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-rw-r--r--audio/decode/ad.h2
-rw-r--r--audio/decode/ad_internal.h2
-rw-r--r--audio/decode/ad_lavc.c4
-rw-r--r--audio/decode/ad_mpg123.c2
-rw-r--r--audio/decode/ad_spdif.c30
-rw-r--r--audio/filter/af.c20
-rw-r--r--audio/filter/af_lavcac3enc.c116
-rw-r--r--audio/filter/af_lavcresample.c213
-rw-r--r--audio/filter/af_lavrresample.c277
-rw-r--r--audio/filter/af_resample.c394
-rw-r--r--audio/filter/af_resample_template.c171
-rw-r--r--audio/mixer.c4
-rw-r--r--audio/mixer.h2
-rw-r--r--audio/out/ao_dsound.c1
-rw-r--r--audio/out/ao_jack.c7
-rw-r--r--audio/out/ao_lavc.c32
16 files changed, 405 insertions, 872 deletions
diff --git a/audio/decode/ad.h b/audio/decode/ad.h
index 3bc3e39267..0bbc11ed9b 100644
--- a/audio/decode/ad.h
+++ b/audio/decode/ad.h
@@ -34,7 +34,7 @@ typedef struct ad_functions
int (*preinit)(sh_audio_t *sh);
int (*init)(sh_audio_t *sh, const char *decoder);
void (*uninit)(sh_audio_t *sh);
- int (*control)(sh_audio_t *sh,int cmd,void* arg, ...);
+ int (*control)(sh_audio_t *sh, int cmd, void *arg);
int (*decode_audio)(sh_audio_t *sh, unsigned char *buffer, int minlen,
int maxlen);
} ad_functions_t;
diff --git a/audio/decode/ad_internal.h b/audio/decode/ad_internal.h
index 7eca629fca..61bf306a5e 100644
--- a/audio/decode/ad_internal.h
+++ b/audio/decode/ad_internal.h
@@ -32,7 +32,7 @@ static void add_decoders(struct mp_decoder_list *list);
static int init(sh_audio_t *sh, const char *decoder);
static int preinit(sh_audio_t *sh);
static void uninit(sh_audio_t *sh);
-static int control(sh_audio_t *sh,int cmd,void* arg, ...);
+static int control(sh_audio_t *sh, int cmd, void *arg);
static int decode_audio(sh_audio_t *sh,unsigned char *buffer,int minlen,int maxlen);
#define LIBAD_EXTERN(x) const ad_functions_t mpcodecs_ad_##x = {\
diff --git a/audio/decode/ad_lavc.c b/audio/decode/ad_lavc.c
index 1f59280275..248df307d7 100644
--- a/audio/decode/ad_lavc.c
+++ b/audio/decode/ad_lavc.c
@@ -160,7 +160,7 @@ static int setup_format(sh_audio_t *sh_audio,
int container_samplerate = sh_audio->container_out_samplerate;
if (!container_samplerate && sh_audio->wf)
container_samplerate = sh_audio->wf->nSamplesPerSec;
- if (lavc_context->codec_id == CODEC_ID_AAC
+ if (lavc_context->codec_id == AV_CODEC_ID_AAC
&& samplerate == 2 * container_samplerate)
broken_srate = true;
else if (container_samplerate)
@@ -310,7 +310,7 @@ static void uninit(sh_audio_t *sh)
sh->context = NULL;
}
-static int control(sh_audio_t *sh, int cmd, void *arg, ...)
+static int control(sh_audio_t *sh, int cmd, void *arg)
{
struct priv *ctx = sh->context;
switch (cmd) {
diff --git a/audio/decode/ad_mpg123.c b/audio/decode/ad_mpg123.c
index 999dc2fbba..c17edc9197 100644
--- a/audio/decode/ad_mpg123.c
+++ b/audio/decode/ad_mpg123.c
@@ -455,7 +455,7 @@ static int decode_audio(sh_audio_t *sh, unsigned char *buf, int minlen,
return bytes;
}
-static int control(sh_audio_t *sh, int cmd, void *arg, ...)
+static int control(sh_audio_t *sh, int cmd, void *arg)
{
switch (cmd) {
case ADCTRL_RESYNC_STREAM:
diff --git a/audio/decode/ad_spdif.c b/audio/decode/ad_spdif.c
index 128b1bd8a9..2101721430 100644
--- a/audio/decode/ad_spdif.c
+++ b/audio/decode/ad_spdif.c
@@ -73,13 +73,13 @@ static int preinit(sh_audio_t *sh)
}
static int codecs[] = {
- CODEC_ID_AAC,
- CODEC_ID_AC3,
- CODEC_ID_DTS,
- CODEC_ID_EAC3,
- CODEC_ID_MP3,
- CODEC_ID_TRUEHD,
- CODEC_ID_NONE
+ AV_CODEC_ID_AAC,
+ AV_CODEC_ID_AC3,
+ AV_CODEC_ID_DTS,
+ AV_CODEC_ID_EAC3,
+ AV_CODEC_ID_MP3,
+ AV_CODEC_ID_TRUEHD,
+ AV_CODEC_ID_NONE
};
static int init(sh_audio_t *sh, const char *decoder)
@@ -149,21 +149,21 @@ static int init(sh_audio_t *sh, const char *decoder)
sh->ds->buffer_pos -= in_size;
switch (lavf_ctx->streams[0]->codec->codec_id) {
- case CODEC_ID_AAC:
+ case AV_CODEC_ID_AAC:
spdif_ctx->iec61937_packet_size = 16384;
sh->sample_format = AF_FORMAT_IEC61937_LE;
sh->samplerate = srate;
sh->channels = 2;
sh->i_bps = bps;
break;
- case CODEC_ID_AC3:
+ case AV_CODEC_ID_AC3:
spdif_ctx->iec61937_packet_size = 6144;
sh->sample_format = AF_FORMAT_AC3_LE;
sh->samplerate = srate;
sh->channels = 2;
sh->i_bps = bps;
break;
- case CODEC_ID_DTS:
+ case AV_CODEC_ID_DTS:
if(sh->opts->dtshd) {
opt = av_opt_find(&lavf_ctx->oformat->priv_class,
"dtshd_rate", NULL, 0, 0);
@@ -185,21 +185,21 @@ static int init(sh_audio_t *sh, const char *decoder)
sh->i_bps = bps;
}
break;
- case CODEC_ID_EAC3:
+ case AV_CODEC_ID_EAC3:
spdif_ctx->iec61937_packet_size = 24576;
sh->sample_format = AF_FORMAT_IEC61937_LE;
sh->samplerate = 192000;
sh->channels = 2;
sh->i_bps = bps;
break;
- case CODEC_ID_MP3:
+ case AV_CODEC_ID_MP3:
spdif_ctx->iec61937_packet_size = 4608;
sh->sample_format = AF_FORMAT_MPEG2;
sh->samplerate = srate;
sh->channels = 2;
sh->i_bps = bps;
break;
- case CODEC_ID_TRUEHD:
+ case AV_CODEC_ID_TRUEHD:
spdif_ctx->iec61937_packet_size = 61440;
sh->sample_format = AF_FORMAT_IEC61937_LE;
sh->samplerate = 192000;
@@ -270,7 +270,7 @@ static int decode_audio(sh_audio_t *sh, unsigned char *buf,
return spdif_ctx->out_buffer_len;
}
-static int control(sh_audio_t *sh, int cmd, void* arg, ...)
+static int control(sh_audio_t *sh, int cmd, void *arg)
{
unsigned char *start;
double pts;
@@ -307,7 +307,7 @@ static void uninit(sh_audio_t *sh)
static void add_decoders(struct mp_decoder_list *list)
{
- for (int n = 0; codecs[n] != CODEC_ID_NONE; n++) {
+ for (int n = 0; codecs[n] != AV_CODEC_ID_NONE; n++) {
const char *format = mp_codec_from_av_codec_id(codecs[n]);
if (format) {
mp_add_decoder(list, "spdif", format, format,
diff --git a/audio/filter/af.c b/audio/filter/af.c
index 71f4c67b51..ad43e5fca7 100644
--- a/audio/filter/af.c
+++ b/audio/filter/af.c
@@ -28,7 +28,6 @@ extern struct af_info af_info_dummy;
extern struct af_info af_info_delay;
extern struct af_info af_info_channels;
extern struct af_info af_info_format;
-extern struct af_info af_info_resample;
extern struct af_info af_info_volume;
extern struct af_info af_info_equalizer;
extern struct af_info af_info_pan;
@@ -38,7 +37,7 @@ extern struct af_info af_info_export;
extern struct af_info af_info_drc;
extern struct af_info af_info_extrastereo;
extern struct af_info af_info_lavcac3enc;
-extern struct af_info af_info_lavcresample;
+extern struct af_info af_info_lavrresample;
extern struct af_info af_info_sweep;
extern struct af_info af_info_hrtf;
extern struct af_info af_info_ladspa;
@@ -53,7 +52,6 @@ static struct af_info* filter_list[]={
&af_info_delay,
&af_info_channels,
&af_info_format,
- &af_info_resample,
&af_info_volume,
&af_info_equalizer,
&af_info_pan,
@@ -65,7 +63,7 @@ static struct af_info* filter_list[]={
&af_info_drc,
&af_info_extrastereo,
&af_info_lavcac3enc,
- &af_info_lavcresample,
+ &af_info_lavrresample,
&af_info_sweep,
&af_info_hrtf,
#ifdef CONFIG_LADSPA
@@ -527,9 +525,7 @@ int af_init(struct af_stream* s)
af = af_control_any_rev(s, AF_CONTROL_RESAMPLE_RATE | AF_CONTROL_SET,
&(s->output.rate));
if (!af) {
- char *resampler = "resample";
- if ((AF_INIT_TYPE_MASK & s->cfg.force) == AF_INIT_SLOW)
- resampler = "lavcresample";
+ char *resampler = "lavrresample";
if((AF_INIT_TYPE_MASK & s->cfg.force) == AF_INIT_SLOW){
if(!strcmp(s->first->info->name,"format"))
af = af_append(s,s->first,resampler);
@@ -546,16 +542,6 @@ int af_init(struct af_stream* s)
if(!af || (AF_OK != af->control(af,AF_CONTROL_RESAMPLE_RATE | AF_CONTROL_SET,
&(s->output.rate))))
return -1;
- // Use lin int if the user wants fast
- if ((AF_INIT_TYPE_MASK & s->cfg.force) == AF_INIT_FAST) {
- char args[32];
- sprintf(args, "%d", s->output.rate);
- if (strcmp(resampler, "lavcresample") == 0)
- strcat(args, ":1");
- else
- strcat(args, ":0:0");
- af->control(af, AF_CONTROL_COMMAND_LINE, args);
- }
}
if(AF_OK != af_reinit(s,af))
return -1;
diff --git a/audio/filter/af_lavcac3enc.c b/audio/filter/af_lavcac3enc.c
index 2b7a4ffb4c..94d59a11ca 100644
--- a/audio/filter/af_lavcac3enc.c
+++ b/audio/filter/af_lavcac3enc.c
@@ -27,7 +27,9 @@
#include <assert.h>
#include <libavcodec/avcodec.h>
+#include <libavutil/audioconvert.h>
#include <libavutil/intreadwrite.h>
+#include <libavutil/common.h>
#include <libavutil/mem.h>
#include "config.h"
@@ -47,6 +49,7 @@ const uint16_t ac3_bitrate_tab[19] = {
typedef struct af_ac3enc_s {
struct AVCodec *lavc_acodec;
struct AVCodecContext *lavc_actx;
+ AVPacket pkt;
bool planarize;
int add_iec61937_header;
int bit_rate;
@@ -105,6 +108,8 @@ static int control(struct af_instance *af, int cmd, void *arg)
// Put sample parameters
s->lavc_actx->channels = af->data->nch;
+ s->lavc_actx->channel_layout =
+ av_get_default_channel_layout(af->data->nch);
s->lavc_actx->sample_rate = af->data->rate;
s->lavc_actx->bit_rate = bit_rate;
@@ -155,12 +160,13 @@ static int control(struct af_instance *af, int cmd, void *arg)
// Deallocate memory
static void uninit(struct af_instance* af)
{
+ af_ac3enc_t *s = af->setup;
+
if (af->data)
free(af->data->audio);
free(af->data);
- if (af->setup) {
- af_ac3enc_t *s = af->setup;
- af->setup = NULL;
+ if (s) {
+ av_free_packet(&s->pkt);
if(s->lavc_actx) {
avcodec_close(s->lavc_actx);
av_free(s->lavc_actx);
@@ -176,8 +182,8 @@ static struct mp_audio* play(struct af_instance* af, struct mp_audio* data)
af_ac3enc_t *s = af->setup;
struct mp_audio *c = data; // Current working data
struct mp_audio *l;
- int len, left, outsize = 0, destsize;
- char *buf, *src, *dest;
+ int left, outsize = 0;
+ char *buf, *src;
int max_output_len;
int frame_num = (data->len + s->pending_len) / s->expect_len;
int samplesize = af_fmt2bits(s->in_sampleformat) / 8;
@@ -207,6 +213,8 @@ static struct mp_audio* play(struct af_instance* af, struct mp_audio* data)
while (left > 0) {
+ int ret;
+
if (left + s->pending_len < s->expect_len) {
memcpy(s->pending_data + s->pending_len, src, left);
src += left;
@@ -215,8 +223,7 @@ static struct mp_audio* play(struct af_instance* af, struct mp_audio* data)
break;
}
- dest = s->add_iec61937_header ? buf + 8 : buf;
- destsize = (char *)l->audio + l->len - buf;
+ char *src2 = src;
if (s->pending_len) {
int needs = s->expect_len - s->pending_len;
@@ -225,58 +232,69 @@ static struct mp_audio* play(struct af_instance* af, struct mp_audio* data)
src += needs;
left -= needs;
}
+ src2= s->pending_data;
+ }
- if (c->nch >= 5)
- reorder_channel_nch(s->pending_data,
- AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
- AF_CHANNEL_LAYOUT_LAVC_DEFAULT,
- c->nch,
- s->expect_len / samplesize, samplesize);
-
- void *data = (void *) s->pending_data;
- if (s->planarize) {
- void *data2 = malloc(s->expect_len);
- reorder_to_planar(data2, data, samplesize,
- c->nch, s->expect_len / samplesize / c->nch);
- data = data2;
- }
+ if (c->nch >= 5) {
+ reorder_channel_nch(src2,
+ AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
+ AF_CHANNEL_LAYOUT_LAVC_DEFAULT,
+ c->nch,
+ s->expect_len / samplesize,
+ samplesize);
+ }
- len = avcodec_encode_audio(s->lavc_actx, dest, destsize, data);
+ void *data = (void *) src2;
+ if (s->planarize) {
+ void *data2 = malloc(s->expect_len);
+ reorder_to_planar(data2, data, samplesize,
+ c->nch, s->expect_len / samplesize / c->nch);
+ data = data2;
+ }
- if (s->planarize)
- free(data);
+ AVFrame *frame = avcodec_alloc_frame();
+ if (!frame) {
+ mp_msg(MSGT_AFILTER, MSGL_FATAL, "[libaf] Could not allocate memory \n");
+ return NULL;
+ }
+ frame->nb_samples = AC3_FRAME_SIZE;
+ frame->format = s->lavc_actx->sample_fmt;
+ frame->channel_layout = s->lavc_actx->channel_layout;
+
+ ret = avcodec_fill_audio_frame(frame, c->nch, s->lavc_actx->sample_fmt,
+ (const uint8_t*)data, s->expect_len, 0);
+ if (ret < 0) {
+ mp_msg(MSGT_AFILTER, MSGL_FATAL, "[lavac3enc] Frame setup failed.\n");
+ return NULL;
+ }
- s->pending_len = 0;
+ int ok;
+ ret = avcodec_encode_audio2(s->lavc_actx, &s->pkt, frame, &ok);
+ if (ret < 0 || !ok) {
+ mp_msg(MSGT_AFILTER, MSGL_FATAL, "[lavac3enc] Encode failed.\n");
+ return NULL;
}
- else {
- if (c->nch >= 5)
- reorder_channel_nch(src,
- AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
- AF_CHANNEL_LAYOUT_LAVC_DEFAULT,
- c->nch,
- s->expect_len / samplesize, samplesize);
-
- void *data = (void *) src;
- if (s->planarize) {
- void *data2 = malloc(s->expect_len);
- reorder_to_planar(data2, data, samplesize,
- c->nch, s->expect_len / samplesize / c->nch);
- data = data2;
- }
- len = avcodec_encode_audio(s->lavc_actx, dest, destsize, data);
+ if (s->planarize)
+ free(data);
- if (s->planarize)
- free(data);
+ avcodec_free_frame(&frame);
+ if (s->pending_len) {
+ s->pending_len = 0;
+ } else {
src += s->expect_len;
left -= s->expect_len;
}
+
mp_msg(MSGT_AFILTER, MSGL_DBG2, "avcodec_encode_audio got %d, pending %d.\n",
- len, s->pending_len);
+ s->pkt.size, s->pending_len);
+ int len = s->pkt.size;
+ int header_len = 0;
if (s->add_iec61937_header) {
- int bsmod = dest[5] & 0x7;
+ assert(s->pkt.size > 5);
+ int bsmod = s->pkt.data[5] & 0x7;
AV_WB16(buf, 0xF872); // iec 61937 syncword 1
AV_WB16(buf + 2, 0x4E1F); // iec 61937 syncword 2
@@ -285,9 +303,15 @@ static struct mp_audio* play(struct af_instance* af, struct mp_audio* data)
AV_WB16(buf + 6, len << 3); // number of bits in payload
memset(buf + 8 + len, 0, AC3_FRAME_SIZE * 2 * 2 - 8 - len);
+ header_len = 8;
len = AC3_FRAME_SIZE * 2 * 2;
}
+ assert(buf + len <= (char *)af->data->audio + af->data->len);
+ assert(s->pkt.size <= len - header_len);
+
+ memcpy(buf + header_len, s->pkt.data, s->pkt.size);
+
outsize += len;
buf += len;
}
@@ -360,6 +384,8 @@ static int af_open(struct af_instance* af){
mp_msg(MSGT_AFILTER, MSGL_WARN,
"[af_lavcac3enc]: need to planarize audio data\n");
+ av_init_packet(&s->pkt);
+
return AF_OK;
}
diff --git a/audio/filter/af_lavcresample.c b/audio/filter/af_lavcresample.c
deleted file mode 100644
index ce777fed31..0000000000
--- a/audio/filter/af_lavcresample.c
+++ /dev/null
@@ -1,213 +0,0 @@
-/*
- * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
- *
- * This file is part of MPlayer.
- *
- * MPlayer is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * MPlayer is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with MPlayer; if not, write to the Free Software Foundation, Inc.,
- * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#include <inttypes.h>
-
-#include "config.h"
-#include "af.h"
-#include "libavcodec/avcodec.h"
-#include "libavutil/rational.h"
-
-// Data for specific instances of this filter
-typedef struct af_resample_s{
- struct AVResampleContext *avrctx;
- int16_t *in[AF_NCH];
- int in_alloc;
- int index;
-
- int filter_length;
- int linear;
- int phase_shift;
- double cutoff;
-
- int ctx_out_rate;
- int ctx_in_rate;
- int ctx_filter_size;
- int ctx_phase_shift;
- int ctx_linear;
- double ctx_cutoff;
-}af_resample_t;
-
-
-// Initialization and runtime control
-static int control(struct af_instance* af, int cmd, void* arg)
-{
- af_resample_t* s = (af_resample_t*)af->setup;
- struct mp_audio *data= (struct mp_audio*)arg;
- int out_rate, test_output_res; // helpers for checking input format
-
- switch(cmd){
- case AF_CONTROL_REINIT:
- if((af->data->rate == data->rate) || (af->data->rate == 0))
- return AF_DETACH;
-
- af->data->nch = data->nch;
- if (af->data->nch > AF_NCH) af->data->nch = AF_NCH;
- af->data->format = AF_FORMAT_S16_NE;
- af->data->bps = 2;
- af->mul = (double)af->data->rate / data->rate;
- af->delay = af->data->nch * s->filter_length / min(af->mul, 1); // *bps*.5
-
- if (s->ctx_out_rate != af->data->rate || s->ctx_in_rate != data->rate || s->ctx_filter_size != s->filter_length ||
- s->ctx_phase_shift != s->phase_shift || s->ctx_linear != s->linear || s->ctx_cutoff != s->cutoff) {
- if(s->avrctx) av_resample_close(s->avrctx);
- s->avrctx= av_resample_init(af->data->rate, /*in_rate*/data->rate, s->filter_length, s->phase_shift, s->linear, s->cutoff);
- s->ctx_out_rate = af->data->rate;
- s->ctx_in_rate = data->rate;
- s->ctx_filter_size = s->filter_length;
- s->ctx_phase_shift = s->phase_shift;
- s->ctx_linear = s->linear;
- s->ctx_cutoff = s->cutoff;
- }
-
- // hack to make af_test_output ignore the samplerate change
- out_rate = af->data->rate;
- af->data->rate = data->rate;
- test_output_res = af_test_output(af, (struct mp_audio*)arg);
- af->data->rate = out_rate;
- return test_output_res;
- case AF_CONTROL_COMMAND_LINE:{
- s->cutoff= 0.0;
- sscanf((char*)arg,"%d:%d:%d:%d:%lf", &af->data->rate, &s->filter_length, &s->linear, &s->phase_shift, &s->cutoff);
- if(s->cutoff <= 0.0) s->cutoff= max(1.0 - 6.5/(s->filter_length+8), 0.80);
- return AF_OK;
- }
- case AF_CONTROL_RESAMPLE_RATE | AF_CONTROL_SET:
- af->data->rate = *(int*)arg;
- return AF_OK;
- }
- return AF_UNKNOWN;
-}
-
-// Deallocate memory
-static void uninit(struct af_instance* af)
-{
- if(af->data)
- free(af->data->audio);
- free(af->data);
- if(af->setup){
- int i;
- af_resample_t *s = af->setup;
- if(s->avrctx) av_resample_close(s->avrctx);
- for (i=0; i < AF_NCH; i++)
- free(s->in[i]);
- free(s);
- }
-}
-
-// Filter data through filter
-static struct mp_audio* play(struct af_instance* af, struct mp_audio* data)
-{
- af_resample_t *s = af->setup;
- int i, j, consumed, ret = 0;
- int16_t *in = (int16_t*)data->audio;
- int16_t *out;
- int chans = data->nch;
- int in_len = data->len/(2*chans);
- int out_len = in_len * af->mul + 10;
- int16_t tmp[AF_NCH][out_len];
-
- if(AF_OK != RESIZE_LOCAL_BUFFER(af,data))
- return NULL;
-
- out= (int16_t*)af->data->audio;
-
- out_len= min(out_len, af->data->len/(2*chans));
-
- if(s->in_alloc < in_len + s->index){
- s->in_alloc= in_len + s->index;
- for(i=0; i<chans; i++){
- s->in[i]= realloc(s->in[i], s->in_alloc*sizeof(int16_t));
- }
- }
-
- if(chans==1){
- memcpy(&s->in[0][s->index], in, in_len * sizeof(int16_t));
- }else if(chans==2){
- for(j=0; j<in_len; j++){
- s->in[0][j + s->index]= *(in++);
- s->in[1][j + s->index]= *(in++);
- }
- }else{
- for(j=0; j<in_len; j++){
- for(i=0; i<chans; i++){
- s->in[i][j + s->index]= *(in++);
- }
- }
- }
- in_len += s->index;
-
- for(i=0; i<chans; i++){
- ret= av_resample(s->avrctx, tmp[i], s->in[i], &consumed, in_len, out_len, i+1 == chans);
- }
- out_len= ret;
-
- s->index= in_len - consumed;
- for(i=0; i<chans; i++){
- memmove(s->in[i], s->in[i] + consumed, s->index*sizeof(int16_t));
- }
-
- if(chans==1){
- memcpy(out, tmp[0], out_len*sizeof(int16_t));
- }else if(chans==2){
- for(j=0; j<out_len; j++){
- *(out++)= tmp[0][j];
- *(out++)= tmp[1][j];
- }
- }else{
- for(j=0; j<out_len; j++){
- for(i=0; i<chans; i++){
- *(out++)= tmp[i][j];
- }
- }
- }
-
- data->audio = af->data->audio;
- data->len = out_len*chans*2;
- data->rate = af->data->rate;
- return data;
-}
-
-static int af_open(struct af_instance* af){
- af_resample_t *s = calloc(1,sizeof(af_resample_t));
- af->control=control;
- af->uninit=uninit;
- af->play=play;
- af->mul=1;
- af->data=calloc(1,sizeof(struct mp_audio));
- s->filter_length= 16;
- s->cutoff= max(1.0 - 6.5/(s->filter_length+8), 0.80);
- s->phase_shift= 10;
-// s->setup = RSMP_INT | FREQ_SLOPPY;
- af->setup=s;
- return AF_OK;
-}
-
-struct af_info af_info_lavcresample = {
- "Sample frequency conversion using libavcodec",
- "lavcresample",
- "Michael Niedermayer",
- "",
- AF_FLAGS_REENTRANT,
- af_open
-};
diff --git a/audio/filter/af_lavrresample.c b/audio/filter/af_lavrresample.c
new file mode 100644
index 0000000000..5b26a0dce6
--- /dev/null
+++ b/audio/filter/af_lavrresample.c
@@ -0,0 +1,277 @@
+/*
+ * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
+ * Copyright (c) 2013 Stefano Pigozzi <stefano.pigozzi@gmail.com>
+ *
+ * This file is part of mpv.
+ *
+ * MPlayer is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * MPlayer is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with MPlayer; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <inttypes.h>
+#include <libavutil/opt.h>
+#include <libavutil/audioconvert.h>
+#include <libavutil/common.h>
+#include <libavutil/samplefmt.h>
+#include <libavutil/mathematics.h>
+
+#include "talloc.h"
+#include "config.h"
+
+#if defined(CONFIG_LIBAVRESAMPLE)
+#include <libavresample/avresample.h>
+#elif defined(CONFIG_LIBSWRESAMPLE)
+#include <libswresample/swresample.h>
+#define AVAudioResampleContext SwrContext
+#define avresample_alloc_context swr_alloc
+#define avresample_open swr_init
+#define avresample_close(x) do { } while(0)
+#define avresample_available(x) 0
+#define avresample_convert(ctx, out, out_planesize, out_samples, in, in_planesize, in_samples) \
+ swr_convert(ctx, out, out_samples, (const uint8_t**)(in), in_samples)
+#else
+#error "config.h broken"
+#endif
+
+#include "core/mp_msg.h"
+#include "core/subopt-helper.h"
+#include "audio/filter/af.h"
+
+struct af_resample_opts {
+ int filter_size;
+ int phase_shift;
+ int linear;
+ double cutoff;
+
+ int out_rate;
+ int in_rate;
+};
+
+struct af_resample {
+ struct AVAudioResampleContext *avrctx;
+ struct af_resample_opts ctx; // opts in the context
+ struct af_resample_opts opts; // opts requested by the user
+};
+
+#ifdef CONFIG_LIBAVRESAMPLE
+static int get_delay(struct af_resample *s)
+{
+ return avresample_get_delay(s->avrctx);
+}
+#else
+static int get_delay(struct af_resample *s)
+{
+ return swr_get_delay(s->avrctx, s->ctx.in_rate);
+}
+#endif
+
+static double af_resample_default_cutoff(int filter_size)
+{
+ return FFMAX(1.0 - 6.5 / (filter_size + 8), 0.80);
+}
+
+static bool needs_lavrctx_reconfigure(struct af_resample *s,
+ struct mp_audio *in,
+ struct mp_audio *out)
+{
+ return s->ctx.out_rate != out->rate ||
+ s->ctx.in_rate != in->rate ||
+ s->ctx.filter_size != s->opts.filter_size ||
+ s->ctx.phase_shift != s->opts.phase_shift ||
+ s->ctx.linear != s->opts.linear ||
+ s->ctx.cutoff != s->opts.cutoff;
+
+}
+
+#define ctx_opt_set_int(a,b) av_opt_set_int(s->avrctx, (a), (b), 0)
+#define ctx_opt_set_dbl(a,b) av_opt_set_double(s->avrctx, (a), (b), 0)
+
+static int control(struct af_instance *af, int cmd, void *arg)
+{
+ struct af_resample *s = (struct af_resample *) af->setup;
+ struct mp_audio *in = (struct mp_audio *) arg;
+ struct mp_audio *out = (struct mp_audio *) af->data;
+
+ switch (cmd) {
+ case AF_CONTROL_REINIT: {
+ if ((out->rate == in->rate) || (out->rate == 0))
+ return AF_DETACH;
+
+ out->nch = FFMIN(in->nch, AF_NCH);
+ out->format = AF_FORMAT_S16_NE;
+ out->bps = 2;
+ af->mul = (double) out->rate / in->rate;
+ af->delay = out->nch * s->opts.filter_size / FFMIN(af->mul, 1);
+
+ if (needs_lavrctx_reconfigure(s, in, out)) {
+ if (s->avrctx)
+ avresample_close(s->avrctx);
+
+ s->ctx.out_rate = out->rate;
+ s->ctx.in_rate = in->rate;
+ s->ctx.filter_size = s->opts.filter_size;
+ s->ctx.phase_shift = s->opts.phase_shift;
+ s->ctx.linear = s->opts.linear;
+ s->ctx.cutoff = s->opts.cutoff;
+
+ int ch_layout = av_get_default_channel_layout(out->nch);
+
+ ctx_opt_set_int("in_channel_layout", ch_layout);
+ ctx_opt_set_int("out_channel_layout", ch_layout);
+
+ ctx_opt_set_int("in_sample_rate", s->ctx.in_rate);
+ ctx_opt_set_int("out_sample_rate", s->ctx.out_rate);
+
+ ctx_opt_set_int("in_sample_fmt", AV_SAMPLE_FMT_S16);
+ ctx_opt_set_int("out_sample_fmt", AV_SAMPLE_FMT_S16);
+
+ ctx_opt_set_int("filter_size", s->ctx.filter_size);
+ ctx_opt_set_int("phase_shift", s->ctx.phase_shift);
+ ctx_opt_set_int("linear_interp", s->ctx.linear);
+
+ ctx_opt_set_dbl("cutoff", s->ctx.cutoff);
+
+ if (avresample_open(s->avrctx) < 0) {
+ mp_msg(MSGT_AFILTER, MSGL_ERR, "[lavrresample] Cannot open "
+ "Libavresample Context. \n");
+ return AF_ERROR;
+ }
+ }
+
+ int out_rate, test_output_res;
+ // hack to make af_test_output ignore the samplerate change
+ out_rate = out->rate;
+ out->rate = in->rate;
+ test_output_res = af_test_output(af, in);
+ out->rate = out_rate;
+ return test_output_res;
+ }
+ case AF_CONTROL_COMMAND_LINE: {
+ s->opts.cutoff = 0.0;
+
+ const opt_t subopts[] = {
+ {"srate", OPT_ARG_INT, &out->rate, NULL},
+ {"filter_size", OPT_ARG_INT, &s->opts.filter_size, NULL},
+ {"phase_shift", OPT_ARG_INT, &s->opts.phase_shift, NULL},
+ {"linear", OPT_ARG_BOOL, &s->opts.linear, NULL},
+ {"cutoff", OPT_ARG_FLOAT, &s->opts.cutoff, NULL},
+ {0}
+ };
+
+ if (subopt_parse(arg, subopts) != 0) {
+ mp_msg(MSGT_AFILTER, MSGL_ERR, "[lavrresample] Invalid option "
+ "specified.\n");
+ return AF_ERROR;
+ }
+
+ if (s->opts.cutoff <= 0.0)
+ s->opts.cutoff = af_resample_default_cutoff(s->opts.filter_size);
+ return AF_OK;
+ }
+ case AF_CONTROL_RESAMPLE_RATE | AF_CONTROL_SET:
+ out->rate = *(int *)arg;
+ return AF_OK;
+ }
+ return AF_UNKNOWN;
+}
+
+#undef ctx_opt_set_int
+#undef ctx_opt_set_dbl
+
+static void uninit(struct af_instance *af)
+{
+ if (af->setup) {
+ struct af_resample *s = af->setup;
+ if (s->avrctx)
+ avresample_close(s->avrctx);
+ talloc_free(af->setup);
+ }
+}
+
+static struct mp_audio *play(struct af_instance *af, struct mp_audio *data)
+{
+ struct af_resample *s = af->setup;
+ struct mp_audio *in = data;
+ struct mp_audio *out = af->data;
+
+
+ int in_size = data->len;
+ int in_samples = in_size / (data->bps * data->nch);
+ int out_samples = avresample_available(s->avrctx) +
+ av_rescale_rnd(get_delay(s) + in_samples,
+ s->ctx.out_rate, s->ctx.in_rate, AV_ROUND_UP);
+ int out_size = out->bps * out_samples * out->nch;
+
+ if (talloc_get_size(out->audio) < out_size)
+ out->audio = talloc_realloc_size(out, out->audio, out_size);
+
+ af->delay = out->bps * av_rescale_rnd(get_delay(s),
+ s->ctx.out_rate, s->ctx.in_rate,
+ AV_ROUND_UP);
+
+ out_samples = avresample_convert(s->avrctx,
+ (uint8_t **) &out->audio, out_size, out_samples,
+ (uint8_t **) &in->audio, in_size, in_samples);
+
+ out_size = out->bps * out_samples * out->nch;
+ in->audio = out->audio;
+ in->len = out_size;
+ in->rate = s->ctx.out_rate;
+ return data;