| /* |
| * Copyright (C) 2017 Paul B Mahol |
| * Copyright (C) 2013-2015 Andreas Fuchs, Wolfgang Hrauda |
| * This file is part of FFmpeg. |
| * |
| * FFmpeg is free software; you can redistribute it and/or |
| * modify it under the terms of the GNU Lesser General Public |
| * License as published by the Free Software Foundation; either |
| * version 2.1 of the License, or (at your option) any later version. |
| * |
| * FFmpeg 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 |
| * Lesser General Public License for more details. |
| * |
| * You should have received a copy of the GNU Lesser General Public |
| * License along with FFmpeg; if not, write to the Free Software |
| * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
| */ |
| |
| #include <math.h> |
| |
| #include "libavutil/avstring.h" |
| #include "libavutil/channel_layout.h" |
| #include "libavutil/float_dsp.h" |
| #include "libavutil/intmath.h" |
| #include "libavutil/opt.h" |
| #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; |
| |
| char *map; |
| int type; |
| |
| int lfe_channel; |
| |
| int have_hrirs; |
| int eof_hrirs; |
| |
| int ir_len; |
| |
| int mapping[64]; |
| |
| int nb_inputs; |
| |
| int nb_irs; |
| |
| float gain; |
| float lfe_gain, gain_lfe; |
| |
| float *ringbuffer[2]; |
| int write[2]; |
| |
| int buffer_length; |
| int n_fft; |
| int size; |
| int hrir_fmt; |
| |
| int *delay[2]; |
| float *data_ir[2]; |
| float *temp_src[2]; |
| FFTComplex *temp_fft[2]; |
| |
| FFTContext *fft[2], *ifft[2]; |
| FFTComplex *data_hrtf[2]; |
| |
| AVFloatDSPContext *fdsp; |
| struct headphone_inputs { |
| AVFrame *frame; |
| int ir_len; |
| int delay_l; |
| int delay_r; |
| int eof; |
| } *in; |
| } HeadphoneContext; |
| |
| static int parse_channel_name(HeadphoneContext *s, int x, char **arg, int *rchannel, char *buf) |
| { |
| int len, i, channel_id = 0; |
| int64_t layout, layout0; |
| |
| if (sscanf(*arg, "%7[A-Z]%n", buf, &len)) { |
| layout0 = layout = av_get_channel_layout(buf); |
| if (layout == AV_CH_LOW_FREQUENCY) |
| s->lfe_channel = x; |
| for (i = 32; i > 0; i >>= 1) { |
| if (layout >= 1LL << i) { |
| channel_id += i; |
| layout >>= i; |
| } |
| } |
| if (channel_id >= 64 || layout0 != 1LL << channel_id) |
| return AVERROR(EINVAL); |
| *rchannel = channel_id; |
| *arg += len; |
| return 0; |
| } |
| return AVERROR(EINVAL); |
| } |
| |
| static void parse_map(AVFilterContext *ctx) |
| { |
| HeadphoneContext *s = ctx->priv; |
| char *arg, *tokenizer, *p, *args = av_strdup(s->map); |
| int i; |
| |
| if (!args) |
| return; |
| p = args; |
| |
| s->lfe_channel = -1; |
| s->nb_inputs = 1; |
| |
| for (i = 0; i < 64; i++) { |
| s->mapping[i] = -1; |
| } |
| |
| while ((arg = av_strtok(p, "|", &tokenizer))) { |
| int out_ch_id; |
| char buf[8]; |
| |
| p = NULL; |
| 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_irs] = out_ch_id; |
| s->nb_irs++; |
| } |
| |
| if (s->hrir_fmt == HRIR_MULTI) |
| s->nb_inputs = 2; |
| else |
| s->nb_inputs = s->nb_irs + 1; |
| |
| av_free(args); |
| } |
| |
| typedef struct ThreadData { |
| AVFrame *in, *out; |
| int *write; |
| int **delay; |
| float **ir; |
| int *n_clippings; |
| float **ringbuffer; |
| float **temp_src; |
| FFTComplex **temp_fft; |
| } 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 int *const delay = td->delay[jobnr]; |
| const float *const ir = td->ir[jobnr]; |
| int *n_clippings = &td->n_clippings[jobnr]; |
| float *ringbuffer = td->ringbuffer[jobnr]; |
| float *temp_src = td->temp_src[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->channels; |
| const int buffer_length = s->buffer_length; |
| const uint32_t modulo = (uint32_t)buffer_length - 1; |
| float *buffer[16]; |
| int wr = *write; |
| int read; |
| int i, l; |
| |
| dst += offset; |
| for (l = 0; l < in_channels; l++) { |
| buffer[l] = ringbuffer + l * buffer_length; |
| } |
| |
| for (i = 0; i < in->nb_samples; i++) { |
| const float *temp_ir = ir; |
| |
| *dst = 0; |
| for (l = 0; l < in_channels; l++) { |
| *(buffer[l] + wr) = src[l]; |
| } |
| |
| for (l = 0; l < in_channels; l++) { |
| const float *const bptr = buffer[l]; |
| |
| if (l == s->lfe_channel) { |
| *dst += *(buffer[s->lfe_channel] + wr) * s->gain_lfe; |
| temp_ir += FFALIGN(ir_len, 16); |
| continue; |
| } |
| |
| read = (wr - *(delay + l) - (ir_len - 1) + buffer_length) & modulo; |
| |
| if (read + ir_len < buffer_length) { |
| memcpy(temp_src, bptr + read, ir_len * sizeof(*temp_src)); |
| } else { |
| int len = FFMIN(ir_len - (read % ir_len), buffer_length - read); |
| |
| memcpy(temp_src, bptr + read, len * sizeof(*temp_src)); |
| memcpy(temp_src + len, bptr, (ir_len - len) * sizeof(*temp_src)); |
| } |
| |
| dst[0] += s->fdsp->scalarproduct_float(temp_ir, temp_src, ir_len); |
| temp_ir += FFALIGN(ir_len, 16); |
| } |
| |
| if (fabs(*dst) > 1) |
| *n_clippings += 1; |
| |
| dst += 2; |
| src += in_channels; |
| wr = (wr + 1) & modulo; |
| } |
| |
| *write = wr; |
| |
| 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]; |
| FFTComplex *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->channels; |
| const int buffer_length = s->buffer_length; |
| const uint32_t modulo = (uint32_t)buffer_length - 1; |
| FFTComplex *fft_in = s->temp_fft[jobnr]; |
| FFTContext *ifft = s->ifft[jobnr]; |
| FFTContext *fft = s->fft[jobnr]; |
| const int n_fft = s->n_fft; |
| const float fft_scale = 1.0f / s->n_fft; |
| FFTComplex *hrtf_offset; |
| int wr = *write; |
| int n_read; |
| int i, j; |
| |
| dst += offset; |
| |
| n_read = FFMIN(s->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; |
| } |
| |
| 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 + offset; |
| |
| memset(fft_in, 0, sizeof(FFTComplex) * n_fft); |
| |
| for (j = 0; j < in->nb_samples; j++) { |
| fft_in[j].re = src[j * in_channels + i]; |
| } |
| |
| av_fft_permute(fft, fft_in); |
| av_fft_calc(fft, fft_in); |
| for (j = 0; j < n_fft; j++) { |
| const FFTComplex *hcomplex = hrtf_offset + j; |
| const float re = fft_in[j].re; |
| const float im = fft_in[j].im; |
| |
| fft_in[j].re = re * hcomplex->re - im * hcomplex->im; |
| fft_in[j].im = re * hcomplex->im + im * hcomplex->re; |
| } |
| |
| av_fft_permute(ifft, fft_in); |
| av_fft_calc(ifft, fft_in); |
| |
| for (j = 0; j < in->nb_samples; j++) { |
| dst[2 * j] += fft_in[j].re * fft_scale; |
| } |
| |
| for (j = 0; j < ir_len - 1; j++) { |
| int write_pos = (wr + j) & modulo; |
| |
| *(ringbuffer + write_pos) += fft_in[in->nb_samples + j].re * fft_scale; |
| } |
| } |
| |
| for (i = 0; i < out->nb_samples; i++) { |
| if (fabs(*dst) > 1) { |
| n_clippings[0]++; |
| } |
| |
| dst += 2; |
| } |
| |
| *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->in[input_number].ir_len = ir_len; |
| s->ir_len = FFMAX(ir_len, s->ir_len); |
| |
| 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.delay = s->delay; td.ir = s->data_ir; td.n_clippings = n_clippings; |
| td.ringbuffer = s->ringbuffer; td.temp_src = s->temp_src; |
| td.temp_fft = s->temp_fft; |
| |
| if (s->type == TIME_DOMAIN) { |
| ctx->internal->execute(ctx, headphone_convolute, &td, NULL, 2); |
| } else { |
| ctx->internal->execute(ctx, headphone_fast_convolute, &td, NULL, 2); |
| } |
| emms_c(); |
| |
| 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_irs = s->nb_irs; |
| int nb_input_channels = ctx->inputs[0]->channels; |
| float gain_lin = expf((s->gain - 3 * nb_input_channels) / 20 * M_LN10); |
| FFTComplex *data_hrtf_l = NULL; |
| FFTComplex *data_hrtf_r = NULL; |
| FFTComplex *fft_in_l = NULL; |
| FFTComplex *fft_in_r = NULL; |
| float *data_ir_l = NULL; |
| float *data_ir_r = NULL; |
| int offset = 0, ret = 0; |
| int n_fft; |
| 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 + s->size)); |
| |
| if (s->type == FREQUENCY_DOMAIN) { |
| fft_in_l = av_calloc(n_fft, sizeof(*fft_in_l)); |
| fft_in_r = av_calloc(n_fft, sizeof(*fft_in_r)); |
| if (!fft_in_l || !fft_in_r) { |
| ret = AVERROR(ENOMEM); |
| goto fail; |
| } |
| |
| av_fft_end(s->fft[0]); |
| av_fft_end(s->fft[1]); |
| s->fft[0] = av_fft_init(log2(s->n_fft), 0); |
| s->fft[1] = av_fft_init(log2(s->n_fft), 0); |
| av_fft_end(s->ifft[0]); |
| av_fft_end(s->ifft[1]); |
| s->ifft[0] = av_fft_init(log2(s->n_fft), 1); |
| s->ifft[1] = av_fft_init(log2(s->n_fft), 1); |
| |
| 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; |
| } |
| } |
| |
| 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_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); |
| 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->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; |
| } |
| } |
| |
| if (!s->data_ir[0] || !s->data_ir[1] || |
| !s->ringbuffer[0] || !s->ringbuffer[1]) { |
| ret = AVERROR(ENOMEM); |
| goto fail; |
| } |
| |
| if (s->type == TIME_DOMAIN) { |
| s->temp_src[0] = av_calloc(FFALIGN(ir_len, 16), sizeof(float)); |
| s->temp_src[1] = av_calloc(FFALIGN(ir_len, 16), sizeof(float)); |
| |
| data_ir_l = av_calloc(nb_irs * FFALIGN(ir_len, 16), sizeof(*data_ir_l)); |
| data_ir_r = av_calloc(nb_irs * FFALIGN(ir_len, 16), sizeof(*data_ir_r)); |
| if (!data_ir_r || !data_ir_l || !s->temp_src[0] || !s->temp_src[1]) { |
| ret = AVERROR(ENOMEM); |
| goto fail; |
| } |
| } else { |
| 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_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; |
| float *ptr; |
| |
| ret = ff_inlink_consume_samples(ctx->inputs[i + 1], len, len, &s->in[i + 1].frame); |
| if (ret < 0) |
| return ret; |
| ptr = (float *)s->in[i + 1].frame->extended_data[0]; |
| |
| if (s->hrir_fmt == HRIR_STEREO) { |
| int idx = -1; |
| |
| 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; |
| } |
| } |
| |
| 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)); |
| } |
| } |
| } |
| |
| av_frame_free(&s->in[i + 1].frame); |
| } |
| |
| if (s->type == TIME_DOMAIN) { |
| 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_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; |
| } |
| |
| memcpy(s->data_hrtf[0], data_hrtf_l, |
| sizeof(FFTComplex) * nb_irs * n_fft); |
| memcpy(s->data_hrtf[1], data_hrtf_r, |
| sizeof(FFTComplex) * nb_irs * n_fft); |
| } |
| |
| s->have_hrirs = 1; |
| |
| fail: |
| |
| av_freep(&data_ir_l); |
| av_freep(&data_ir_r); |
| |
| av_freep(&data_hrtf_l); |
| av_freep(&data_hrtf_r); |
| |
| av_freep(&fft_in_l); |
| av_freep(&fft_in_r); |
| |
| 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(ctx->outputs[0], ctx); |
| if (!s->eof_hrirs) { |
| for (i = 1; i < s->nb_inputs; i++) { |
| if (s->in[i].eof) |
| continue; |
| |
| if ((ret = check_ir(ctx->inputs[i], i)) < 0) |
| return ret; |
| |
| if (!s->in[i].eof) { |
| if (ff_outlink_get_status(ctx->inputs[i]) == 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); |
| 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; |
| } |
| |
| 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]->out_channel_layouts); |
| if (ret) |
| return ret; |
| |
| ret = ff_add_channel_layout(&stereo_layout, AV_CH_LAYOUT_STEREO); |
| if (ret) |
| return ret; |
| |
| 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(stereo_layout, &ctx->outputs[0]->in_channel_layouts); |
| if (ret) |
| return ret; |
| |
| formats = ff_all_samplerates(); |
| if (!formats) |
| return AVERROR(ENOMEM); |
| return ff_set_common_samplerates(ctx, formats); |
| } |
| |
| static int config_input(AVFilterLink *inlink) |
| { |
| AVFilterContext *ctx = inlink->dst; |
| HeadphoneContext *s = ctx->priv; |
| |
| if (s->nb_irs < inlink->channels) { |
| av_log(ctx, AV_LOG_ERROR, "Number of HRIRs must be >= %d.\n", inlink->channels); |
| return AVERROR(EINVAL); |
| } |
| |
| 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_insert_inpad(ctx, 0, &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); |
| |
| s->in = av_calloc(s->nb_inputs, sizeof(*s->in)); |
| if (!s->in) |
| return AVERROR(ENOMEM); |
| |
| for (i = 1; i < s->nb_inputs; i++) { |
| char *name = av_asprintf("hrir%d", i - 1); |
| AVFilterPad pad = { |
| .name = name, |
| .type = AVMEDIA_TYPE_AUDIO, |
| }; |
| if (!name) |
| return AVERROR(ENOMEM); |
| if ((ret = ff_insert_inpad(ctx, i, &pad)) < 0) { |
| av_freep(&pad.name); |
| return ret; |
| } |
| } |
| |
| s->fdsp = avpriv_float_dsp_alloc(0); |
| if (!s->fdsp) |
| return AVERROR(ENOMEM); |
| |
| 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->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); |
| } |
| } |
| |
| s->gain_lfe = expf((s->gain - 3 * inlink->channels - 6 + s->lfe_gain) / 20 * M_LN10); |
| |
| return 0; |
| } |
| |
| static av_cold void uninit(AVFilterContext *ctx) |
| { |
| HeadphoneContext *s = ctx->priv; |
| int i; |
| |
| av_fft_end(s->ifft[0]); |
| av_fft_end(s->ifft[1]); |
| av_fft_end(s->fft[0]); |
| av_fft_end(s->fft[1]); |
| av_freep(&s->delay[0]); |
| av_freep(&s->delay[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->temp_fft[0]); |
| av_freep(&s->temp_fft[1]); |
| av_freep(&s->data_hrtf[0]); |
| av_freep(&s->data_hrtf[1]); |
| av_freep(&s->fdsp); |
| |
| for (i = 0; i < s->nb_inputs; i++) { |
| if (ctx->input_pads && i) |
| av_freep(&ctx->input_pads[i].name); |
| } |
| av_freep(&s->in); |
| } |
| |
| #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, |
| }, |
| { NULL } |
| }; |
| |
| 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, |
| .query_formats = query_formats, |
| .activate = activate, |
| .inputs = NULL, |
| .outputs = outputs, |
| .flags = AVFILTER_FLAG_SLICE_THREADS | AVFILTER_FLAG_DYNAMIC_INPUTS, |
| }; |
