#include "libavresample/avresample.h"
#include "avcodec.h"
+#include "bitstream.h"
#include "celp_filters.h"
#include "fft.h"
-#include "get_bits.h"
#include "internal.h"
#include "mathops.h"
#include "opus.h"
*/
static int opus_rc_init(OpusRangeCoder *rc, const uint8_t *data, int size)
{
- int ret = init_get_bits8(&rc->gb, data, size);
+ int ret = bitstream_init8(&rc->bc, data, size);
if (ret < 0)
return ret;
rc->range = 128;
- rc->value = 127 - get_bits(&rc->gb, 7);
+ rc->value = 127 - bitstream_read(&rc->bc, 7);
rc->total_read_bits = 9;
opus_rc_normalize(rc);
static int opus_decode_subpacket(OpusStreamContext *s,
const uint8_t *buf, int buf_size,
+ float **out, int out_size,
int nb_samples)
{
int output_samples = 0;
int flush_needed = 0;
int i, j, ret;
+ s->out[0] = out[0];
+ s->out[1] = out[1];
+ s->out_size = out_size;
+
/* check if we need to flush the resampler */
if (avresample_is_open(s->avr)) {
if (buf) {
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
int coded_samples = 0;
- int decoded_samples = 0;
+ int decoded_samples = INT_MAX;
+ int delayed_samples = 0;
int i, ret;
+ /* calculate the number of delayed samples */
+ for (i = 0; i < c->nb_streams; i++) {
+ delayed_samples = FFMAX(delayed_samples,
+ c->streams[i].delayed_samples + av_audio_fifo_size(c->sync_buffers[i]));
+ }
+
/* decode the header of the first sub-packet to find out the sample count */
if (buf) {
OpusPacket *pkt = &c->streams[0].packet;
c->streams[0].silk_samplerate = get_silk_samplerate(pkt->config);
}
- frame->nb_samples = coded_samples + c->streams[0].delayed_samples;
+ frame->nb_samples = coded_samples + delayed_samples;
/* no input or buffered data => nothing to do */
if (!frame->nb_samples) {
}
frame->nb_samples = 0;
+ memset(c->out, 0, c->nb_streams * 2 * sizeof(*c->out));
for (i = 0; i < avctx->channels; i++) {
ChannelMap *map = &c->channel_maps[i];
if (!map->copy)
- c->streams[map->stream_idx].out[map->channel_idx] = (float*)frame->extended_data[i];
+ c->out[2 * map->stream_idx + map->channel_idx] = (float*)frame->extended_data[i];
}
- for (i = 0; i < c->nb_streams; i++)
- c->streams[i].out_size = frame->linesize[0];
+ /* read the data from the sync buffers */
+ for (i = 0; i < c->nb_streams; i++) {
+ float **out = c->out + 2 * i;
+ int sync_size = av_audio_fifo_size(c->sync_buffers[i]);
+
+ float sync_dummy[32];
+ int out_dummy = (!out[0]) | ((!out[1]) << 1);
+
+ if (!out[0])
+ out[0] = sync_dummy;
+ if (!out[1])
+ out[1] = sync_dummy;
+ if (out_dummy && sync_size > FF_ARRAY_ELEMS(sync_dummy))
+ return AVERROR_BUG;
+
+ ret = av_audio_fifo_read(c->sync_buffers[i], (void**)out, sync_size);
+ if (ret < 0)
+ return ret;
+
+ if (out_dummy & 1)
+ out[0] = NULL;
+ else
+ out[0] += ret;
+ if (out_dummy & 2)
+ out[1] = NULL;
+ else
+ out[1] += ret;
+
+ c->out_size[i] = frame->linesize[0] - ret * sizeof(float);
+ }
/* decode each sub-packet */
for (i = 0; i < c->nb_streams; i++) {
s->silk_samplerate = get_silk_samplerate(s->packet.config);
}
- ret = opus_decode_subpacket(&c->streams[i], buf,
- s->packet.data_size, coded_samples);
+ ret = opus_decode_subpacket(&c->streams[i], buf, s->packet.data_size,
+ c->out + 2 * i, c->out_size[i], coded_samples);
if (ret < 0)
return ret;
- if (decoded_samples && ret != decoded_samples) {
- av_log(avctx, AV_LOG_ERROR, "Different numbers of decoded samples "
- "in a multi-channel stream\n");
- return AVERROR_INVALIDDATA;
- }
- decoded_samples = ret;
+ c->decoded_samples[i] = ret;
+ decoded_samples = FFMIN(decoded_samples, ret);
+
buf += s->packet.packet_size;
buf_size -= s->packet.packet_size;
}
+ /* buffer the extra samples */
+ for (i = 0; i < c->nb_streams; i++) {
+ int buffer_samples = c->decoded_samples[i] - decoded_samples;
+ if (buffer_samples) {
+ float *buf[2] = { c->out[2 * i + 0] ? c->out[2 * i + 0] : (float*)frame->extended_data[0],
+ c->out[2 * i + 1] ? c->out[2 * i + 1] : (float*)frame->extended_data[0] };
+ buf[0] += decoded_samples;
+ buf[1] += decoded_samples;
+ ret = av_audio_fifo_write(c->sync_buffers[i], (void**)buf, buffer_samples);
+ if (ret < 0)
+ return ret;
+ }
+ }
+
for (i = 0; i < avctx->channels; i++) {
ChannelMap *map = &c->channel_maps[i];
memset(frame->extended_data[i], 0, frame->linesize[0]);
}
- if (c->gain_i) {
+ if (c->gain_i && decoded_samples > 0) {
c->fdsp.vector_fmul_scalar((float*)frame->extended_data[i],
(float*)frame->extended_data[i],
c->gain, FFALIGN(decoded_samples, 8));
av_audio_fifo_drain(s->celt_delay, av_audio_fifo_size(s->celt_delay));
avresample_close(s->avr);
+ av_audio_fifo_drain(c->sync_buffers[i], av_audio_fifo_size(c->sync_buffers[i]));
+
ff_silk_flush(s->silk);
ff_celt_flush(s->celt);
}
}
av_freep(&c->streams);
+
+ if (c->sync_buffers) {
+ for (i = 0; i < c->nb_streams; i++)
+ av_audio_fifo_free(c->sync_buffers[i]);
+ }
+ av_freep(&c->sync_buffers);
+ av_freep(&c->decoded_samples);
+ av_freep(&c->out);
+ av_freep(&c->out_size);
+
c->nb_streams = 0;
av_freep(&c->channel_maps);
/* allocate and init each independent decoder */
c->streams = av_mallocz_array(c->nb_streams, sizeof(*c->streams));
- if (!c->streams) {
+ c->out = av_mallocz_array(c->nb_streams, 2 * sizeof(*c->out));
+ c->out_size = av_mallocz_array(c->nb_streams, sizeof(*c->out_size));
+ c->sync_buffers = av_mallocz_array(c->nb_streams, sizeof(*c->sync_buffers));
+ c->decoded_samples = av_mallocz_array(c->nb_streams, sizeof(*c->decoded_samples));
+ if (!c->streams || !c->sync_buffers || !c->decoded_samples || !c->out || !c->out_size) {
c->nb_streams = 0;
ret = AVERROR(ENOMEM);
goto fail;
ret = AVERROR(ENOMEM);
goto fail;
}
+
+ c->sync_buffers[i] = av_audio_fifo_alloc(avctx->sample_fmt,
+ s->output_channels, 32);
+ if (!c->sync_buffers[i]) {
+ ret = AVERROR(ENOMEM);
+ goto fail;
+ }
}
return 0;
projects / ffmpeg / blobdiff