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-rw-r--r--audio/aconverter.c653
-rw-r--r--audio/aconverter.h41
-rw-r--r--audio/aframe.c130
-rw-r--r--audio/aframe.h11
-rw-r--r--audio/audio.c625
-rw-r--r--audio/audio.h102
-rw-r--r--audio/filter/af.c824
-rw-r--r--audio/filter/af.h163
-rw-r--r--audio/filter/af_format.c173
-rw-r--r--audio/filter/af_lavcac3enc.c504
-rw-r--r--audio/filter/af_lavfi.c413
-rw-r--r--audio/filter/af_lavrresample.c187
-rw-r--r--audio/filter/af_rubberband.c446
-rw-r--r--audio/filter/af_scaletempo.c577
-rw-r--r--audio/filter/tools.c72
-rw-r--r--audio/format.c2
-rw-r--r--audio/format.h1
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(