[FFmpeg-devel] [PATCH] avcodec/vorbisenc: Fix memory leak on errors

Rostislav Pehlivanov atomnuker at gmail.com
Tue Jun 6 19:59:26 EEST 2017


On 6 June 2017 at 15:06, Tyler Jones <tdjones879 at gmail.com> wrote:

> Switches temporary samples for processing to be stored in the encoder's
> context, avoids memory leaks if any errors occur while encoding a frame.
> Fixes CID1412026
>
> Signed-off-by: Tyler Jones <tdjones879 at gmail.com>
> ---
>  libavcodec/vorbisenc.c | 49 ++++++++++++------------------
> -------------------
>  1 file changed, 12 insertions(+), 37 deletions(-)
>
> diff --git a/libavcodec/vorbisenc.c b/libavcodec/vorbisenc.c
> index 856f590..afded40 100644
> --- a/libavcodec/vorbisenc.c
> +++ b/libavcodec/vorbisenc.c
> @@ -112,6 +112,7 @@ typedef struct vorbis_enc_context {
>      float *samples;
>      float *floor;  // also used for tmp values for mdct
>      float *coeffs; // also used for residue after floor
> +    float *scratch; // used for tmp values for psy model
>      float quality;
>
>      AudioFrameQueue afq;
> @@ -452,7 +453,9 @@ static int create_vorbis_context(vorbis_enc_context
> *venc,
>      venc->samples    = av_malloc_array(sizeof(float) * venc->channels, (1
> << venc->log2_blocksize[1]));
>      venc->floor      = av_malloc_array(sizeof(float) * venc->channels, (1
> << venc->log2_blocksize[1]) / 2);
>      venc->coeffs     = av_malloc_array(sizeof(float) * venc->channels, (1
> << venc->log2_blocksize[1]) / 2);
> -    if (!venc->saved || !venc->samples || !venc->floor || !venc->coeffs)
> +    venc->scratch    = av_malloc_array(sizeof(float) * venc->channels, (1
> << venc->log2_blocksize[1]) / 2);
> +
> +    if (!venc->saved || !venc->samples || !venc->floor || !venc->coeffs
> || !venc->scratch)
>          return AVERROR(ENOMEM);
>
>      if ((ret = dsp_init(avctx, venc)) < 0)
> @@ -992,7 +995,7 @@ static int residue_encode(vorbis_enc_context *venc,
> vorbis_enc_residue *rc,
>  }
>
>  static int apply_window_and_mdct(vorbis_enc_context *venc,
> -                                 float **audio, int samples)
> +                                 float *audio, int samples)
>  {
>      int channel;
>      const float * win = venc->win[0];
> @@ -1017,7 +1020,7 @@ static int apply_window_and_mdct(vorbis_enc_context
> *venc,
>          for (channel = 0; channel < venc->channels; channel++) {
>              float *offset = venc->samples + channel * window_len * 2 +
> window_len;
>
> -            fdsp->vector_fmul_reverse(offset, audio[channel], win,
> samples);
> +            fdsp->vector_fmul_reverse(offset, audio + channel *
> window_len, win, samples);
>              fdsp->vector_fmul_scalar(offset, offset, 1/n, samples);
>          }
>      } else {
> @@ -1034,7 +1037,7 @@ static int apply_window_and_mdct(vorbis_enc_context
> *venc,
>          for (channel = 0; channel < venc->channels; channel++) {
>              float *offset = venc->saved + channel * window_len;
>
> -            fdsp->vector_fmul(offset, audio[channel], win, samples);
> +            fdsp->vector_fmul(offset, audio + channel * window_len, win,
> samples);
>              fdsp->vector_fmul_scalar(offset, offset, 1/n, samples);
>          }
>          venc->have_saved = 1;
> @@ -1068,28 +1071,8 @@ static AVFrame *spawn_empty_frame(AVCodecContext
> *avctx, int channels)
>      return f;
>  }
>
> -static float **alloc_audio_arrays(int channels, int frame_size)
> -{
> -    float **audio = av_mallocz_array(channels, sizeof(float *));
> -    if (!audio)
> -        return NULL;
> -
> -    for (int ch = 0; ch < channels; ch++) {
> -        audio[ch] = av_mallocz_array(frame_size, sizeof(float));
> -        if (!audio[ch]) {
> -            // alloc has failed, free everything allocated thus far
> -            for (ch--; ch >= 0; ch--)
> -                av_free(audio[ch]);
> -            av_free(audio);
> -            return NULL;
> -        }
> -    }
> -
> -    return audio;
> -}
> -
>  /* Concatenate audio frames into an appropriately sized array of samples
> */
> -static void move_audio(vorbis_enc_context *venc, float **audio, int
> *samples, int sf_size)
> +static void move_audio(vorbis_enc_context *venc, float *audio, int
> *samples, int sf_size)
>  {
>      AVFrame *cur = NULL;
>      int frame_size = 1 << (venc->log2_blocksize[1] - 1);
> @@ -1102,7 +1085,7 @@ static void move_audio(vorbis_enc_context *venc,
> float **audio, int *samples, in
>          for (int ch = 0; ch < venc->channels; ch++) {
>              const float *input = (float *) cur->extended_data[ch];
>              const size_t len  = cur->nb_samples * sizeof(float);
> -            memcpy(&audio[ch][sf*sf_size], input, len);
> +            memcpy(audio + ch*frame_size + sf*sf_size, input, len);
>          }
>          av_frame_free(&cur);
>      }
> @@ -1112,7 +1095,6 @@ static int vorbis_encode_frame(AVCodecContext
> *avctx, AVPacket *avpkt,
>                                 const AVFrame *frame, int *got_packet_ptr)
>  {
>      vorbis_enc_context *venc = avctx->priv_data;
> -    float **audio = NULL;
>      int i, ret, need_more;
>      int samples = 0, frame_size = 1 << (venc->log2_blocksize[1] - 1);
>      vorbis_enc_mode *mode;
> @@ -1132,10 +1114,6 @@ static int vorbis_encode_frame(AVCodecContext
> *avctx, AVPacket *avpkt,
>      if (need_more)
>          return 0;
>
> -    audio = alloc_audio_arrays(venc->channels, frame_size);
> -    if (!audio)
> -        return AVERROR(ENOMEM);
> -
>      /* Pad the bufqueue with empty frames for encoding the last packet. */
>      if (!frame) {
>          if (venc->bufqueue.available * avctx->frame_size < frame_size) {
> @@ -1151,9 +1129,9 @@ static int vorbis_encode_frame(AVCodecContext
> *avctx, AVPacket *avpkt,
>          }
>      }
>
> -    move_audio(venc, audio, &samples, avctx->frame_size);
> +    move_audio(venc, venc->scratch, &samples, avctx->frame_size);
>
> -    if (!apply_window_and_mdct(venc, audio, samples))
> +    if (!apply_window_and_mdct(venc, venc->scratch, samples))
>          return 0;
>
>      if ((ret = ff_alloc_packet2(avctx, avpkt, 8192, 0)) < 0)
> @@ -1213,10 +1191,6 @@ static int vorbis_encode_frame(AVCodecContext
> *avctx, AVPacket *avpkt,
>      flush_put_bits(&pb);
>      avpkt->size = put_bits_count(&pb) >> 3;
>
> -    for (int ch = 0; ch < venc->channels; ch++)
> -        av_free(audio[ch]);
> -    av_free(audio);
> -
>      ff_af_queue_remove(&venc->afq, frame_size, &avpkt->pts,
> &avpkt->duration);
>
>      if (frame_size > avpkt->duration) {
> @@ -1281,6 +1255,7 @@ static av_cold int vorbis_encode_close(AVCodecContext
> *avctx)
>      av_freep(&venc->samples);
>      av_freep(&venc->floor);
>      av_freep(&venc->coeffs);
> +    av_freep(&venc->scratch);
>      av_freep(&venc->fdsp);
>
>      ff_mdct_end(&venc->mdct[0]);
> --
> 2.7.4
>
>
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>
>
Pushed, thanks


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