#include "libavcodec/avfft.h"
#include "avfilter.h"
+#include "filters.h"
#include "internal.h"
#include "audio.h"
#define TIME_DOMAIN 0
#define FREQUENCY_DOMAIN 1
+#define HRIR_STEREO 0
+#define HRIR_MULTI 1
+
typedef struct HeadphoneContext {
const AVClass *class;
int have_hrirs;
int eof_hrirs;
- int64_t pts;
int ir_len;
int buffer_length;
int n_fft;
int size;
+ int hrir_fmt;
int *delay[2];
float *data_ir[2];
char buf[8];
p = NULL;
- if (parse_channel_name(s, s->nb_inputs - 1, &arg, &out_ch_id, buf)) {
+ if (parse_channel_name(s, s->nb_irs, &arg, &out_ch_id, buf)) {
av_log(ctx, AV_LOG_WARNING, "Failed to parse \'%s\' as channel name.\n", buf);
continue;
}
- s->mapping[s->nb_inputs - 1] = out_ch_id;
- s->nb_inputs++;
+ s->mapping[s->nb_irs] = out_ch_id;
+ s->nb_irs++;
}
- s->nb_irs = s->nb_inputs - 1;
+
+ if (s->hrir_fmt == HRIR_MULTI)
+ s->nb_inputs = 2;
+ else
+ s->nb_inputs = s->nb_irs + 1;
av_free(args);
}
return 0;
}
-static int read_ir(AVFilterLink *inlink, AVFrame *frame)
+static int read_ir(AVFilterLink *inlink, int input_number, AVFrame *frame)
{
AVFilterContext *ctx = inlink->dst;
HeadphoneContext *s = ctx->priv;
- int ir_len, max_ir_len, input_number, ret;
-
- for (input_number = 0; input_number < s->nb_inputs; input_number++)
- if (inlink == ctx->inputs[input_number])
- break;
+ int ir_len, max_ir_len, ret;
ret = av_audio_fifo_write(s->in[input_number].fifo, (void **)frame->extended_data,
frame->nb_samples);
return 0;
}
-static int headphone_frame(HeadphoneContext *s, AVFilterLink *outlink)
+static int headphone_frame(HeadphoneContext *s, AVFrame *in, AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
- AVFrame *in = s->in[0].frame;
int n_clippings[2] = { 0 };
ThreadData td;
AVFrame *out;
- av_audio_fifo_read(s->in[0].fifo, (void **)in->extended_data, s->size);
-
out = ff_get_audio_buffer(outlink, in->nb_samples);
- if (!out)
+ if (!out) {
+ av_frame_free(&in);
return AVERROR(ENOMEM);
- out->pts = s->pts;
- if (s->pts != AV_NOPTS_VALUE)
- s->pts += av_rescale_q(out->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
+ }
+ out->pts = in->pts;
td.in = in; td.out = out; td.write = s->write;
td.delay = s->delay; td.ir = s->data_ir; td.n_clippings = n_clippings;
n_clippings[0] + n_clippings[1], out->nb_samples * 2);
}
+ av_frame_free(&in);
return ff_filter_frame(outlink, out);
}
float *data_ir_r = NULL;
int offset = 0, ret = 0;
int n_fft;
- int i, j;
+ int i, j, k;
s->buffer_length = 1 << (32 - ff_clz(s->ir_len));
- s->n_fft = n_fft = 1 << (32 - ff_clz(s->ir_len + inlink->sample_rate));
+ s->n_fft = n_fft = 1 << (32 - ff_clz(s->ir_len + s->size));
if (s->type == FREQUENCY_DOMAIN) {
fft_in_l = av_calloc(n_fft, sizeof(*fft_in_l));
s->data_ir[0] = av_calloc(FFALIGN(s->ir_len, 16), sizeof(float) * s->nb_irs);
s->data_ir[1] = av_calloc(FFALIGN(s->ir_len, 16), sizeof(float) * s->nb_irs);
- s->delay[0] = av_malloc_array(s->nb_irs, sizeof(float));
- s->delay[1] = av_malloc_array(s->nb_irs, sizeof(float));
+ s->delay[0] = av_calloc(s->nb_irs, sizeof(float));
+ s->delay[1] = av_calloc(s->nb_irs, sizeof(float));
if (s->type == TIME_DOMAIN) {
s->ringbuffer[0] = av_calloc(s->buffer_length, sizeof(float) * nb_input_channels);
} else {
s->ringbuffer[0] = av_calloc(s->buffer_length, sizeof(float));
s->ringbuffer[1] = av_calloc(s->buffer_length, sizeof(float));
- s->temp_fft[0] = av_malloc_array(s->n_fft, sizeof(FFTComplex));
- s->temp_fft[1] = av_malloc_array(s->n_fft, sizeof(FFTComplex));
+ s->temp_fft[0] = av_calloc(s->n_fft, sizeof(FFTComplex));
+ s->temp_fft[1] = av_calloc(s->n_fft, sizeof(FFTComplex));
if (!s->temp_fft[0] || !s->temp_fft[1]) {
ret = AVERROR(ENOMEM);
goto fail;
goto fail;
}
- s->in[0].frame = ff_get_audio_buffer(ctx->inputs[0], s->size);
- if (!s->in[0].frame) {
- ret = AVERROR(ENOMEM);
- goto fail;
- }
- for (i = 0; i < s->nb_irs; i++) {
+ for (i = 0; i < s->nb_inputs - 1; i++) {
s->in[i + 1].frame = ff_get_audio_buffer(ctx->inputs[i + 1], s->ir_len);
if (!s->in[i + 1].frame) {
ret = AVERROR(ENOMEM);
goto fail;
}
} else {
- data_hrtf_l = av_malloc_array(n_fft, sizeof(*data_hrtf_l) * nb_irs);
- data_hrtf_r = av_malloc_array(n_fft, sizeof(*data_hrtf_r) * nb_irs);
+ data_hrtf_l = av_calloc(n_fft, sizeof(*data_hrtf_l) * nb_irs);
+ data_hrtf_r = av_calloc(n_fft, sizeof(*data_hrtf_r) * nb_irs);
if (!data_hrtf_r || !data_hrtf_l) {
ret = AVERROR(ENOMEM);
goto fail;
}
}
- for (i = 0; i < s->nb_irs; i++) {
+ for (i = 0; i < s->nb_inputs - 1; i++) {
int len = s->in[i + 1].ir_len;
int delay_l = s->in[i + 1].delay_l;
int delay_r = s->in[i + 1].delay_r;
- int idx = -1;
float *ptr;
- for (j = 0; j < inlink->channels; j++) {
- if (s->mapping[i] < 0) {
- continue;
- }
-
- if ((av_channel_layout_extract_channel(inlink->channel_layout, j)) == (1LL << s->mapping[i])) {
- idx = j;
- break;
- }
- }
- if (idx == -1)
- continue;
-
av_audio_fifo_read(s->in[i + 1].fifo, (void **)s->in[i + 1].frame->extended_data, len);
ptr = (float *)s->in[i + 1].frame->extended_data[0];
- if (s->type == TIME_DOMAIN) {
- offset = idx * FFALIGN(len, 16);
- for (j = 0; j < len; j++) {
- data_ir_l[offset + j] = ptr[len * 2 - j * 2 - 2] * gain_lin;
- data_ir_r[offset + j] = ptr[len * 2 - j * 2 - 1] * gain_lin;
- }
- } else {
- memset(fft_in_l, 0, n_fft * sizeof(*fft_in_l));
- memset(fft_in_r, 0, n_fft * sizeof(*fft_in_r));
+ if (s->hrir_fmt == HRIR_STEREO) {
+ int idx = -1;
- offset = idx * n_fft;
- for (j = 0; j < len; j++) {
- fft_in_l[delay_l + j].re = ptr[j * 2 ] * gain_lin;
- fft_in_r[delay_r + j].re = ptr[j * 2 + 1] * gain_lin;
+ for (j = 0; j < inlink->channels; j++) {
+ if (s->mapping[i] < 0) {
+ continue;
+ }
+
+ if ((av_channel_layout_extract_channel(inlink->channel_layout, j)) == (1LL << s->mapping[i])) {
+ idx = i;
+ break;
+ }
}
- av_fft_permute(s->fft[0], fft_in_l);
- av_fft_calc(s->fft[0], fft_in_l);
- memcpy(data_hrtf_l + offset, fft_in_l, n_fft * sizeof(*fft_in_l));
- av_fft_permute(s->fft[0], fft_in_r);
- av_fft_calc(s->fft[0], fft_in_r);
- memcpy(data_hrtf_r + offset, fft_in_r, n_fft * sizeof(*fft_in_r));
+ if (idx == -1)
+ continue;
+ if (s->type == TIME_DOMAIN) {
+ offset = idx * FFALIGN(len, 16);
+ for (j = 0; j < len; j++) {
+ data_ir_l[offset + j] = ptr[len * 2 - j * 2 - 2] * gain_lin;
+ data_ir_r[offset + j] = ptr[len * 2 - j * 2 - 1] * gain_lin;
+ }
+ } else {
+ memset(fft_in_l, 0, n_fft * sizeof(*fft_in_l));
+ memset(fft_in_r, 0, n_fft * sizeof(*fft_in_r));
+
+ offset = idx * n_fft;
+ for (j = 0; j < len; j++) {
+ fft_in_l[delay_l + j].re = ptr[j * 2 ] * gain_lin;
+ fft_in_r[delay_r + j].re = ptr[j * 2 + 1] * gain_lin;
+ }
+
+ av_fft_permute(s->fft[0], fft_in_l);
+ av_fft_calc(s->fft[0], fft_in_l);
+ memcpy(data_hrtf_l + offset, fft_in_l, n_fft * sizeof(*fft_in_l));
+ av_fft_permute(s->fft[0], fft_in_r);
+ av_fft_calc(s->fft[0], fft_in_r);
+ memcpy(data_hrtf_r + offset, fft_in_r, n_fft * sizeof(*fft_in_r));
+ }
+ } else {
+ int I, N = ctx->inputs[1]->channels;
+
+ for (k = 0; k < N / 2; k++) {
+ int idx = -1;
+
+ for (j = 0; j < inlink->channels; j++) {
+ if (s->mapping[k] < 0) {
+ continue;
+ }
+
+ if ((av_channel_layout_extract_channel(inlink->channel_layout, j)) == (1LL << s->mapping[k])) {
+ idx = k;
+ break;
+ }
+ }
+ if (idx == -1)
+ continue;
+
+ I = idx * 2;
+ if (s->type == TIME_DOMAIN) {
+ offset = idx * FFALIGN(len, 16);
+ for (j = 0; j < len; j++) {
+ data_ir_l[offset + j] = ptr[len * N - j * N - N + I ] * gain_lin;
+ data_ir_r[offset + j] = ptr[len * N - j * N - N + I + 1] * gain_lin;
+ }
+ } else {
+ memset(fft_in_l, 0, n_fft * sizeof(*fft_in_l));
+ memset(fft_in_r, 0, n_fft * sizeof(*fft_in_r));
+
+ offset = idx * n_fft;
+ for (j = 0; j < len; j++) {
+ fft_in_l[delay_l + j].re = ptr[j * N + I ] * gain_lin;
+ fft_in_r[delay_r + j].re = ptr[j * N + I + 1] * gain_lin;
+ }
+
+ av_fft_permute(s->fft[0], fft_in_l);
+ av_fft_calc(s->fft[0], fft_in_l);
+ memcpy(data_hrtf_l + offset, fft_in_l, n_fft * sizeof(*fft_in_l));
+ av_fft_permute(s->fft[0], fft_in_r);
+ av_fft_calc(s->fft[0], fft_in_r);
+ memcpy(data_hrtf_r + offset, fft_in_r, n_fft * sizeof(*fft_in_r));
+ }
+ }
}
}
memcpy(s->data_ir[0], data_ir_l, sizeof(float) * nb_irs * FFALIGN(ir_len, 16));
memcpy(s->data_ir[1], data_ir_r, sizeof(float) * nb_irs * FFALIGN(ir_len, 16));
} else {
- s->data_hrtf[0] = av_malloc_array(n_fft * s->nb_irs, sizeof(FFTComplex));
- s->data_hrtf[1] = av_malloc_array(n_fft * s->nb_irs, sizeof(FFTComplex));
+ s->data_hrtf[0] = av_calloc(n_fft * s->nb_irs, sizeof(FFTComplex));
+ s->data_hrtf[1] = av_calloc(n_fft * s->nb_irs, sizeof(FFTComplex));
if (!s->data_hrtf[0] || !s->data_hrtf[1]) {
ret = AVERROR(ENOMEM);
goto fail;
return ret;
}
-static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+static int activate(AVFilterContext *ctx)
{
- AVFilterContext *ctx = inlink->dst;
HeadphoneContext *s = ctx->priv;
+ AVFilterLink *inlink = ctx->inputs[0];
AVFilterLink *outlink = ctx->outputs[0];
- int ret = 0;
+ AVFrame *in = NULL;
+ int i, ret;
- ret = av_audio_fifo_write(s->in[0].fifo, (void **)in->extended_data,
- in->nb_samples);
- if (s->pts == AV_NOPTS_VALUE)
- s->pts = in->pts;
+ FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx);
+ if (!s->eof_hrirs) {
+ for (i = 1; i < s->nb_inputs; i++) {
+ AVFrame *ir = NULL;
+ int64_t pts;
+ int status;
- av_frame_free(&in);
+ if (s->in[i].eof)
+ continue;
- if (ret < 0)
- return ret;
+ if ((ret = ff_inlink_consume_frame(ctx->inputs[i], &ir)) > 0) {
+ ret = read_ir(ctx->inputs[i], i, ir);
+ if (ret < 0)
+ return ret;
+ }
+ if (ret < 0)
+ return ret;
+
+ if (!s->in[i].eof) {
+ if (ff_inlink_acknowledge_status(ctx->inputs[i], &status, &pts)) {
+ if (status == AVERROR_EOF) {
+ s->in[i].eof = 1;
+ }
+ }
+ }
+ }
+
+ for (i = 1; i < s->nb_inputs; i++) {
+ if (!s->in[i].eof)
+ break;
+ }
+
+ if (i != s->nb_inputs) {
+ if (ff_outlink_frame_wanted(ctx->outputs[0])) {
+ for (i = 1; i < s->nb_inputs; i++) {
+ if (!s->in[i].eof)
+ ff_inlink_request_frame(ctx->inputs[i]);
+ }
+ }
+ return 0;
+ } else {
+ s->eof_hrirs = 1;
+ }
+ }
if (!s->have_hrirs && s->eof_hrirs) {
ret = convert_coeffs(ctx, inlink);
return ret;
}
- if (s->have_hrirs) {
- while (av_audio_fifo_size(s->in[0].fifo) >= s->size) {
- ret = headphone_frame(s, outlink);
- if (ret < 0)
- return ret;
- }
+ if ((ret = ff_inlink_consume_samples(ctx->inputs[0], s->size, s->size, &in)) > 0) {
+ ret = headphone_frame(s, in, outlink);
+ if (ret < 0)
+ return ret;
}
+ if (ret < 0)
+ return ret;
+
+ FF_FILTER_FORWARD_STATUS(ctx->inputs[0], ctx->outputs[0]);
+ if (ff_outlink_frame_wanted(ctx->outputs[0]))
+ ff_inlink_request_frame(ctx->inputs[0]);
+
return 0;
}
struct HeadphoneContext *s = ctx->priv;
AVFilterFormats *formats = NULL;
AVFilterChannelLayouts *layouts = NULL;
+ AVFilterChannelLayouts *stereo_layout = NULL;
+ AVFilterChannelLayouts *hrir_layouts = NULL;
int ret, i;
ret = ff_add_format(&formats, AV_SAMPLE_FMT_FLT);
if (ret)
return ret;
- layouts = NULL;
- ret = ff_add_channel_layout(&layouts, AV_CH_LAYOUT_STEREO);
+ ret = ff_add_channel_layout(&stereo_layout, AV_CH_LAYOUT_STEREO);
if (ret)
return ret;
- for (i = 1; i < s->nb_inputs; i++) {
- ret = ff_channel_layouts_ref(layouts, &ctx->inputs[i]->out_channel_layouts);
+ if (s->hrir_fmt == HRIR_MULTI) {
+ hrir_layouts = ff_all_channel_counts();
+ if (!hrir_layouts)
+ ret = AVERROR(ENOMEM);
+ ret = ff_channel_layouts_ref(hrir_layouts, &ctx->inputs[1]->out_channel_layouts);
if (ret)
return ret;
+ } else {
+ for (i = 1; i < s->nb_inputs; i++) {
+ ret = ff_channel_layouts_ref(stereo_layout, &ctx->inputs[i]->out_channel_layouts);
+ if (ret)
+ return ret;
+ }
}
- ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->in_channel_layouts);
+ ret = ff_channel_layouts_ref(stereo_layout, &ctx->outputs[0]->in_channel_layouts);
if (ret)
return ret;
AVFilterContext *ctx = inlink->dst;
HeadphoneContext *s = ctx->priv;
- if (s->type == FREQUENCY_DOMAIN) {
- inlink->partial_buf_size =
- inlink->min_samples =
- inlink->max_samples = inlink->sample_rate;
- }
-
if (s->nb_irs < inlink->channels) {
- av_log(ctx, AV_LOG_ERROR, "Number of inputs must be >= %d.\n", inlink->channels + 1);
+ av_log(ctx, AV_LOG_ERROR, "Number of HRIRs must be >= %d.\n", inlink->channels);
return AVERROR(EINVAL);
}
.name = "in0",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_input,
- .filter_frame = filter_frame,
};
if ((ret = ff_insert_inpad(ctx, 0, &pad)) < 0)
return ret;
AVFilterPad pad = {
.name = name,
.type = AVMEDIA_TYPE_AUDIO,
- .filter_frame = read_ir,
};
if (!name)
return AVERROR(ENOMEM);
s->fdsp = avpriv_float_dsp_alloc(0);
if (!s->fdsp)
return AVERROR(ENOMEM);
- s->pts = AV_NOPTS_VALUE;
return 0;
}
AVFilterLink *inlink = ctx->inputs[0];
int i;
- if (s->type == TIME_DOMAIN)
- s->size = 1024;
- else
- s->size = inlink->sample_rate;
+ if (s->hrir_fmt == HRIR_MULTI) {
+ AVFilterLink *hrir_link = ctx->inputs[1];
+
+ if (hrir_link->channels < inlink->channels * 2) {
+ av_log(ctx, AV_LOG_ERROR, "Number of channels in HRIR stream must be >= %d.\n", inlink->channels * 2);
+ return AVERROR(EINVAL);
+ }
+ }
for (i = 0; i < s->nb_inputs; i++) {
s->in[i].fifo = av_audio_fifo_alloc(ctx->inputs[i]->format, ctx->inputs[i]->channels, 1024);
return 0;
}
-static int request_frame(AVFilterLink *outlink)
-{
- AVFilterContext *ctx = outlink->src;
- HeadphoneContext *s = ctx->priv;
- int i, ret;
-
- for (i = 1; !s->eof_hrirs && i < s->nb_inputs; i++) {
- if (!s->in[i].eof) {
- ret = ff_request_frame(ctx->inputs[i]);
- if (ret == AVERROR_EOF) {
- s->in[i].eof = 1;
- ret = 0;
- }
- return ret;
- } else {
- if (i == s->nb_inputs - 1)
- s->eof_hrirs = 1;
- }
- }
- return ff_request_frame(ctx->inputs[0]);
-}
-
static av_cold void uninit(AVFilterContext *ctx)
{
HeadphoneContext *s = ctx->priv;
{ "type", "set processing", OFFSET(type), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, .flags = FLAGS, "type" },
{ "time", "time domain", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, .flags = FLAGS, "type" },
{ "freq", "frequency domain", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, .flags = FLAGS, "type" },
+ { "size", "set frame size", OFFSET(size), AV_OPT_TYPE_INT, {.i64=1024},1024,96000, .flags = FLAGS },
+ { "hrir", "set hrir format", OFFSET(hrir_fmt), AV_OPT_TYPE_INT, {.i64=HRIR_STEREO}, 0, 1, .flags = FLAGS, "hrir" },
+ { "stereo", "hrir files have exactly 2 channels", 0, AV_OPT_TYPE_CONST, {.i64=HRIR_STEREO}, 0, 0, .flags = FLAGS, "hrir" },
+ { "multich", "single multichannel hrir file", 0, AV_OPT_TYPE_CONST, {.i64=HRIR_MULTI}, 0, 0, .flags = FLAGS, "hrir" },
{ NULL }
};
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output,
- .request_frame = request_frame,
},
{ NULL }
};
.init = init,
.uninit = uninit,
.query_formats = query_formats,
+ .activate = activate,
.inputs = NULL,
.outputs = outputs,
.flags = AVFILTER_FLAG_SLICE_THREADS | AVFILTER_FLAG_DYNAMIC_INPUTS,
projects / ffmpeg / blobdiff