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opusdec: properly handle mismatching configurations in multichannel streams

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The substreams can have different resampling delays, so an additional
level of buffering is needed to synchronize them.

Bug-Id: 876
This commit is contained in:
Anton Khirnov
2015-07-27 11:13:53 +02:00
parent 859ce02c98
commit c49b88b93b
2 changed files with 99 additions and 14 deletions
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10
libavcodec/opus.h
View File

@@ -164,6 +164,16 @@ typedef struct ChannelMap {
typedef struct OpusContext {
OpusStreamContext *streams;
/* current output buffers for each streams */
float **out;
int *out_size;
/* Buffers for synchronizing the streams when they have different
* resampling delays */
AVAudioFifo **sync_buffers;
/* number of decoded samples for each stream */
int *decoded_samples;
int nb_streams;
int nb_stereo_streams;

103
libavcodec/opusdec.c
View File

@@ -367,12 +367,17 @@ static int opus_decode_frame(OpusStreamContext *s, const uint8_t *data, int size
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) {
@@ -450,9 +455,16 @@ static int opus_decode_packet(AVCodecContext *avctx, void *data,
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;
@@ -465,7 +477,7 @@ static int opus_decode_packet(AVCodecContext *avctx, void *data,
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) {
@@ -481,14 +493,43 @@ static int opus_decode_packet(AVCodecContext *avctx, void *data,
}
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++) {
@@ -509,20 +550,31 @@ static int opus_decode_packet(AVCodecContext *avctx, void *data,
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] += buffer_samples;
buf[1] += buffer_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];
@@ -563,6 +615,8 @@ static av_cold void opus_decode_flush(AVCodecContext *ctx)
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);
}
@@ -587,6 +641,16 @@ static av_cold int opus_decode_close(AVCodecContext *avctx)
}
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);
@@ -611,7 +675,11 @@ static av_cold int opus_decode_init(AVCodecContext *avctx)
/* 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;
@@ -658,6 +726,13 @@ static av_cold int opus_decode_init(AVCodecContext *avctx)
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;