static int control(struct af_instance *af, int cmd, void *arg)
{
struct af_resample *s = (struct af_resample *) af->priv;
struct mp_audio *in = (struct mp_audio *) arg;
struct mp_audio *out = (struct mp_audio *) af->data;
switch (cmd) {
case AF_CONTROL_REINIT: {
struct mp_audio orig_in = *in;
if (((out->rate == in->rate) || (out->rate == 0)) &&
(out->format == in->format) &&
(mp_chmap_equals(&out->channels, &in->channels) || out->nch == 0) &&
s->allow_detach)
return AF_DETACH;
if (out->rate == 0)
out->rate = in->rate;
if (mp_chmap_is_empty(&out->channels))
mp_audio_set_channels(out, &in->channels);
if (af_to_avformat(in->format) == AV_SAMPLE_FMT_NONE)
mp_audio_set_format(in, AF_FORMAT_FLOAT);
if (af_to_avformat(out->format) == AV_SAMPLE_FMT_NONE)
mp_audio_set_format(out, in->format);
af->mul = out->rate / (double)in->rate;
int r = ((in->format == orig_in.format) &&
mp_chmap_equals(&in->channels, &orig_in.channels))
? AF_OK : AF_FALSE;
if (r == AF_OK && needs_lavrctx_reconfigure(s, in, out))
r = configure_lavrr(af, in, out);
return r;
}
case AF_CONTROL_SET_FORMAT: {
if (af_to_avformat(*(int*)arg) == AV_SAMPLE_FMT_NONE)
return AF_FALSE;
mp_audio_set_format(af->data, *(int*)arg);
return AF_OK;
}
case AF_CONTROL_SET_CHANNELS: {
mp_audio_set_channels(af->data, (struct mp_chmap *)arg);
return AF_OK;
}
case AF_CONTROL_SET_RESAMPLE_RATE:
out->rate = *(int *)arg;
return AF_OK;
case AF_CONTROL_RESET:
drop_all_output(s);
return AF_OK;
}
return AF_UNKNOWN;
}
static int control(struct af_instance *af, int cmd, void *arg)
{
struct af_resample *s = af->priv;
switch (cmd) {
case AF_CONTROL_REINIT: {
struct mp_audio *in = arg;
struct mp_audio *out = af->data;
struct mp_audio orig_in = *in;
if (((out->rate == in->rate) || (out->rate == 0)) &&
(out->format == in->format) &&
(mp_chmap_equals(&out->channels, &in->channels) || out->nch == 0) &&
s->allow_detach && s->playback_speed == 1.0)
return AF_DETACH;
if (out->rate == 0)
out->rate = in->rate;
if (mp_chmap_is_empty(&out->channels))
mp_audio_set_channels(out, &in->channels);
if (af_to_avformat(in->format) == AV_SAMPLE_FMT_NONE)
mp_audio_set_format(in, AF_FORMAT_FLOAT);
if (check_output_conversion(out->format) == AV_SAMPLE_FMT_NONE)
mp_audio_set_format(out, in->format);
int r = ((in->format == orig_in.format) &&
mp_chmap_equals(&in->channels, &orig_in.channels))
? AF_OK : AF_FALSE;
if (r == AF_OK)
r = configure_lavrr(af, in, out, true);
return r;
}
case AF_CONTROL_SET_PLAYBACK_SPEED_RESAMPLE: {
s->playback_speed = *(double *)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
}
return AF_OK;
}
return AF_UNKNOWN;
}
static bool needs_lavrctx_reconfigure(struct af_resample *s,
struct mp_audio *in,
struct mp_audio *out)
{
return s->ctx.in_rate != in->rate ||
s->ctx.in_format != in->format ||
!mp_chmap_equals(&s->ctx.in_channels, &in->channels) ||
s->ctx.out_rate != out->rate ||
s->ctx.out_format != out->format ||
!mp_chmap_equals(&s->ctx.out_channels, &out->channels) ||
s->ctx.filter_size != s->opts.filter_size ||
s->ctx.phase_shift != s->opts.phase_shift ||
s->ctx.linear != s->opts.linear ||
s->ctx.cutoff != s->opts.cutoff;
}
// Whether they use the same speakers (even if in different order).
bool mp_chmap_equals_reordered(const struct mp_chmap *a, const struct mp_chmap *b)
{
struct mp_chmap t1 = *a, t2 = *b;
mp_chmap_reorder_norm(&t1);
mp_chmap_reorder_norm(&t2);
return mp_chmap_equals(&t1, &t2);
}
bool mp_chmap_is_compatible(const struct mp_chmap *a, const struct mp_chmap *b)
{
if (mp_chmap_equals(a, b))
return true;
if (a->num == b->num && (mp_chmap_is_unknown(a) || mp_chmap_is_unknown(b)))
return true;
return false;
}
// libavfilter allows changing some parameters on the fly, but not
// others.
static bool is_aformat_ok(struct mp_aframe *a, struct mp_aframe *b)
{
struct mp_chmap ca = {0}, cb = {0};
mp_aframe_get_chmap(a, &ca);
mp_aframe_get_chmap(b, &cb);
return mp_chmap_equals(&ca, &cb) &&
mp_aframe_get_rate(a) == mp_aframe_get_rate(b) &&
mp_aframe_get_format(a) == mp_aframe_get_format(b);
}
/* Helper function for testing the output format */
int af_test_output(struct af_instance* af, struct mp_audio* out)
{
if((af->data->format != out->format) ||
(af->data->bps != out->bps) ||
(af->data->rate != out->rate) ||
!mp_chmap_equals(&af->data->channels, &out->channels)){
*out = *af->data;
return AF_FALSE;
}
return AF_OK;
}
static bool chmap_pa_from_mp(pa_channel_map *dst, struct mp_chmap *src)
{
if (src->num > PA_CHANNELS_MAX)
return false;
dst->channels = src->num;
if (mp_chmap_equals(src, &(const struct mp_chmap)MP_CHMAP_INIT_MONO)) {
dst->map[0] = PA_CHANNEL_POSITION_MONO;
return true;
}
for (int n = 0; n < src->num; n++) {
int mp_speaker = src->speaker[n];
int pa_speaker = PA_CHANNEL_POSITION_INVALID;
for (int i = 0; speaker_map[i][1] != -1; i++) {
if (speaker_map[i][1] == mp_speaker) {
pa_speaker = speaker_map[i][0];
break;
}
}
if (pa_speaker == PA_CHANNEL_POSITION_INVALID)
return false;
dst->map[n] = pa_speaker;
}
return true;
}
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 = check_output_conversion(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;
mp_audio_set_format(&s->pre_out_fmt, af_from_avformat(out_samplefmt));
// 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);
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 filter_n_bytes(sh_audio_t *sh, struct bstr *outbuf, int len)
{
assert(len - 1 + sh->audio_out_minsize <= sh->a_buffer_size);
int error = 0;
// Decode more bytes if needed
int old_samplerate = sh->samplerate;
struct mp_chmap old_channels = sh->channels;
int old_sample_format = sh->sample_format;
while (sh->a_buffer_len < len) {
unsigned char *buf = sh->a_buffer + sh->a_buffer_len;
int minlen = len - sh->a_buffer_len;
int maxlen = sh->a_buffer_size - sh->a_buffer_len;
int ret = sh->ad_driver->decode_audio(sh, buf, minlen, maxlen);
int format_change = sh->samplerate != old_samplerate
|| !mp_chmap_equals(&sh->channels, &old_channels)
|| sh->sample_format != old_sample_format;
if (ret <= 0 || format_change) {
error = format_change ? -2 : -1;
// samples from format-changing call get discarded too
len = sh->a_buffer_len;
break;
}
sh->a_buffer_len += ret;
}
// Filter
struct mp_audio filter_input = {
.audio = sh->a_buffer,
.len = len,
.rate = sh->samplerate,
};
mp_audio_set_format(&filter_input, sh->sample_format);
mp_audio_set_channels(&filter_input, &sh->channels);
struct mp_audio *filter_output = af_play(sh->afilter, &filter_input);
if (!filter_output)
return -1;
set_min_out_buffer_size(outbuf, outbuf->len + filter_output->len);
memcpy(outbuf->start + outbuf->len, filter_output->audio,
filter_output->len);
outbuf->len += filter_output->len;
// remove processed data from decoder buffer:
sh->a_buffer_len -= len;
memmove(sh->a_buffer, sh->a_buffer + len, sh->a_buffer_len);
return error;
}
/* Try to get at least minlen decoded+filtered bytes in outbuf
* (total length including possible existing data).
* Return 0 on success, -1 on error/EOF (not distinguished).
* In the former case outbuf->len is always >= minlen on return.
* In case of EOF/error it might or might not be.
* Outbuf.start must be talloc-allocated, and will be reallocated
* if needed to fit all filter output. */
int decode_audio(sh_audio_t *sh_audio, struct bstr *outbuf, int minlen)
{
// Indicates that a filter seems to be buffering large amounts of data
int huge_filter_buffer = 0;
// Decoded audio must be cut at boundaries of this many bytes
int unitsize = sh_audio->channels.num * sh_audio->samplesize * 16;
/* Filter output size will be about filter_multiplier times input size.
* If some filter buffers audio in big blocks this might only hold
* as average over time. */
double filter_multiplier = af_calc_filter_multiplier(sh_audio->afilter);
/* If the decoder set audio_out_minsize then it can do the equivalent of
* "while (output_len < target_len) output_len += audio_out_minsize;",
* so we must guarantee there is at least audio_out_minsize-1 bytes
* more space in the output buffer than the minimum length we try to
* decode. */
int max_decode_len = sh_audio->a_buffer_size - sh_audio->audio_out_minsize;
if (!unitsize)
return -1;
max_decode_len -= max_decode_len % unitsize;
while (minlen >= 0 && outbuf->len < minlen) {
// + some extra for possible filter buffering
int declen = (minlen - outbuf->len) / filter_multiplier + (unitsize << 5);
if (huge_filter_buffer)
/* Some filter must be doing significant buffering if the estimated
* input length didn't produce enough output from filters.
* Feed the filters 2k bytes at a time until we have enough output.
* Very small amounts could make filtering inefficient while large
* amounts can make MPlayer demux the file unnecessarily far ahead
* to get audio data and buffer video frames in memory while doing
* so. However the performance impact of either is probably not too
* significant as long as the value is not completely insane. */
declen = 2000;
declen -= declen % unitsize;
if (declen > max_decode_len)
declen = max_decode_len;
else
/* if this iteration does not fill buffer, we must have lots
* of buffering in filters */
huge_filter_buffer = 1;
int res = filter_n_bytes(sh_audio, outbuf, declen);
if (res < 0)
return res;
}
return 0;
}
bool mp_chmap_is_stereo(const struct mp_chmap *src)
{
static const struct mp_chmap stereo = MP_CHMAP_INIT_STEREO;
return mp_chmap_equals(src, &stereo);
}
static int control(struct af_instance *af, int cmd, void *arg)
{
struct af_resample *s = (struct af_resample *) af->priv;
struct mp_audio *in = (struct mp_audio *) arg;
struct mp_audio *out = (struct mp_audio *) af->data;
switch (cmd) {
case AF_CONTROL_REINIT: {
struct mp_audio orig_in = *in;
if (((out->rate == in->rate) || (out->rate == 0)) &&
(out->format == in->format) &&
(mp_chmap_equals(&out->channels, &in->channels) || out->nch == 0) &&
s->allow_detach)
return AF_DETACH;
if (out->rate == 0)
out->rate = in->rate;
if (mp_chmap_is_empty(&out->channels))
mp_audio_set_channels(out, &in->channels);
enum AVSampleFormat in_samplefmt = af_to_avformat(in->format);
if (in_samplefmt == AV_SAMPLE_FMT_NONE) {
mp_audio_set_format(in, AF_FORMAT_FLOAT_NE);
in_samplefmt = af_to_avformat(in->format);
}
enum AVSampleFormat out_samplefmt = af_to_avformat(out->format);
if (out_samplefmt == AV_SAMPLE_FMT_NONE) {
mp_audio_set_format(out, in->format);
out_samplefmt = in_samplefmt;
}
af->mul = (double) (out->rate * out->nch) / (in->rate * in->nch);
af->delay = out->nch * s->opts.filter_size / FFMIN(af->mul, 1);
if (needs_lavrctx_reconfigure(s, in, out)) {
avresample_close(s->avrctx);
avresample_close(s->avrctx_out);
s->ctx.out_rate = out->rate;
s->ctx.in_rate = in->rate;
s->ctx.out_format = out->format;
s->ctx.in_format = in->format;
s->ctx.out_channels= out->channels;
s->ctx.in_channels = in->channels;
s->ctx.filter_size = s->opts.filter_size;
s->ctx.phase_shift = s->opts.phase_shift;
s->ctx.linear = s->opts.linear;
s->ctx.cutoff = s->opts.cutoff;
ctx_opt_set_int("filter_size", s->ctx.filter_size);
ctx_opt_set_int("phase_shift", s->ctx.phase_shift);
ctx_opt_set_int("linear_interp", s->ctx.linear);
ctx_opt_set_dbl("cutoff", s->ctx.cutoff);
if (parse_avopts(s->avrctx, s->avopts) < 0) {
mp_msg(MSGT_VFILTER, MSGL_FATAL,
"af_lavrresample: could not set opts: '%s'\n", s->avopts);
return AF_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);
ctx_opt_set_int("in_channel_layout", in_ch_layout);
ctx_opt_set_int("out_channel_layout", out_ch_layout);
ctx_opt_set_int("in_sample_rate", s->ctx.in_rate);
ctx_opt_set_int("out_sample_rate", s->ctx.out_rate);
ctx_opt_set_int("in_sample_fmt", in_samplefmt);
ctx_opt_set_int("out_sample_fmt", out_samplefmt);
struct mp_chmap in_lavc;
mp_chmap_from_lavc(&in_lavc, in_ch_layout);
mp_chmap_get_reorder(s->reorder_in, &map_in, &in_lavc);
struct mp_chmap out_lavc;
mp_chmap_from_lavc(&out_lavc, out_ch_layout);
mp_chmap_get_reorder(s->reorder_out, &out_lavc, &map_out);
// Same configuration; we just reorder.
av_opt_set_int(s->avrctx_out, "in_channel_layout", out_ch_layout, 0);
av_opt_set_int(s->avrctx_out, "out_channel_layout", out_ch_layout, 0);
av_opt_set_int(s->avrctx_out, "in_sample_fmt", out_samplefmt, 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->ctx.out_rate, 0);
av_opt_set_int(s->avrctx_out, "out_sample_rate", s->ctx.out_rate, 0);
#if USE_SET_CHANNEL_MAPPING
// 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);
avresample_set_channel_mapping(s->avrctx_out, s->reorder_out);
#endif
if (avresample_open(s->avrctx) < 0 ||
avresample_open(s->avrctx_out) < 0)
{
mp_msg(MSGT_AFILTER, MSGL_ERR, "[lavrresample] Cannot open "
"Libavresample Context. \n");
return AF_ERROR;
}
}
return ((in->format == orig_in.format) &&
mp_chmap_equals(&in->channels, &orig_in.channels))
? AF_OK : AF_FALSE;
}
case AF_CONTROL_FORMAT_FMT | AF_CONTROL_SET: {
if (af_to_avformat(*(int*)arg) == AV_SAMPLE_FMT_NONE)
return AF_FALSE;
mp_audio_set_format(af->data, *(int*)arg);
return AF_OK;
}
case AF_CONTROL_CHANNELS | AF_CONTROL_SET: {
mp_audio_set_channels(af->data, (struct mp_chmap *)arg);
return AF_OK;
}
case AF_CONTROL_RESAMPLE_RATE | AF_CONTROL_SET:
out->rate = *(int *)arg;
return AF_OK;
}
return AF_UNKNOWN;
}
static int control(struct af_instance *af, int cmd, void *arg)
{
struct af_resample *s = af->priv;
switch (cmd) {
case AF_CONTROL_REINIT: {
struct mp_audio *in = arg;
struct mp_audio *out = af->data;
struct mp_audio orig_in = *in;
if (((out->rate == in->rate) || (out->rate == 0)) &&
(out->format == in->format) &&
(mp_chmap_equals(&out->channels, &in->channels) || out->nch == 0) &&
s->allow_detach && s->playback_speed == 1.0)
return AF_DETACH;
if (out->rate == 0)
out->rate = in->rate;
if (mp_chmap_is_empty(&out->channels))
mp_audio_set_channels(out, &in->channels);
if (af_to_avformat(in->format) == AV_SAMPLE_FMT_NONE)
mp_audio_set_format(in, AF_FORMAT_FLOAT);
if (check_output_conversion(out->format) == AV_SAMPLE_FMT_NONE)
mp_audio_set_format(out, in->format);
int r = ((in->format == orig_in.format) &&
mp_chmap_equals(&in->channels, &orig_in.channels))
? AF_OK : AF_FALSE;
if (r == AF_OK && needs_lavrctx_reconfigure(s, in, out))
r = configure_lavrr(af, in, out);
return r;
}
case AF_CONTROL_SET_FORMAT: {
int format = *(int *)arg;
if (format && check_output_conversion(format) == AV_SAMPLE_FMT_NONE)
return AF_FALSE;
mp_audio_set_format(af->data, format);
return AF_OK;
}
case AF_CONTROL_SET_CHANNELS: {
mp_audio_set_channels(af->data, (struct mp_chmap *)arg);
return AF_OK;
}
case AF_CONTROL_SET_RESAMPLE_RATE:
af->data->rate = *(int *)arg;
return AF_OK;
case AF_CONTROL_SET_PLAYBACK_SPEED_RESAMPLE: {
s->playback_speed = *(double *)arg;
int new_rate = rate_from_speed(s->ctx.in_rate_af, s->playback_speed);
if (new_rate != s->ctx.in_rate && s->avrctx && af->fmt_out.format) {
// Before reconfiguring, drain the audio that is still buffered
// in the resampler.
af->filter_frame(af, NULL);
// Reinitialize resampler.
configure_lavrr(af, &af->fmt_in, &af->fmt_out);
}
return AF_OK;
}
case AF_CONTROL_RESET:
if (s->avrctx)
drop_all_output(s);
return AF_OK;
}
return AF_UNKNOWN;
}
static int control(struct af_instance *af, int cmd, void *arg)
{
struct af_resample *s = af->priv;
switch (cmd) {
case AF_CONTROL_REINIT: {
struct mp_audio *in = arg;
struct mp_audio *out = af->data;
struct mp_audio orig_in = *in;
if (((out->rate == in->rate) || (out->rate == 0)) &&
(out->format == in->format) &&
(mp_chmap_equals(&out->channels, &in->channels) || out->nch == 0) &&
s->allow_detach && s->playback_speed == 1.0)
return AF_DETACH;
if (out->rate == 0)
out->rate = in->rate;
if (mp_chmap_is_empty(&out->channels))
mp_audio_set_channels(out, &in->channels);
if (af_to_avformat(in->format) == AV_SAMPLE_FMT_NONE)
mp_audio_set_format(in, AF_FORMAT_FLOAT);
if (check_output_conversion(out->format) == AV_SAMPLE_FMT_NONE)
mp_audio_set_format(out, in->format);
int r = ((in->format == orig_in.format) &&
mp_chmap_equals(&in->channels, &orig_in.channels))
? AF_OK : AF_FALSE;
if (r == AF_OK)
r = configure_lavrr(af, in, out, true);
return r;
}
case AF_CONTROL_SET_FORMAT: {
int format = *(int *)arg;
if (format && check_output_conversion(format) == AV_SAMPLE_FMT_NONE)
return AF_FALSE;
mp_audio_set_format(af->data, format);
return AF_OK;
}
case AF_CONTROL_SET_CHANNELS: {
mp_audio_set_channels(af->data, (struct mp_chmap *)arg);
return AF_OK;
}
case AF_CONTROL_SET_RESAMPLE_RATE:
af->data->rate = *(int *)arg;
return AF_OK;
case AF_CONTROL_SET_PLAYBACK_SPEED_RESAMPLE: {
s->playback_speed = *(double *)arg;
return AF_OK;
}
case AF_CONTROL_RESET:
if (s->avrctx)
drop_all_output(s);
return AF_OK;
}
return AF_UNKNOWN;
}
bool mp_audio_config_equals(const struct mp_audio *a, const struct mp_audio *b)
{
return a->format == b->format && a->rate == b->rate &&
mp_chmap_equals(&a->channels, &b->channels);
}
