[FFmpeg-cvslog] libfdk-aacdec: Flush delayed samples at the end
Martin Storsjö
git at videolan.org
Fri Feb 4 00:04:28 EET 2022
ffmpeg | branch: master | Martin Storsjö <martin at martin.st> | Thu Aug 5 14:54:33 2021 +0300| [c69b1a12bb6231bd648cf29c1d99282e2d5e68d0] | committer: Martin Storsjö
libfdk-aacdec: Flush delayed samples at the end
The fdk-aac decoder can return decoded audio data with a delay.
(Whether it does this or not depends on the options set; by default
it does add some delay.) Previously, this delay was handled by
adjusting the timestamps of the decoded frames, but the last delayed
samples weren't returned.
Set the AV_CODEC_CAP_DELAY flag to indicate that the caller should
flush remaining samples at the end. Also trim off the corresponding
amount of samples at the start instead of adjusting timestamps.
Signed-off-by: Martin Storsjö <martin at martin.st>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=c69b1a12bb6231bd648cf29c1d99282e2d5e68d0
---
libavcodec/libfdk-aacdec.c | 80 +++++++++++++++++++++++++++++++++++++---------
1 file changed, 65 insertions(+), 15 deletions(-)
diff --git a/libavcodec/libfdk-aacdec.c b/libavcodec/libfdk-aacdec.c
index 93b52023b0..d560e313ca 100644
--- a/libavcodec/libfdk-aacdec.c
+++ b/libavcodec/libfdk-aacdec.c
@@ -58,7 +58,11 @@ typedef struct FDKAACDecContext {
int drc_cut;
int album_mode;
int level_limit;
- int output_delay;
+#if FDKDEC_VER_AT_LEAST(2, 5) // 2.5.10
+ int output_delay_set;
+ int flush_samples;
+ int delay_samples;
+#endif
} FDKAACDecContext;
@@ -123,7 +127,12 @@ static int get_stream_info(AVCodecContext *avctx)
avctx->sample_rate = info->sampleRate;
avctx->frame_size = info->frameSize;
#if FDKDEC_VER_AT_LEAST(2, 5) // 2.5.10
- s->output_delay = info->outputDelay;
+ if (!s->output_delay_set && info->outputDelay) {
+ // Set this only once.
+ s->flush_samples = info->outputDelay;
+ s->delay_samples = info->outputDelay;
+ s->output_delay_set = 1;
+ }
#endif
for (i = 0; i < info->numChannels; i++) {
@@ -367,14 +376,31 @@ static int fdk_aac_decode_frame(AVCodecContext *avctx, void *data,
int ret;
AAC_DECODER_ERROR err;
UINT valid = avpkt->size;
+ UINT flags = 0;
+ int input_offset = 0;
- err = aacDecoder_Fill(s->handle, &avpkt->data, &avpkt->size, &valid);
- if (err != AAC_DEC_OK) {
- av_log(avctx, AV_LOG_ERROR, "aacDecoder_Fill() failed: %x\n", err);
- return AVERROR_INVALIDDATA;
+ if (avpkt->size) {
+ err = aacDecoder_Fill(s->handle, &avpkt->data, &avpkt->size, &valid);
+ if (err != AAC_DEC_OK) {
+ av_log(avctx, AV_LOG_ERROR, "aacDecoder_Fill() failed: %x\n", err);
+ return AVERROR_INVALIDDATA;
+ }
+ } else {
+#if FDKDEC_VER_AT_LEAST(2, 5) // 2.5.10
+ /* Handle decoder draining */
+ if (s->flush_samples > 0) {
+ flags |= AACDEC_FLUSH;
+ } else {
+ return AVERROR_EOF;
+ }
+#else
+ return AVERROR_EOF;
+#endif
}
- err = aacDecoder_DecodeFrame(s->handle, (INT_PCM *) s->decoder_buffer, s->decoder_buffer_size / sizeof(INT_PCM), 0);
+ err = aacDecoder_DecodeFrame(s->handle, (INT_PCM *) s->decoder_buffer,
+ s->decoder_buffer_size / sizeof(INT_PCM),
+ flags);
if (err == AAC_DEC_NOT_ENOUGH_BITS) {
ret = avpkt->size - valid;
goto end;
@@ -390,16 +416,36 @@ static int fdk_aac_decode_frame(AVCodecContext *avctx, void *data,
goto end;
frame->nb_samples = avctx->frame_size;
+#if FDKDEC_VER_AT_LEAST(2, 5) // 2.5.10
+ if (flags & AACDEC_FLUSH) {
+ // Only return the right amount of samples at the end; if calling the
+ // decoder with AACDEC_FLUSH, it will keep returning frames indefinitely.
+ frame->nb_samples = FFMIN(s->flush_samples, frame->nb_samples);
+ av_log(s, AV_LOG_DEBUG, "Returning %d/%d delayed samples.\n",
+ frame->nb_samples, s->flush_samples);
+ s->flush_samples -= frame->nb_samples;
+ } else {
+ // Trim off samples from the start to compensate for extra decoder
+ // delay. We could also just adjust the pts, but this avoids
+ // including the extra samples in the output altogether.
+ if (s->delay_samples) {
+ int drop_samples = FFMIN(s->delay_samples, frame->nb_samples);
+ av_log(s, AV_LOG_DEBUG, "Dropping %d/%d delayed samples.\n",
+ drop_samples, s->delay_samples);
+ s->delay_samples -= drop_samples;
+ frame->nb_samples -= drop_samples;
+ input_offset = drop_samples * avctx->channels;
+ if (frame->nb_samples <= 0)
+ return 0;
+ }
+ }
+#endif
+
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
goto end;
- if (frame->pts != AV_NOPTS_VALUE)
- frame->pts -= av_rescale_q(s->output_delay,
- (AVRational){1, avctx->sample_rate},
- avctx->time_base);
-
- memcpy(frame->extended_data[0], s->decoder_buffer,
- avctx->channels * avctx->frame_size *
+ memcpy(frame->extended_data[0], s->decoder_buffer + input_offset,
+ avctx->channels * frame->nb_samples *
av_get_bytes_per_sample(avctx->sample_fmt));
*got_frame_ptr = 1;
@@ -432,7 +478,11 @@ const AVCodec ff_libfdk_aac_decoder = {
.decode = fdk_aac_decode_frame,
.close = fdk_aac_decode_close,
.flush = fdk_aac_decode_flush,
- .capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF,
+ .capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF
+#if FDKDEC_VER_AT_LEAST(2, 5) // 2.5.10
+ | AV_CODEC_CAP_DELAY
+#endif
+ ,
.priv_class = &fdk_aac_dec_class,
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE |
FF_CODEC_CAP_INIT_CLEANUP,
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