static int control(struct af_instance *af, int cmd, void *arg)
{
struct priv *s = af->priv;
switch (cmd) {
case AF_CONTROL_REINIT: {
struct mp_audio *in = arg;
mp_audio_copy_config(af->data, in);
mp_audio_force_interleaved_format(af->data);
if (s->fast && af_fmt_from_planar(in->format) != AF_FORMAT_FLOAT) {
mp_audio_set_format(af->data, AF_FORMAT_S16);
} else {
mp_audio_set_format(af->data, AF_FORMAT_FLOAT);
}
if (af_fmt_is_planar(in->format))
mp_audio_set_format(af->data, af_fmt_to_planar(af->data->format));
return af_test_output(af, in);
}
case AF_CONTROL_SET_VOLUME:
s->level = *(float *)arg;
return AF_OK;
case AF_CONTROL_GET_VOLUME:
*(float *)arg = s->level;
return AF_OK;
}
return AF_UNKNOWN;
}
static void do_reorder(struct mp_audio *mpa, int *reorder)
{
if (af_fmt_is_planar(mpa->format)) {
reorder_planes(mpa, reorder);
} else {
reorder_channels(mpa->planes[0], reorder, mpa->bps, mpa->nch, mpa->samples);
}
}
int af_fmt_seconds_to_bytes(int format, float seconds, int channels, int samplerate)
{
assert(!af_fmt_is_planar(format));
int bps = (af_fmt2bits(format) / 8);
int framelen = channels * bps;
int bytes = seconds * bps * samplerate;
if (bytes % framelen)
bytes += framelen - (bytes % framelen);
return bytes;
}
// 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 struct mp_audio *play(struct af_instance *af, struct mp_audio *data)
{
struct af_resample *s = af->priv;
struct mp_audio *in = data;
struct mp_audio *out = af->data;
out->samples = avresample_available(s->avrctx) +
av_rescale_rnd(get_delay(s) + in->samples,
s->ctx.out_rate, s->ctx.in_rate, AV_ROUND_UP);
mp_audio_realloc_min(out, out->samples);
af->delay = get_delay(s) / (double)s->ctx.in_rate;
#if !USE_SET_CHANNEL_MAPPING
do_reorder(in, s->reorder_in);
#endif
if (out->samples) {
out->samples = avresample_convert(s->avrctx,
(uint8_t **) out->planes, out->samples * out->sstride, out->samples,
(uint8_t **) in->planes, in->samples * in->sstride, in->samples);
if (out->samples < 0)
return NULL; // error
}
*data = *out;
#if USE_SET_CHANNEL_MAPPING
if (needs_reorder(s->reorder_out, out->nch)) {
if (af_fmt_is_planar(out->format)) {
reorder_planes(data, s->reorder_out);
} else {
int out_size = out->samples * out->sstride;
if (talloc_get_size(s->reorder_buffer) < out_size)
s->reorder_buffer = talloc_realloc_size(s, s->reorder_buffer, out_size);
data->planes[0] = s->reorder_buffer;
int out_samples = avresample_convert(s->avrctx_out,
(uint8_t **) data->planes, out_size, out->samples,
(uint8_t **) out->planes, out_size, out->samples);
assert(out_samples == data->samples);
}
}
#else
do_reorder(data, s->reorder_out);
#endif
return data;
}
// this should round samples down to frame sizes
// return: number of samples played
static int play(struct ao *ao, void **data, int samples, int flags)
{
struct priv *ac = ao->priv;
struct encode_lavc_context *ectx = ao->encode_lavc_ctx;
int bufpos = 0;
double nextpts;
double outpts;
int orig_samples = samples;
pthread_mutex_lock(&ectx->lock);
if (!encode_lavc_start(ectx)) {
MP_WARN(ao, "not ready yet for encoding audio\n");
pthread_mutex_unlock(&ectx->lock);
return 0;
}
double pts = ectx->last_audio_in_pts;
pts += ectx->samples_since_last_pts / (double)ao->samplerate;
size_t num_planes = af_fmt_is_planar(ao->format) ? ao->channels.num : 1;
void *tempdata = NULL;
void *padded[MP_NUM_CHANNELS];
if ((flags & AOPLAY_FINAL_CHUNK) && (samples % ac->aframesize)) {
tempdata = talloc_new(NULL);
size_t bytelen = samples * ao->sstride;
size_t extralen = (ac->aframesize - 1) * ao->sstride;
for (int n = 0; n < num_planes; n++) {
padded[n] = talloc_size(tempdata, bytelen + extralen);
memcpy(padded[n], data[n], bytelen);
af_fill_silence((char *)padded[n] + bytelen, extralen, ao->format);
}
data = padded;
samples = (bytelen + extralen) / ao->sstride;
}
if (pts == MP_NOPTS_VALUE) {
MP_WARN(ao, "frame without pts, please report; synthesizing pts instead\n");
// synthesize pts from previous expected next pts
pts = ac->expected_next_pts;
}
if (ac->worst_time_base.den == 0) {
//if (ac->codec->time_base.num / ac->codec->time_base.den >= ac->stream->time_base.num / ac->stream->time_base.den)
if (ac->codec->time_base.num * (double) ac->stream->time_base.den >=
ac->stream->time_base.num * (double) ac->codec->time_base.den) {
MP_VERBOSE(ao, "NOTE: using codec time base (%d/%d) for pts "
"adjustment; the stream base (%d/%d) is not worse.\n",
(int)ac->codec->time_base.num,
(int)ac->codec->time_base.den,
(int)ac->stream->time_base.num,
(int)ac->stream->time_base.den);
ac->worst_time_base = ac->codec->time_base;
ac->worst_time_base_is_stream = 0;
} else {
MP_WARN(ao, "NOTE: not using codec time base (%d/%d) for pts "
"adjustment; the stream base (%d/%d) is worse.\n",
(int)ac->codec->time_base.num,
(int)ac->codec->time_base.den,
(int)ac->stream->time_base.num,
(int)ac->stream->time_base.den);
ac->worst_time_base = ac->stream->time_base;
ac->worst_time_base_is_stream = 1;
}
// NOTE: we use the following "axiom" of av_rescale_q:
// if time base A is worse than time base B, then
// av_rescale_q(av_rescale_q(x, A, B), B, A) == x
// this can be proven as long as av_rescale_q rounds to nearest, which
// it currently does
// av_rescale_q(x, A, B) * B = "round x*A to nearest multiple of B"
// and:
// av_rescale_q(av_rescale_q(x, A, B), B, A) * A
// == "round av_rescale_q(x, A, B)*B to nearest multiple of A"
// == "round 'round x*A to nearest multiple of B' to nearest multiple of A"
//
// assume this fails. Then there is a value of x*A, for which the
// nearest multiple of B is outside the range [(x-0.5)*A, (x+0.5)*A[.
// Absurd, as this range MUST contain at least one multiple of B.
}
// Fix and apply the discontinuity pts offset.
if (!ectx->options->rawts && ectx->options->copyts) {
// fix the discontinuity pts offset
nextpts = pts;
if (ectx->discontinuity_pts_offset == MP_NOPTS_VALUE) {
ectx->discontinuity_pts_offset = ectx->next_in_pts - nextpts;
}
else if (fabs(nextpts + ectx->discontinuity_pts_offset - ectx->next_in_pts) > 30) {
MP_WARN(ao, "detected an unexpected discontinuity (pts jumped by "
"%f seconds)\n",
nextpts + ectx->discontinuity_pts_offset - ectx->next_in_pts);
ectx->discontinuity_pts_offset = ectx->next_in_pts - nextpts;
}
outpts = pts + ectx->discontinuity_pts_offset;
}
else {
outpts = pts;
}
// Shift pts by the pts offset first.
outpts += encode_lavc_getoffset(ectx, ac->codec);
while (samples - bufpos >= ac->aframesize) {
void *start[MP_NUM_CHANNELS] = {0};
for (int n = 0; n < num_planes; n++)
start[n] = (char *)data[n] + bufpos * ao->sstride;
encode(ao, outpts + bufpos / (double) ao->samplerate, start);
bufpos += ac->aframesize;
}
// Calculate expected pts of next audio frame (input side).
ac->expected_next_pts = pts + bufpos / (double) ao->samplerate;
// Set next allowed input pts value (input side).
if (!ectx->options->rawts && ectx->options->copyts) {
nextpts = ac->expected_next_pts + ectx->discontinuity_pts_offset;
if (nextpts > ectx->next_in_pts)
ectx->next_in_pts = nextpts;
}
talloc_free(tempdata);
int taken = FFMIN(bufpos, orig_samples);
ectx->samples_since_last_pts += taken;
pthread_mutex_unlock(&ectx->lock);
if (flags & AOPLAY_FINAL_CHUNK) {
if (bufpos < orig_samples) {
MP_ERR(ao, "did not write enough data at the end\n");
}
} else {
if (bufpos > orig_samples) {
MP_ERR(ao, "audio buffer overflow (should never happen)\n");
}
}
return taken;
}
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;
}