diff options
Diffstat (limited to 'audio')
-rw-r--r-- | audio/aconverter.c | 653 | ||||
-rw-r--r-- | audio/aconverter.h | 41 | ||||
-rw-r--r-- | audio/aframe.c | 130 | ||||
-rw-r--r-- | audio/aframe.h | 11 | ||||
-rw-r--r-- | audio/audio.c | 625 | ||||
-rw-r--r-- | audio/audio.h | 102 | ||||
-rw-r--r-- | audio/filter/af.c | 824 | ||||
-rw-r--r-- | audio/filter/af.h | 163 | ||||
-rw-r--r-- | audio/filter/af_format.c | 173 | ||||
-rw-r--r-- | audio/filter/af_lavcac3enc.c | 504 | ||||
-rw-r--r-- | audio/filter/af_lavfi.c | 413 | ||||
-rw-r--r-- | audio/filter/af_lavrresample.c | 187 | ||||
-rw-r--r-- | audio/filter/af_rubberband.c | 446 | ||||
-rw-r--r-- | audio/filter/af_scaletempo.c | 577 | ||||
-rw-r--r-- | audio/filter/tools.c | 72 | ||||
-rw-r--r-- | audio/format.c | 2 | ||||
-rw-r--r-- | audio/format.h | 1 |
17 files changed, 1119 insertions, 3805 deletions
diff --git a/audio/aconverter.c b/audio/aconverter.c deleted file mode 100644 index 2475df878d..0000000000 --- a/audio/aconverter.c +++ /dev/null @@ -1,653 +0,0 @@ -/* - * This file is part of mpv. - * - * mpv 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. - * - * mpv 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 mpv. If not, see <http://www.gnu.org/licenses/>. - */ - -#include <libavutil/opt.h> -#include <libavutil/common.h> -#include <libavutil/samplefmt.h> -#include <libavutil/channel_layout.h> -#include <libavutil/mathematics.h> - -#include "config.h" - -#include "common/common.h" -#include "common/av_common.h" -#include "common/msg.h" -#include "options/m_config.h" -#include "options/m_option.h" -#include "aconverter.h" -#include "aframe.h" -#include "fmt-conversion.h" -#include "format.h" - -#define HAVE_LIBSWRESAMPLE (!HAVE_LIBAV) -#define HAVE_LIBAVRESAMPLE HAVE_LIBAV - -#if HAVE_LIBAVRESAMPLE -#include <libavresample/avresample.h> -#elif HAVE_LIBSWRESAMPLE -#include <libswresample/swresample.h> -#define AVAudioResampleContext SwrContext -#define avresample_alloc_context swr_alloc -#define avresample_open swr_init -#define avresample_close(x) do { } while(0) -#define avresample_free swr_free -#define avresample_available(x) 0 -#define avresample_convert(ctx, out, out_planesize, out_samples, in, in_planesize, in_samples) \ - swr_convert(ctx, out, out_samples, (const uint8_t**)(in), in_samples) -#define avresample_set_channel_mapping swr_set_channel_mapping -#define avresample_set_compensation swr_set_compensation -#else -#error "config.h broken or no resampler found" -#endif - -struct mp_aconverter { - struct mp_log *log; - struct mpv_global *global; - double playback_speed; - bool is_resampling; - bool passthrough_mode; - struct AVAudioResampleContext *avrctx; - struct mp_aframe *avrctx_fmt; // output format of avrctx - struct mp_aframe *pool_fmt; // format used to allocate frames for avrctx output - struct mp_aframe *pre_out_fmt; // format before final conversion - struct AVAudioResampleContext *avrctx_out; // for output channel reordering - const struct mp_resample_opts *opts; // opts requested by the user - // At least libswresample keeps a pointer around for this: - int reorder_in[MP_NUM_CHANNELS]; - int reorder_out[MP_NUM_CHANNELS]; - struct mp_aframe_pool *reorder_buffer; - struct mp_aframe_pool *out_pool; - - int in_rate_user; // user input sample rate - int in_rate; // actual rate (used by lavr), adjusted for playback speed - int in_format; - struct mp_chmap in_channels; - int out_rate; - int out_format; - struct mp_chmap out_channels; - - struct mp_aframe *input; // queued input frame - bool input_eof; // queued input EOF - struct mp_aframe *output; // queued output frame - bool output_eof; // queued output EOF -}; - -#define OPT_BASE_STRUCT struct mp_resample_opts -const struct m_sub_options resample_config = { - .opts = (const m_option_t[]) { - OPT_INTRANGE("audio-resample-filter-size", filter_size, 0, 0, 32), - OPT_INTRANGE("audio-resample-phase-shift", phase_shift, 0, 0, 30), - OPT_FLAG("audio-resample-linear", linear, 0), - OPT_DOUBLE("audio-resample-cutoff", cutoff, M_OPT_RANGE, - .min = 0, .max = 1), - OPT_FLAG("audio-normalize-downmix", normalize, 0), - OPT_KEYVALUELIST("audio-swresample-o", avopts, 0), - {0} - }, - .size = sizeof(struct mp_resample_opts), - .defaults = &(const struct mp_resample_opts)MP_RESAMPLE_OPTS_DEF, - .change_flags = UPDATE_AUDIO, -}; - -#if HAVE_LIBAVRESAMPLE -static double get_delay(struct mp_aconverter *p) -{ - return avresample_get_delay(p->avrctx) / (double)p->in_rate + - avresample_available(p->avrctx) / (double)p->out_rate; -} -static int get_out_samples(struct mp_aconverter *p, int in_samples) -{ - return avresample_get_out_samples(p->avrctx, in_samples); -} -#else -static double get_delay(struct mp_aconverter *p) -{ - int64_t base = p->in_rate * (int64_t)p->out_rate; - return swr_get_delay(p->avrctx, base) / (double)base; -} -static int get_out_samples(struct mp_aconverter *p, int in_samples) -{ - return swr_get_out_samples(p->avrctx, in_samples); -} -#endif - -static void close_lavrr(struct mp_aconverter *p) -{ - if (p->avrctx) - avresample_close(p->avrctx); - avresample_free(&p->avrctx); - if (p->avrctx_out) - avresample_close(p->avrctx_out); - avresample_free(&p->avrctx_out); - - TA_FREEP(&p->pre_out_fmt); - TA_FREEP(&p->avrctx_fmt); - TA_FREEP(&p->pool_fmt); -} - -static int rate_from_speed(int rate, double speed) -{ - return lrint(rate * speed); -} - -static struct mp_chmap fudge_pairs[][2] = { - {MP_CHMAP2(BL, BR), MP_CHMAP2(SL, SR)}, - {MP_CHMAP2(SL, SR), MP_CHMAP2(BL, BR)}, - {MP_CHMAP2(SDL, SDR), MP_CHMAP2(SL, SR)}, - {MP_CHMAP2(SL, SR), MP_CHMAP2(SDL, SDR)}, -}; - -// Modify out_layout and return the new value. The intention is reducing the -// loss libswresample's rematrixing will cause by exchanging similar, but -// strictly speaking incompatible channel pairs. For example, 7.1 should be -// changed to 7.1(wide) without dropping the SL/SR channels. (We still leave -// it to libswresample to create the remix matrix.) -static uint64_t fudge_layout_conversion(struct mp_aconverter *p, - uint64_t in, uint64_t out) -{ - for (int n = 0; n < MP_ARRAY_SIZE(fudge_pairs); n++) { - uint64_t a = mp_chmap_to_lavc(&fudge_pairs[n][0]); - uint64_t b = mp_chmap_to_lavc(&fudge_pairs[n][1]); - if ((in & a) == a && (in & b) == 0 && - (out & a) == 0 && (out & b) == b) - { - out = (out & ~b) | a; - - MP_VERBOSE(p, "Fudge: %s -> %s\n", - mp_chmap_to_str(&fudge_pairs[n][0]), - mp_chmap_to_str(&fudge_pairs[n][1])); - } - } - return out; -} - -// mp_chmap_get_reorder() performs: -// to->speaker[n] = from->speaker[src[n]] -// but libavresample does: -// to->speaker[dst[n]] = from->speaker[n] -static void transpose_order(int *map, int num) -{ - int nmap[MP_NUM_CHANNELS] = {0}; - for (int n = 0; n < num; n++) { - for (int i = 0; i < num; i++) { - if (map[n] == i) - nmap[i] = n; - } - } - memcpy(map, nmap, sizeof(nmap)); -} - -static bool configure_lavrr(struct mp_aconverter *p, bool verbose) -{ - close_lavrr(p); - - p->in_rate = rate_from_speed(p->in_rate_user, p->playback_speed); - - p->passthrough_mode = p->opts->allow_passthrough && - p->in_rate == p->out_rate && - p->in_format == p->out_format && - mp_chmap_equals(&p->in_channels, &p->out_channels); - - if (p->passthrough_mode) - return true; - - p->avrctx = avresample_alloc_context(); - p->avrctx_out = avresample_alloc_context(); - if (!p->avrctx || !p->avrctx_out) - goto error; - - enum AVSampleFormat in_samplefmt = af_to_avformat(p->in_format); - enum AVSampleFormat out_samplefmt = af_to_avformat(p->out_format); - enum AVSampleFormat out_samplefmtp = av_get_planar_sample_fmt(out_samplefmt); - - if (in_samplefmt == AV_SAMPLE_FMT_NONE || - out_samplefmt == AV_SAMPLE_FMT_NONE || - out_samplefmtp == AV_SAMPLE_FMT_NONE) - goto error; - - av_opt_set_int(p->avrctx, "filter_size", p->opts->filter_size, 0); - av_opt_set_int(p->avrctx, "phase_shift", p->opts->phase_shift, 0); - av_opt_set_int(p->avrctx, "linear_interp", p->opts->linear, 0); - - double cutoff = p->opts->cutoff; - if (cutoff <= 0.0) - cutoff = MPMAX(1.0 - 6.5 / (p->opts->filter_size + 8), 0.80); - av_opt_set_double(p->avrctx, "cutoff", cutoff, 0); - - int normalize = p->opts->normalize; -#if HAVE_LIBSWRESAMPLE - av_opt_set_double(p->avrctx, "rematrix_maxval", normalize ? 1 : 1000, 0); -#else - av_opt_set_int(p->avrctx, "normalize_mix_level", !!normalize, 0); -#endif - - if (mp_set_avopts(p->log, p->avrctx, p->opts->avopts) < 0) - goto error; - - struct mp_chmap map_in = p->in_channels; - struct mp_chmap map_out = p->out_channels; - - // Try not to do any remixing if at least one is "unknown". Some corner - // cases also benefit from disabling all channel handling logic if the - // src/dst layouts are the same (like fl-fr-na -> fl-fr-na). - if (mp_chmap_is_unknown(&map_in) || mp_chmap_is_unknown(&map_out) || - mp_chmap_equals(&map_in, &map_out)) - { - mp_chmap_set_unknown(&map_in, map_in.num); - mp_chmap_set_unknown(&map_out, map_out.num); - } - - // unchecked: don't take any channel reordering into account - uint64_t in_ch_layout = mp_chmap_to_lavc_unchecked(&map_in); - uint64_t out_ch_layout = mp_chmap_to_lavc_unchecked(&map_out); - - struct mp_chmap in_lavc, out_lavc; - mp_chmap_from_lavc(&in_lavc, in_ch_layout); - mp_chmap_from_lavc(&out_lavc, out_ch_layout); - - if (verbose && !mp_chmap_equals(&in_lavc, &out_lavc)) { - MP_VERBOSE(p, "Remix: %s -> %s\n", mp_chmap_to_str(&in_lavc), - mp_chmap_to_str(&out_lavc)); - } - - if (in_lavc.num != map_in.num) { - // For handling NA channels, we would have to add a planarization step. - MP_FATAL(p, "Unsupported input channel layout %s.\n", - mp_chmap_to_str(&map_in)); - goto error; - } - - mp_chmap_get_reorder(p->reorder_in, &map_in, &in_lavc); - transpose_order(p->reorder_in, map_in.num); - - if (mp_chmap_equals(&out_lavc, &map_out)) { - // No intermediate step required - output new format directly. - out_samplefmtp = out_samplefmt; - } else { - // Verify that we really just reorder and/or insert NA channels. - struct mp_chmap withna = out_lavc; - mp_chmap_fill_na(&withna, map_out.num); - if (withna.num != map_out.num) - goto error; - } - mp_chmap_get_reorder(p->reorder_out, &out_lavc, &map_out); - - p->pre_out_fmt = mp_aframe_create(); - mp_aframe_set_rate(p->pre_out_fmt, p->out_rate); - mp_aframe_set_chmap(p->pre_out_fmt, &p->out_channels); - mp_aframe_set_format(p->pre_out_fmt, p->out_format); - - p->avrctx_fmt = mp_aframe_create(); - mp_aframe_config_copy(p->avrctx_fmt, p->pre_out_fmt); - mp_aframe_set_chmap(p->avrctx_fmt, &out_lavc); - mp_aframe_set_format(p->avrctx_fmt, af_from_avformat(out_samplefmtp)); - - // If there are NA channels, the final output will have more channels than - // the avrctx output. Also, avrctx will output planar (out_samplefmtp was - // not overwritten). Allocate the output frame with more channels, so the - // NA channels can be trivially added. - p->pool_fmt = mp_aframe_create(); - mp_aframe_config_copy(p->pool_fmt, p->avrctx_fmt); - if (map_out.num > out_lavc.num) - mp_aframe_set_chmap(p->pool_fmt, &map_out); - - out_ch_layout = fudge_layout_conversion(p, in_ch_layout, out_ch_layout); - - // Real conversion; output is input to avrctx_out. - av_opt_set_int(p->avrctx, "in_channel_layout", in_ch_layout, 0); - av_opt_set_int(p->avrctx, "out_channel_layout", out_ch_layout, 0); - av_opt_set_int(p->avrctx, "in_sample_rate", p->in_rate, 0); - av_opt_set_int(p->avrctx, "out_sample_rate", p->out_rate, 0); - av_opt_set_int(p->avrctx, "in_sample_fmt", in_samplefmt, 0); - av_opt_set_int(p->avrctx, "out_sample_fmt", out_samplefmtp, 0); - - // Just needs the correct number of channels for deplanarization. - struct mp_chmap fake_chmap; - mp_chmap_set_unknown(&fake_chmap, map_out.num); - uint64_t fake_out_ch_layout = mp_chmap_to_lavc_unchecked(&fake_chmap); - if (!fake_out_ch_layout) - goto error; - av_opt_set_int(p->avrctx_out, "in_channel_layout", fake_out_ch_layout, 0); - av_opt_set_int(p->avrctx_out, "out_channel_layout", fake_out_ch_layout, 0); - - av_opt_set_int(p->avrctx_out, "in_sample_fmt", out_samplefmtp, 0); - av_opt_set_int(p->avrctx_out, "out_sample_fmt", out_samplefmt, 0); - av_opt_set_int(p->avrctx_out, "in_sample_rate", p->out_rate, 0); - av_opt_set_int(p->avrctx_out, "out_sample_rate", p->out_rate, 0); - - // API has weird requirements, quoting avresample.h: - // * This function can only be called when the allocated context is not open. - // * Also, the input channel layout must have already been set. - avresample_set_channel_mapping(p->avrctx, p->reorder_in); - - p->is_resampling = false; - - if (avresample_open(p->avrctx) < 0 || avresample_open(p->avrctx_out) < 0) { - MP_ERR(p, "Cannot open Libavresample context.\n"); - goto error; - } - return true; - -error: - close_lavrr(p); - return false; -} - -bool mp_aconverter_reconfig(struct mp_aconverter *p, - int in_rate, int in_format, struct mp_chmap in_channels, - int out_rate, int out_format, struct mp_chmap out_channels) -{ - close_lavrr(p); - - TA_FREEP(&p->input); - TA_FREEP(&p->output); - p->input_eof = p->output_eof = false; - - p->playback_speed = 1.0; - - p->in_rate_user = in_rate; - p->in_format = in_format; - p->in_channels = in_channels; - p->out_rate = out_rate; - p->out_format = out_format; - p->out_channels = out_channels; - - return configure_lavrr(p, true); -} - -void mp_aconverter_flush(struct mp_aconverter *p) -{ - if (!p->avrctx) - return; -#if HAVE_LIBSWRESAMPLE - swr_close(p->avrctx); - if (swr_init(p->avrctx) < 0) - close_lavrr(p); -#else - while (avresample_read(p->avrctx, NULL, 1000) > 0) {} -#endif -} - -void mp_aconverter_set_speed(struct mp_aconverter *p, double speed) -{ - p->playback_speed = speed; -} - -static void extra_output_conversion(struct mp_aframe *mpa) -{ - int format = af_fmt_from_planar(mp_aframe_get_format(mpa)); - int num_planes = mp_aframe_get_planes(mpa); - uint8_t **planes = mp_aframe_get_data_rw(mpa); - if (!planes) - return; - for (int p = 0; p < num_planes; p++) { - void *ptr = planes[p]; - int total = mp_aframe_get_total_plane_samples(mpa); - if (format == AF_FORMAT_FLOAT) { - for (int s = 0; s < total; s++) - ((float *)ptr)[s] = av_clipf(((float *)ptr)[s], -1.0f, 1.0f); - } else if (format == AF_FORMAT_DOUBLE) { - for (int s = 0; s < total; s++) - ((double *)ptr)[s] = MPCLAMP(((double *)ptr)[s], -1.0, 1.0); - } - } -} - -// This relies on the tricky way mpa was allocated. -static bool reorder_planes(struct mp_aframe *mpa, int *reorder, - struct mp_chmap *newmap) -{ - if (!mp_aframe_set_chmap(mpa, newmap)) - return false; - - int num_planes = newmap->num; - uint8_t **planes = mp_aframe_get_data_rw(mpa); - uint8_t *old_planes[MP_NUM_CHANNELS]; - assert(num_planes <= MP_NUM_CHANNELS); - for (int n = 0; n < num_planes; n++) - old_planes[n] = planes[n]; - - int next_na = 0; - for (int n = 0; n < num_planes; n++) - next_na += newmap->speaker[n] != MP_SPEAKER_ID_NA; - - for (int n = 0; n < num_planes; n++) { - int src = reorder[n]; - assert(src >= -1 && src < num_planes); - if (src >= 0) { - planes[n] = old_planes[src]; - } else { - assert(next_na < num_planes); - planes[n] = old_planes[next_na++]; - // The NA planes were never written by avrctx, so clear them. - af_fill_silence(planes[n], - mp_aframe_get_sstride(mpa) * mp_aframe_get_size(mpa), - mp_aframe_get_format(mpa)); - } - } - - return true; -} - -static int resample_frame(struct AVAudioResampleContext *r, - struct mp_aframe *out, struct mp_aframe *in) -{ - // Be aware that the channel layout and count can be different for in and - // out frames. In some situations the caller will fix up the frames before - // or after conversion. The sample rates can also be different. - AVFrame *av_i = in ? mp_aframe_get_raw_avframe(in) : NULL; - AVFrame *av_o = out ? mp_aframe_get_raw_avframe(out) : NULL; - return avresample_convert(r, - av_o ? av_o->extended_data : NULL, - av_o ? av_o->linesize[0] : 0, - av_o ? av_o->nb_samples : 0, - av_i ? av_i->extended_data : NULL, - av_i ? av_i->linesize[0] : 0, - av_i ? av_i->nb_samples : 0); -} - -static void filter_resample(struct mp_aconverter *p, struct mp_aframe *in) -{ - struct mp_aframe *out = NULL; - - if (!p->avrctx) - goto error; - - int samples = get_out_samples(p, in ? mp_aframe_get_size(in) : 0); - out = mp_aframe_create(); - mp_aframe_config_copy(out, p->pool_fmt); - if (mp_aframe_pool_allocate(p->out_pool, out, samples) < 0) - goto error; - - int out_samples = 0; - if (samples) { - out_samples = resample_frame(p->avrctx, out, in); - if (out_samples < 0 || out_samples > samples) - goto error; - mp_aframe_set_size(out, out_samples); - } - - struct mp_chmap out_chmap; - if (!mp_aframe_get_chmap(p->pool_fmt, &out_chmap)) - goto error; - if (!reorder_planes(out, p->reorder_out, &out_chmap)) - goto error; - - if (!mp_aframe_config_equals(out, p->pre_out_fmt)) { - struct mp_aframe *new = mp_aframe_create(); - mp_aframe_config_copy(new, p->pre_out_fmt); - if (mp_aframe_pool_allocate(p->reorder_buffer, new, out_samples) < 0) { - talloc_free(new); - goto error; - } - int got = 0; - if (out_samples) - got = resample_frame(p->avrctx_out, new, out); - talloc_free(out); - out = new; - if (got != out_samples) - goto error; - } - - extra_output_conversion(out); - - if (in) - mp_aframe_copy_attributes(out, in); - - if (out_samples) { - p->output = out; - } else { - talloc_free(out); - } - p->output_eof = !in; // we've read everything - - return; -error: - talloc_free(out); - MP_ERR(p, "Error on resampling.\n"); -} - -static void filter(struct mp_aconverter *p) -{ - if (p->output || p->output_eof || !(p->input || p->input_eof)) - return; - - int new_rate = rate_from_speed(p->in_rate_user, p->playback_speed); - - if (p->passthrough_mode && new_rate != p->in_rate) - configure_lavrr(p, false); - - if (p->passthrough_mode) { - p->output = p->input; - p->input = NULL; - p->output_eof = p->input_eof; - p->input_eof = false; - return; - } - - if (p->avrctx && !(!p->is_resampling && new_rate == p->in_rate)) { - AVRational r = av_d2q(p->playback_speed * p->in_rate_user / p->in_rate, - INT_MAX / 2); - // Essentially, swr/avresample_set_compensation() does 2 things: - // - adjust output sample rate by sample_delta/compensation_distance - // - reset the adjustment after compensation_distance output samples - // Increase the compensation_distance to avoid undesired reset - // semantics - we want to keep the ratio for the whole frame we're - // feeding it, until the next filter() call. - int mult = INT_MAX / 2 / MPMAX(MPMAX(abs(r.num), abs(r.den)), 1); - r = (AVRational){ r.num * mult, r.den * mult }; - if (avresample_set_compensation(p->avrctx, r.den - r.num, r.den) >= 0) { - new_rate = p->in_rate; - p->is_resampling = true; - } - } - - bool need_reinit = fabs(new_rate / (double)p->in_rate - 1) > 0.01; - if (need_reinit && new_rate != p->in_rate) { - // Before reconfiguring, drain the audio that is still buffered - // in the resampler. - filter_resample(p, NULL); - // Reinitialize resampler. - configure_lavrr(p, false); - p->output_eof = false; - if (p->output) - return; // need to read output before continuing filtering - } - - filter_resample(p, p->input); - TA_FREEP(&p->input); - p->input_eof = false; -} - -// Queue input. If true, ownership of in passes to mp_aconverted and the input -// was accepted. Otherwise, return false and reject in. -// in==NULL means trigger EOF. -bool mp_aconverter_write_input(struct mp_aconverter *p, struct mp_aframe *in) -{ - if (p->input || p->input_eof) - return false; - - p->input = in; - p->input_eof = !in; - return true; -} - -// Return output frame, or NULL if nothing available. -// *eof is set to true if NULL is returned, and it was due to EOF. -struct mp_aframe *mp_aconverter_read_output(struct mp_aconverter *p, bool *eof) -{ - *eof = false; - - filter(p); - - if (p->output) { - struct mp_aframe *out = p->output; - p->output = NULL; - return out; - } - - *eof = p->output_eof; - p->output_eof = false; - return NULL; -} - -double mp_aconverter_get_latency(struct mp_aconverter *p) -{ - double delay = get_delay(p); - - if (p->input) - delay += mp_aframe_duration(p->input); - - // In theory this is influenced by playback speed, but other parts of the - // player get it wrong anyway. - if (p->output) - delay += mp_aframe_duration(p->output); - - return delay; -} - -static void destroy_aconverter(void *ptr) -{ - struct mp_aconverter *p = ptr; - - close_lavrr(p); - - talloc_free(p->input); - talloc_free(p->output); -} - -// If opts is not NULL, the pointer must be valid for the lifetime of the -// mp_aconverter. -struct mp_aconverter *mp_aconverter_create(struct mpv_global *global, - struct mp_log *log, - const struct mp_resample_opts *opts) -{ - struct mp_aconverter *p = talloc_zero(NULL, struct mp_aconverter); - p->log = log; - p->global = global; - - p->opts = opts; - if (!p->opts) - p->opts = mp_get_config_group(p, global, &resample_config); - - p->reorder_buffer = mp_aframe_pool_create(p); - p->out_pool = mp_aframe_pool_create(p); - - talloc_set_destructor(p, destroy_aconverter); - - return p; -} diff --git a/audio/aconverter.h b/audio/aconverter.h deleted file mode 100644 index 22ca93e4c1..0000000000 --- a/audio/aconverter.h +++ /dev/null @@ -1,41 +0,0 @@ -#pragma once - -#include <stdbool.h> - -#include "chmap.h" - -struct mp_aconverter; -struct mp_aframe; -struct mpv_global; -struct mp_log; - -struct mp_resample_opts { - int filter_size; - int phase_shift; - int linear; - double cutoff; - int normalize; - int allow_passthrough; - char **avopts; -}; - -#define MP_RESAMPLE_OPTS_DEF { \ - .filter_size = 16, \ - .cutoff = 0.0, \ - .phase_shift = 10, \ - .normalize = 0, \ - } - -extern const struct m_sub_options resample_config; - -struct mp_aconverter *mp_aconverter_create(struct mpv_global *global, - struct mp_log *log, - const struct mp_resample_opts *opts); -bool mp_aconverter_reconfig(struct mp_aconverter *p, - int in_rate, int in_format, struct mp_chmap in_channels, - int out_rate, int out_format, struct mp_chmap out_channels); -void mp_aconverter_flush(struct mp_aconverter *p); -void mp_aconverter_set_speed(struct mp_aconverter *p, double speed); -bool mp_aconverter_write_input(struct mp_aconverter *p, struct mp_aframe *in); -struct mp_aframe *mp_aconverter_read_output(struct mp_aconverter *p, bool *eof); -double mp_aconverter_get_latency(struct mp_aconverter *p); diff --git a/audio/aframe.c b/audio/aframe.c index 1f053a6715..9115cf67fd 100644 --- a/audio/aframe.c +++ b/audio/aframe.c @@ -32,6 +32,11 @@ struct mp_aframe { // We support spdif formats, which are allocated as AV_SAMPLE_FMT_S16. int format; double pts; + double speed; +}; + +struct avframe_opaque { + double speed; }; static void free_frame(void *ptr) @@ -43,11 +48,11 @@ static void free_frame(void *ptr) struct mp_aframe *mp_aframe_create(void) { struct mp_aframe *frame = talloc_zero(NULL, struct mp_aframe); - frame->pts = MP_NOPTS_VALUE; frame->av_frame = av_frame_alloc(); if (!frame->av_frame) abort(); talloc_set_destructor(frame, free_frame); + mp_aframe_reset(frame); return frame; } @@ -61,6 +66,7 @@ struct mp_aframe *mp_aframe_new_ref(struct mp_aframe *frame) dst->chmap = frame->chmap; dst->format = frame->format; dst->pts = frame->pts; + dst->speed = frame->speed; if (mp_aframe_is_allocated(frame)) { if (av_frame_ref(dst->av_frame, frame->av_frame) < 0) @@ -80,6 +86,7 @@ void mp_aframe_reset(struct mp_aframe *frame) frame->chmap.num = 0; frame->format = 0; frame->pts = MP_NOPTS_VALUE; + frame->speed = 1.0; } // Remove all actual audio data and leave only the metadata. @@ -120,6 +127,11 @@ struct mp_aframe *mp_aframe_from_avframe(struct AVFrame *av_frame) mp_chmap_from_channels(&frame->chmap, av_frame->channels); #endif + if (av_frame->opaque_ref) { + struct avframe_opaque *op = (void *)av_frame->opaque_ref->data; + frame->speed = op->speed; + } + return frame; } @@ -137,6 +149,16 @@ struct AVFrame *mp_aframe_to_avframe(struct mp_aframe *frame) if (!mp_chmap_is_lavc(&frame->chmap)) return NULL; + if (!frame->av_frame->opaque_ref && frame->speed != 1.0) { + frame->av_frame->opaque_ref = + av_buffer_alloc(sizeof(struct avframe_opaque)); + if (!frame->av_frame->opaque_ref) + return NULL; + + struct avframe_opaque *op = (void *)frame->av_frame->opaque_ref->data; + op->speed = frame->speed; + } + return av_frame_clone(frame->av_frame); } @@ -183,6 +205,7 @@ void mp_aframe_config_copy(struct mp_aframe *dst, struct mp_aframe *src) void mp_aframe_copy_attributes(struct mp_aframe *dst, struct mp_aframe *src) { dst->pts = src->pts; + dst->speed = src->speed; int rate = dst->av_frame->sample_rate; @@ -316,6 +339,37 @@ void mp_aframe_set_pts(struct mp_aframe *frame, double pts) frame->pts = pts; } +// Set a speed factor. This is multiplied with the sample rate to get the +// "effective" samplerate (mp_aframe_get_effective_rate()), which will be used +// to do PTS calculations. If speed!=1.0, the PTS values always refer to the +// original PTS (before changing speed), and if you want reasonably continuous +// PTS between frames, you need to use the effective samplerate. +void mp_aframe_set_speed(struct mp_aframe *frame, double factor) +{ + frame->speed = factor; +} + +// Adjust current speed factor. +void mp_aframe_mul_speed(struct mp_aframe *frame, double factor) +{ + frame->speed *= factor; +} + +double mp_aframe_get_speed(struct mp_aframe *frame) +{ + return frame->speed; +} + +// Matters for speed changed frames (such as a frame which has been resampled +// to play at a different speed). +// Return the sample rate at which the frame would have to be played to result +// in the same duration as the original frame before the speed change. +// This is used for A/V sync. +double mp_aframe_get_effective_rate(struct mp_aframe *frame) +{ + return mp_aframe_get_rate(frame) / frame->speed; +} + // Return number of data pointers. int mp_aframe_get_planes(struct mp_aframe *frame) { @@ -339,6 +393,18 @@ int mp_aframe_get_total_plane_samples(struct mp_aframe *frame) ? 1 : mp_aframe_get_channels(frame)); } +char *mp_aframe_format_str_buf(char *buf, size_t buf_size, struct mp_aframe *fmt) +{ + char ch[128]; + mp_chmap_to_str_buf(ch, sizeof(ch), &fmt->chmap); + char *hr_ch = mp_chmap_to_str_hr(&fmt->chmap); + if (strcmp(hr_ch, ch) != 0) + mp_snprintf_cat(ch, sizeof(ch), " (%s)", hr_ch); + snprintf(buf, buf_size, "%dHz %s %dch %s", fmt->av_frame->sample_rate, + ch, fmt->chmap.num, af_fmt_to_str(fmt->format)); + return buf; +} + // Set data to the audio after the given number of samples (i.e. slice it). void mp_aframe_skip_samples(struct mp_aframe *f, int samples) { @@ -352,25 +418,25 @@ void mp_aframe_skip_samples(struct mp_aframe *f, int samples) f->av_frame->nb_samples -= samples; if (f->pts != MP_NOPTS_VALUE) - f->pts += samples / (double)mp_aframe_get_rate(f); + f->pts += samples / mp_aframe_get_effective_rate(f); } // Return the timestamp of the sample just after the end of this frame. double mp_aframe_end_pts(struct mp_aframe *f) { - int rate = mp_aframe_get_rate(f); - if (f->pts == MP_NOPTS_VALUE || rate < 1) + double rate = mp_aframe_get_effective_rate(f); + if (f->pts == MP_NOPTS_VALUE || rate <= 0) return MP_NOPTS_VALUE; - return f->pts + f->av_frame->nb_samples / (double)rate; + return f->pts + f->av_frame->nb_samples / rate; } // Return the duration in seconds of the frame (0 if invalid). double mp_aframe_duration(struct mp_aframe *f) { - int rate = mp_aframe_get_rate(f); - if (rate < 1) + double rate = mp_aframe_get_effective_rate(f); + if (rate <= 0) return 0; - return f->av_frame->nb_samples / (double)rate; + return f->av_frame->nb_samples / rate; } // Clip the given frame to the given timestamp range. Adjusts the frame size @@ -378,7 +444,7 @@ double mp_aframe_duration(struct mp_aframe *f) void mp_aframe_clip_timestamps(struct mp_aframe *f, double start, double end) { double f_end = mp_aframe_end_pts(f); - int rate = mp_aframe_get_rate(f); + double rate = mp_aframe_get_effective_rate(f); if (f_end == MP_NOPTS_VALUE) return; if (end != MP_NOPTS_VALUE) { @@ -405,6 +471,52 @@ void mp_aframe_clip_timestamps(struct mp_aframe *f, double start, double end) } } +bool mp_aframe_copy_samples(struct mp_aframe *dst, int dst_offset, + struct mp_aframe *src, int src_offset, + int samples) +{ |