void ff_af_queue_remove(AudioFrameQueue *afq, int nb_samples, int64_t *pts,
int *duration)
{
int64_t out_pts = AV_NOPTS_VALUE;
int removed_samples = 0;
#ifdef DEBUG
af_queue_log_state(afq);
#endif
/* get output pts from the next frame or generated pts */
if (afq->frame_queue) {
if (afq->frame_queue->pts != AV_NOPTS_VALUE)
out_pts = afq->frame_queue->pts - afq->remaining_delay;
} else {
if (afq->next_pts != AV_NOPTS_VALUE)
out_pts = afq->next_pts - afq->remaining_delay;
}
if (pts) {
if (out_pts != AV_NOPTS_VALUE)
*pts = ff_samples_to_time_base(afq->avctx, out_pts);
else
*pts = AV_NOPTS_VALUE;
}
/* if the delay is larger than the packet duration, we use up delay samples
for the output packet and leave all frames in the queue */
if (afq->remaining_delay >= nb_samples) {
removed_samples += nb_samples;
afq->remaining_delay -= nb_samples;
}
/* remove frames from the queue until we have enough to cover the
requested number of samples or until the queue is empty */
while (removed_samples < nb_samples && afq->frame_queue) {
removed_samples += afq->frame_queue->duration;
delete_next_frame(afq);
}
afq->remaining_samples -= removed_samples;
/* if there are no frames left and we have room for more samples, use
any remaining delay samples */
if (removed_samples < nb_samples && afq->remaining_samples > 0) {
int add_samples = FFMIN(afq->remaining_samples,
nb_samples - removed_samples);
removed_samples += add_samples;
afq->remaining_samples -= add_samples;
}
if (removed_samples > nb_samples)
av_log(afq->avctx, AV_LOG_WARNING, "frame_size is too large\n");
if (duration)
*duration = ff_samples_to_time_base(afq->avctx, removed_samples);
}
static int encode_superframe(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
WMACodecContext *s = avctx->priv_data;
int i, total_gain, ret, error;
s->block_len_bits = s->frame_len_bits; // required by non variable block len
s->block_len = 1 << s->block_len_bits;
ret = apply_window_and_mdct(avctx, frame);
if (ret < 0)
return ret;
if (s->ms_stereo) {
float a, b;
int i;
for (i = 0; i < s->block_len; i++) {
a = s->coefs[0][i] * 0.5;
b = s->coefs[1][i] * 0.5;
s->coefs[0][i] = a + b;
s->coefs[1][i] = a - b;
}
}
if ((ret = ff_alloc_packet2(avctx, avpkt, 2 * MAX_CODED_SUPERFRAME_SIZE, 0)) < 0)
return ret;
total_gain = 128;
for (i = 64; i; i >>= 1) {
error = encode_frame(s, s->coefs, avpkt->data, avpkt->size,
total_gain - i);
if (error <= 0)
total_gain -= i;
}
while(total_gain <= 128 && error > 0)
error = encode_frame(s, s->coefs, avpkt->data, avpkt->size, total_gain++);
if (error > 0) {
av_log(avctx, AV_LOG_ERROR, "Invalid input data or requested bitrate too low, cannot encode\n");
avpkt->size = 0;
return AVERROR(EINVAL);
}
av_assert0((put_bits_count(&s->pb) & 7) == 0);
i= avctx->block_align - (put_bits_count(&s->pb)+7)/8;
av_assert0(i>=0);
while(i--)
put_bits(&s->pb, 8, 'N');
flush_put_bits(&s->pb);
av_assert0(put_bits_ptr(&s->pb) - s->pb.buf == avctx->block_align);
if (frame->pts != AV_NOPTS_VALUE)
avpkt->pts = frame->pts - ff_samples_to_time_base(avctx, avctx->initial_padding);
avpkt->size = avctx->block_align;
*got_packet_ptr = 1;
return 0;
}
static int encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
DCAEncContext *c = avctx->priv_data;
const int32_t *samples;
int ret, i;
if ((ret = ff_alloc_packet2(avctx, avpkt, c->frame_size, 0)) < 0)
return ret;
samples = (const int32_t *)frame->data[0];
subband_transform(c, samples);
if (c->lfe_channel)
lfe_downsample(c, samples);
calc_masking(c, samples);
find_peaks(c);
assign_bits(c);
calc_scales(c);
quantize_all(c);
shift_history(c, samples);
init_put_bits(&c->pb, avpkt->data, avpkt->size);
put_frame_header(c);
put_primary_audio_header(c);
for (i = 0; i < SUBFRAMES; i++)
put_subframe(c, i);
for (i = put_bits_count(&c->pb); i < 8*c->frame_size; i++)
put_bits(&c->pb, 1, 0);
flush_put_bits(&c->pb);
avpkt->pts = frame->pts;
avpkt->duration = ff_samples_to_time_base(avctx, frame->nb_samples);
avpkt->size = put_bits_count(&c->pb) >> 3;
*got_packet_ptr = 1;
return 0;
}
static int encode_superframe(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
WMACodecContext *s = avctx->priv_data;
int i, total_gain, ret;
s->block_len_bits= s->frame_len_bits; //required by non variable block len
s->block_len = 1 << s->block_len_bits;
apply_window_and_mdct(avctx, frame);
if (s->ms_stereo) {
float a, b;
int i;
for(i = 0; i < s->block_len; i++) {
a = s->coefs[0][i]*0.5;
b = s->coefs[1][i]*0.5;
s->coefs[0][i] = a + b;
s->coefs[1][i] = a - b;
}
}
if ((ret = ff_alloc_packet(avpkt, 2 * MAX_CODED_SUPERFRAME_SIZE))) {
av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
return ret;
}
#if 1
total_gain= 128;
for(i=64; i; i>>=1){
int error = encode_frame(s, s->coefs, avpkt->data, avpkt->size,
total_gain - i);
if(error<0)
total_gain-= i;
}
#else
total_gain= 90;
best = encode_frame(s, s->coefs, avpkt->data, avpkt->size, total_gain);
for(i=32; i; i>>=1){
int scoreL = encode_frame(s, s->coefs, avpkt->data, avpkt->size, total_gain - i);
int scoreR = encode_frame(s, s->coefs, avpkt->data, avpkt->size, total_gain + i);
av_log(NULL, AV_LOG_ERROR, "%d %d %d (%d)\n", scoreL, best, scoreR, total_gain);
if(scoreL < FFMIN(best, scoreR)){
best = scoreL;
total_gain -= i;
}else if(scoreR < best){
best = scoreR;
total_gain += i;
}
}
#endif
if ((i = encode_frame(s, s->coefs, avpkt->data, avpkt->size, total_gain)) >= 0) {
av_log(avctx, AV_LOG_ERROR, "required frame size too large. please "
"use a higher bit rate.\n");
return AVERROR(EINVAL);
}
assert((put_bits_count(&s->pb) & 7) == 0);
while (i++)
put_bits(&s->pb, 8, 'N');
flush_put_bits(&s->pb);
if (frame->pts != AV_NOPTS_VALUE)
avpkt->pts = frame->pts - ff_samples_to_time_base(avctx, avctx->delay);
avpkt->size = s->block_align;
*got_packet_ptr = 1;
return 0;
}
