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

[Steven Liu]

2018-01-02 16:59 GMT+08:00  <zhangjiejun1992 at gmail.com>:
> From: Jiejun Zhang <zhangjiejun1992 at gmail.com>
>
> This fixes #6940
> ---
>  libavcodec/audiotoolboxenc.c | 34 +++++++++++++++++++++++++++++-----
>  1 file changed, 29 insertions(+), 5 deletions(-)
>
> diff --git a/libavcodec/audiotoolboxenc.c b/libavcodec/audiotoolboxenc.c
> index 71885d1530..0c1e5feadc 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;
> +
>      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",
> +               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);
>      if (*nb_packets > frame->nb_samples)
>          *nb_packets = frame->nb_samples;
>
> @@ -565,6 +585,10 @@ 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);
> +    if (at->audio_data_buf_size > 0) {
> +        at->audio_data_buf_size = 0;
> +        av_free(at->audio_data_buf);
> +    }
>      return 0;
>  }


LGTM


Thanks

Steven