[FFmpeg-devel] [PATCH 2/2] libfdk-aacdec: Flush delayed samples at the end

[Andreas Rheinhardt]

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?

- Andreas