On 6 June 2017 at 15:06, Tyler Jones <tdjones...@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 <tdjones...@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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> ffmpeg-devel@ffmpeg.org
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>
>
Pushed, thanks
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