static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
{
AVFilterContext *ctx = inlink->dst;
ResampleContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
if (s->avr) {
AVFilterBufferRef *buf_out;
int delay, nb_samples, ret;
/* maximum possible samples lavr can output */
delay = avresample_get_delay(s->avr);
nb_samples = av_rescale_rnd(buf->audio->nb_samples + delay,
outlink->sample_rate, inlink->sample_rate,
AV_ROUND_UP);
buf_out = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
ret = avresample_convert(s->avr, (void**)buf_out->extended_data,
buf_out->linesize[0], nb_samples,
(void**)buf->extended_data, buf->linesize[0],
buf->audio->nb_samples);
av_assert0(!avresample_available(s->avr));
if (s->next_pts == AV_NOPTS_VALUE) {
if (buf->pts == AV_NOPTS_VALUE) {
av_log(ctx, AV_LOG_WARNING, "First timestamp is missing, "
"assuming 0.\n");
s->next_pts = 0;
} else
s->next_pts = av_rescale_q(buf->pts, inlink->time_base,
outlink->time_base);
}
if (ret > 0) {
buf_out->audio->nb_samples = ret;
if (buf->pts != AV_NOPTS_VALUE) {
buf_out->pts = av_rescale_q(buf->pts, inlink->time_base,
outlink->time_base) -
av_rescale(delay, outlink->sample_rate,
inlink->sample_rate);
} else
buf_out->pts = s->next_pts;
s->next_pts = buf_out->pts + buf_out->audio->nb_samples;
ff_filter_samples(outlink, buf_out);
}
avfilter_unref_buffer(buf);
} else
ff_filter_samples(outlink, buf);
}
static int request_frame(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
ResampleContext *s = ctx->priv;
int ret = avfilter_request_frame(ctx->inputs[0]);
/* flush the lavr delay buffer */
if (ret == AVERROR_EOF && s->avr) {
AVFilterBufferRef *buf;
int nb_samples = av_rescale_rnd(avresample_get_delay(s->avr),
outlink->sample_rate,
ctx->inputs[0]->sample_rate,
AV_ROUND_UP);
if (!nb_samples)
return ret;
buf = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
if (!buf)
return AVERROR(ENOMEM);
ret = avresample_convert(s->avr, (void**)buf->extended_data,
buf->linesize[0], nb_samples,
NULL, 0, 0);
if (ret <= 0) {
avfilter_unref_buffer(buf);
return (ret == 0) ? AVERROR_EOF : ret;
}
buf->pts = s->next_pts;
ff_filter_samples(outlink, buf);
return 0;
}
return ret;
}
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
{
AVFilterLink *outlink = inlink->dst->outputs[0];
int16_t *taps, *endin, *in, *out;
AVFilterBufferRef *outsamples =
ff_get_audio_buffer(inlink, AV_PERM_WRITE,
insamples->audio->nb_samples);
int ret;
avfilter_copy_buffer_ref_props(outsamples, insamples);
taps = ((EarwaxContext *)inlink->dst->priv)->taps;
out = (int16_t *)outsamples->data[0];
in = (int16_t *)insamples ->data[0];
// copy part of new input and process with saved input
memcpy(taps+NUMTAPS, in, NUMTAPS * sizeof(*taps));
out = scalarproduct(taps, taps + NUMTAPS, out);
// process current input
endin = in + insamples->audio->nb_samples * 2 - NUMTAPS;
out = scalarproduct(in, endin, out);
// save part of input for next round
memcpy(taps, endin, NUMTAPS * sizeof(*taps));
ret = ff_filter_samples(outlink, outsamples);
avfilter_unref_buffer(insamples);
return ret;
}
static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *samplesref)
{
AVFilterContext *ctx = inlink->dst;
int i;
for (i = 0; i < ctx->nb_outputs; i++)
ff_filter_samples(inlink->dst->outputs[i],
avfilter_ref_buffer(samplesref, ~AV_PERM_WRITE));
}
static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *inpicref)
{
SetPTSContext *setpts = inlink->dst->priv;
double d;
AVFilterBufferRef *outpicref = avfilter_ref_buffer(inpicref, ~0);
if (!outpicref)
return AVERROR(ENOMEM);
if (isnan(setpts->var_values[VAR_STARTPTS])) {
setpts->var_values[VAR_STARTPTS] = TS2D(inpicref->pts);
setpts->var_values[VAR_STARTT ] = TS2T(inpicref->pts, inlink->time_base);
}
setpts->var_values[VAR_PTS ] = TS2D(inpicref->pts);
setpts->var_values[VAR_T ] = TS2T(inpicref->pts, inlink->time_base);
setpts->var_values[VAR_POS ] = inpicref->pos == -1 ? NAN : inpicref->pos;
switch (inlink->type) {
case AVMEDIA_TYPE_VIDEO:
setpts->var_values[VAR_INTERLACED] = inpicref->video->interlaced;
break;
case AVMEDIA_TYPE_AUDIO:
setpts->var_values[VAR_NB_SAMPLES] = inpicref->audio->nb_samples;
break;
}
d = av_expr_eval(setpts->expr, setpts->var_values, NULL);
outpicref->pts = D2TS(d);
setpts->var_values[VAR_PREV_INPTS ] = TS2D(inpicref ->pts);
setpts->var_values[VAR_PREV_INT ] = TS2T(inpicref ->pts, inlink->time_base);
setpts->var_values[VAR_PREV_OUTPTS] = TS2D(outpicref->pts);
setpts->var_values[VAR_PREV_OUTT] = TS2T(outpicref->pts, inlink->time_base);
av_dlog(inlink->dst,
"n:%"PRId64" interlaced:%d nb_samples:%d nb_consumed_samples:%d "
"pos:%"PRId64" pts:%"PRId64" t:%f -> pts:%"PRId64" t:%f\n",
(int64_t)setpts->var_values[VAR_N],
(int)setpts->var_values[VAR_INTERLACED],
(int)setpts->var_values[VAR_NB_SAMPLES],
(int)setpts->var_values[VAR_NB_CONSUMED_SAMPLES],
(int64_t)setpts->var_values[VAR_POS],
(int64_t)setpts->var_values[VAR_PREV_INPTS],
setpts->var_values[VAR_PREV_INT],
(int64_t)setpts->var_values[VAR_PREV_OUTPTS],
setpts->var_values[VAR_PREV_OUTT]);
setpts->var_values[VAR_N] += 1.0;
if (setpts->type == AVMEDIA_TYPE_AUDIO) {
setpts->var_values[VAR_NB_CONSUMED_SAMPLES] += inpicref->audio->nb_samples;
return ff_filter_samples(inlink->dst->outputs[0], outpicref);
} else
return ff_start_frame (inlink->dst->outputs[0], outpicref);
}
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *samplesref)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
if (av_cmp_q(inlink->time_base, outlink->time_base)) {
int64_t orig_pts = samplesref->pts;
samplesref->pts = av_rescale_q(samplesref->pts, inlink->time_base, outlink->time_base);
av_log(ctx, AV_LOG_DEBUG, "tb:%d/%d pts:%"PRId64" -> tb:%d/%d pts:%"PRId64"\n",
inlink ->time_base.num, inlink ->time_base.den, orig_pts,
outlink->time_base.num, outlink->time_base.den, samplesref->pts);
}
return ff_filter_samples(outlink, samplesref);
}
static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
{
AConvertContext *aconvert = inlink->dst->priv;
const int n = insamplesref->audio->nb_samples;
AVFilterLink *const outlink = inlink->dst->outputs[0];
AVFilterBufferRef *outsamplesref = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n);
swr_convert(aconvert->swr, outsamplesref->data, n,
(void *)insamplesref->data, n);
avfilter_copy_buffer_ref_props(outsamplesref, insamplesref);
outsamplesref->audio->channel_layout = outlink->channel_layout;
ff_filter_samples(outlink, outsamplesref);
avfilter_unref_buffer(insamplesref);
}
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
AVFilterBufferRef *outsamples = insamples;
if (av_cmp_q(inlink->time_base, outlink->time_base)) {
outsamples = avfilter_ref_buffer(insamples, ~0);
outsamples->pts = av_rescale_q(insamples->pts, inlink->time_base, outlink->time_base);
av_log(ctx, AV_LOG_DEBUG, "tb:%d/%d pts:%"PRId64" -> tb:%d/%d pts:%"PRId64"\n",
inlink ->time_base.num, inlink ->time_base.den, insamples ->pts,
outlink->time_base.num, outlink->time_base.den, outsamples->pts);
avfilter_unref_buffer(insamples);
}
return ff_filter_samples(outlink, outsamples);
}
static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *samplesref)
{
AVFilterContext *ctx = inlink->dst;
ShowInfoContext *showinfo = ctx->priv;
uint32_t plane_checksum[8] = {0}, checksum = 0;
char chlayout_str[128];
int plane;
int linesize =
samplesref->audio->nb_samples *
av_get_bytes_per_sample(samplesref->format);
if (!av_sample_fmt_is_planar(samplesref->format))
linesize *= av_get_channel_layout_nb_channels(samplesref->audio->channel_layout);
for (plane = 0; samplesref->data[plane] && plane < 8; plane++) {
uint8_t *data = samplesref->data[plane];
plane_checksum[plane] = av_adler32_update(plane_checksum[plane],
data, linesize);
checksum = av_adler32_update(checksum, data, linesize);
}
av_get_channel_layout_string(chlayout_str, sizeof(chlayout_str), -1,
samplesref->audio->channel_layout);
av_log(ctx, AV_LOG_INFO,
"n:%d pts:%s pts_time:%s pos:%"PRId64" "
"fmt:%s chlayout:%s nb_samples:%d rate:%d "
"checksum:%08X plane_checksum[%08X",
showinfo->frame,
av_ts2str(samplesref->pts), av_ts2timestr(samplesref->pts, &inlink->time_base),
samplesref->pos,
av_get_sample_fmt_name(samplesref->format),
chlayout_str,
samplesref->audio->nb_samples,
samplesref->audio->sample_rate,
checksum,
plane_checksum[0]);
for (plane = 1; samplesref->data[plane] && plane < 8; plane++)
av_log(ctx, AV_LOG_INFO, " %08X", plane_checksum[plane]);
av_log(ctx, AV_LOG_INFO, "]\n");
showinfo->frame++;
ff_filter_samples(inlink->dst->outputs[0], samplesref);
}
/* FIXME: samplesref is same as link->cur_buf. Need to consider removing the redundant parameter. */
void ff_default_filter_samples(AVFilterLink *inlink, AVFilterBufferRef *samplesref)
{
AVFilterLink *outlink = NULL;
if (inlink->dst->output_count)
outlink = inlink->dst->outputs[0];
if (outlink) {
outlink->out_buf = ff_default_get_audio_buffer(inlink, AV_PERM_WRITE,
samplesref->audio->nb_samples);
outlink->out_buf->pts = samplesref->pts;
outlink->out_buf->audio->sample_rate = samplesref->audio->sample_rate;
ff_filter_samples(outlink, avfilter_ref_buffer(outlink->out_buf, ~0));
avfilter_unref_buffer(outlink->out_buf);
outlink->out_buf = NULL;
}
avfilter_unref_buffer(samplesref);
inlink->cur_buf = NULL;
}
int ff_filter_frame(AVFilterLink *link, AVFilterBufferRef *frame)
{
int ret;
FF_TPRINTF_START(NULL, filter_frame); ff_tlog_link(NULL, link, 1); ff_tlog(NULL, " "); ff_tlog_ref(NULL, frame, 1);
switch (link->type) {
case AVMEDIA_TYPE_VIDEO:
if((ret = ff_start_frame(link, frame)) < 0)
return ret;
if((ret = ff_draw_slice(link, 0, frame->video->h, 1)) < 0)
return ret;
if((ret = ff_end_frame(link)) < 0)
return ret;
return ret;
case AVMEDIA_TYPE_AUDIO:
return ff_filter_samples(link, frame);
default: return AVERROR(EINVAL);
}
}
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *samplesref)
{
AVFilterContext *ctx = inlink->dst;
int i, ret = 0;
for (i = 0; i < ctx->nb_outputs; i++) {
AVFilterBufferRef *buf_out = avfilter_ref_buffer(samplesref,
~AV_PERM_WRITE);
if (!buf_out) {
ret = AVERROR(ENOMEM);
break;
}
ret = ff_filter_samples(inlink->dst->outputs[i], buf_out);
if (ret < 0)
break;
}
avfilter_unref_buffer(samplesref);
return ret;
}
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
{
AVFilterContext *ctx = inlink->dst;
AShowInfoContext *s = ctx->priv;
char chlayout_str[128];
uint32_t checksum = 0;
int channels = av_get_channel_layout_nb_channels(buf->audio->channel_layout);
int planar = av_sample_fmt_is_planar(buf->format);
int block_align = av_get_bytes_per_sample(buf->format) * (planar ? 1 : channels);
int data_size = buf->audio->nb_samples * block_align;
int planes = planar ? channels : 1;
int i;
for (i = 0; i < planes; i++) {
uint8_t *data = buf->extended_data[i];
s->plane_checksums[i] = av_adler32_update(0, data, data_size);
checksum = i ? av_adler32_update(checksum, data, data_size) :
s->plane_checksums[0];
}
av_get_channel_layout_string(chlayout_str, sizeof(chlayout_str), -1,
buf->audio->channel_layout);
av_log(ctx, AV_LOG_INFO,
"n:%"PRIu64" pts:%"PRId64" pts_time:%f "
"fmt:%s chlayout:%s rate:%d nb_samples:%d "
"checksum:%08X ",
s->frame, buf->pts, buf->pts * av_q2d(inlink->time_base),
av_get_sample_fmt_name(buf->format), chlayout_str,
buf->audio->sample_rate, buf->audio->nb_samples,
checksum);
av_log(ctx, AV_LOG_INFO, "plane_checksums: [ ");
for (i = 0; i < planes; i++)
av_log(ctx, AV_LOG_INFO, "%08X ", s->plane_checksums[i]);
av_log(ctx, AV_LOG_INFO, "]\n");
s->frame++;
return ff_filter_samples(inlink->dst->outputs[0], buf);
}
static int send_out(AVFilterContext *ctx, int out_id)
{
AStreamSyncContext *as = ctx->priv;
struct buf_queue *queue = &as->queue[out_id];
AVFilterBufferRef *buf = queue->buf[queue->tail];
int ret;
queue->buf[queue->tail] = NULL;
as->var_values[VAR_B1 + out_id]++;
as->var_values[VAR_S1 + out_id] += buf->audio->nb_samples;
if (buf->pts != AV_NOPTS_VALUE)
as->var_values[VAR_T1 + out_id] =
av_q2d(ctx->outputs[out_id]->time_base) * buf->pts;
as->var_values[VAR_T1 + out_id] += buf->audio->nb_samples /
(double)ctx->inputs[out_id]->sample_rate;
ret = ff_filter_samples(ctx->outputs[out_id], buf);
queue->nb--;
queue->tail = (queue->tail + 1) % QUEUE_SIZE;
if (as->req[out_id])
as->req[out_id]--;
return ret;
}
static int request_frame(AVFilterLink *outlink)
{
FifoContext *fifo = outlink->src->priv;
int ret = 0;
if (!fifo->root.next) {
if ((ret = ff_request_frame(outlink->src->inputs[0])) < 0)
return ret;
av_assert0(fifo->root.next);
}
/* by doing this, we give ownership of the reference to the next filter,
* so we don't have to worry about dereferencing it ourselves. */
switch (outlink->type) {
case AVMEDIA_TYPE_VIDEO:
if ((ret = ff_start_frame(outlink, fifo->root.next->buf)) < 0 ||
(ret = ff_draw_slice(outlink, 0, outlink->h, 1)) < 0 ||
(ret = ff_end_frame(outlink)) < 0)
return ret;
queue_pop(fifo);
break;
case AVMEDIA_TYPE_AUDIO:
if (outlink->request_samples) {
return return_audio_frame(outlink->src);
} else {
ret = ff_filter_samples(outlink, fifo->root.next->buf);
queue_pop(fifo);
}
break;
default:
return AVERROR(EINVAL);
}
return ret;
}
static int return_audio_frame(AVFilterContext *ctx)
{
AVFilterLink *link = ctx->outputs[0];
FifoContext *s = ctx->priv;
AVFilterBufferRef *head = s->root.next->buf;
AVFilterBufferRef *buf_out;
int ret;
if (!s->buf_out &&
head->audio->nb_samples >= link->request_samples &&
calc_ptr_alignment(head) >= 32) {
if (head->audio->nb_samples == link->request_samples) {
buf_out = head;
queue_pop(s);
} else {
buf_out = avfilter_ref_buffer(head, AV_PERM_READ);
if (!buf_out)
return AVERROR(ENOMEM);
buf_out->audio->nb_samples = link->request_samples;
buffer_offset(link, head, link->request_samples);
}
} else {
int nb_channels = av_get_channel_layout_nb_channels(link->channel_layout);
if (!s->buf_out) {
s->buf_out = ff_get_audio_buffer(link, AV_PERM_WRITE,
link->request_samples);
if (!s->buf_out)
return AVERROR(ENOMEM);
s->buf_out->audio->nb_samples = 0;
s->buf_out->pts = head->pts;
s->allocated_samples = link->request_samples;
} else if (link->request_samples != s->allocated_samples) {
av_log(ctx, AV_LOG_ERROR, "request_samples changed before the "
"buffer was returned.\n");
return AVERROR(EINVAL);
}
while (s->buf_out->audio->nb_samples < s->allocated_samples) {
int len = FFMIN(s->allocated_samples - s->buf_out->audio->nb_samples,
head->audio->nb_samples);
av_samples_copy(s->buf_out->extended_data, head->extended_data,
s->buf_out->audio->nb_samples, 0, len, nb_channels,
link->format);
s->buf_out->audio->nb_samples += len;
if (len == head->audio->nb_samples) {
avfilter_unref_buffer(head);
queue_pop(s);
if (!s->root.next &&
(ret = ff_request_frame(ctx->inputs[0])) < 0) {
if (ret == AVERROR_EOF) {
av_samples_set_silence(s->buf_out->extended_data,
s->buf_out->audio->nb_samples,
s->allocated_samples -
s->buf_out->audio->nb_samples,
nb_channels, link->format);
s->buf_out->audio->nb_samples = s->allocated_samples;
break;
}
return ret;
}
head = s->root.next->buf;
} else {
buffer_offset(link, head, len);
}
}
buf_out = s->buf_out;
s->buf_out = NULL;
}
return ff_filter_samples(link, buf_out);
}
void ff_null_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
{
ff_filter_samples(link->dst->outputs[0], samplesref);
}
static void default_filter_samples(AVFilterLink *link,
AVFilterBufferRef *samplesref)
{
ff_filter_samples(link->dst->outputs[0], samplesref);
}
