, const uint8_t *data, int size
/* decode the silk frame */
if (s->packet.mode == OPUS_MODE_SILK || s->packet.mode == OPUS_MODE_HYBRID) {
if (!avresample_is_open(s->avr)) {
if (!swr_is_initialized(s->swr)) {
ret = opus_init_resample(s);
if (ret < 0)
return ret;
@ -232,12 +233,9 @@ static int opus_decode_frame(OpusStreamContext *s, const uint8_t *data, int size
av_log(s->avctx, AV_LOG_ERROR, "Error decoding a SILK frame.\n");
return samples;
}
samples = avresample_convert(s->avr, (uint8_t**)s->out, s->out_size,
s->packet.frame_duration,
(uint8_t**)s->silk_output,
sizeof(s->silk_buf[0]),
samples);
samples = swr_convert(s->swr,
(uint8_t**)s->out, s->packet.frame_duration,
(uint8_t**)s->silk_output, samples);
if (samples < 0) {
av_log(s->avctx, AV_LOG_ERROR, "Error resampling SILK data.\n");
return samples;
@ -374,10 +372,10 @@ static int opus_decode_subpacket(OpusStreamContext *s,
int i, j, ret;
/* check if we need to flush the resampler */
if (avresample_is_open(s->avr)) {
if (swr_is_initialized(s->swr)) {
if (buf) {
int64_t cur_samplerate;
av_opt_get_int(s->avr, "in_sample_rate", 0, &cur_samplerate);
av_opt_get_int(s->swr, "in_sample_rate", 0, &cur_samplerate);
flush_needed = (s->packet.mode == OPUS_MODE_CELT) || (cur_samplerate != s->silk_samplerate);
} else {
flush_needed = !!s->delayed_samples;
@ -406,7 +404,7 @@ static int opus_decode_subpacket(OpusStreamContext *s,
av_log(s->avctx, AV_LOG_ERROR, "Error flushing the resampler.\n");
return ret;
}
avresample_close(s->avr);
swr_close(s->swr);
output_samples += s->delayed_samples;
s->delayed_samples = 0;
@ -555,7 +553,7 @@ static av_cold void opus_decode_flush(AVCodecContext *ctx)
if (s->celt_delay)
av_audio_fifo_drain(s->celt_delay, av_audio_fifo_size(s->celt_delay));
avresample_close(s->avr);
swr_close(s->swr);
ff_silk_flush(s->silk);
ff_celt_flush(s->celt);
@ -577,7 +575,7 @@ static av_cold int opus_decode_close(AVCodecContext *avctx)
s->out_dummy_allocated_size = 0;
av_audio_fifo_free(s->celt_delay);
avresample_free(&s->avr);
swr_free(&s->swr);
}
av_freep(&c->streams);
@ -627,16 +625,17 @@ static av_cold int opus_decode_init(AVCodecContext *avctx)
s->fdsp = &c->fdsp;
s->avr = avresample_alloc_context();
if (!s->avr)
s->swr =swr_alloc();
if (!s->swr)
goto fail;
layout = (s->output_channels == 1) ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;
av_opt_set_int(s->avr, "in_sample_fmt", avctx->sample_fmt, 0);
av_opt_set_int(s->avr, "out_sample_fmt", avctx->sample_fmt, 0);
av_opt_set_int(s->avr, "in_channel_layout", layout, 0);
av_opt_set_int(s->avr, "out_channel_layout", layout, 0);
av_opt_set_int(s->avr, "out_sample_rate", avctx->sample_rate, 0);
av_opt_set_int(s->swr, "in_sample_fmt", avctx->sample_fmt, 0);
av_opt_set_int(s->swr, "out_sample_fmt", avctx->sample_fmt, 0);
av_opt_set_int(s->swr, "in_channel_layout", layout, 0);
av_opt_set_int(s->swr, "out_channel_layout", layout, 0);
av_opt_set_int(s->swr, "out_sample_rate", avctx->sample_rate, 0);
av_opt_set_int(s->swr, "filter_size", 16, 0);
ret = ff_silk_init(avctx, &s->silk, s->output_channels);
if (ret < 0)