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 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;
}
// must get exactly ac->aframesize amount of data
static void encode(struct ao *ao, double apts, void **data)
{
struct priv *ac = ao->priv;
struct encode_lavc_context *ectx = ao->encode_lavc_ctx;
double realapts = ac->aframecount * (double) ac->aframesize /
ao->samplerate;
ac->aframecount++;
if (data)
ectx->audio_pts_offset = realapts - apts;
if(data) {
AVFrame *frame = av_frame_alloc();
frame->format = af_to_avformat(ao->format);
frame->nb_samples = ac->aframesize;
size_t num_planes = af_fmt_is_planar(ao->format) ? ao->channels.num : 1;
assert(num_planes <= AV_NUM_DATA_POINTERS);
for (int n = 0; n < num_planes; n++)
frame->extended_data[n] = data[n];
frame->linesize[0] = frame->nb_samples * ao->sstride;
if (ectx->options->rawts || ectx->options->copyts) {
// real audio pts
frame->pts = floor(apts * ac->codec->time_base.den / ac->codec->time_base.num + 0.5);
} else {
// audio playback time
frame->pts = floor(realapts * ac->codec->time_base.den / ac->codec->time_base.num + 0.5);
}
int64_t frame_pts = av_rescale_q(frame->pts, ac->codec->time_base, ac->worst_time_base);
if (ac->lastpts != AV_NOPTS_VALUE && frame_pts <= ac->lastpts) {
// this indicates broken video
// (video pts failing to increase fast enough to match audio)
MP_WARN(ao, "audio frame pts went backwards (%d <- %d), autofixed\n",
(int)frame->pts, (int)ac->lastpts);
frame_pts = ac->lastpts + 1;
frame->pts = av_rescale_q(frame_pts, ac->worst_time_base, ac->codec->time_base);
}
ac->lastpts = frame_pts;
frame->quality = ac->codec->global_quality;
encode_audio_and_write(ao, frame);
av_frame_free(&frame);
}
else
encode_audio_and_write(ao, NULL);
}
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;
}
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;
}
// Initialization and runtime control
static int control(struct af_instance *af, int cmd, void *arg)
{
af_ac3enc_t *s = af->priv;
static const int default_bit_rate[AC3_MAX_CHANNELS+1] = \
{0, 96000, 192000, 256000, 384000, 448000, 448000};
switch (cmd){
case AF_CONTROL_REINIT: {
struct mp_audio *in = arg;
struct mp_audio orig_in = *in;
if (!af_fmt_is_pcm(in->format) || in->nch < s->cfg_min_channel_num)
return AF_DETACH;
// At least currently, the AC3 encoder doesn't export sample rates.
in->rate = 48000;
select_encode_format(s->lavc_actx, in);
af->data->rate = in->rate;
mp_audio_set_format(af->data, AF_FORMAT_S_AC3);
mp_audio_set_num_channels(af->data, 2);
if (!mp_audio_config_equals(in, &orig_in))
return AF_FALSE;
if (s->cfg_add_iec61937_header) {
s->out_samples = AC3_FRAME_SIZE;
} else {
s->out_samples = AC3_MAX_CODED_FRAME_SIZE / af->data->sstride;
}
mp_audio_copy_config(s->input, in);
talloc_free(s->pending);
s->pending = NULL;
MP_DBG(af, "reinit: %d, %d, %d.\n", in->nch, in->rate, s->in_samples);
int bit_rate = s->bit_rate ? s->bit_rate : default_bit_rate[in->nch];
if (s->lavc_actx->channels != in->nch ||
s->lavc_actx->sample_rate != in->rate ||
s->lavc_actx->bit_rate != bit_rate)
{
avcodec_close(s->lavc_actx);
// Put sample parameters
s->lavc_actx->sample_fmt = af_to_avformat(in->format);
s->lavc_actx->channels = in->nch;
s->lavc_actx->channel_layout = mp_chmap_to_lavc(&in->channels);
s->lavc_actx->sample_rate = in->rate;
s->lavc_actx->bit_rate = bit_rate;
if (avcodec_open2(s->lavc_actx, s->lavc_acodec, NULL) < 0) {
MP_ERR(af, "Couldn't open codec %s, br=%d.\n", "ac3", bit_rate);
return AF_ERROR;
}
if (s->lavc_actx->frame_size < 1) {
MP_ERR(af, "encoder didn't specify input frame size\n");
return AF_ERROR;
}
}
s->in_samples = s->lavc_actx->frame_size;
mp_audio_realloc(s->input, s->in_samples);
s->input->samples = 0;
s->encoder_buffered = 0;
return AF_OK;
}
case AF_CONTROL_RESET:
if (avcodec_is_open(s->lavc_actx))
avcodec_flush_buffers(s->lavc_actx);
talloc_free(s->pending);
s->pending = NULL;
s->input->samples = 0;
s->encoder_buffered = 0;
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;
}
bool af_lavrresample_test_conversion(int src_format, int dst_format)
{
return af_to_avformat(src_format) != AV_SAMPLE_FMT_NONE &&
check_output_conversion(dst_format) != AV_SAMPLE_FMT_NONE;
}
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 bool recreate_graph(struct af_instance *af, struct mp_audio *config)
{
void *tmp = talloc_new(NULL);
struct priv *p = af->priv;
AVFilterContext *in = NULL, *out = NULL;
int r;
if (bstr0(p->cfg_graph).len == 0) {
mp_msg(MSGT_AFILTER, MSGL_FATAL, "lavfi: no filter graph set\n");
return false;
}
destroy_graph(af);
mp_msg(MSGT_AFILTER, MSGL_V, "lavfi: create graph: '%s'\n", p->cfg_graph);
AVFilterGraph *graph = avfilter_graph_alloc();
if (!graph)
goto error;
if (parse_avopts(graph, p->cfg_avopts) < 0) {
mp_msg(MSGT_VFILTER, MSGL_FATAL, "lavfi: could not set opts: '%s'\n",
p->cfg_avopts);
goto error;
}
AVFilterInOut *outputs = avfilter_inout_alloc();
AVFilterInOut *inputs = avfilter_inout_alloc();
if (!outputs || !inputs)
goto error;
char *src_args = talloc_asprintf(tmp,
"sample_rate=%d:sample_fmt=%s:channels=%d:time_base=%d/%d:"
"channel_layout=0x%"PRIx64, config->rate,
av_get_sample_fmt_name(af_to_avformat(config->format)),
config->channels.num, 1, config->rate,
mp_chmap_to_lavc(&config->channels));
if (avfilter_graph_create_filter(&in, avfilter_get_by_name("abuffer"),
"src", src_args, NULL, graph) < 0)
goto error;
if (avfilter_graph_create_filter(&out, avfilter_get_by_name("abuffersink"),
"out", NULL, NULL, graph) < 0)
goto error;
static const enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32,
AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DBL,
AV_SAMPLE_FMT_U8P, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S32P,
AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP,
AV_SAMPLE_FMT_NONE
};
r = av_opt_set_int_list(out, "sample_fmts", sample_fmts,
AV_SAMPLE_FMT_NONE, AV_OPT_SEARCH_CHILDREN);
if (r < 0)
goto error;
r = av_opt_set_int(out, "all_channel_counts", 1, AV_OPT_SEARCH_CHILDREN);
if (r < 0)
goto error;
outputs->name = av_strdup("in");
outputs->filter_ctx = in;
inputs->name = av_strdup("out");
inputs->filter_ctx = out;
if (graph_parse(graph, p->cfg_graph, inputs, outputs, NULL) < 0)
goto error;
if (avfilter_graph_config(graph, NULL) < 0)
goto error;
p->in = in;
p->out = out;
p->graph = graph;
assert(out->nb_inputs == 1);
assert(in->nb_outputs == 1);
talloc_free(tmp);
return true;
error:
mp_msg(MSGT_AFILTER, MSGL_FATAL, "Can't configure libavfilter graph.\n");
avfilter_graph_free(&graph);
talloc_free(tmp);
return false;
}
// open & setup audio device
static int init(struct ao *ao)
{
struct priv *ac = talloc_zero(ao, struct priv);
AVCodec *codec;
ao->priv = ac;
if (!encode_lavc_available(ao->encode_lavc_ctx)) {
MP_ERR(ao, "the option --o (output file) must be specified\n");
return -1;
}
pthread_mutex_lock(&ao->encode_lavc_ctx->lock);
if (encode_lavc_alloc_stream(ao->encode_lavc_ctx,
AVMEDIA_TYPE_AUDIO,
&ac->stream, &ac->codec) < 0) {
MP_ERR(ao, "could not get a new audio stream\n");
goto fail;
}
codec = ao->encode_lavc_ctx->ac;
int samplerate = af_select_best_samplerate(ao->samplerate,
codec->supported_samplerates);
if (samplerate > 0)
ao->samplerate = samplerate;
// TODO: Remove this redundancy with encode_lavc_alloc_stream also
// setting the time base.
// Using codec->time_bvase is deprecated, but needed for older lavf.
ac->stream->time_base.num = 1;
ac->stream->time_base.den = ao->samplerate;
ac->codec->time_base.num = 1;
ac->codec->time_base.den = ao->samplerate;
ac->codec->sample_rate = ao->samplerate;
struct mp_chmap_sel sel = {0};
mp_chmap_sel_add_any(&sel);
if (!ao_chmap_sel_adjust2(ao, &sel, &ao->channels, false))
goto fail;
mp_chmap_reorder_to_lavc(&ao->channels);
ac->codec->channels = ao->channels.num;
ac->codec->channel_layout = mp_chmap_to_lavc(&ao->channels);
ac->codec->sample_fmt = AV_SAMPLE_FMT_NONE;
select_format(ao, codec);
ac->sample_size = af_fmt_to_bytes(ao->format);
ac->codec->sample_fmt = af_to_avformat(ao->format);
ac->codec->bits_per_raw_sample = ac->sample_size * 8;
if (encode_lavc_open_codec(ao->encode_lavc_ctx, ac->codec) < 0)
goto fail;
ac->pcmhack = 0;
if (ac->codec->frame_size <= 1)
ac->pcmhack = av_get_bits_per_sample(ac->codec->codec_id) / 8;
if (ac->pcmhack)
ac->aframesize = 16384; // "enough"
else
ac->aframesize = ac->codec->frame_size;
// enough frames for at least 0.25 seconds
ac->framecount = ceil(ao->samplerate * 0.25 / ac->aframesize);
// but at least one!
ac->framecount = FFMAX(ac->framecount, 1);
ac->savepts = AV_NOPTS_VALUE;
ac->lastpts = AV_NOPTS_VALUE;
ao->untimed = true;
if (ao->channels.num > AV_NUM_DATA_POINTERS)
goto fail;
pthread_mutex_unlock(&ao->encode_lavc_ctx->lock);
return 0;
fail:
pthread_mutex_unlock(&ao->encode_lavc_ctx->lock);
ac->shutdown = true;
return -1;
}
// open & setup audio device
static int init(struct ao *ao)
{
struct priv *ac = talloc_zero(ao, struct priv);
AVCodec *codec;
ao->priv = ac;
if (!encode_lavc_available(ao->encode_lavc_ctx)) {
MP_ERR(ao, "the option --o (output file) must be specified\n");
return -1;
}
pthread_mutex_lock(&ao->encode_lavc_ctx->lock);
ac->stream = encode_lavc_alloc_stream(ao->encode_lavc_ctx,
AVMEDIA_TYPE_AUDIO);
if (!ac->stream) {
MP_ERR(ao, "could not get a new audio stream\n");
goto fail;
}
codec = encode_lavc_get_codec(ao->encode_lavc_ctx, ac->stream);
// ac->stream->time_base.num = 1;
// ac->stream->time_base.den = ao->samplerate;
// doing this breaks mpeg2ts in ffmpeg
// which doesn't properly force the time base to be 90000
// furthermore, ffmpeg.c doesn't do this either and works
ac->stream->codec->time_base.num = 1;
ac->stream->codec->time_base.den = ao->samplerate;
ac->stream->codec->sample_rate = ao->samplerate;
struct mp_chmap_sel sel = {0};
mp_chmap_sel_add_any(&sel);
if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels))
goto fail;
mp_chmap_reorder_to_lavc(&ao->channels);
ac->stream->codec->channels = ao->channels.num;
ac->stream->codec->channel_layout = mp_chmap_to_lavc(&ao->channels);
ac->stream->codec->sample_fmt = AV_SAMPLE_FMT_NONE;
select_format(ao, codec);
ac->sample_size = af_fmt2bits(ao->format) / 8;
ac->stream->codec->sample_fmt = af_to_avformat(ao->format);
ac->stream->codec->bits_per_raw_sample = ac->sample_size * 8;
if (encode_lavc_open_codec(ao->encode_lavc_ctx, ac->stream) < 0)
goto fail;
ac->pcmhack = 0;
if (ac->stream->codec->frame_size <= 1)
ac->pcmhack = av_get_bits_per_sample(ac->stream->codec->codec_id) / 8;
if (ac->pcmhack) {
ac->aframesize = 16384; // "enough"
ac->buffer_size =
ac->aframesize * ac->pcmhack * ao->channels.num * 2 + 200;
} else {
ac->aframesize = ac->stream->codec->frame_size;
ac->buffer_size =
ac->aframesize * ac->sample_size * ao->channels.num * 2 + 200;
}
if (ac->buffer_size < FF_MIN_BUFFER_SIZE)
ac->buffer_size = FF_MIN_BUFFER_SIZE;
ac->buffer = talloc_size(ac, ac->buffer_size);
// enough frames for at least 0.25 seconds
ac->framecount = ceil(ao->samplerate * 0.25 / ac->aframesize);
// but at least one!
ac->framecount = FFMAX(ac->framecount, 1);
ac->savepts = MP_NOPTS_VALUE;
ac->lastpts = MP_NOPTS_VALUE;
ao->untimed = true;
pthread_mutex_unlock(&ao->encode_lavc_ctx->lock);
return 0;
fail:
pthread_mutex_unlock(&ao->encode_lavc_ctx->lock);
return -1;
}
// must get exactly ac->aframesize amount of data
static int encode(struct ao *ao, double apts, void **data)
{
AVPacket packet;
struct priv *ac = ao->priv;
struct encode_lavc_context *ectx = ao->encode_lavc_ctx;
double realapts = ac->aframecount * (double) ac->aframesize /
ao->samplerate;
int status, gotpacket;
ac->aframecount++;
if (data)
ectx->audio_pts_offset = realapts - apts;
av_init_packet(&packet);
packet.data = ac->buffer;
packet.size = ac->buffer_size;
if(data) {
AVFrame *frame = av_frame_alloc();
frame->format = af_to_avformat(ao->format);
frame->nb_samples = ac->aframesize;
assert(ao->channels.num <= AV_NUM_DATA_POINTERS);
for (int n = 0; n < ao->channels.num; n++)
frame->extended_data[n] = data[n];
frame->linesize[0] = frame->nb_samples * ao->sstride;
if (ectx->options->rawts || ectx->options->copyts) {
// real audio pts
frame->pts = floor(apts * ac->stream->codec->time_base.den / ac->stream->codec->time_base.num + 0.5);
} else {
// audio playback time
frame->pts = floor(realapts * ac->stream->codec->time_base.den / ac->stream->codec->time_base.num + 0.5);
}
int64_t frame_pts = av_rescale_q(frame->pts, ac->stream->codec->time_base, ac->worst_time_base);
if (ac->lastpts != MP_NOPTS_VALUE && frame_pts <= ac->lastpts) {
// this indicates broken video
// (video pts failing to increase fast enough to match audio)
MP_WARN(ao, "audio frame pts went backwards (%d <- %d), autofixed\n",
(int)frame->pts, (int)ac->lastpts);
frame_pts = ac->lastpts + 1;
frame->pts = av_rescale_q(frame_pts, ac->worst_time_base, ac->stream->codec->time_base);
}
ac->lastpts = frame_pts;
frame->quality = ac->stream->codec->global_quality;
status = avcodec_encode_audio2(ac->stream->codec, &packet, frame, &gotpacket);
if (!status) {
if (ac->savepts == MP_NOPTS_VALUE)
ac->savepts = frame->pts;
}
av_frame_free(&frame);
}
else
{
status = avcodec_encode_audio2(ac->stream->codec, &packet, NULL, &gotpacket);
}
if(status) {
MP_ERR(ao, "error encoding\n");
return -1;
}
if(!gotpacket)
return 0;
MP_DBG(ao, "got pts %f (playback time: %f); out size: %d\n",
apts, realapts, packet.size);
encode_lavc_write_stats(ao->encode_lavc_ctx, ac->stream);
packet.stream_index = ac->stream->index;
// Do we need this at all? Better be safe than sorry...
if (packet.pts == AV_NOPTS_VALUE) {
MP_WARN(ao, "encoder lost pts, why?\n");
if (ac->savepts != MP_NOPTS_VALUE)
packet.pts = ac->savepts;
}
if (packet.pts != AV_NOPTS_VALUE)
packet.pts = av_rescale_q(packet.pts, ac->stream->codec->time_base,
ac->stream->time_base);
if (packet.dts != AV_NOPTS_VALUE)
packet.dts = av_rescale_q(packet.dts, ac->stream->codec->time_base,
ac->stream->time_base);
if(packet.duration > 0)
packet.duration = av_rescale_q(packet.duration, ac->stream->codec->time_base,
ac->stream->time_base);
ac->savepts = MP_NOPTS_VALUE;
if (encode_lavc_write_frame(ao->encode_lavc_ctx, &packet) < 0) {
MP_ERR(ao, "error writing at %f %f/%f\n",
realapts, (double) ac->stream->time_base.num,
(double) ac->stream->time_base.den);
return -1;
}
return packet.size;
}
static bool test_conversion(int src_format, int dst_format)
{
return af_to_avformat(src_format) != AV_SAMPLE_FMT_NONE &&
af_to_avformat(dst_format) != AV_SAMPLE_FMT_NONE;
}
static int configure_lavrr(struct af_instance *af, struct mp_audio *in,
struct mp_audio *out)
{
struct af_resample *s = af->priv;
enum AVSampleFormat in_samplefmt = af_to_avformat(in->format);
enum AVSampleFormat out_samplefmt = af_to_avformat(out->format);
if (in_samplefmt == AV_SAMPLE_FMT_NONE || out_samplefmt == AV_SAMPLE_FMT_NONE)
return AF_ERROR;
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;
av_opt_set_int(s->avrctx, "filter_size", s->ctx.filter_size, 0);
av_opt_set_int(s->avrctx, "phase_shift", s->ctx.phase_shift, 0);
av_opt_set_int(s->avrctx, "linear_interp", s->ctx.linear, 0);
av_opt_set_double(s->avrctx, "cutoff", s->ctx.cutoff, 0);
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);
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->ctx.in_rate, 0);
av_opt_set_int(s->avrctx, "out_sample_rate", s->ctx.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_samplefmt, 0);
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 AF_OK;
}
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;
}
// Return the format libavresample should convert to, given the final output
// format mp_format. In some cases (S24) we perform an extra conversion step,
// and signal here what exactly libavresample should output. It will be the
// input to the final conversion to mp_format.
static int check_output_conversion(int mp_format)
{
if (mp_format == AF_FORMAT_S24)
return AV_SAMPLE_FMT_S32;
return af_to_avformat(mp_format);
}
static bool recreate_graph(struct af_instance *af, struct mp_audio *config)
{
void *tmp = talloc_new(NULL);
struct priv *p = af->priv;
AVFilterContext *in = NULL, *out = NULL, *f_format = NULL;
if (bstr0(p->cfg_graph).len == 0) {
MP_FATAL(af, "lavfi: no filter graph set\n");
return false;
}
destroy_graph(af);
MP_VERBOSE(af, "lavfi: create graph: '%s'\n", p->cfg_graph);
AVFilterGraph *graph = avfilter_graph_alloc();
if (!graph)
goto error;
if (mp_set_avopts(af->log, graph, p->cfg_avopts) < 0)
goto error;
AVFilterInOut *outputs = avfilter_inout_alloc();
AVFilterInOut *inputs = avfilter_inout_alloc();
if (!outputs || !inputs)
goto error;
// Build list of acceptable output sample formats. libavfilter will insert
// conversion filters if needed.
static const enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32,
AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DBL,
AV_SAMPLE_FMT_U8P, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S32P,
AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP,
AV_SAMPLE_FMT_NONE
};
char *fmtstr = talloc_strdup(tmp, "");
for (int n = 0; sample_fmts[n] != AV_SAMPLE_FMT_NONE; n++) {
const char *name = av_get_sample_fmt_name(sample_fmts[n]);
if (name) {
const char *s = fmtstr[0] ? "|" : "";
fmtstr = talloc_asprintf_append_buffer(fmtstr, "%s%s", s, name);
}
}
char *src_args = talloc_asprintf(tmp,
"sample_rate=%d:sample_fmt=%s:time_base=%d/%d:"
"channel_layout=0x%"PRIx64, config->rate,
av_get_sample_fmt_name(af_to_avformat(config->format)),
1, config->rate, mp_chmap_to_lavc(&config->channels));
if (avfilter_graph_create_filter(&in, avfilter_get_by_name("abuffer"),
"src", src_args, NULL, graph) < 0)
goto error;
if (avfilter_graph_create_filter(&out, avfilter_get_by_name("abuffersink"),
"out", NULL, NULL, graph) < 0)
goto error;
if (avfilter_graph_create_filter(&f_format, avfilter_get_by_name("aformat"),
"format", fmtstr, NULL, graph) < 0)
goto error;
if (avfilter_link(f_format, 0, out, 0) < 0)
goto error;
outputs->name = av_strdup("in");
outputs->filter_ctx = in;
inputs->name = av_strdup("out");
inputs->filter_ctx = f_format;
if (graph_parse(graph, p->cfg_graph, inputs, outputs, NULL) < 0)
goto error;
if (avfilter_graph_config(graph, NULL) < 0)
goto error;
p->in = in;
p->out = out;
p->graph = graph;
assert(out->nb_inputs == 1);
assert(in->nb_outputs == 1);
talloc_free(tmp);
return true;
error:
MP_FATAL(af, "Can't configure libavfilter graph.\n");
avfilter_graph_free(&graph);
talloc_free(tmp);
return false;
}
// Attempt to initialize all pads. Return true if all are initialized, or
// false if more data is needed (or on error).
static bool init_pads(struct lavfi *c)
{
if (!c->graph)
goto error;
for (int n = 0; n < c->num_out_pads; n++) {
struct lavfi_pad *pad = c->out_pads[n];
if (pad->buffer)
continue;
const AVFilter *dst_filter = NULL;
if (pad->type == MP_FRAME_AUDIO) {
dst_filter = avfilter_get_by_name("abuffersink");
} else if (pad->type == MP_FRAME_VIDEO) {
dst_filter = avfilter_get_by_name("buffersink");
} else {
assert(0);
}
if (!dst_filter)
goto error;
char name[256];
snprintf(name, sizeof(name), "mpv_sink_%s", pad->name);
if (avfilter_graph_create_filter(&pad->buffer, dst_filter,
name, NULL, NULL, c->graph) < 0)
goto error;
if (avfilter_link(pad->filter, pad->filter_pad, pad->buffer, 0) < 0)
goto error;
}
for (int n = 0; n < c->num_in_pads; n++) {
struct lavfi_pad *pad = c->in_pads[n];
if (pad->buffer)
continue;
mp_frame_unref(&pad->in_fmt);
read_pad_input(c, pad);
// no input data, format unknown, can't init, wait longer.
if (!pad->pending.type)
return false;
if (mp_frame_is_data(pad->pending)) {
assert(pad->pending.type == pad->type);
pad->in_fmt = mp_frame_ref(pad->pending);
if (!pad->in_fmt.type)
goto error;
if (pad->in_fmt.type == MP_FRAME_VIDEO)
mp_image_unref_data(pad->in_fmt.data);
if (pad->in_fmt.type == MP_FRAME_AUDIO)
mp_aframe_unref_data(pad->in_fmt.data);
}
if (pad->pending.type == MP_FRAME_EOF && !pad->in_fmt.type) {
// libavfilter makes this painful. Init it with a dummy config,
// just so we can tell it the stream is EOF.
if (pad->type == MP_FRAME_AUDIO) {
struct mp_aframe *fmt = mp_aframe_create();
mp_aframe_set_format(fmt, AF_FORMAT_FLOAT);
mp_aframe_set_chmap(fmt, &(struct mp_chmap)MP_CHMAP_INIT_STEREO);
mp_aframe_set_rate(fmt, 48000);
pad->in_fmt = (struct mp_frame){MP_FRAME_AUDIO, fmt};
}
if (pad->type == MP_FRAME_VIDEO) {
struct mp_image *fmt = talloc_zero(NULL, struct mp_image);
mp_image_setfmt(fmt, IMGFMT_420P);
mp_image_set_size(fmt, 64, 64);
pad->in_fmt = (struct mp_frame){MP_FRAME_VIDEO, fmt};
}
}
if (pad->in_fmt.type != pad->type)
goto error;
AVBufferSrcParameters *params = av_buffersrc_parameters_alloc();
if (!params)
goto error;
pad->timebase = AV_TIME_BASE_Q;
char *filter_name = NULL;
if (pad->type == MP_FRAME_AUDIO) {
struct mp_aframe *fmt = pad->in_fmt.data;
params->format = af_to_avformat(mp_aframe_get_format(fmt));
params->sample_rate = mp_aframe_get_rate(fmt);
struct mp_chmap chmap = {0};
mp_aframe_get_chmap(fmt, &chmap);
params->channel_layout = mp_chmap_to_lavc(&chmap);
pad->timebase = (AVRational){1, mp_aframe_get_rate(fmt)};
filter_name = "abuffer";
} else if (pad->type == MP_FRAME_VIDEO) {
struct mp_image *fmt = pad->in_fmt.data;
params->format = imgfmt2pixfmt(fmt->imgfmt);
params->width = fmt->w;
params->height = fmt->h;
params->sample_aspect_ratio.num = fmt->params.p_w;
params->sample_aspect_ratio.den = fmt->params.p_h;
params->hw_frames_ctx = fmt->hwctx;
params->frame_rate = av_d2q(fmt->nominal_fps, 1000000);
filter_name = "buffer";
} else {
assert(0);
}
params->time_base = pad->timebase;
const AVFilter *filter = avfilter_get_by_name(filter_name);
if (filter) {
char name[256];
snprintf(name, sizeof(name), "mpv_src_%s", pad->name);
pad->buffer = avfilter_graph_alloc_filter(c->graph, filter, name);
}
if (!pad->buffer) {
av_free(params);
goto error;
}
int ret = av_buffersrc_parameters_set(pad->buffer, params);
av_free(params);
if (ret < 0)
goto error;
if (avfilter_init_str(pad->buffer, NULL) < 0)
goto error;
if (avfilter_link(pad->buffer, 0, pad->filter, pad->filter_pad) < 0)
goto error;
}
return true;
error:
MP_FATAL(c, "could not initialize filter pads\n");
c->failed = true;
mp_filter_internal_mark_failed(c->f);
return false;
}
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;
}