* resampling audio filter
*/
+#include "libavutil/audioconvert.h"
+#include "libavutil/avstring.h"
+#include "libavutil/opt.h"
+#include "libavutil/samplefmt.h"
+#include "libavutil/avassert.h"
#include "libswresample/swresample.h"
#include "avfilter.h"
+#include "audio.h"
#include "internal.h"
typedef struct {
- int out_rate;
double ratio;
struct SwrContext *swr;
+ int64_t next_pts;
+ int req_fullfilled;
} AResampleContext;
-static av_cold int init(AVFilterContext *ctx, const char *args, void *opaque)
+static av_cold int init(AVFilterContext *ctx, const char *args)
{
AResampleContext *aresample = ctx->priv;
- int ret;
+ int ret = 0;
+ char *argd = av_strdup(args);
+
+ aresample->next_pts = AV_NOPTS_VALUE;
+ aresample->swr = swr_alloc();
+ if (!aresample->swr)
+ return AVERROR(ENOMEM);
if (args) {
- if ((ret = ff_parse_sample_rate(&aresample->out_rate, args, ctx)) < 0)
- return ret;
- } else {
- aresample->out_rate = -1;
- }
+ char *ptr=argd, *token;
- return 0;
+ while(token = av_strtok(ptr, ":", &ptr)) {
+ char *value;
+ av_strtok(token, "=", &value);
+
+ if(value) {
+ if((ret=av_opt_set(aresample->swr, token, value, 0)) < 0)
+ goto end;
+ } else {
+ int out_rate;
+ if ((ret = ff_parse_sample_rate(&out_rate, token, ctx)) < 0)
+ goto end;
+ if((ret = av_opt_set_int(aresample->swr, "osr", out_rate, 0)) < 0)
+ goto end;
+ }
+ }
+ }
+end:
+ av_free(argd);
+ return ret;
}
static av_cold void uninit(AVFilterContext *ctx)
swr_free(&aresample->swr);
}
+static int query_formats(AVFilterContext *ctx)
+{
+ AResampleContext *aresample = ctx->priv;
+ int out_rate = av_get_int(aresample->swr, "osr", NULL);
+ uint64_t out_layout = av_get_int(aresample->swr, "ocl", NULL);
+ enum AVSampleFormat out_format = av_get_int(aresample->swr, "osf", NULL);
+
+ AVFilterLink *inlink = ctx->inputs[0];
+ AVFilterLink *outlink = ctx->outputs[0];
+
+ AVFilterFormats *in_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
+ AVFilterFormats *out_formats;
+ AVFilterFormats *in_samplerates = ff_all_samplerates();
+ AVFilterFormats *out_samplerates;
+ AVFilterChannelLayouts *in_layouts = ff_all_channel_layouts();
+ AVFilterChannelLayouts *out_layouts;
+
+ ff_formats_ref (in_formats, &inlink->out_formats);
+ ff_formats_ref (in_samplerates, &inlink->out_samplerates);
+ ff_channel_layouts_ref(in_layouts, &inlink->out_channel_layouts);
+
+ if(out_rate > 0) {
+ out_samplerates = ff_make_format_list((int[]){ out_rate, -1 });
+ } else {
+ out_samplerates = ff_all_samplerates();
+ }
+ ff_formats_ref(out_samplerates, &outlink->in_samplerates);
+
+ if(out_format != AV_SAMPLE_FMT_NONE) {
+ out_formats = ff_make_format_list((int[]){ out_format, -1 });
+ } else
+ out_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
+ ff_formats_ref(out_formats, &outlink->in_formats);
+
+ if(out_layout) {
+ out_layouts = avfilter_make_format64_list((int64_t[]){ out_layout, -1 });
+ } else
+ out_layouts = ff_all_channel_layouts();
+ ff_channel_layouts_ref(out_layouts, &outlink->in_channel_layouts);
+
+ return 0;
+}
+
+
static int config_output(AVFilterLink *outlink)
{
int ret;
AVFilterContext *ctx = outlink->src;
AVFilterLink *inlink = ctx->inputs[0];
AResampleContext *aresample = ctx->priv;
+ int out_rate;
+ uint64_t out_layout;
+ enum AVSampleFormat out_format;
+ char inchl_buf[128], outchl_buf[128];
- if (aresample->out_rate == -1)
- aresample->out_rate = outlink->sample_rate;
- else
- outlink->sample_rate = aresample->out_rate;
- outlink->time_base = (AVRational) {1, aresample->out_rate};
-
- //TODO: make the resampling parameters (filter size, phrase shift, linear, cutoff) configurable
aresample->swr = swr_alloc_set_opts(aresample->swr,
- inlink->channel_layout, inlink->format, aresample->out_rate,
+ outlink->channel_layout, outlink->format, outlink->sample_rate,
inlink->channel_layout, inlink->format, inlink->sample_rate,
0, ctx);
if (!aresample->swr)
return AVERROR(ENOMEM);
+
ret = swr_init(aresample->swr);
if (ret < 0)
return ret;
+ out_rate = av_get_int(aresample->swr, "osr", NULL);
+ out_layout = av_get_int(aresample->swr, "ocl", NULL);
+ out_format = av_get_int(aresample->swr, "osf", NULL);
+ outlink->time_base = (AVRational) {1, out_rate};
+
+ av_assert0(outlink->sample_rate == out_rate);
+ av_assert0(outlink->channel_layout == out_layout);
+ av_assert0(outlink->format == out_format);
+
aresample->ratio = (double)outlink->sample_rate / inlink->sample_rate;
- av_log(ctx, AV_LOG_INFO, "r:%"PRId64"Hz -> r:%"PRId64"Hz\n",
- inlink->sample_rate, outlink->sample_rate);
+ av_get_channel_layout_string(inchl_buf, sizeof(inchl_buf), -1, inlink ->channel_layout);
+ av_get_channel_layout_string(outchl_buf, sizeof(outchl_buf), -1, outlink->channel_layout);
+
+ av_log(ctx, AV_LOG_INFO, "chl:%s fmt:%s r:%dHz -> chl:%s fmt:%s r:%dHz\n",
+ inchl_buf, av_get_sample_fmt_name(inlink->format), inlink->sample_rate,
+ outchl_buf, av_get_sample_fmt_name(outlink->format), outlink->sample_rate);
return 0;
}
{
AResampleContext *aresample = inlink->dst->priv;
const int n_in = insamplesref->audio->nb_samples;
- int n_out = n_in * aresample->ratio;
+ int n_out = n_in * aresample->ratio * 2 ;
AVFilterLink *const outlink = inlink->dst->outputs[0];
- AVFilterBufferRef *outsamplesref = avfilter_get_audio_buffer(outlink, AV_PERM_WRITE, n_out);
+ AVFilterBufferRef *outsamplesref = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n_out);
- n_out = swr_convert(aresample->swr, outsamplesref->data, n_out,
- (void *)insamplesref->data, n_in);
avfilter_copy_buffer_ref_props(outsamplesref, insamplesref);
+
+ if(insamplesref->pts != AV_NOPTS_VALUE) {
+ int64_t inpts = av_rescale(insamplesref->pts, inlink->time_base.num * (int64_t)outlink->sample_rate * inlink->sample_rate, inlink->time_base.den);
+ int64_t outpts= swr_next_pts(aresample->swr, inpts);
+ aresample->next_pts =
+ outsamplesref->pts = (outpts + inlink->sample_rate/2) / inlink->sample_rate;
+ } else {
+ outsamplesref->pts = AV_NOPTS_VALUE;
+ }
+
+ n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out,
+ (void *)insamplesref->extended_data, n_in);
+ if (n_out <= 0) {
+ avfilter_unref_buffer(outsamplesref);
+ avfilter_unref_buffer(insamplesref);
+ return;
+ }
+
outsamplesref->audio->sample_rate = outlink->sample_rate;
outsamplesref->audio->nb_samples = n_out;
- outsamplesref->pts = insamplesref->pts == AV_NOPTS_VALUE ? AV_NOPTS_VALUE :
- av_rescale(outlink->sample_rate, insamplesref->pts, inlink ->sample_rate);
- avfilter_filter_samples(outlink, outsamplesref);
+ ff_filter_samples(outlink, outsamplesref);
+ aresample->req_fullfilled= 1;
avfilter_unref_buffer(insamplesref);
}
+static int request_frame(AVFilterLink *outlink)
+{
+ AVFilterContext *ctx = outlink->src;
+ AResampleContext *aresample = ctx->priv;
+ AVFilterLink *const inlink = outlink->src->inputs[0];
+ int ret;
+
+ aresample->req_fullfilled = 0;
+ do{
+ ret = ff_request_frame(ctx->inputs[0]);
+ }while(!aresample->req_fullfilled && ret>=0);
+
+ if (ret == AVERROR_EOF) {
+ AVFilterBufferRef *outsamplesref;
+ int n_out = 4096;
+
+ outsamplesref = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n_out);
+ if (!outsamplesref)
+ return AVERROR(ENOMEM);
+ n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out, 0, 0);
+ if (n_out <= 0) {
+ avfilter_unref_buffer(outsamplesref);
+ return (n_out == 0) ? AVERROR_EOF : n_out;
+ }
+
+ outsamplesref->audio->sample_rate = outlink->sample_rate;
+ outsamplesref->audio->nb_samples = n_out;
+#if 0
+ outsamplesref->pts = aresample->next_pts;
+ if(aresample->next_pts != AV_NOPTS_VALUE)
+ aresample->next_pts += av_rescale_q(n_out, (AVRational){1 ,outlink->sample_rate}, outlink->time_base);
+#else
+ outsamplesref->pts = (swr_next_pts(aresample->swr, INT64_MIN) + inlink->sample_rate/2) / inlink->sample_rate;
+#endif
+
+ ff_filter_samples(outlink, outsamplesref);
+ return 0;
+ }
+ return ret;
+}
+
AVFilter avfilter_af_aresample = {
.name = "aresample",
.description = NULL_IF_CONFIG_SMALL("Resample audio data."),
.init = init,
.uninit = uninit,
+ .query_formats = query_formats,
.priv_size = sizeof(AResampleContext),
.inputs = (const AVFilterPad[]) {{ .name = "default",
{ .name = NULL}},
.outputs = (const AVFilterPad[]) {{ .name = "default",
.config_props = config_output,
+ .request_frame = request_frame,
.type = AVMEDIA_TYPE_AUDIO, },
{ .name = NULL}},
};
projects / ffmpeg / blobdiff