static int filter_out(struct af_instance *af)
{
struct priv *p = af->priv;
while (rubberband_available(p->rubber) <= 0) {
const float *dummy[MP_NUM_CHANNELS] = {0};
const float **in_data = dummy;
size_t in_samples = 0;
if (p->pending) {
if (!p->pending->samples)
break;
// recover from previous EOF
if (p->needs_reset) {
rubberband_reset(p->rubber);
p->rubber_delay = 0;
}
p->needs_reset = false;
size_t needs = rubberband_get_samples_required(p->rubber);
in_data = (void *)&p->pending->planes;
in_samples = MPMIN(p->pending->samples, needs);
}
if (p->needs_reset)
break; // previous EOF
p->needs_reset = !p->pending; // EOF
rubberband_process(p->rubber, in_data, in_samples, p->needs_reset);
p->rubber_delay += in_samples;
if (!p->pending)
break;
mp_audio_skip_samples(p->pending, in_samples);
}
int out_samples = rubberband_available(p->rubber);
if (out_samples > 0) {
struct mp_audio *out =
mp_audio_pool_get(af->out_pool, af->data, out_samples);
if (!out)
return -1;
if (p->pending)
mp_audio_copy_config(out, p->pending);
float **out_data = (void *)&out->planes;
out->samples = rubberband_retrieve(p->rubber, out_data, out->samples);
p->rubber_delay -= out->samples * p->speed;
af_add_output_frame(af, out);
}
int delay_samples = p->rubber_delay;
if (p->pending)
delay_samples += p->pending->samples;
af->delay = delay_samples / (af->data->rate * p->speed);
return 0;
}
// Copy data from input frame to encode frame (because libavcodec wants a full
// AC3 frame for encoding, while filter input frames can be smaller or larger).
// Return true if the frame is complete.
static bool fill_buffer(struct af_instance *af)
{
af_ac3enc_t *s = af->priv;
af->delay = 0;
if (s->pending) {
int copy = MPMIN(s->in_samples - s->input->samples, s->pending->samples);
s->input->samples += copy;
mp_audio_copy(s->input, s->input->samples - copy, s->pending, 0, copy);
mp_audio_skip_samples(s->pending, copy);
}
update_delay(af);
return s->input->samples >= s->in_samples;
}
// Copy data from input frame to encode frame (because libavcodec wants a full
// AC3 frame for encoding, while filter input frames can be smaller or larger).
// Return true if the frame is complete.
static bool fill_buffer(struct af_instance *af)
{
af_ac3enc_t *s = af->priv;
af->delay = 0;
if (s->pending) {
if (!mp_audio_is_writeable(s->input))
assert(s->input->samples == 0); // we can't have sent a partial frame
mp_audio_realloc_min(s->input, s->in_samples);
int copy = MPMIN(s->in_samples - s->input->samples, s->pending->samples);
s->input->samples += copy;
mp_audio_copy(s->input, s->input->samples - copy, s->pending, 0, copy);
mp_audio_skip_samples(s->pending, copy);
}
update_delay(af);
return s->input->samples >= s->in_samples;
}
// Filter data through filter
static int filter(struct af_instance* af, struct mp_audio* audio, int flags)
{
struct mp_audio *out = af->data;
af_ac3enc_t *s = af->priv;
int num_frames = (audio->samples + mp_audio_buffer_samples(s->pending))
/ s->in_samples;
int max_out_samples = s->out_samples * num_frames;
mp_audio_realloc_min(out, max_out_samples);
out->samples = 0;
while (audio->samples > 0) {
int ret;
int consumed_pending = 0;
struct mp_audio in_frame;
int pending = mp_audio_buffer_samples(s->pending);
if (pending == 0 && audio->samples >= s->in_samples) {
in_frame = *audio;
mp_audio_skip_samples(audio, s->in_samples);
} else {
if (pending > 0 && pending < s->in_samples) {
struct mp_audio tmp = *audio;
tmp.samples = MPMIN(tmp.samples, s->in_samples);
mp_audio_buffer_append(s->pending, &tmp);
mp_audio_skip_samples(audio, tmp.samples);
}
mp_audio_buffer_peek(s->pending, &in_frame);
if (in_frame.samples < s->in_samples)
break;
consumed_pending = s->in_samples;
}
in_frame.samples = s->in_samples;
AVFrame *frame = av_frame_alloc();
if (!frame) {
MP_FATAL(af, "Could not allocate memory \n");
return -1;
}
frame->nb_samples = s->in_samples;
frame->format = s->lavc_actx->sample_fmt;
frame->channel_layout = s->lavc_actx->channel_layout;
assert(in_frame.num_planes <= AV_NUM_DATA_POINTERS);
frame->extended_data = frame->data;
for (int n = 0; n < in_frame.num_planes; n++)
frame->data[n] = in_frame.planes[n];
frame->linesize[0] = s->in_samples * audio->sstride;
int ok;
ret = avcodec_encode_audio2(s->lavc_actx, &s->pkt, frame, &ok);
av_frame_free(&frame);
if (ret < 0 || !ok) {
MP_FATAL(af, "Encode failed.\n");
return -1;
}
mp_audio_buffer_skip(s->pending, consumed_pending);
MP_DBG(af, "avcodec_encode_audio got %d, pending %d.\n",
s->pkt.size, mp_audio_buffer_samples(s->pending));
int frame_size = s->pkt.size;
int header_len = 0;
char hdr[8];
if (s->cfg_add_iec61937_header && s->pkt.size > 5) {
int bsmod = s->pkt.data[5] & 0x7;
int len = frame_size;
frame_size = AC3_FRAME_SIZE * 2 * 2;
header_len = 8;
AV_WB16(hdr, 0xF872); // iec 61937 syncword 1
AV_WB16(hdr + 2, 0x4E1F); // iec 61937 syncword 2
hdr[4] = bsmod; // bsmod
hdr[5] = 0x01; // data-type ac3
AV_WB16(hdr + 6, len << 3); // number of bits in payload
}
size_t max_size = (max_out_samples - out->samples) * out->sstride;
if (frame_size > max_size)
abort();
char *buf = (char *)out->planes[0] + out->samples * out->sstride;
memcpy(buf, hdr, header_len);
memcpy(buf + header_len, s->pkt.data, s->pkt.size);
memset(buf + header_len + s->pkt.size, 0,
frame_size - (header_len + s->pkt.size));
out->samples += frame_size / out->sstride;
}
mp_audio_buffer_append(s->pending, audio);
*audio = *out;
return 0;
}
// Filter len bytes of input, put result into outbuf.
static int filter_n_bytes(struct dec_audio *da, struct mp_audio_buffer *outbuf,
int len)
{
bool format_change = false;
int error = 0;
assert(len > 0); // would break EOF logic below
while (mp_audio_buffer_samples(da->decode_buffer) < len) {
// Check for a format change
struct mp_audio config;
mp_audio_buffer_get_format(da->decode_buffer, &config);
format_change = !mp_audio_config_equals(&da->decoded, &config);
if (format_change) {
error = AD_EOF; // drain remaining data left in the current buffer
break;
}
if (da->decoded.samples > 0) {
int copy = MPMIN(da->decoded.samples, len);
struct mp_audio append = da->decoded;
append.samples = copy;
mp_audio_buffer_append(da->decode_buffer, &append);
mp_audio_skip_samples(&da->decoded, copy);
da->pts_offset += copy;
continue;
}
error = da->ad_driver->decode_packet(da);
if (error < 0)
break;
}
if (error == AD_WAIT)
return error;
// Filter
struct mp_audio filter_data;
mp_audio_buffer_peek(da->decode_buffer, &filter_data);
filter_data.rate = da->afilter->input.rate; // due to playback speed change
len = MPMIN(filter_data.samples, len);
filter_data.samples = len;
bool eof = error == AD_EOF && filter_data.samples == 0;
if (af_filter(da->afilter, &filter_data, eof ? AF_FILTER_FLAG_EOF : 0) < 0)
return AD_ERR;
mp_audio_buffer_append(outbuf, &filter_data);
if (error == AD_EOF && filter_data.samples > 0)
error = 0; // don't end playback yet
// remove processed data from decoder buffer:
mp_audio_buffer_skip(da->decode_buffer, len);
// if format was changed, and all data was drained, execute the format change
if (format_change && eof) {
error = AD_NEW_FMT;
if (!reinit_audio_buffer(da))
error = AD_ERR; // switch to invalid format
}
return error;
}