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authorwm4 <wm4@nowhere>2017-09-21 11:07:02 +0200
committerwm4 <wm4@nowhere>2017-09-21 12:42:09 +0200
commit3a2d5e68acb2ac0f8b09b896907a692b1c48c6b3 (patch)
tree13a7e321ce6accceca720fb654cdf7ea8eee0afc /audio
parentcaaa1189ba75a7df9a4d02f7747d6c0bf3b05012 (diff)
downloadmpv-3a2d5e68acb2ac0f8b09b896907a692b1c48c6b3.tar.bz2
mpv-3a2d5e68acb2ac0f8b09b896907a692b1c48c6b3.tar.xz
audio: move libswresample wrapper out of audio filter code
Move it from af_lavrresample.c to a new aconverter.c file, which is independent from the filter chain code. It also doesn't use mp_audio, and thus has no GPL dependencies. Preparation for later commits. Not particularly well tested, so have fun.
Diffstat (limited to 'audio')
-rw-r--r--audio/aconverter.c641
-rw-r--r--audio/aconverter.h39
-rw-r--r--audio/aframe.c23
-rw-r--r--audio/aframe.h3
-rw-r--r--audio/audio.c31
-rw-r--r--audio/audio.h1
-rw-r--r--audio/filter/af.c2
-rw-r--r--audio/filter/af.h2
-rw-r--r--audio/filter/af_lavrresample.c492
9 files changed, 773 insertions, 461 deletions
diff --git a/audio/aconverter.c b/audio/aconverter.c
new file mode 100644
index 0000000000..f4efc61dc6
--- /dev/null
+++ b/audio/aconverter.c
@@ -0,0 +1,641 @@
+/*
+ * 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_IS_FFMPEG
+#define HAVE_LIBAVRESAMPLE HAVE_IS_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
+};
+
+#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 global_normalize;
+ mp_read_option_raw(p->global, "audio-normalize-downmix", &m_option_type_flag,
+ &global_normalize);
+ int normalize = p->opts->normalize;
+ if (normalize < 0)
+ normalize = global_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;
+
+ static const struct mp_resample_opts defs = MP_RESAMPLE_OPTS_DEF;
+
+ p->opts = opts ? opts : &defs;
+
+ 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
new file mode 100644
index 0000000000..57c5524c2f
--- /dev/null
+++ b/audio/aconverter.h
@@ -0,0 +1,39 @@
+#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 = -1, \
+ }
+
+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 cf785f6657..4168f92124 100644
--- a/audio/aframe.c
+++ b/audio/aframe.c
@@ -166,10 +166,9 @@ void mp_aframe_config_copy(struct mp_aframe *dst, struct mp_aframe *src)
dst->chmap = src->chmap;
dst->format = src->format;
- dst->pts = src->pts;
- if (av_frame_copy_props(dst->av_frame, src->av_frame) < 0)
- abort();
+ mp_aframe_copy_attributes(dst, src);
+
dst->av_frame->format = src->av_frame->format;
dst->av_frame->channel_layout = src->av_frame->channel_layout;
#if LIBAVUTIL_VERSION_MICRO >= 100
@@ -178,6 +177,16 @@ void mp_aframe_config_copy(struct mp_aframe *dst, struct mp_aframe *src)
#endif
}
+// Copy "soft" attributes from src to dst, excluding things which affect
+// frame allocation and organization.
+void mp_aframe_copy_attributes(struct mp_aframe *dst, struct mp_aframe *src)
+{
+ dst->pts = src->pts;
+
+ if (av_frame_copy_props(dst->av_frame, src->av_frame) < 0)
+ abort();
+}
+
// Return whether a and b use the same physical audio format. Extra metadata
// such as PTS, per-frame signalling, and AVFrame side data is not compared.
bool mp_aframe_config_equals(struct mp_aframe *a, struct mp_aframe *b)
@@ -317,6 +326,14 @@ size_t mp_aframe_get_sstride(struct mp_aframe *frame)
(af_fmt_is_planar(format) ? 1 : mp_aframe_get_channels(frame));
}
+// Return total number of samples on each plane.
+int mp_aframe_get_total_plane_samples(struct mp_aframe *frame)
+{
+ return frame->av_frame->nb_samples *
+ (af_fmt_is_planar(mp_aframe_get_format(frame))
+ ? 1 : mp_aframe_get_channels(frame));
+}
+
// 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)
{
diff --git a/audio/aframe.h b/audio/aframe.h
index 5661178419..9ce19cdee1 100644
--- a/audio/aframe.h
+++ b/audio/aframe.h
@@ -23,6 +23,8 @@ void mp_aframe_config_copy(struct mp_aframe *dst, struct mp_aframe *src);
bool mp_aframe_config_equals(struct mp_aframe *a, struct mp_aframe *b);
bool mp_aframe_config_is_valid(struct mp_aframe *frame);
+void mp_aframe_copy_attributes(struct mp_aframe *dst, struct mp_aframe *src);
+
uint8_t **mp_aframe_get_data_ro(struct mp_aframe *frame);
uint8_t **mp_aframe_get_data_rw(struct mp_aframe *frame);
@@ -40,6 +42,7 @@ bool mp_aframe_set_size(struct mp_aframe *frame, int samples);
void mp_aframe_set_pts(struct mp_aframe *frame, double pts);
int mp_aframe_get_planes(struct mp_aframe *frame);
+int mp_aframe_get_total_plane_samples(struct mp_aframe *frame);
size_t mp_aframe_get_sstride(struct mp_aframe *frame);
void mp_aframe_skip_samples(struct mp_aframe *f, int samples);
diff --git a/audio/audio.c b/audio/audio.c
index 008aa1883b..b636c66620 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -406,6 +406,9 @@ fail:
struct mp_audio *mp_audio_from_aframe(struct mp_aframe *aframe)
{
+ if (!aframe)
+ return NULL;
+
struct AVFrame *av = mp_aframe_get_raw_avframe(aframe);
struct mp_audio *res = mp_audio_from_avframe(av);
if (!res)
@@ -428,6 +431,34 @@ void mp_audio_config_from_aframe(struct mp_audio *dst, struct mp_aframe *src)
dst->rate = mp_aframe_get_rate(src);
}
+struct mp_aframe *mp_audio_to_aframe(struct mp_audio *mpa)
+{
+ if (!mpa)
+ return NULL;
+
+ struct mp_aframe *aframe = mp_aframe_create();
+ struct AVFrame *av = mp_aframe_get_raw_avframe(aframe);
+ mp_aframe_set_format(aframe, mpa->format);
+ mp_aframe_set_chmap(aframe, &mpa->channels);
+ mp_aframe_set_rate(aframe, mpa->rate);
+
+ // bullshit it into ffmpeg-compatible parameters
+ struct mp_audio mpb = *mpa;
+ struct mp_chmap chmap;
+ mp_chmap_set_unknown(&chmap, mpb.channels.num);
+ mp_audio_set_channels(&mpb, &chmap);
+ if (af_fmt_is_spdif(mpb.format))
+ mp_audio_set_format(&mpb, AF_FORMAT_S16);
+
+ // put the reference into av, which magically puts it into aframe
+ // aframe keeps its parameters, so the bullshit doesn't matter
+ if (mp_audio_to_avframe(&mpb, av) < 0) {
+ talloc_free(aframe);
+ return NULL;
+ }
+ return aframe;
+}
+
int mp_audio_to_avframe(struct mp_audio *frame, struct AVFrame *avframe)
{
av_frame_unref(avframe);
diff --git a/audio/audio.h b/audio/audio.h
index a8370a0eb7..a4d9134a20 100644
--- a/audio/audio.h
+++ b/audio/audio.h
@@ -86,6 +86,7 @@ int mp_audio_to_avframe(struct mp_audio *frame, struct AVFrame *avframe);
struct mp_aframe;
struct mp_audio *mp_audio_from_aframe(struct mp_aframe *aframe);
void mp_audio_config_from_aframe(struct mp_audio *dst, struct mp_aframe *src);
+struct mp_aframe *mp_audio_to_aframe(struct mp_audio *mpa);
struct mp_audio_pool;
struct mp_audio_pool *mp_audio_pool_create(void *ta_parent);
diff --git a/audio/filter/af.c b/audio/filter/af.c
index 0df0f28bb6..a76945feea 100644
--- a/audio/filter/af.c
+++ b/audio/filter/af.c
@@ -175,6 +175,7 @@ static struct af_instance *af_create(struct af_stream *s, char *name,
.data = talloc_zero(af, struct mp_audio),
.log = mp_log_new(af, s->log, name),
.opts = s->opts,
+ .global = s->global,
.replaygain_data = s->replaygain_data,
.out_pool = mp_audio_pool_create(af),
};
@@ -546,6 +547,7 @@ struct af_stream *af_new(struct mpv_global *global)
s->first->next = s->last;
s->last->prev = s->first;
s->opts = global->opts;
+ s->global = global;
return s;
}
diff --git a/audio/filter/af.h b/audio/filter/af.h
index 553fc03e32..bbf7b53bad 100644
--- a/audio/filter/af.h
+++ b/audio/filter/af.h
@@ -55,6 +55,7 @@ struct af_instance {
char *full_name;
struct mp_log *log;
struct MPOpts *opts;
+ struct mpv_global *global;
struct replaygain_data *replaygain_data;
int (*control)(struct af_instance *af, int cmd, void *arg);
void (*uninit)(struct af_instance *af);
@@ -98,6 +99,7 @@ struct af_stream {
struct mp_log *log;
struct MPOpts *opts;
+ struct mpv_global *global;
struct replaygain_data *replaygain_data;
};
diff --git a/audio/filter/af_lavrresample.c b/audio/filter/af_lavrresample.c
index c18f8cc28a..55eb6b0f20 100644
--- a/audio/filter/af_lavrresample.c
+++ b/audio/filter/af_lavrresample.c
@@ -27,326 +27,24 @@
#include <math.h>
#include <assert.h>
-#include <libavutil/opt.h>
-#include <libavutil/common.h>
-#include <libavutil/samplefmt.h>
-#include <libavutil/channel_layout.h>
-#include <libavutil/mathematics.h>
-
#include "common/common.h"
#include "config.h"
-#define HAVE_LIBSWRESAMPLE HAVE_IS_FFMPEG
-#define HAVE_LIBAVRESAMPLE HAVE_IS_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
-
#include "common/av_common.h"
#include "common/msg.h"
#include "options/m_option.h"
#include "audio/filter/af.h"
#include "audio/fmt-conversion.h"
#include "osdep/endian.h"
-
-struct af_resample_opts {
- int filter_size;
- int phase_shift;
- int linear;
- double cutoff;
- int normalize;
-};
+#include "audio/aconverter.h"
struct af_resample {
int allow_detach;
- char **avopts;
double playback_speed;
- bool is_resampling;
- struct AVAudioResampleContext *avrctx;
- struct mp_audio avrctx_fmt; // output format of avrctx
- struct mp_audio pool_fmt; // format used to allocate frames for avrctx output
- struct mp_audio pre_out_fmt; // format before final conversion (S24)
- struct AVAudioResampleContext *avrctx_out; // for output channel reordering
- struct af_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_audio_pool *reorder_buffer;
-
- int in_rate_af; // filter 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;
-};
-
-#if HAVE_LIBAVRESAMPLE
-static double get_delay(struct af_resample *s)
-{
- return avresample_get_delay(s->avrctx) / (double)s->in_rate +
- avresample_available(s->avrctx) / (double)s->out_rate;
-}
-static int get_out_samples(struct af_resample *s, int in_samples)
-{
- return avresample_get_out_samples(s->avrctx, in_samples);
-}
-#else
-static double get_delay(struct af_resample *s)
-{
- int64_t base = s->in_rate * (int64_t)s->out_rate;
- return swr_get_delay(s->avrctx, base) / (double)base;
-}
-static int get_out_samples(struct af_resample *s, int in_samples)
-{
- return swr_get_out_samples(s->avrctx, in_samples);
-}
-#endif
-
-static void close_lavrr(struct af_instance *af)
-{
- struct af_resample *s = af->priv;
-
- if (s->avrctx)
- avresample_close(s->avrctx);
- avresample_free(&s->avrctx);
- if (s->avrctx_out)
- avresample_close(s->avrctx_out);
- avresample_free(&s->avrctx_out);
-}
-
-static int resample_frame(struct AVAudioResampleContext *r,
- struct mp_audio *out, struct mp_audio *in)
-{
- return avresample_convert(r,
- out ? (uint8_t **)out->planes : NULL,
- out ? mp_audio_get_allocated_size(out) : 0,
- out ? out->samples : 0,
- in ? (uint8_t **)in->planes : NULL,
- in ? mp_audio_get_allocated_size(in) : 0,
- in ? in->samples : 0);
-}
-
-static double af_resample_default_cutoff(int filter_size)
-{
- return FFMAX(1.0 - 6.5 / (filter_size + 8), 0.80);
-}
-
-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)},
+ struct mp_resample_opts opts;
+ struct mp_aconverter *converter;
};
-// 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 af_instance *af,
- 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(af, "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 int configure_lavrr(struct af_instance *af, struct mp_audio *in,
- struct mp_audio *out, bool verbose)
-{
- struct af_resample *s = af->priv;
-
- close_lavrr(af);
-
- s->avrctx = avresample_alloc_context();
- s->avrctx_out = avresample_alloc_context();
- if (!s->avrctx || !s->avrctx_out)
- goto error;
-
- enum AVSampleFormat in_samplefmt = af_to_avformat(in->format);
- enum AVSampleFormat out_samplefmt = af_to_avformat(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;
-
- s->out_rate = out->rate;
- s->in_rate_af = in->rate;
- s->in_rate = rate_from_speed(in->rate, s->playback_speed);
- s->out_format = out->format;
- s->in_format = in->format;
- s->out_channels= out->channels;
- s->in_channels = in->channels;
-
- av_opt_set_int(s->avrctx, "filter_size", s->opts.filter_size, 0);
- av_opt_set_int(s->avrctx, "phase_shift", s->opts.phase_shift, 0);
- av_opt_set_int(s->avrctx, "linear_interp", s->opts.linear, 0);
-
- av_opt_set_double(s->avrctx, "cutoff", s->opts.cutoff, 0);
-
- int normalize = s->opts.normalize;
- if (normalize < 0)
- normalize = af->opts->audio_normalize;
-#if HAVE_LIBSWRESAMPLE
- av_opt_set_double(s->avrctx, "rematrix_maxval", normalize ? 1 : 1000, 0);
-#else
- av_opt_set_int(s->avrctx, "normalize_mix_level", !!normalize, 0);
-#endif
-
- if (mp_set_avopts(af->log, s->avrctx, s->avopts) < 0)
- goto error;
-
- struct mp_chmap map_in = in->channels;
- struct mp_chmap map_out = out->channels;
-
- // Try not to do any remixing if at least one is "unknown".
- if (mp_chmap_is_unknown(&map_in) || mp_chmap_is_unknown(&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(af, "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(af, "Unsupported channel remapping.\n");
- goto error;
- }
-
- mp_chmap_get_reorder(s->reorder_in, &map_in, &in_lavc);
- transpose_order(s->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(s->reorder_out, &out_lavc, &map_out);
-
- s->avrctx_fmt = *out;
- mp_audio_set_channels(&s->avrctx_fmt, &out_lavc);
- mp_audio_set_format(&s->avrctx_fmt, af_from_avformat(out_samplefmtp));
-
- s->pre_out_fmt = *out;
-
- // 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.
- s->pool_fmt = s->avrctx_fmt;
- if (map_out.num > out_lavc.num)
- mp_audio_set_channels(&s->pool_fmt, &map_out);
-
- out_ch_layout = fudge_layout_conversion(af, in_ch_layout, out_ch_layout);
-
- // Real conversion; output is input to avrctx_out.
- av_opt_set_int(s->avrctx, "in_channel_layout", in_ch_layout, 0);
- av_opt_set_int(s->avrctx, "out_channel_layout", out_ch_layout, 0);
- av_opt_set_int(s->avrctx, "in_sample_rate", s->in_rate, 0);
- av_opt_set_int(s->avrctx, "out_sample_rate", s->out_rate, 0);
- av_opt_set_int(s->avrctx, "in_sample_fmt", in_samplefmt, 0);
- av_opt_set_int(s->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(s->avrctx_out, "in_channel_layout", fake_out_ch_layout, 0);
- av_opt_set_int(s->avrctx_out, "out_channel_layout", fake_out_ch_layout, 0);
-
- av_opt_set_int(s->avrctx_out, "in_sample_fmt", out_samplefmtp, 0);
- av_opt_set_int(s->avrctx_out, "out_sample_fmt", out_samplefmt, 0);
- av_opt_set_int(s->avrctx_out, "in_sample_rate", s->out_rate, 0);
- av_opt_set_int(s->avrctx_out, "out_sample_rate", s->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(s->avrctx, s->reorder_in);
-
- s->is_resampling = false;
-
- if (avresample_open(s->avrctx) < 0 || avresample_open(s->avrctx_out) < 0) {
- MP_ERR(af, "Cannot open Libavresample Context. \n");
- goto error;
- }
- return AF_OK;
-
-error:
- close_lavrr(af);
- return AF_ERROR;
-}
-
-
static int control(struct af_instance *af, int cmd, void *arg)
{
struct af_resample *s = af->priv;
@@ -378,8 +76,12 @@ static int control(struct af_instance *af, int cmd, void *arg)
mp_chmap_equals(&in->channels, &orig_in.channels))
? AF_OK : AF_FALSE;
- if (r == AF_OK)
- r = configure_lavrr(af, in, out, true);
+ if (r == AF_OK) {
+ if (!mp_aconverter_reconfig(s->converter,
+ in->rate, in->format, in->channels,
+ out->rate, out->format, out->channels))
+ r = AF_ERROR;
+ }
return r;
}
case AF_CONTROL_SET_PLAYBACK_SPEED_RESAMPLE: {
@@ -387,17 +89,7 @@ static int control(struct af_instance *af, int cmd, void *arg)
return AF_OK;
}
case AF_CONTROL_RESET:
- if (s->avrctx) {
-#if HAVE_LIBSWRESAMPLE
- swr_close(s->avrctx);
- if (swr_init(s->avrctx) < 0) {
- close_lavrr(af);
- return AF_ERROR;
- }
-#else
- while (avresample_read(s->avrctx, NULL, 1000) > 0) {}
-#endif
- }
+ mp_aconverter_flush(s->converter);
return AF_OK;
}
return AF_UNKNOWN;
@@ -405,149 +97,40 @@ static int control(struct af_instance *af, int cmd, void *arg)
static void uninit(struct af_instance *af)
{
- close_lavrr(af);
-}
-
-// The LSB is always ignored.
-#if BYTE_ORDER == BIG_ENDIAN
-#define SHIFT24(x) ((3-(x))*8)
-#else
-#define SHIFT24(x) (((x)+1)*8)
-#endif
-
-static void extra_output_conversion(struct af_instance *af, struct mp_audio *mpa)
-{
- for (int p = 0; p < mpa->num_planes; p++) {
- void *ptr = mpa->planes[p];
- int total = mpa->samples * mpa->spf;
- if (af_fmt_from_planar(mpa->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 (af_fmt_from_planar(mpa->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 void reorder_planes(struct mp_audio *mpa, int *reorder,
- struct mp_chmap *newmap)
-{
- struct mp_audio prev = *mpa;
- mp_audio_set_channels(mpa, newmap);
-
- // The trailing planes were never written by avrctx, they're the NA channels.
- int next_na = prev.num_planes;
+ struct af_resample *s = af->priv;
- for (int n = 0; n < mpa->num_planes; n++) {
- int src = reorder[n];
- assert(src >= -1 && src < prev.num_planes);
- if (src >= 0) {
- mpa->planes[n] = prev.planes[src];
- } else {
- assert(next_na < mpa->num_planes);
- mpa->planes[n] = prev.planes[next_na++];
- af_fill_silence(mpa->planes[n], mpa->sstride * mpa->samples,
- mpa->format);
- }
- }
+ talloc_free(s->converter);
}
-static int filter_resample(struct af_instance *af, struct mp_audio *in)
+static int filter(struct af_instance *af, struct mp_audio *in)
{
struct af_resample *s = af->priv;
- struct mp_audio *out = NULL;
-
- if (!s->avrctx)
- goto error;
- int samples = get_out_samples(s, in ? in->samples : 0);
-
- struct mp_audio out_format = s->pool_fmt;
- out = mp_audio_pool_get(af->out_pool, &out_format, samples);
- if (!out)
- goto error;
- if (in)
- mp_audio_copy_attributes(out, in);
-
- if (out->samples) {
- out->samples = resample_frame(s->avrctx, out, in);
- if (out->samples < 0)
- goto error;
- }
-
- struct mp_audio real_out = *out;
- mp_audio_copy_config(out, &s->avrctx_fmt);
-
- if (out->samples && !mp_audio_config_equals(out, &s->pre_out_fmt)) {
- assert(af_fmt_is_planar(out->format) && out->format == real_out.format);
- reorder_planes(out, s->reorder_out, &s->pool_fmt.channels);
- if (!mp_audio_config_equals(out, &s->pre_out_fmt)) {
- struct mp_audio *new = mp_audio_pool_get(s->reorder_buffer,
- &s->pre_out_fmt,
- out->samples);
- if (!new)
- goto error;
- mp_audio_copy_attributes(new, out);
- int out_samples = resample_frame(s->avrctx_out, new, out);
- talloc_free(out);
- out = new;
- if (out_samples != new->samples)
- goto error;
- }
- }
+ mp_aconverter_set_speed(s->converter, s->playback_speed);
- extra_output_conversion(af, out);
+ af->filter_out(af);
+ struct mp_aframe *aframe = mp_audio_to_aframe(in);
+ if (!aframe && in)
+ return -1;
talloc_free(in);
- if (out->samples) {
- af_add_output_frame(af, out);
- } else {
- talloc_free(out);
- }
-
- af->delay = get_delay(s);
+ bool ok = mp_aconverter_write_input(s->converter, aframe);
+ if (!ok)
+ talloc_free(aframe);
- return 0;
-error:
- talloc_free(in);
- talloc_free(out);
- return -1;
+ return ok ? 0 : -1;
}
-static int filter(struct af_instance *af, struct mp_audio *in)
+static int filter_out(struct af_instance *af)
{
struct af_resample *s = af->priv;
-
- int new_rate = rate_from_speed(s->in_rate_af, s->playback_speed);
- if (s->avrctx && !(!s->is_resampling && new_rate == s->in_rate)) {
- AVRational r = av_d2q(s->playback_speed * s->in_rate_af / s->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(s->avrctx, r.den - r.num, r.den) >= 0) {
- new_rate = s->in_rate;
- s->is_resampling = true;
- }
- }
-
- bool need_reinit = fabs(new_rate / (double)s->in_rate - 1) > 0.01;
- if (need_reinit && new_rate != s->in_rate) {
- // Before reconfiguring, drain the audio that is still buffered
- // in the resampler.
- filter_resample(af, NULL);
- // Reinitialize resampler.
- configure_lavrr(af, &af->fmt_in, &af->fmt_out, false);
- }
-
- return filter_resample(af, in);
+ bool eof;
+ struct mp_aframe *out = mp_aconverter_read_output(s->converter, &eof);
+ if (out)
+ af_add_output_frame(af, mp_audio_from_aframe(out));
+ talloc_free(out);
+ af->delay = mp_aconverter_get_latency(s->converter);
+ return 0;
}
static int af_open(struct af_instance *af)
@@ -557,11 +140,9 @@ static int af_open(struct af_instance *af)
af->control = control;
af->uninit = uninit;
af->filter_frame = filter;
+ af->filter_out = filter_out;
- if (s->opts.cutoff <= 0.0)
- s->opts.cutoff = af_resample_default_cutoff(s->opts.filter_size);
-
- s->reorder_buffer = mp_audio_pool_create(s);
+ s->converter = mp_aconverter_create(af->global, af->log, &s->opts);
return AF_OK;
}
@@ -574,12 +155,7 @@ const struct af_info af_info_lavrresample = {
.open = af_open,
.priv_size = sizeof(struct af_resample),
.priv_defaults = &(const struct af_resample) {
- .opts = {
- .filter_size = 16,
- .cutoff = 0.0,
- .phase_shift = 10,
- .normalize = -1,
- },
+ .opts = MP_RESAMPLE_OPTS_DEF,
.playback_speed = 1.0,
.allow_detach = 1,
},
@@ -591,7 +167,7 @@ const struct af_info af_info_lavrresample = {
OPT_FLAG("detach", allow_detach, 0),
OPT_CHOICE("normalize", opts.normalize, 0,
({"no", 0}, {"yes", 1}, {"auto", -1})),
- OPT_KEYVALUELIST("o", avopts, 0),
+ OPT_KEYVALUELIST("o", opts.avopts, 0),
{0}
},
};