[FFmpeg-devel] [PATCH v2] lavc/audiotoolboxenc: fix noise in encoded audio

[James Almer]

On 1/2/2018 12:03 PM, zhangjiejun1992 at gmail.com wrote:
> From: Jiejun Zhang <zhangjiejun1992 at gmail.com>
> 
> This fixes #6940
> 
> Although undocumented, AudioToolbox seems to require the data supplied
> by the callback (i.e. ffat_encode_callback) being unchanged until the
> next time the callback is called. In the old implementation, the
> AVBuffer backing the frame is recycled after the frame is freed, and
> somebody else (maybe the decoder) will write into the AVBuffer and
> change the data. AudioToolbox then encodes some wrong data and noise
> is produced. Copying the data to a separate buffer solves this
> problem.
> ---
>  libavcodec/audiotoolboxenc.c | 32 +++++++++++++++++++++++++++-----
>  1 file changed, 27 insertions(+), 5 deletions(-)
> 
> diff --git a/libavcodec/audiotoolboxenc.c b/libavcodec/audiotoolboxenc.c
> index 71885d1530..dcac88cdde 100644
> --- a/libavcodec/audiotoolboxenc.c
> +++ b/libavcodec/audiotoolboxenc.c
> @@ -48,6 +48,9 @@ typedef struct ATDecodeContext {
>      AudioFrameQueue afq;
>      int eof;
>      int frame_size;
> +
> +    uint8_t* audio_data_buf;
> +    uint32_t audio_data_buf_size;
>  } ATDecodeContext;
>  
>  static UInt32 ffat_get_format_id(enum AVCodecID codec, int profile)
> @@ -442,6 +445,9 @@ static av_cold int ffat_init_encoder(AVCodecContext *avctx)
>  
>      ff_af_queue_init(avctx, &at->afq);
>  
> +    at->audio_data_buf_size = 0;
> +    at->audio_data_buf = NULL;

The entire ATDecodeContext struct is zero initialized when allocated, so
this is not needed.

> +
>      return 0;
>  }
>  
> @@ -465,13 +471,27 @@ static OSStatus ffat_encode_callback(AudioConverterRef converter, UInt32 *nb_pac
>      }
>  
>      frame = ff_bufqueue_get(&at->frame_queue);
> -
> +    int audio_data_size = frame->nb_samples *
> +                          av_get_bytes_per_sample(avctx->sample_fmt) *
> +                          avctx->channels;
> +    if (at->audio_data_buf_size < audio_data_size) {
> +        av_log(avctx, AV_LOG_INFO, "Increasing audio data buffer size to %d\n",

Verbose or debug level, please.

> +               audio_data_size);
> +        av_free(at->audio_data_buf);
> +        at->audio_data_buf_size = audio_data_size;
> +        at->audio_data_buf = av_malloc(at->audio_data_buf_size);
> +        if (!at->audio_data_buf) {
> +            at->audio_data_buf_size = 0;
> +            data->mNumberBuffers = 0;
> +            *nb_packets = 0;
> +            return AVERROR(ENOMEM);
> +        }
> +    }
>      data->mNumberBuffers              = 1;
>      data->mBuffers[0].mNumberChannels = avctx->channels;
> -    data->mBuffers[0].mDataByteSize   = frame->nb_samples *
> -                                        av_get_bytes_per_sample(avctx->sample_fmt) *
> -                                        avctx->channels;
> -    data->mBuffers[0].mData           = frame->data[0];
> +    data->mBuffers[0].mDataByteSize   = audio_data_size;
> +    data->mBuffers[0].mData           = at->audio_data_buf;
> +    memcpy(at->audio_data_buf, frame->data[0], data->mBuffers[0].mDataByteSize);

Can't you instead create a new reference for the frame buffer? Or will
making it non writable break things further into the process? It would
save you a memcpy per frame.

>      if (*nb_packets > frame->nb_samples)
>          *nb_packets = frame->nb_samples;
>  
> @@ -565,6 +585,8 @@ static av_cold int ffat_close_encoder(AVCodecContext *avctx)
>      ff_bufqueue_discard_all(&at->frame_queue);
>      ff_bufqueue_discard_all(&at->used_frame_queue);
>      ff_af_queue_close(&at->afq);
> +    at->audio_data_buf_size = 0;
> +    av_freep(&at->audio_data_buf);
>      return 0;
>  }
>  
>