| /* |
| * Copyright (c) 2013-2015 Paul B Mahol |
| * |
| * 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 |
| */ |
| |
| /** |
| * @file |
| * fade audio filter |
| */ |
| |
| #include "libavutil/audio_fifo.h" |
| #include "libavutil/opt.h" |
| #include "audio.h" |
| #include "avfilter.h" |
| #include "internal.h" |
| |
| typedef struct { |
| const AVClass *class; |
| int type; |
| int curve, curve2; |
| int nb_samples; |
| int64_t start_sample; |
| int64_t duration; |
| int64_t start_time; |
| int overlap; |
| int cf0_eof; |
| int crossfade_is_over; |
| AVAudioFifo *fifo[2]; |
| int64_t pts; |
| |
| void (*fade_samples)(uint8_t **dst, uint8_t * const *src, |
| int nb_samples, int channels, int direction, |
| int64_t start, int range, int curve); |
| void (*crossfade_samples)(uint8_t **dst, uint8_t * const *cf0, |
| uint8_t * const *cf1, |
| int nb_samples, int channels, |
| int curve0, int curve1); |
| } AudioFadeContext; |
| |
| enum CurveType { TRI, QSIN, ESIN, HSIN, LOG, IPAR, QUA, CUB, SQU, CBR, PAR, EXP, IQSIN, IHSIN, DESE, DESI, NB_CURVES }; |
| |
| #define OFFSET(x) offsetof(AudioFadeContext, x) |
| #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM |
| |
| static int query_formats(AVFilterContext *ctx) |
| { |
| AVFilterFormats *formats; |
| AVFilterChannelLayouts *layouts; |
| static const enum AVSampleFormat sample_fmts[] = { |
| AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P, |
| AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32P, |
| AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP, |
| AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP, |
| AV_SAMPLE_FMT_NONE |
| }; |
| int ret; |
| |
| layouts = ff_all_channel_counts(); |
| if (!layouts) |
| return AVERROR(ENOMEM); |
| ret = ff_set_common_channel_layouts(ctx, layouts); |
| if (ret < 0) |
| return ret; |
| |
| formats = ff_make_format_list(sample_fmts); |
| if (!formats) |
| return AVERROR(ENOMEM); |
| ret = ff_set_common_formats(ctx, formats); |
| if (ret < 0) |
| return ret; |
| |
| formats = ff_all_samplerates(); |
| if (!formats) |
| return AVERROR(ENOMEM); |
| return ff_set_common_samplerates(ctx, formats); |
| } |
| |
| static double fade_gain(int curve, int64_t index, int range) |
| { |
| #define CUBE(a) ((a)*(a)*(a)) |
| double gain; |
| |
| gain = av_clipd(1.0 * index / range, 0, 1.0); |
| |
| switch (curve) { |
| case QSIN: |
| gain = sin(gain * M_PI / 2.0); |
| break; |
| case IQSIN: |
| /* 0.6... = 2 / M_PI */ |
| gain = 0.6366197723675814 * asin(gain); |
| break; |
| case ESIN: |
| gain = 1.0 - cos(M_PI / 4.0 * (CUBE(2.0*gain - 1) + 1)); |
| break; |
| case HSIN: |
| gain = (1.0 - cos(gain * M_PI)) / 2.0; |
| break; |
| case IHSIN: |
| /* 0.3... = 1 / M_PI */ |
| gain = 0.3183098861837907 * acos(1 - 2 * gain); |
| break; |
| case EXP: |
| /* -11.5... = 5*ln(0.1) */ |
| gain = exp(-11.512925464970227 * (1 - gain)); |
| break; |
| case LOG: |
| gain = av_clipd(1 + 0.2 * log10(gain), 0, 1.0); |
| break; |
| case PAR: |
| gain = 1 - sqrt(1 - gain); |
| break; |
| case IPAR: |
| gain = (1 - (1 - gain) * (1 - gain)); |
| break; |
| case QUA: |
| gain *= gain; |
| break; |
| case CUB: |
| gain = CUBE(gain); |
| break; |
| case SQU: |
| gain = sqrt(gain); |
| break; |
| case CBR: |
| gain = cbrt(gain); |
| break; |
| case DESE: |
| gain = gain <= 0.5 ? cbrt(2 * gain) / 2: 1 - cbrt(2 * (1 - gain)) / 2; |
| break; |
| case DESI: |
| gain = gain <= 0.5 ? CUBE(2 * gain) / 2: 1 - CUBE(2 * (1 - gain)) / 2; |
| break; |
| } |
| |
| return gain; |
| } |
| |
| #define FADE_PLANAR(name, type) \ |
| static void fade_samples_## name ##p(uint8_t **dst, uint8_t * const *src, \ |
| int nb_samples, int channels, int dir, \ |
| int64_t start, int range, int curve) \ |
| { \ |
| int i, c; \ |
| \ |
| for (i = 0; i < nb_samples; i++) { \ |
| double gain = fade_gain(curve, start + i * dir, range); \ |
| for (c = 0; c < channels; c++) { \ |
| type *d = (type *)dst[c]; \ |
| const type *s = (type *)src[c]; \ |
| \ |
| d[i] = s[i] * gain; \ |
| } \ |
| } \ |
| } |
| |
| #define FADE(name, type) \ |
| static void fade_samples_## name (uint8_t **dst, uint8_t * const *src, \ |
| int nb_samples, int channels, int dir, \ |
| int64_t start, int range, int curve) \ |
| { \ |
| type *d = (type *)dst[0]; \ |
| const type *s = (type *)src[0]; \ |
| int i, c, k = 0; \ |
| \ |
| for (i = 0; i < nb_samples; i++) { \ |
| double gain = fade_gain(curve, start + i * dir, range); \ |
| for (c = 0; c < channels; c++, k++) \ |
| d[k] = s[k] * gain; \ |
| } \ |
| } |
| |
| FADE_PLANAR(dbl, double) |
| FADE_PLANAR(flt, float) |
| FADE_PLANAR(s16, int16_t) |
| FADE_PLANAR(s32, int32_t) |
| |
| FADE(dbl, double) |
| FADE(flt, float) |
| FADE(s16, int16_t) |
| FADE(s32, int32_t) |
| |
| static int config_output(AVFilterLink *outlink) |
| { |
| AVFilterContext *ctx = outlink->src; |
| AudioFadeContext *s = ctx->priv; |
| |
| switch (outlink->format) { |
| case AV_SAMPLE_FMT_DBL: s->fade_samples = fade_samples_dbl; break; |
| case AV_SAMPLE_FMT_DBLP: s->fade_samples = fade_samples_dblp; break; |
| case AV_SAMPLE_FMT_FLT: s->fade_samples = fade_samples_flt; break; |
| case AV_SAMPLE_FMT_FLTP: s->fade_samples = fade_samples_fltp; break; |
| case AV_SAMPLE_FMT_S16: s->fade_samples = fade_samples_s16; break; |
| case AV_SAMPLE_FMT_S16P: s->fade_samples = fade_samples_s16p; break; |
| case AV_SAMPLE_FMT_S32: s->fade_samples = fade_samples_s32; break; |
| case AV_SAMPLE_FMT_S32P: s->fade_samples = fade_samples_s32p; break; |
| } |
| |
| if (s->duration) |
| s->nb_samples = av_rescale(s->duration, outlink->sample_rate, AV_TIME_BASE); |
| if (s->start_time) |
| s->start_sample = av_rescale(s->start_time, outlink->sample_rate, AV_TIME_BASE); |
| |
| return 0; |
| } |
| |
| #if CONFIG_AFADE_FILTER |
| |
| static const AVOption afade_options[] = { |
| { "type", "set the fade direction", OFFSET(type), AV_OPT_TYPE_INT, {.i64 = 0 }, 0, 1, FLAGS, "type" }, |
| { "t", "set the fade direction", OFFSET(type), AV_OPT_TYPE_INT, {.i64 = 0 }, 0, 1, FLAGS, "type" }, |
| { "in", "fade-in", 0, AV_OPT_TYPE_CONST, {.i64 = 0 }, 0, 0, FLAGS, "type" }, |
| { "out", "fade-out", 0, AV_OPT_TYPE_CONST, {.i64 = 1 }, 0, 0, FLAGS, "type" }, |
| { "start_sample", "set number of first sample to start fading", OFFSET(start_sample), AV_OPT_TYPE_INT64, {.i64 = 0 }, 0, INT64_MAX, FLAGS }, |
| { "ss", "set number of first sample to start fading", OFFSET(start_sample), AV_OPT_TYPE_INT64, {.i64 = 0 }, 0, INT64_MAX, FLAGS }, |
| { "nb_samples", "set number of samples for fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 44100}, 1, INT32_MAX, FLAGS }, |
| { "ns", "set number of samples for fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 44100}, 1, INT32_MAX, FLAGS }, |
| { "start_time", "set time to start fading", OFFSET(start_time), AV_OPT_TYPE_DURATION, {.i64 = 0. }, 0, INT32_MAX, FLAGS }, |
| { "st", "set time to start fading", OFFSET(start_time), AV_OPT_TYPE_DURATION, {.i64 = 0. }, 0, INT32_MAX, FLAGS }, |
| { "duration", "set fade duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0. }, 0, INT32_MAX, FLAGS }, |
| { "d", "set fade duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0. }, 0, INT32_MAX, FLAGS }, |
| { "curve", "set fade curve type", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, 0, NB_CURVES - 1, FLAGS, "curve" }, |
| { "c", "set fade curve type", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, 0, NB_CURVES - 1, FLAGS, "curve" }, |
| { "tri", "linear slope", 0, AV_OPT_TYPE_CONST, {.i64 = TRI }, 0, 0, FLAGS, "curve" }, |
| { "qsin", "quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = QSIN }, 0, 0, FLAGS, "curve" }, |
| { "esin", "exponential sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = ESIN }, 0, 0, FLAGS, "curve" }, |
| { "hsin", "half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = HSIN }, 0, 0, FLAGS, "curve" }, |
| { "log", "logarithmic", 0, AV_OPT_TYPE_CONST, {.i64 = LOG }, 0, 0, FLAGS, "curve" }, |
| { "ipar", "inverted parabola", 0, AV_OPT_TYPE_CONST, {.i64 = IPAR }, 0, 0, FLAGS, "curve" }, |
| { "qua", "quadratic", 0, AV_OPT_TYPE_CONST, {.i64 = QUA }, 0, 0, FLAGS, "curve" }, |
| { "cub", "cubic", 0, AV_OPT_TYPE_CONST, {.i64 = CUB }, 0, 0, FLAGS, "curve" }, |
| { "squ", "square root", 0, AV_OPT_TYPE_CONST, {.i64 = SQU }, 0, 0, FLAGS, "curve" }, |
| { "cbr", "cubic root", 0, AV_OPT_TYPE_CONST, {.i64 = CBR }, 0, 0, FLAGS, "curve" }, |
| { "par", "parabola", 0, AV_OPT_TYPE_CONST, {.i64 = PAR }, 0, 0, FLAGS, "curve" }, |
| { "exp", "exponential", 0, AV_OPT_TYPE_CONST, {.i64 = EXP }, 0, 0, FLAGS, "curve" }, |
| { "iqsin", "inverted quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = IQSIN}, 0, 0, FLAGS, "curve" }, |
| { "ihsin", "inverted half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = IHSIN}, 0, 0, FLAGS, "curve" }, |
| { "dese", "double-exponential seat", 0, AV_OPT_TYPE_CONST, {.i64 = DESE }, 0, 0, FLAGS, "curve" }, |
| { "desi", "double-exponential sigmoid", 0, AV_OPT_TYPE_CONST, {.i64 = DESI }, 0, 0, FLAGS, "curve" }, |
| { NULL } |
| }; |
| |
| AVFILTER_DEFINE_CLASS(afade); |
| |
| static av_cold int init(AVFilterContext *ctx) |
| { |
| AudioFadeContext *s = ctx->priv; |
| |
| if (INT64_MAX - s->nb_samples < s->start_sample) |
| return AVERROR(EINVAL); |
| |
| return 0; |
| } |
| |
| static int filter_frame(AVFilterLink *inlink, AVFrame *buf) |
| { |
| AudioFadeContext *s = inlink->dst->priv; |
| AVFilterLink *outlink = inlink->dst->outputs[0]; |
| int nb_samples = buf->nb_samples; |
| AVFrame *out_buf; |
| int64_t cur_sample = av_rescale_q(buf->pts, inlink->time_base, (AVRational){1, inlink->sample_rate}); |
| |
| if ((!s->type && (s->start_sample + s->nb_samples < cur_sample)) || |
| ( s->type && (cur_sample + nb_samples < s->start_sample))) |
| return ff_filter_frame(outlink, buf); |
| |
| if (av_frame_is_writable(buf)) { |
| out_buf = buf; |
| } else { |
| out_buf = ff_get_audio_buffer(inlink, nb_samples); |
| if (!out_buf) |
| return AVERROR(ENOMEM); |
| av_frame_copy_props(out_buf, buf); |
| } |
| |
| if ((!s->type && (cur_sample + nb_samples < s->start_sample)) || |
| ( s->type && (s->start_sample + s->nb_samples < cur_sample))) { |
| av_samples_set_silence(out_buf->extended_data, 0, nb_samples, |
| av_frame_get_channels(out_buf), out_buf->format); |
| } else { |
| int64_t start; |
| |
| if (!s->type) |
| start = cur_sample - s->start_sample; |
| else |
| start = s->start_sample + s->nb_samples - cur_sample; |
| |
| s->fade_samples(out_buf->extended_data, buf->extended_data, |
| nb_samples, av_frame_get_channels(buf), |
| s->type ? -1 : 1, start, |
| s->nb_samples, s->curve); |
| } |
| |
| if (buf != out_buf) |
| av_frame_free(&buf); |
| |
| return ff_filter_frame(outlink, out_buf); |
| } |
| |
| static const AVFilterPad avfilter_af_afade_inputs[] = { |
| { |
| .name = "default", |
| .type = AVMEDIA_TYPE_AUDIO, |
| .filter_frame = filter_frame, |
| }, |
| { NULL } |
| }; |
| |
| static const AVFilterPad avfilter_af_afade_outputs[] = { |
| { |
| .name = "default", |
| .type = AVMEDIA_TYPE_AUDIO, |
| .config_props = config_output, |
| }, |
| { NULL } |
| }; |
| |
| AVFilter ff_af_afade = { |
| .name = "afade", |
| .description = NULL_IF_CONFIG_SMALL("Fade in/out input audio."), |
| .query_formats = query_formats, |
| .priv_size = sizeof(AudioFadeContext), |
| .init = init, |
| .inputs = avfilter_af_afade_inputs, |
| .outputs = avfilter_af_afade_outputs, |
| .priv_class = &afade_class, |
| .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC, |
| }; |
| |
| #endif /* CONFIG_AFADE_FILTER */ |
| |
| #if CONFIG_ACROSSFADE_FILTER |
| |
| static const AVOption acrossfade_options[] = { |
| { "nb_samples", "set number of samples for cross fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 44100}, 1, INT32_MAX/10, FLAGS }, |
| { "ns", "set number of samples for cross fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 44100}, 1, INT32_MAX/10, FLAGS }, |
| { "duration", "set cross fade duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0. }, 0, 60, FLAGS }, |
| { "d", "set cross fade duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0. }, 0, 60, FLAGS }, |
| { "overlap", "overlap 1st stream end with 2nd stream start", OFFSET(overlap), AV_OPT_TYPE_BOOL, {.i64 = 1 }, 0, 1, FLAGS }, |
| { "o", "overlap 1st stream end with 2nd stream start", OFFSET(overlap), AV_OPT_TYPE_BOOL, {.i64 = 1 }, 0, 1, FLAGS }, |
| { "curve1", "set fade curve type for 1st stream", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, 0, NB_CURVES - 1, FLAGS, "curve" }, |
| { "c1", "set fade curve type for 1st stream", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, 0, NB_CURVES - 1, FLAGS, "curve" }, |
| { "tri", "linear slope", 0, AV_OPT_TYPE_CONST, {.i64 = TRI }, 0, 0, FLAGS, "curve" }, |
| { "qsin", "quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = QSIN }, 0, 0, FLAGS, "curve" }, |
| { "esin", "exponential sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = ESIN }, 0, 0, FLAGS, "curve" }, |
| { "hsin", "half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = HSIN }, 0, 0, FLAGS, "curve" }, |
| { "log", "logarithmic", 0, AV_OPT_TYPE_CONST, {.i64 = LOG }, 0, 0, FLAGS, "curve" }, |
| { "ipar", "inverted parabola", 0, AV_OPT_TYPE_CONST, {.i64 = IPAR }, 0, 0, FLAGS, "curve" }, |
| { "qua", "quadratic", 0, AV_OPT_TYPE_CONST, {.i64 = QUA }, 0, 0, FLAGS, "curve" }, |
| { "cub", "cubic", 0, AV_OPT_TYPE_CONST, {.i64 = CUB }, 0, 0, FLAGS, "curve" }, |
| { "squ", "square root", 0, AV_OPT_TYPE_CONST, {.i64 = SQU }, 0, 0, FLAGS, "curve" }, |
| { "cbr", "cubic root", 0, AV_OPT_TYPE_CONST, {.i64 = CBR }, 0, 0, FLAGS, "curve" }, |
| { "par", "parabola", 0, AV_OPT_TYPE_CONST, {.i64 = PAR }, 0, 0, FLAGS, "curve" }, |
| { "exp", "exponential", 0, AV_OPT_TYPE_CONST, {.i64 = EXP }, 0, 0, FLAGS, "curve" }, |
| { "iqsin", "inverted quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = IQSIN}, 0, 0, FLAGS, "curve" }, |
| { "ihsin", "inverted half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = IHSIN}, 0, 0, FLAGS, "curve" }, |
| { "dese", "double-exponential seat", 0, AV_OPT_TYPE_CONST, {.i64 = DESE }, 0, 0, FLAGS, "curve" }, |
| { "desi", "double-exponential sigmoid", 0, AV_OPT_TYPE_CONST, {.i64 = DESI }, 0, 0, FLAGS, "curve" }, |
| { "curve2", "set fade curve type for 2nd stream", OFFSET(curve2), AV_OPT_TYPE_INT, {.i64 = TRI }, 0, NB_CURVES - 1, FLAGS, "curve" }, |
| { "c2", "set fade curve type for 2nd stream", OFFSET(curve2), AV_OPT_TYPE_INT, {.i64 = TRI }, 0, NB_CURVES - 1, FLAGS, "curve" }, |
| { NULL } |
| }; |
| |
| AVFILTER_DEFINE_CLASS(acrossfade); |
| |
| #define CROSSFADE_PLANAR(name, type) \ |
| static void crossfade_samples_## name ##p(uint8_t **dst, uint8_t * const *cf0, \ |
| uint8_t * const *cf1, \ |
| int nb_samples, int channels, \ |
| int curve0, int curve1) \ |
| { \ |
| int i, c; \ |
| \ |
| for (i = 0; i < nb_samples; i++) { \ |
| double gain0 = fade_gain(curve0, nb_samples - 1 - i, nb_samples); \ |
| double gain1 = fade_gain(curve1, i, nb_samples); \ |
| for (c = 0; c < channels; c++) { \ |
| type *d = (type *)dst[c]; \ |
| const type *s0 = (type *)cf0[c]; \ |
| const type *s1 = (type *)cf1[c]; \ |
| \ |
| d[i] = s0[i] * gain0 + s1[i] * gain1; \ |
| } \ |
| } \ |
| } |
| |
| #define CROSSFADE(name, type) \ |
| static void crossfade_samples_## name (uint8_t **dst, uint8_t * const *cf0, \ |
| uint8_t * const *cf1, \ |
| int nb_samples, int channels, \ |
| int curve0, int curve1) \ |
| { \ |
| type *d = (type *)dst[0]; \ |
| const type *s0 = (type *)cf0[0]; \ |
| const type *s1 = (type *)cf1[0]; \ |
| int i, c, k = 0; \ |
| \ |
| for (i = 0; i < nb_samples; i++) { \ |
| double gain0 = fade_gain(curve0, nb_samples - 1 - i, nb_samples); \ |
| double gain1 = fade_gain(curve1, i, nb_samples); \ |
| for (c = 0; c < channels; c++, k++) \ |
| d[k] = s0[k] * gain0 + s1[k] * gain1; \ |
| } \ |
| } |
| |
| CROSSFADE_PLANAR(dbl, double) |
| CROSSFADE_PLANAR(flt, float) |
| CROSSFADE_PLANAR(s16, int16_t) |
| CROSSFADE_PLANAR(s32, int32_t) |
| |
| CROSSFADE(dbl, double) |
| CROSSFADE(flt, float) |
| CROSSFADE(s16, int16_t) |
| CROSSFADE(s32, int32_t) |
| |
| static int acrossfade_filter_frame(AVFilterLink *inlink, AVFrame *in) |
| { |
| AVFilterContext *ctx = inlink->dst; |
| AudioFadeContext *s = ctx->priv; |
| AVFilterLink *outlink = ctx->outputs[0]; |
| AVFrame *out, *cf[2] = { NULL }; |
| int ret = 0, nb_samples; |
| |
| if (s->crossfade_is_over) { |
| in->pts = s->pts; |
| s->pts += av_rescale_q(in->nb_samples, |
| (AVRational){ 1, outlink->sample_rate }, outlink->time_base); |
| return ff_filter_frame(outlink, in); |
| } else if (inlink == ctx->inputs[0]) { |
| av_audio_fifo_write(s->fifo[0], (void **)in->extended_data, in->nb_samples); |
| |
| nb_samples = av_audio_fifo_size(s->fifo[0]) - s->nb_samples; |
| if (nb_samples > 0) { |
| out = ff_get_audio_buffer(outlink, nb_samples); |
| if (!out) { |
| ret = AVERROR(ENOMEM); |
| goto fail; |
| } |
| av_audio_fifo_read(s->fifo[0], (void **)out->extended_data, nb_samples); |
| out->pts = s->pts; |
| s->pts += av_rescale_q(nb_samples, |
| (AVRational){ 1, outlink->sample_rate }, outlink->time_base); |
| ret = ff_filter_frame(outlink, out); |
| } |
| } else if (av_audio_fifo_size(s->fifo[1]) < s->nb_samples) { |
| if (!s->overlap && av_audio_fifo_size(s->fifo[0]) > 0) { |
| nb_samples = av_audio_fifo_size(s->fifo[0]); |
| |
| cf[0] = ff_get_audio_buffer(outlink, nb_samples); |
| out = ff_get_audio_buffer(outlink, nb_samples); |
| if (!out || !cf[0]) { |
| ret = AVERROR(ENOMEM); |
| goto fail; |
| } |
| av_audio_fifo_read(s->fifo[0], (void **)cf[0]->extended_data, nb_samples); |
| |
| s->fade_samples(out->extended_data, cf[0]->extended_data, nb_samples, |
| outlink->channels, -1, nb_samples - 1, nb_samples, s->curve); |
| out->pts = s->pts; |
| s->pts += av_rescale_q(nb_samples, |
| (AVRational){ 1, outlink->sample_rate }, outlink->time_base); |
| ret = ff_filter_frame(outlink, out); |
| if (ret < 0) |
| goto fail; |
| } |
| |
| av_audio_fifo_write(s->fifo[1], (void **)in->extended_data, in->nb_samples); |
| } else if (av_audio_fifo_size(s->fifo[1]) >= s->nb_samples) { |
| av_audio_fifo_write(s->fifo[1], (void **)in->extended_data, in->nb_samples); |
| |
| if (s->overlap) { |
| cf[0] = ff_get_audio_buffer(outlink, s->nb_samples); |
| cf[1] = ff_get_audio_buffer(outlink, s->nb_samples); |
| out = ff_get_audio_buffer(outlink, s->nb_samples); |
| if (!out || !cf[0] || !cf[1]) { |
| av_frame_free(&out); |
| ret = AVERROR(ENOMEM); |
| goto fail; |
| } |
| |
| av_audio_fifo_read(s->fifo[0], (void **)cf[0]->extended_data, s->nb_samples); |
| av_audio_fifo_read(s->fifo[1], (void **)cf[1]->extended_data, s->nb_samples); |
| |
| s->crossfade_samples(out->extended_data, cf[0]->extended_data, |
| cf[1]->extended_data, |
| s->nb_samples, av_frame_get_channels(in), |
| s->curve, s->curve2); |
| out->pts = s->pts; |
| s->pts += av_rescale_q(s->nb_samples, |
| (AVRational){ 1, outlink->sample_rate }, outlink->time_base); |
| ret = ff_filter_frame(outlink, out); |
| if (ret < 0) |
| goto fail; |
| } else { |
| out = ff_get_audio_buffer(outlink, s->nb_samples); |
| cf[1] = ff_get_audio_buffer(outlink, s->nb_samples); |
| if (!out || !cf[1]) { |
| ret = AVERROR(ENOMEM); |
| av_frame_free(&out); |
| goto fail; |
| } |
| |
| av_audio_fifo_read(s->fifo[1], (void **)cf[1]->extended_data, s->nb_samples); |
| |
| s->fade_samples(out->extended_data, cf[1]->extended_data, s->nb_samples, |
| outlink->channels, 1, 0, s->nb_samples, s->curve2); |
| out->pts = s->pts; |
| s->pts += av_rescale_q(s->nb_samples, |
| (AVRational){ 1, outlink->sample_rate }, outlink->time_base); |
| ret = ff_filter_frame(outlink, out); |
| if (ret < 0) |
| goto fail; |
| } |
| |
| nb_samples = av_audio_fifo_size(s->fifo[1]); |
| if (nb_samples > 0) { |
| out = ff_get_audio_buffer(outlink, nb_samples); |
| if (!out) { |
| ret = AVERROR(ENOMEM); |
| goto fail; |
| } |
| |
| av_audio_fifo_read(s->fifo[1], (void **)out->extended_data, nb_samples); |
| out->pts = s->pts; |
| s->pts += av_rescale_q(nb_samples, |
| (AVRational){ 1, outlink->sample_rate }, outlink->time_base); |
| ret = ff_filter_frame(outlink, out); |
| } |
| s->crossfade_is_over = 1; |
| } |
| |
| fail: |
| av_frame_free(&in); |
| av_frame_free(&cf[0]); |
| av_frame_free(&cf[1]); |
| return ret; |
| } |
| |
| static int acrossfade_request_frame(AVFilterLink *outlink) |
| { |
| AVFilterContext *ctx = outlink->src; |
| AudioFadeContext *s = ctx->priv; |
| int ret = 0; |
| |
| if (!s->cf0_eof) { |
| AVFilterLink *cf0 = ctx->inputs[0]; |
| ret = ff_request_frame(cf0); |
| if (ret < 0 && ret != AVERROR_EOF) |
| return ret; |
| if (ret == AVERROR_EOF) { |
| s->cf0_eof = 1; |
| ret = 0; |
| } |
| } else { |
| AVFilterLink *cf1 = ctx->inputs[1]; |
| int nb_samples = av_audio_fifo_size(s->fifo[1]); |
| |
| ret = ff_request_frame(cf1); |
| if (ret == AVERROR_EOF && nb_samples > 0) { |
| AVFrame *out = ff_get_audio_buffer(outlink, nb_samples); |
| if (!out) |
| return AVERROR(ENOMEM); |
| |
| av_audio_fifo_read(s->fifo[1], (void **)out->extended_data, nb_samples); |
| ret = ff_filter_frame(outlink, out); |
| } |
| } |
| |
| return ret; |
| } |
| |
| static int acrossfade_config_output(AVFilterLink *outlink) |
| { |
| AVFilterContext *ctx = outlink->src; |
| AudioFadeContext *s = ctx->priv; |
| |
| if (ctx->inputs[0]->sample_rate != ctx->inputs[1]->sample_rate) { |
| av_log(ctx, AV_LOG_ERROR, |
| "Inputs must have the same sample rate " |
| "%d for in0 vs %d for in1\n", |
| ctx->inputs[0]->sample_rate, ctx->inputs[1]->sample_rate); |
| return AVERROR(EINVAL); |
| } |
| |
| outlink->sample_rate = ctx->inputs[0]->sample_rate; |
| outlink->time_base = ctx->inputs[0]->time_base; |
| outlink->channel_layout = ctx->inputs[0]->channel_layout; |
| outlink->channels = ctx->inputs[0]->channels; |
| |
| switch (outlink->format) { |
| case AV_SAMPLE_FMT_DBL: s->crossfade_samples = crossfade_samples_dbl; break; |
| case AV_SAMPLE_FMT_DBLP: s->crossfade_samples = crossfade_samples_dblp; break; |
| case AV_SAMPLE_FMT_FLT: s->crossfade_samples = crossfade_samples_flt; break; |
| case AV_SAMPLE_FMT_FLTP: s->crossfade_samples = crossfade_samples_fltp; break; |
| case AV_SAMPLE_FMT_S16: s->crossfade_samples = crossfade_samples_s16; break; |
| case AV_SAMPLE_FMT_S16P: s->crossfade_samples = crossfade_samples_s16p; break; |
| case AV_SAMPLE_FMT_S32: s->crossfade_samples = crossfade_samples_s32; break; |
| case AV_SAMPLE_FMT_S32P: s->crossfade_samples = crossfade_samples_s32p; break; |
| } |
| |
| config_output(outlink); |
| |
| s->fifo[0] = av_audio_fifo_alloc(outlink->format, outlink->channels, s->nb_samples); |
| s->fifo[1] = av_audio_fifo_alloc(outlink->format, outlink->channels, s->nb_samples); |
| if (!s->fifo[0] || !s->fifo[1]) |
| return AVERROR(ENOMEM); |
| |
| return 0; |
| } |
| |
| static av_cold void uninit(AVFilterContext *ctx) |
| { |
| AudioFadeContext *s = ctx->priv; |
| |
| av_audio_fifo_free(s->fifo[0]); |
| av_audio_fifo_free(s->fifo[1]); |
| } |
| |
| static const AVFilterPad avfilter_af_acrossfade_inputs[] = { |
| { |
| .name = "crossfade0", |
| .type = AVMEDIA_TYPE_AUDIO, |
| .filter_frame = acrossfade_filter_frame, |
| }, |
| { |
| .name = "crossfade1", |
| .type = AVMEDIA_TYPE_AUDIO, |
| .filter_frame = acrossfade_filter_frame, |
| }, |
| { NULL } |
| }; |
| |
| static const AVFilterPad avfilter_af_acrossfade_outputs[] = { |
| { |
| .name = "default", |
| .type = AVMEDIA_TYPE_AUDIO, |
| .request_frame = acrossfade_request_frame, |
| .config_props = acrossfade_config_output, |
| }, |
| { NULL } |
| }; |
| |
| AVFilter ff_af_acrossfade = { |
| .name = "acrossfade", |
| .description = NULL_IF_CONFIG_SMALL("Cross fade two input audio streams."), |
| .query_formats = query_formats, |
| .priv_size = sizeof(AudioFadeContext), |
| .uninit = uninit, |
| .priv_class = &acrossfade_class, |
| .inputs = avfilter_af_acrossfade_inputs, |
| .outputs = avfilter_af_acrossfade_outputs, |
| }; |
| |
| #endif /* CONFIG_ACROSSFADE_FILTER */ |
