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)
{
switch (cmd) {
case AF_CONTROL_REINIT: {
struct mp_audio *in = arg;
struct mp_audio orig_in = *in;
struct mp_audio *out = af->data;
if (!test_conversion(in->format, out->format))
return AF_DETACH;
if ((in->format & AF_FORMAT_BITS_MASK) == AF_FORMAT_24BIT) {
mp_audio_set_format(out, af_fmt_change_bits(in->format, 32));
} else if ((in->format & AF_FORMAT_BITS_MASK) == AF_FORMAT_32BIT) {
mp_audio_set_format(out, af_fmt_change_bits(in->format, 24));
} else {
abort();
}
out->rate = in->rate;
mp_audio_set_channels(out, &in->channels);
assert(test_conversion(in->format, out->format));
return mp_audio_config_equals(in, &orig_in) ? AF_OK : AF_FALSE;
}
case AF_CONTROL_SET_FORMAT: {
mp_audio_set_format(af->data, *(int*)arg);
return AF_OK;
}
}
return AF_UNKNOWN;
}
// fmt carries the input format. Change it to the best next-possible format
// the encoder likely accepts.
static void select_encode_format(AVCodecContext *c, struct mp_audio *fmt)
{
int formats[AF_FORMAT_COUNT];
af_get_best_sample_formats(fmt->format, formats);
for (int n = 0; formats[n]; n++) {
const enum AVSampleFormat *lf = c->codec->sample_fmts;
for (int i = 0; lf && lf[i] != AV_SAMPLE_FMT_NONE; i++) {
int mpfmt = af_from_avformat(lf[i]);
if (mpfmt && mpfmt == formats[n]) {
mp_audio_set_format(fmt, mpfmt);
goto done_fmt;
}
}
}
done_fmt: ;
int rate =
af_select_best_samplerate(fmt->rate, c->codec->supported_samplerates);
if (rate > 0)
fmt->rate = rate;
struct mp_chmap_sel sel = {0};
const uint64_t *lch = c->codec->channel_layouts;
for (int n = 0; lch && lch[n]; n++) {
struct mp_chmap chmap = {0};
mp_chmap_from_lavc(&chmap, lch[n]);
mp_chmap_sel_add_map(&sel, &chmap);
}
struct mp_chmap res = fmt->channels;
mp_chmap_sel_adjust(&sel, &res);
if (!mp_chmap_is_empty(&res))
mp_audio_set_channels(fmt, &res);
}
static void set_channels(struct mp_audio *mpa, int num)
{
struct mp_chmap map;
// "unknown" channel layouts make it easier to pass through audio data,
// without triggering remixing.
mp_chmap_set_unknown(&map, num);
mp_audio_set_channels(mpa, &map);
}
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 int control(struct af_instance *af, int cmd, void *arg)
{
struct priv *p = af->priv;
switch (cmd) {
case AF_CONTROL_REINIT: {
struct mp_audio *in = arg;
struct mp_audio orig_in = *in;
struct mp_audio *out = af->data;
if (af_to_avformat(in->format) == AV_SAMPLE_FMT_NONE)
mp_audio_set_format(in, AF_FORMAT_FLOAT);
// Removing this requires fixing AVFrame.data vs. AVFrame.extended_data
if (in->channels.num > AV_NUM_DATA_POINTERS)
return AF_ERROR;
if (!mp_chmap_is_lavc(&in->channels))
mp_chmap_reorder_to_lavc(&in->channels); // will always work
if (!recreate_graph(af, in))
return AF_ERROR;
AVFilterLink *l_out = p->out->inputs[0];
out->rate = l_out->sample_rate;
mp_audio_set_format(out, af_from_avformat(l_out->format));
struct mp_chmap out_cm;
mp_chmap_from_lavc(&out_cm, l_out->channel_layout);
mp_audio_set_channels(out, &out_cm);
if (!mp_audio_config_valid(out) || out->channels.num > AV_NUM_DATA_POINTERS)
return AF_ERROR;
p->timebase_out = l_out->time_base;
return mp_audio_config_equals(in, &orig_in) ? AF_OK : AF_FALSE;
}
case AF_CONTROL_GET_METADATA:
if (p->metadata) {
*(struct mp_tags *)arg = *p->metadata;
return CONTROL_OK;
}
return CONTROL_NA;
case AF_CONTROL_RESET:
reset(af);
return AF_OK;
}
return AF_UNKNOWN;
}
static void force_in_params(struct af_instance *af, struct mp_audio *in)
{
struct priv *priv = af->priv;
if (priv->in_format != AF_FORMAT_UNKNOWN)
mp_audio_set_format(in, priv->in_format);
if (priv->in_channels.num)
mp_audio_set_channels(in, &priv->in_channels);
if (priv->in_srate)
in->rate = priv->in_srate;
}
static void force_out_params(struct af_instance *af, struct mp_audio *out)
{
struct priv *priv = af->priv;
if (priv->out_format != AF_FORMAT_UNKNOWN)
mp_audio_set_format(out, priv->out_format);
if (priv->out_channels.num)
mp_audio_set_channels(out, &priv->out_channels);
if (priv->out_srate)
out->rate = priv->out_srate;
}
int init_audio_filters(sh_audio_t *sh_audio, int in_samplerate,
int *out_samplerate, struct mp_chmap *out_channels,
int *out_format)
{
if (!sh_audio->afilter)
sh_audio->afilter = af_new(sh_audio->opts);
struct af_stream *afs = sh_audio->afilter;
// input format: same as codec's output format:
afs->input.rate = in_samplerate;
mp_audio_set_channels(&afs->input, &sh_audio->channels);
mp_audio_set_format(&afs->input, sh_audio->sample_format);
// output format: same as ao driver's input format (if missing, fallback to input)
afs->output.rate = *out_samplerate;
mp_audio_set_channels(&afs->output, out_channels);
mp_audio_set_format(&afs->output, *out_format);
char *s_from = mp_audio_config_to_str(&afs->input);
char *s_to = mp_audio_config_to_str(&afs->output);
mp_tmsg(MSGT_DECAUDIO, MSGL_V,
"Building audio filter chain for %s -> %s...\n", s_from, s_to);
talloc_free(s_from);
talloc_free(s_to);
// let's autoprobe it!
if (af_init(afs) != 0) {
af_destroy(afs);
sh_audio->afilter = NULL;
return 0; // failed :(
}
*out_samplerate = afs->output.rate;
*out_channels = afs->output.channels;
*out_format = afs->output.format;
return 1;
}
static int control(struct af_instance *af, int cmd, void *arg)
{
struct priv *p = af->priv;
switch (cmd) {
case AF_CONTROL_REINIT: {
struct mp_audio *in = arg;
struct mp_audio orig_in = *in;
struct mp_audio *out = af->data;
if (af_to_avformat(in->format) == AV_SAMPLE_FMT_NONE)
mp_audio_set_format(in, AF_FORMAT_FLOAT);
if (!mp_chmap_is_lavc(&in->channels))
mp_chmap_reorder_to_lavc(&in->channels); // will always work
if (!recreate_graph(af, in))
return AF_ERROR;
AVFilterLink *l_out = p->out->inputs[0];
out->rate = l_out->sample_rate;
mp_audio_set_format(out, af_from_avformat(l_out->format));
struct mp_chmap out_cm;
mp_chmap_from_lavc(&out_cm, l_out->channel_layout);
if (!out_cm.num || out_cm.num != l_out->channels)
mp_chmap_from_channels(&out_cm, l_out->channels);
mp_audio_set_channels(out, &out_cm);
if (!mp_audio_config_valid(out))
return AF_ERROR;
p->timebase_out = l_out->time_base;
// Blatantly incorrect; we don't know what the filters do.
af->mul = out->rate / (double)in->rate;
return mp_audio_config_equals(in, &orig_in) ? AF_OK : AF_FALSE;
}
}
return AF_UNKNOWN;
}
int audio_init_filters(struct dec_audio *d_audio, int in_samplerate,
int *out_samplerate, struct mp_chmap *out_channels,
int *out_format)
{
if (!d_audio->afilter)
d_audio->afilter = af_new(d_audio->global);
struct af_stream *afs = d_audio->afilter;
// input format: same as codec's output format:
mp_audio_buffer_get_format(d_audio->decode_buffer, &afs->input);
// Sample rate can be different when adjusting playback speed
afs->input.rate = in_samplerate;
// output format: same as ao driver's input format (if missing, fallback to input)
afs->output.rate = *out_samplerate;
mp_audio_set_channels(&afs->output, out_channels);
mp_audio_set_format(&afs->output, *out_format);
afs->metadata = d_audio->metadata;
afs->replaygain_data = d_audio->replaygain_data;
char *s_from = mp_audio_config_to_str(&afs->input);
char *s_to = mp_audio_config_to_str(&afs->output);
MP_VERBOSE(d_audio, "Building audio filter chain for %s -> %s...\n", s_from, s_to);
talloc_free(s_from);
talloc_free(s_to);
// let's autoprobe it!
if (af_init(afs) != 0) {
af_destroy(afs);
d_audio->afilter = NULL;
return 0; // failed :(
}
*out_samplerate = afs->output.rate;
*out_channels = afs->output.channels;
*out_format = afs->output.format;
return 1;
}
// Initialization and runtime control
static int control(struct af_instance* af, int cmd, void* arg)
{
af_channels_t* s = af->priv;
switch(cmd){
case AF_CONTROL_REINIT: ;
struct mp_chmap chmap;
mp_chmap_set_unknown(&chmap, s->nch);
mp_audio_set_channels(af->data, &chmap);
// Set default channel assignment
if(!s->router){
int i;
// Make sure this filter isn't redundant
if(af->data->nch == ((struct mp_audio*)arg)->nch)
return AF_DETACH;
// If mono: fake stereo
if(((struct mp_audio*)arg)->nch == 1){
s->nr = MPMIN(af->data->nch,2);
for(i=0;i<s->nr;i++){
s->route[i][FR] = 0;
s->route[i][TO] = i;
}
}
else{
s->nr = MPMIN(af->data->nch, ((struct mp_audio*)arg)->nch);
for(i=0;i<s->nr;i++){
s->route[i][FR] = i;
s->route[i][TO] = i;
}
}
}
af->data->rate = ((struct mp_audio*)arg)->rate;
mp_audio_force_interleaved_format((struct mp_audio*)arg);
mp_audio_set_format(af->data, ((struct mp_audio*)arg)->format);
return check_routes(af,((struct mp_audio*)arg)->nch,af->data->nch);
}
return AF_UNKNOWN;
}
// Filter data through filter
static int filter(struct af_instance* af, struct mp_audio* data, int flags)
{
struct mp_audio* c = data; // Current working data
struct mp_audio* l = af->data; // Local data
af_channels_t* s = af->priv;
int i;
mp_audio_realloc_min(af->data, data->samples);
// Reset unused channels
memset(l->planes[0],0,mp_audio_psize(c) / c->nch * l->nch);
if(AF_OK == check_routes(af,c->nch,l->nch))
for(i=0;i<s->nr;i++)
copy(af, c->planes[0],l->planes[0],c->nch,s->route[i][FR],
l->nch,s->route[i][TO],mp_audio_psize(c),c->bps);
// Set output data
c->planes[0] = l->planes[0];
mp_audio_set_channels(c, &l->channels);
return 0;
}
static int control(struct af_instance *af, int cmd, void *arg)
{
switch (cmd) {
case AF_CONTROL_REINIT: {
struct mp_audio *in = arg;
struct mp_audio orig_in = *in;
struct mp_audio *out = af->data;
if (!test_conversion(in->format, out->format))
return AF_DETACH;
out->rate = in->rate;
mp_audio_set_channels(out, &in->channels);
return mp_audio_config_equals(in, &orig_in) ? AF_OK : AF_FALSE;
}
case AF_CONTROL_SET_FORMAT: {
mp_audio_set_format(af->data, *(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 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;
}
int AudioController::reinitialize(mp_audio *in) {
if (!in)
return AF_ERROR;
auto makeFormat = [] (const mp_audio *audio) {
AudioFormat format;
format.m_samplerate = audio->rate/1000.0; // kHz
format.m_bitrate = audio->rate*audio->nch*audio->bps*8;
format.m_bits = audio->bps*8;
format.m_channels = ChannelLayoutInfo::description(ChannelLayoutMap::toLayout(audio->channels));
format.m_type = af_fmt_to_str(audio->format);
return format;
};
d->input = makeFormat(in);
auto out = d->af->data;
out->rate = in->rate;
bool ret = true;
if (!isSupported(in->format)) {
ret = false;
mp_audio_set_format(in, af_fmt_is_planar(in->format) ? AF_FORMAT_FLOATP : AF_FORMAT_FLOAT);
}
if (d->fmt_conv) {
mp_audio_set_format(out, d->fmt_conv);
d->fmt_conv = AF_FORMAT_UNKNOWN;
} else
mp_audio_set_format(out, in->format);
d->chmap = in->channels;
if (!mp_chmap_from_str(&d->chmap, bstr0(ChannelLayoutInfo::data(d->layout).constData())))
_Error("Cannot find matched channel layout for '%%'", ChannelLayoutInfo::description(d->layout));
mp_audio_set_channels(out, &d->chmap);
if (d->outrate != 0)
out->rate = d->outrate;
if (!ret)
return false;
d->af->mul = (double)out->channels.num/in->channels.num;
if (d->tempoScalerActivated)
d->af->mul /= d->scale;
if ((d->resample = out->rate != in->rate)) {
d->af->mul *= (double)out->rate/in->rate;
const auto nch = in->channels.num;/*mp_chmap_to_lavc_unchecked(&in->channels);*/
const auto fmt = af_to_avformat(in->format);
if (!d->swr)
d->swr = swr_alloc();
av_opt_set_int(d->swr, "in_channel_count", nch, 0);
av_opt_set_int(d->swr, "out_channel_count", nch, 0);
av_opt_set_int(d->swr, "in_sample_rate", in->rate, 0);
av_opt_set_int(d->swr, "out_sample_rate", out->rate, 0);
av_opt_set_sample_fmt(d->swr, "in_sample_fmt", fmt, 0);
av_opt_set_sample_fmt(d->swr, "out_sample_fmt", fmt, 0);
swr_init(d->swr);
if (!d->resampled)
d->resampled = talloc_zero(nullptr, mp_audio);
*d->resampled = *in;
d->resampled->rate = out->rate;
in = d->resampled;
}
d->output = makeFormat(out);
const AudioDataFormat fmt_in(*in), fmt_out(*out);
check(d->mixer, d->clip, fmt_in, fmt_out);
d->mixer->setOutput(out);
d->mixer->setChannelLayoutMap(d->map);
d->dirty = 0xffffffff;
return true;
}
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;
}
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;
}
// Use old MPlayer/ALSA channel layout.
void mp_audio_set_channels_old(struct mp_audio *mpa, int num_channels)
{
struct mp_chmap map;
mp_chmap_from_channels_alsa(&map, num_channels);
mp_audio_set_channels(mpa, &map);
}
void mp_audio_copy_config(struct mp_audio *dst, const struct mp_audio *src)
{
mp_audio_set_format(dst, src->format);
mp_audio_set_channels(dst, &src->channels);
dst->rate = src->rate;
}
// Initialization and runtime control
static int control(struct af_instance* af, int cmd, void* arg)
{
switch(cmd){
case AF_CONTROL_REINIT:{
char buf1[256];
char buf2[256];
struct mp_audio *data = arg;
int supported_ac3 = 0;
// Make sure this filter isn't redundant
if(af->data->format == data->format)
return AF_DETACH;
// A bit complex because we can convert AC3
// to generic iec61937 but not the other way
// round.
if (AF_FORMAT_IS_AC3(af->data->format))
supported_ac3 = AF_FORMAT_IS_AC3(data->format);
else if (AF_FORMAT_IS_IEC61937(af->data->format))
supported_ac3 = AF_FORMAT_IS_IEC61937(data->format);
// Allow trivial AC3-endianness conversion
if (!supported_ac3)
// Check for errors in configuration
if((AF_OK != check_bps(data->bps)) ||
(AF_OK != check_format(data->format)) ||
(AF_OK != check_bps(af->data->bps)) ||
(AF_OK != check_format(af->data->format)))
return AF_ERROR;
af_fmt2str(data->format,buf1,256);
af_fmt2str(af->data->format,buf2,256);
mp_msg(MSGT_AFILTER, MSGL_V, "[format] Changing sample format from %s to %s\n",
buf1, buf2);
af->data->rate = data->rate;
mp_audio_set_channels(af->data, &data->channels);
af->mul = (double)af->data->bps / data->bps;
af->play = play; // set default
// look whether only endianness differences are there
if ((af->data->format & ~AF_FORMAT_END_MASK) ==
(data->format & ~AF_FORMAT_END_MASK))
{
mp_msg(MSGT_AFILTER, MSGL_V, "[format] Accelerated endianness conversion only\n");
af->play = play_swapendian;
}
if ((data->format == AF_FORMAT_FLOAT_NE) &&
(af->data->format == AF_FORMAT_S16_NE))
{
mp_msg(MSGT_AFILTER, MSGL_V, "[format] Accelerated %s to %s conversion\n",
buf1, buf2);
af->play = play_float_s16;
}
if ((data->format == AF_FORMAT_S16_NE) &&
(af->data->format == AF_FORMAT_FLOAT_NE))
{
mp_msg(MSGT_AFILTER, MSGL_V, "[format] Accelerated %s to %s conversion\n",
buf1, buf2);
af->play = play_s16_float;
}
return AF_OK;
}
case AF_CONTROL_COMMAND_LINE:{
int format = af_str2fmt_short(bstr0(arg));
if (!format) {
mp_msg(MSGT_AFILTER, MSGL_ERR, "[format] %s is not a valid format\n", (char *)arg);
return AF_ERROR;
}
if(AF_OK != af->control(af, AF_CONTROL_FORMAT_FMT | AF_CONTROL_SET,&format))
return AF_ERROR;
return AF_OK;
}
case AF_CONTROL_FORMAT_FMT | AF_CONTROL_SET:{
// Check for errors in configuration
if(!AF_FORMAT_IS_AC3(*(int*)arg) && AF_OK != check_format(*(int*)arg))
return AF_ERROR;
mp_audio_set_format(af->data, *(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)
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;
}
