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#include <math.h> |
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|
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#include "libavutil/avstring.h" |
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#include "libavutil/channel_layout.h" |
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#include "libavutil/float_dsp.h" |
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#include "libavutil/intmath.h" |
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#include "libavutil/opt.h" |
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#include "libavutil/tx.h" |
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|
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#include "avfilter.h" |
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#include "filters.h" |
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#include "formats.h" |
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#include "internal.h" |
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#include "audio.h" |
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|
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#define TIME_DOMAIN 0 |
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#define FREQUENCY_DOMAIN 1 |
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|
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#define HRIR_STEREO 0 |
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#define HRIR_MULTI 1 |
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|
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typedef struct HeadphoneContext { |
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const AVClass *class; |
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|
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char *map; |
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int type; |
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|
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int lfe_channel; |
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|
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int have_hrirs; |
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int eof_hrirs; |
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|
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int ir_len; |
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int air_len; |
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|
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int nb_hrir_inputs; |
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|
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int nb_irs; |
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|
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float gain; |
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float lfe_gain, gain_lfe; |
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|
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float *ringbuffer[2]; |
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int write[2]; |
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|
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int buffer_length; |
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int n_fft; |
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int size; |
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int hrir_fmt; |
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|
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float *data_ir[2]; |
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float *temp_src[2]; |
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AVComplexFloat *out_fft[2]; |
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AVComplexFloat *in_fft[2]; |
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AVComplexFloat *temp_afft[2]; |
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|
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AVTXContext *fft[2], *ifft[2]; |
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av_tx_fn tx_fn[2], itx_fn[2]; |
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AVComplexFloat *data_hrtf[2]; |
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|
|
float (*scalarproduct_float)(const float *v1, const float *v2, int len); |
|
struct hrir_inputs { |
|
int ir_len; |
|
int eof; |
|
} hrir_in[64]; |
|
AVChannelLayout map_channel_layout; |
|
enum AVChannel mapping[64]; |
|
uint8_t hrir_map[64]; |
|
} HeadphoneContext; |
|
|
|
static int parse_channel_name(const char *arg, enum AVChannel *rchannel) |
|
{ |
|
int channel = av_channel_from_string(arg); |
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|
|
if (channel < 0 || channel >= 64) |
|
return AVERROR(EINVAL); |
|
*rchannel = channel; |
|
return 0; |
|
} |
|
|
|
static void parse_map(AVFilterContext *ctx) |
|
{ |
|
HeadphoneContext *s = ctx->priv; |
|
char *arg, *tokenizer, *p; |
|
uint64_t used_channels = 0; |
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|
|
p = s->map; |
|
while ((arg = av_strtok(p, "|", &tokenizer))) { |
|
enum AVChannel out_channel; |
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|
|
p = NULL; |
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if (parse_channel_name(arg, &out_channel)) { |
|
av_log(ctx, AV_LOG_WARNING, "Failed to parse \'%s\' as channel name.\n", arg); |
|
continue; |
|
} |
|
if (used_channels & (1ULL << out_channel)) { |
|
av_log(ctx, AV_LOG_WARNING, "Ignoring duplicate channel '%s'.\n", arg); |
|
continue; |
|
} |
|
used_channels |= (1ULL << out_channel); |
|
s->mapping[s->nb_irs] = out_channel; |
|
s->nb_irs++; |
|
} |
|
av_channel_layout_from_mask(&s->map_channel_layout, used_channels); |
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|
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if (s->hrir_fmt == HRIR_MULTI) |
|
s->nb_hrir_inputs = 1; |
|
else |
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s->nb_hrir_inputs = s->nb_irs; |
|
} |
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|
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typedef struct ThreadData { |
|
AVFrame *in, *out; |
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int *write; |
|
float **ir; |
|
int *n_clippings; |
|
float **ringbuffer; |
|
float **temp_src; |
|
AVComplexFloat **out_fft; |
|
AVComplexFloat **in_fft; |
|
AVComplexFloat **temp_afft; |
|
} ThreadData; |
|
|
|
static int headphone_convolute(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs) |
|
{ |
|
HeadphoneContext *s = ctx->priv; |
|
ThreadData *td = arg; |
|
AVFrame *in = td->in, *out = td->out; |
|
int offset = jobnr; |
|
int *write = &td->write[jobnr]; |
|
const float *const ir = td->ir[jobnr]; |
|
int *n_clippings = &td->n_clippings[jobnr]; |
|
float *ringbuffer = td->ringbuffer[jobnr]; |
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float *temp_src = td->temp_src[jobnr]; |
|
const int ir_len = s->ir_len; |
|
const int air_len = s->air_len; |
|
const float *src = (const float *)in->data[0]; |
|
float *dst = (float *)out->data[0]; |
|
const int in_channels = in->ch_layout.nb_channels; |
|
const int buffer_length = s->buffer_length; |
|
const uint32_t modulo = (uint32_t)buffer_length - 1; |
|
float *buffer[64]; |
|
int wr = *write; |
|
int read; |
|
int i, l; |
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|
|
dst += offset; |
|
for (l = 0; l < in_channels; l++) { |
|
buffer[l] = ringbuffer + l * buffer_length; |
|
} |
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for (i = 0; i < in->nb_samples; i++) { |
|
const float *cur_ir = ir; |
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|
|
*dst = 0; |
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for (l = 0; l < in_channels; l++) { |
|
*(buffer[l] + wr) = src[l]; |
|
} |
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for (l = 0; l < in_channels; cur_ir += air_len, l++) { |
|
const float *const bptr = buffer[l]; |
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|
|
if (l == s->lfe_channel) { |
|
*dst += *(buffer[s->lfe_channel] + wr) * s->gain_lfe; |
|
continue; |
|
} |
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|
|
read = (wr - (ir_len - 1)) & modulo; |
|
|
|
if (read + ir_len < buffer_length) { |
|
memcpy(temp_src, bptr + read, ir_len * sizeof(*temp_src)); |
|
} else { |
|
int len = FFMIN(air_len - (read % ir_len), buffer_length - read); |
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|
|
memcpy(temp_src, bptr + read, len * sizeof(*temp_src)); |
|
memcpy(temp_src + len, bptr, (air_len - len) * sizeof(*temp_src)); |
|
} |
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|
|
dst[0] += s->scalarproduct_float(cur_ir, temp_src, FFALIGN(ir_len, 32)); |
|
} |
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|
|
if (fabsf(dst[0]) > 1) |
|
n_clippings[0]++; |
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|
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dst += 2; |
|
src += in_channels; |
|
wr = (wr + 1) & modulo; |
|
} |
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|
|
*write = wr; |
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|
|
return 0; |
|
} |
|
|
|
static int headphone_fast_convolute(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs) |
|
{ |
|
HeadphoneContext *s = ctx->priv; |
|
ThreadData *td = arg; |
|
AVFrame *in = td->in, *out = td->out; |
|
int offset = jobnr; |
|
int *write = &td->write[jobnr]; |
|
AVComplexFloat *hrtf = s->data_hrtf[jobnr]; |
|
int *n_clippings = &td->n_clippings[jobnr]; |
|
float *ringbuffer = td->ringbuffer[jobnr]; |
|
const int ir_len = s->ir_len; |
|
const float *src = (const float *)in->data[0]; |
|
float *dst = (float *)out->data[0]; |
|
const int in_channels = in->ch_layout.nb_channels; |
|
const int buffer_length = s->buffer_length; |
|
const uint32_t modulo = (uint32_t)buffer_length - 1; |
|
AVComplexFloat *fft_out = s->out_fft[jobnr]; |
|
AVComplexFloat *fft_in = s->in_fft[jobnr]; |
|
AVComplexFloat *fft_acc = s->temp_afft[jobnr]; |
|
AVTXContext *ifft = s->ifft[jobnr]; |
|
AVTXContext *fft = s->fft[jobnr]; |
|
av_tx_fn tx_fn = s->tx_fn[jobnr]; |
|
av_tx_fn itx_fn = s->itx_fn[jobnr]; |
|
const int n_fft = s->n_fft; |
|
const float fft_scale = 1.0f / s->n_fft; |
|
AVComplexFloat *hrtf_offset; |
|
int wr = *write; |
|
int n_read; |
|
int i, j; |
|
|
|
dst += offset; |
|
|
|
n_read = FFMIN(ir_len, in->nb_samples); |
|
for (j = 0; j < n_read; j++) { |
|
dst[2 * j] = ringbuffer[wr]; |
|
ringbuffer[wr] = 0.0; |
|
wr = (wr + 1) & modulo; |
|
} |
|
|
|
for (j = n_read; j < in->nb_samples; j++) { |
|
dst[2 * j] = 0; |
|
} |
|
|
|
memset(fft_acc, 0, sizeof(AVComplexFloat) * n_fft); |
|
|
|
for (i = 0; i < in_channels; i++) { |
|
if (i == s->lfe_channel) { |
|
for (j = 0; j < in->nb_samples; j++) { |
|
dst[2 * j] += src[i + j * in_channels] * s->gain_lfe; |
|
} |
|
continue; |
|
} |
|
|
|
offset = i * n_fft; |
|
hrtf_offset = hrtf + s->hrir_map[i] * n_fft; |
|
|
|
memset(fft_in, 0, sizeof(AVComplexFloat) * n_fft); |
|
|
|
for (j = 0; j < in->nb_samples; j++) { |
|
fft_in[j].re = src[j * in_channels + i]; |
|
} |
|
|
|
tx_fn(fft, fft_out, fft_in, sizeof(*fft_in)); |
|
|
|
for (j = 0; j < n_fft; j++) { |
|
const AVComplexFloat *hcomplex = hrtf_offset + j; |
|
const float re = fft_out[j].re; |
|
const float im = fft_out[j].im; |
|
|
|
fft_acc[j].re += re * hcomplex->re - im * hcomplex->im; |
|
fft_acc[j].im += re * hcomplex->im + im * hcomplex->re; |
|
} |
|
} |
|
|
|
itx_fn(ifft, fft_out, fft_acc, sizeof(*fft_acc)); |
|
|
|
for (j = 0; j < in->nb_samples; j++) { |
|
dst[2 * j] += fft_out[j].re * fft_scale; |
|
if (fabsf(dst[2 * j]) > 1) |
|
n_clippings[0]++; |
|
} |
|
|
|
for (j = 0; j < ir_len - 1; j++) { |
|
int write_pos = (wr + j) & modulo; |
|
|
|
*(ringbuffer + write_pos) += fft_out[in->nb_samples + j].re * fft_scale; |
|
} |
|
|
|
*write = wr; |
|
|
|
return 0; |
|
} |
|
|
|
static int check_ir(AVFilterLink *inlink, int input_number) |
|
{ |
|
AVFilterContext *ctx = inlink->dst; |
|
HeadphoneContext *s = ctx->priv; |
|
int ir_len, max_ir_len; |
|
|
|
ir_len = ff_inlink_queued_samples(inlink); |
|
max_ir_len = 65536; |
|
if (ir_len > max_ir_len) { |
|
av_log(ctx, AV_LOG_ERROR, "Too big length of IRs: %d > %d.\n", ir_len, max_ir_len); |
|
return AVERROR(EINVAL); |
|
} |
|
s->hrir_in[input_number].ir_len = ir_len; |
|
s->ir_len = FFMAX(ir_len, s->ir_len); |
|
|
|
if (ff_inlink_check_available_samples(inlink, ir_len + 1) == 1) { |
|
s->hrir_in[input_number].eof = 1; |
|
return 1; |
|
} |
|
|
|
if (!s->hrir_in[input_number].eof) { |
|
ff_inlink_request_frame(inlink); |
|
return 0; |
|
} |
|
|
|
return 0; |
|
} |
|
|
|
static int headphone_frame(HeadphoneContext *s, AVFrame *in, AVFilterLink *outlink) |
|
{ |
|
AVFilterContext *ctx = outlink->src; |
|
int n_clippings[2] = { 0 }; |
|
ThreadData td; |
|
AVFrame *out; |
|
|
|
out = ff_get_audio_buffer(outlink, in->nb_samples); |
|
if (!out) { |
|
av_frame_free(&in); |
|
return AVERROR(ENOMEM); |
|
} |
|
out->pts = in->pts; |
|
|
|
td.in = in; td.out = out; td.write = s->write; |
|
td.ir = s->data_ir; td.n_clippings = n_clippings; |
|
td.ringbuffer = s->ringbuffer; td.temp_src = s->temp_src; |
|
td.out_fft = s->out_fft; |
|
td.in_fft = s->in_fft; |
|
td.temp_afft = s->temp_afft; |
|
|
|
if (s->type == TIME_DOMAIN) { |
|
ff_filter_execute(ctx, headphone_convolute, &td, NULL, 2); |
|
} else { |
|
ff_filter_execute(ctx, headphone_fast_convolute, &td, NULL, 2); |
|
} |
|
|
|
if (n_clippings[0] + n_clippings[1] > 0) { |
|
av_log(ctx, AV_LOG_WARNING, "%d of %d samples clipped. Please reduce gain.\n", |
|
n_clippings[0] + n_clippings[1], out->nb_samples * 2); |
|
} |
|
|
|
av_frame_free(&in); |
|
return ff_filter_frame(outlink, out); |
|
} |
|
|
|
static int convert_coeffs(AVFilterContext *ctx, AVFilterLink *inlink) |
|
{ |
|
struct HeadphoneContext *s = ctx->priv; |
|
const int ir_len = s->ir_len; |
|
int nb_input_channels = ctx->inputs[0]->ch_layout.nb_channels; |
|
const int nb_hrir_channels = s->nb_hrir_inputs == 1 ? ctx->inputs[1]->ch_layout.nb_channels : s->nb_hrir_inputs * 2; |
|
float gain_lin = expf((s->gain - 3 * nb_input_channels) / 20 * M_LN10); |
|
AVFrame *frame; |
|
int ret = 0; |
|
int n_fft; |
|
int i, j, k; |
|
|
|
s->air_len = 1 << (32 - ff_clz(ir_len)); |
|
if (s->type == TIME_DOMAIN) { |
|
s->air_len = FFALIGN(s->air_len, 32); |
|
} |
|
s->buffer_length = 1 << (32 - ff_clz(s->air_len)); |
|
s->n_fft = n_fft = 1 << (32 - ff_clz(ir_len + s->size)); |
|
|
|
if (s->type == FREQUENCY_DOMAIN) { |
|
float scale = 1.f; |
|
|
|
ret = av_tx_init(&s->fft[0], &s->tx_fn[0], AV_TX_FLOAT_FFT, 0, s->n_fft, &scale, 0); |
|
if (ret < 0) |
|
goto fail; |
|
ret = av_tx_init(&s->fft[1], &s->tx_fn[1], AV_TX_FLOAT_FFT, 0, s->n_fft, &scale, 0); |
|
if (ret < 0) |
|
goto fail; |
|
ret = av_tx_init(&s->ifft[0], &s->itx_fn[0], AV_TX_FLOAT_FFT, 1, s->n_fft, &scale, 0); |
|
if (ret < 0) |
|
goto fail; |
|
ret = av_tx_init(&s->ifft[1], &s->itx_fn[1], AV_TX_FLOAT_FFT, 1, s->n_fft, &scale, 0); |
|
if (ret < 0) |
|
goto fail; |
|
|
|
if (!s->fft[0] || !s->fft[1] || !s->ifft[0] || !s->ifft[1]) { |
|
av_log(ctx, AV_LOG_ERROR, "Unable to create FFT contexts of size %d.\n", s->n_fft); |
|
ret = AVERROR(ENOMEM); |
|
goto fail; |
|
} |
|
} |
|
|
|
if (s->type == TIME_DOMAIN) { |
|
s->ringbuffer[0] = av_calloc(s->buffer_length, sizeof(float) * nb_input_channels); |
|
s->ringbuffer[1] = 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->out_fft[0] = av_calloc(s->n_fft, sizeof(AVComplexFloat)); |
|
s->out_fft[1] = av_calloc(s->n_fft, sizeof(AVComplexFloat)); |
|
s->in_fft[0] = av_calloc(s->n_fft, sizeof(AVComplexFloat)); |
|
s->in_fft[1] = av_calloc(s->n_fft, sizeof(AVComplexFloat)); |
|
s->temp_afft[0] = av_calloc(s->n_fft, sizeof(AVComplexFloat)); |
|
s->temp_afft[1] = av_calloc(s->n_fft, sizeof(AVComplexFloat)); |
|
if (!s->in_fft[0] || !s->in_fft[1] || |
|
!s->out_fft[0] || !s->out_fft[1] || |
|
!s->temp_afft[0] || !s->temp_afft[1]) { |
|
ret = AVERROR(ENOMEM); |
|
goto fail; |
|
} |
|
} |
|
|
|
if (!s->ringbuffer[0] || !s->ringbuffer[1]) { |
|
ret = AVERROR(ENOMEM); |
|
goto fail; |
|
} |
|
|
|
if (s->type == TIME_DOMAIN) { |
|
s->temp_src[0] = av_calloc(s->air_len, sizeof(float)); |
|
s->temp_src[1] = av_calloc(s->air_len, sizeof(float)); |
|
|
|
s->data_ir[0] = av_calloc(nb_hrir_channels * s->air_len, sizeof(*s->data_ir[0])); |
|
s->data_ir[1] = av_calloc(nb_hrir_channels * s->air_len, sizeof(*s->data_ir[1])); |
|
if (!s->data_ir[0] || !s->data_ir[1] || !s->temp_src[0] || !s->temp_src[1]) { |
|
ret = AVERROR(ENOMEM); |
|
goto fail; |
|
} |
|
} else { |
|
s->data_hrtf[0] = av_calloc(n_fft, sizeof(*s->data_hrtf[0]) * nb_hrir_channels); |
|
s->data_hrtf[1] = av_calloc(n_fft, sizeof(*s->data_hrtf[1]) * nb_hrir_channels); |
|
if (!s->data_hrtf[0] || !s->data_hrtf[1]) { |
|
ret = AVERROR(ENOMEM); |
|
goto fail; |
|
} |
|
} |
|
|
|
for (i = 0; i < s->nb_hrir_inputs; av_frame_free(&frame), i++) { |
|
int len = s->hrir_in[i].ir_len; |
|
float *ptr; |
|
|
|
ret = ff_inlink_consume_samples(ctx->inputs[i + 1], len, len, &frame); |
|
if (ret < 0) |
|
goto fail; |
|
ptr = (float *)frame->extended_data[0]; |
|
|
|
if (s->hrir_fmt == HRIR_STEREO) { |
|
int idx = av_channel_layout_index_from_channel(&s->map_channel_layout, |
|
s->mapping[i]); |
|
if (idx < 0) |
|
continue; |
|
|
|
s->hrir_map[i] = idx; |
|
if (s->type == TIME_DOMAIN) { |
|
float *data_ir_l = s->data_ir[0] + idx * s->air_len; |
|
float *data_ir_r = s->data_ir[1] + idx * s->air_len; |
|
|
|
for (j = 0; j < len; j++) { |
|
data_ir_l[j] = ptr[len * 2 - j * 2 - 2] * gain_lin; |
|
data_ir_r[j] = ptr[len * 2 - j * 2 - 1] * gain_lin; |
|
} |
|
} else { |
|
AVComplexFloat *fft_out_l = s->data_hrtf[0] + idx * n_fft; |
|
AVComplexFloat *fft_out_r = s->data_hrtf[1] + idx * n_fft; |
|
AVComplexFloat *fft_in_l = s->in_fft[0]; |
|
AVComplexFloat *fft_in_r = s->in_fft[1]; |
|
|
|
for (j = 0; j < len; j++) { |
|
fft_in_l[j].re = ptr[j * 2 ] * gain_lin; |
|
fft_in_r[j].re = ptr[j * 2 + 1] * gain_lin; |
|
} |
|
|
|
s->tx_fn[0](s->fft[0], fft_out_l, fft_in_l, sizeof(*fft_in_l)); |
|
s->tx_fn[0](s->fft[0], fft_out_r, fft_in_r, sizeof(*fft_in_r)); |
|
} |
|
} else { |
|
int I, N = ctx->inputs[1]->ch_layout.nb_channels; |
|
|
|
for (k = 0; k < N / 2; k++) { |
|
int idx = av_channel_layout_index_from_channel(&inlink->ch_layout, |
|
s->mapping[k]); |
|
if (idx < 0) |
|
continue; |
|
|
|
s->hrir_map[k] = idx; |
|
I = k * 2; |
|
if (s->type == TIME_DOMAIN) { |
|
float *data_ir_l = s->data_ir[0] + idx * s->air_len; |
|
float *data_ir_r = s->data_ir[1] + idx * s->air_len; |
|
|
|
for (j = 0; j < len; j++) { |
|
data_ir_l[j] = ptr[len * N - j * N - N + I ] * gain_lin; |
|
data_ir_r[j] = ptr[len * N - j * N - N + I + 1] * gain_lin; |
|
} |
|
} else { |
|
AVComplexFloat *fft_out_l = s->data_hrtf[0] + idx * n_fft; |
|
AVComplexFloat *fft_out_r = s->data_hrtf[1] + idx * n_fft; |
|
AVComplexFloat *fft_in_l = s->in_fft[0]; |
|
AVComplexFloat *fft_in_r = s->in_fft[1]; |
|
|
|
for (j = 0; j < len; j++) { |
|
fft_in_l[j].re = ptr[j * N + I ] * gain_lin; |
|
fft_in_r[j].re = ptr[j * N + I + 1] * gain_lin; |
|
} |
|
|
|
s->tx_fn[0](s->fft[0], fft_out_l, fft_in_l, sizeof(*fft_in_l)); |
|
s->tx_fn[0](s->fft[0], fft_out_r, fft_in_r, sizeof(*fft_in_r)); |
|
} |
|
} |
|
} |
|
} |
|
|
|
s->have_hrirs = 1; |
|
|
|
fail: |
|
return ret; |
|
} |
|
|
|
static int activate(AVFilterContext *ctx) |
|
{ |
|
HeadphoneContext *s = ctx->priv; |
|
AVFilterLink *inlink = ctx->inputs[0]; |
|
AVFilterLink *outlink = ctx->outputs[0]; |
|
AVFrame *in = NULL; |
|
int i, ret; |
|
|
|
FF_FILTER_FORWARD_STATUS_BACK_ALL(outlink, ctx); |
|
if (!s->eof_hrirs) { |
|
int eof = 1; |
|
for (i = 0; i < s->nb_hrir_inputs; i++) { |
|
AVFilterLink *input = ctx->inputs[i + 1]; |
|
|
|
if (s->hrir_in[i].eof) |
|
continue; |
|
|
|
if ((ret = check_ir(input, i)) <= 0) |
|
return ret; |
|
|
|
if (s->hrir_in[i].eof) { |
|
if (!ff_inlink_queued_samples(input)) { |
|
av_log(ctx, AV_LOG_ERROR, "No samples provided for " |
|
"HRIR stream %d.\n", i); |
|
return AVERROR_INVALIDDATA; |
|
} |
|
} else { |
|
eof = 0; |
|
} |
|
} |
|
if (!eof) { |
|
ff_filter_set_ready(ctx, 100); |
|
return 0; |
|
} |
|
s->eof_hrirs = 1; |
|
|
|
ret = convert_coeffs(ctx, inlink); |
|
if (ret < 0) |
|
return ret; |
|
} else if (!s->have_hrirs) |
|
return AVERROR_EOF; |
|
|
|
if ((ret = ff_inlink_consume_samples(inlink, 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(inlink, outlink); |
|
if (ff_outlink_frame_wanted(outlink)) |
|
ff_inlink_request_frame(inlink); |
|
|
|
return 0; |
|
} |
|
|
|
static int query_formats(AVFilterContext *ctx) |
|
{ |
|
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; |
|
ret = ff_set_common_formats(ctx, formats); |
|
if (ret) |
|
return ret; |
|
|
|
layouts = ff_all_channel_layouts(); |
|
if (!layouts) |
|
return AVERROR(ENOMEM); |
|
|
|
ret = ff_channel_layouts_ref(layouts, &ctx->inputs[0]->outcfg.channel_layouts); |
|
if (ret) |
|
return ret; |
|
|
|
ret = ff_add_channel_layout(&stereo_layout, &(AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO); |
|
if (ret) |
|
return ret; |
|
ret = ff_channel_layouts_ref(stereo_layout, &ctx->outputs[0]->incfg.channel_layouts); |
|
if (ret) |
|
return ret; |
|
|
|
if (s->hrir_fmt == HRIR_MULTI) { |
|
hrir_layouts = ff_all_channel_counts(); |
|
if (!hrir_layouts) |
|
return AVERROR(ENOMEM); |
|
ret = ff_channel_layouts_ref(hrir_layouts, &ctx->inputs[1]->outcfg.channel_layouts); |
|
if (ret) |
|
return ret; |
|
} else { |
|
for (i = 1; i <= s->nb_hrir_inputs; i++) { |
|
ret = ff_channel_layouts_ref(stereo_layout, &ctx->inputs[i]->outcfg.channel_layouts); |
|
if (ret) |
|
return ret; |
|
} |
|
} |
|
|
|
return ff_set_common_all_samplerates(ctx); |
|
} |
|
|
|
static int config_input(AVFilterLink *inlink) |
|
{ |
|
AVFilterContext *ctx = inlink->dst; |
|
HeadphoneContext *s = ctx->priv; |
|
|
|
if (s->nb_irs < inlink->ch_layout.nb_channels) { |
|
av_log(ctx, AV_LOG_ERROR, "Number of HRIRs must be >= %d.\n", inlink->ch_layout.nb_channels); |
|
return AVERROR(EINVAL); |
|
} |
|
|
|
s->lfe_channel = av_channel_layout_index_from_channel(&inlink->ch_layout, |
|
AV_CHAN_LOW_FREQUENCY); |
|
return 0; |
|
} |
|
|
|
static av_cold int init(AVFilterContext *ctx) |
|
{ |
|
HeadphoneContext *s = ctx->priv; |
|
int i, ret; |
|
|
|
AVFilterPad pad = { |
|
.name = "in0", |
|
.type = AVMEDIA_TYPE_AUDIO, |
|
.config_props = config_input, |
|
}; |
|
if ((ret = ff_append_inpad(ctx, &pad)) < 0) |
|
return ret; |
|
|
|
if (!s->map) { |
|
av_log(ctx, AV_LOG_ERROR, "Valid mapping must be set.\n"); |
|
return AVERROR(EINVAL); |
|
} |
|
|
|
parse_map(ctx); |
|
|
|
for (i = 0; i < s->nb_hrir_inputs; i++) { |
|
char *name = av_asprintf("hrir%d", i); |
|
AVFilterPad pad = { |
|
.name = name, |
|
.type = AVMEDIA_TYPE_AUDIO, |
|
}; |
|
if (!name) |
|
return AVERROR(ENOMEM); |
|
if ((ret = ff_append_inpad_free_name(ctx, &pad)) < 0) |
|
return ret; |
|
} |
|
|
|
if (s->type == TIME_DOMAIN) { |
|
AVFloatDSPContext *fdsp = avpriv_float_dsp_alloc(0); |
|
if (!fdsp) |
|
return AVERROR(ENOMEM); |
|
s->scalarproduct_float = fdsp->scalarproduct_float; |
|
av_free(fdsp); |
|
} |
|
|
|
return 0; |
|
} |
|
|
|
static int config_output(AVFilterLink *outlink) |
|
{ |
|
AVFilterContext *ctx = outlink->src; |
|
HeadphoneContext *s = ctx->priv; |
|
AVFilterLink *inlink = ctx->inputs[0]; |
|
|
|
if (s->hrir_fmt == HRIR_MULTI) { |
|
AVFilterLink *hrir_link = ctx->inputs[1]; |
|
|
|
if (hrir_link->ch_layout.nb_channels < inlink->ch_layout.nb_channels * 2) { |
|
av_log(ctx, AV_LOG_ERROR, "Number of channels in HRIR stream must be >= %d.\n", inlink->ch_layout.nb_channels * 2); |
|
return AVERROR(EINVAL); |
|
} |
|
} |
|
|
|
s->gain_lfe = expf((s->gain - 3 * inlink->ch_layout.nb_channels + s->lfe_gain) / 20 * M_LN10); |
|
|
|
return 0; |
|
} |
|
|
|
static av_cold void uninit(AVFilterContext *ctx) |
|
{ |
|
HeadphoneContext *s = ctx->priv; |
|
|
|
av_tx_uninit(&s->ifft[0]); |
|
av_tx_uninit(&s->ifft[1]); |
|
av_tx_uninit(&s->fft[0]); |
|
av_tx_uninit(&s->fft[1]); |
|
av_freep(&s->data_ir[0]); |
|
av_freep(&s->data_ir[1]); |
|
av_freep(&s->ringbuffer[0]); |
|
av_freep(&s->ringbuffer[1]); |
|
av_freep(&s->temp_src[0]); |
|
av_freep(&s->temp_src[1]); |
|
av_freep(&s->out_fft[0]); |
|
av_freep(&s->out_fft[1]); |
|
av_freep(&s->in_fft[0]); |
|
av_freep(&s->in_fft[1]); |
|
av_freep(&s->temp_afft[0]); |
|
av_freep(&s->temp_afft[1]); |
|
av_freep(&s->data_hrtf[0]); |
|
av_freep(&s->data_hrtf[1]); |
|
} |
|
|
|
#define OFFSET(x) offsetof(HeadphoneContext, x) |
|
#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM |
|
|
|
static const AVOption headphone_options[] = { |
|
{ "map", "set channels convolution mappings", OFFSET(map), AV_OPT_TYPE_STRING, {.str=NULL}, .flags = FLAGS }, |
|
{ "gain", "set gain in dB", OFFSET(gain), AV_OPT_TYPE_FLOAT, {.dbl=0}, -20, 40, .flags = FLAGS }, |
|
{ "lfe", "set lfe gain in dB", OFFSET(lfe_gain), AV_OPT_TYPE_FLOAT, {.dbl=0}, -20, 40, .flags = FLAGS }, |
|
{ "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 } |
|
}; |
|
|
|
AVFILTER_DEFINE_CLASS(headphone); |
|
|
|
static const AVFilterPad outputs[] = { |
|
{ |
|
.name = "default", |
|
.type = AVMEDIA_TYPE_AUDIO, |
|
.config_props = config_output, |
|
}, |
|
}; |
|
|
|
const AVFilter ff_af_headphone = { |
|
.name = "headphone", |
|
.description = NULL_IF_CONFIG_SMALL("Apply headphone binaural spatialization with HRTFs in additional streams."), |
|
.priv_size = sizeof(HeadphoneContext), |
|
.priv_class = &headphone_class, |
|
.init = init, |
|
.uninit = uninit, |
|
.activate = activate, |
|
.inputs = NULL, |
|
FILTER_OUTPUTS(outputs), |
|
FILTER_QUERY_FUNC(query_formats), |
|
.flags = AVFILTER_FLAG_SLICE_THREADS | AVFILTER_FLAG_DYNAMIC_INPUTS, |
|
}; |
|
|