Martin Storsjö:
> On Fri, 21 Jan 2022, Andreas Rheinhardt wrote:
>
>> Martin Storsjö:
>>> Also trim off delay samples at the start instead of adjusting pts
>>> to compensate for them; this avoids unwanted offsets if working
>>> with raw samples without considering their pts.
>>> ---
>>> 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,
>>>
>>
>> When I use the libfdk-aac decoder I get the exact number of samples like
>> with the native aac decoder (namely number of frames * 1024, as
>> expected). What makes you believe this is necessary?
>
> The fdk-aac decoder can have, depending on combination of options, some
> amount of extra internal delay, that the libavcodec internal aac decoder
> doesn't have. (It's also possible to set the options in a state where
> the fdk-aac decoder doesn't induce any extra delay.)
>
> Currently, we compensate for that extra delay by just offsetting pts
> backwards, so for a stream with N packets, we return samples with
> timestamps [-delay,N*framesize-delay].
>
> In order not to lose data at the end, we must make the decoder flushable
> and flush up to (delay) samples at the end. And since one doesn't
> normally expect extra delay samples at the start of an AAC decoder
> output, we also trim out the same amount of samples at the start (to
> simplify for users that don't observe the pts, who otherwise are
> surprised by the stream starting from pts -delay instead of at pts 0).
>
Interesting: There is indeed a delay at the start (720 samples in a
quick test) compared to the native AAC decoder.
Furthermore, the current code is buggy, as it believes that
avcodec->time_base to be the time_base of the returned AVFrames (it is
in reality avcodec->pkt_timebase; just test with AAC-in-Matroska for this).
I haven't tested your patches, but I have now realized that there is
indeed an issue. And your patch should also fix the wrong timebase issue.
- Andreas
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