> On Jun 2, 2016, at 2:32 AM, Rick Kern <ker...@gmail.com> wrote:
>
> AudioConverterFillComplexBuffer() doesn't always call its callback. A frame
> queue is used to prevent skipped audio samples.
>
> Signed-off-by: Rick Kern <ker...@gmail.com>
> ---
> libavcodec/audiotoolboxenc.c | 78 +++++++++++++++++++++++++++++---------------
> 1 file changed, 52 insertions(+), 26 deletions(-)
>
> diff --git a/libavcodec/audiotoolboxenc.c b/libavcodec/audiotoolboxenc.c
> index 855df0c..262308a 100644
> --- a/libavcodec/audiotoolboxenc.c
> +++ b/libavcodec/audiotoolboxenc.c
> @@ -22,6 +22,9 @@
>
> #include <AudioToolbox/AudioToolbox.h>
>
> +#define FF_BUFQUEUE_SIZE 256
> +#include "libavfilter/bufferqueue.h"
> +
> #include "config.h"
> #include "audio_frame_queue.h"
> #include "avcodec.h"
> @@ -38,8 +41,8 @@ typedef struct ATDecodeContext {
> int quality;
>
> AudioConverterRef converter;
> - AVFrame in_frame;
> - AVFrame new_in_frame;
> + struct FFBufQueue frame_queue;
> + struct FFBufQueue used_frame_queue;
>
> unsigned pkt_size;
> AudioFrameQueue afq;
> @@ -449,28 +452,30 @@ static OSStatus ffat_encode_callback(AudioConverterRef
> converter, UInt32 *nb_pac
> {
> AVCodecContext *avctx = inctx;
> ATDecodeContext *at = avctx->priv_data;
> + AVFrame *frame;
>
> - if (at->eof) {
> - *nb_packets = 0;
> - return 0;
> + if (!at->frame_queue.available) {
> + if (at->eof) {
> + *nb_packets = 0;
> + return 0;
> + } else {
> + *nb_packets = 0;
> + return 1;
> + }
> }
>
> - av_frame_unref(&at->in_frame);
> - av_frame_move_ref(&at->in_frame, &at->new_in_frame);
> -
> - if (!at->in_frame.data[0]) {
> - *nb_packets = 0;
> - return 1;
> - }
> + frame = ff_bufqueue_get(&at->frame_queue);
>
> data->mNumberBuffers = 1;
> data->mBuffers[0].mNumberChannels = avctx->channels;
> - data->mBuffers[0].mDataByteSize = at->in_frame.nb_samples *
> + data->mBuffers[0].mDataByteSize = frame->nb_samples *
>
> av_get_bytes_per_sample(avctx->sample_fmt) *
> avctx->channels;
> - data->mBuffers[0].mData = at->in_frame.data[0];
> - if (*nb_packets > at->in_frame.nb_samples)
> - *nb_packets = at->in_frame.nb_samples;
> + data->mBuffers[0].mData = frame->data[0];
> + if (*nb_packets > frame->nb_samples)
> + *nb_packets = frame->nb_samples;
> +
> + ff_bufqueue_add(avctx, &at->used_frame_queue, frame);
>
> return 0;
> }
> @@ -492,20 +497,38 @@ static int ffat_encode(AVCodecContext *avctx, AVPacket
> *avpkt,
> };
> AudioStreamPacketDescription out_pkt_desc = {0};
>
> - if ((ret = ff_alloc_packet2(avctx, avpkt, at->pkt_size, 0)) < 0)
> - return ret;
> -
> - av_frame_unref(&at->new_in_frame);
> -
> if (frame) {
> + AVFrame *in_frame;
> +
> + if (ff_bufqueue_is_full(&at->frame_queue)) {
> + /*
> + * The frame queue is significantly larger than needed in
> practice,
> + * but no clear way to determine the minimum number of samples to
> + * get output from AudioConverterFillComplexBuffer().
> + */
> + av_log(avctx, AV_LOG_ERROR, "Bug: frame queue is too small.\n");
> + return AVERROR_BUG;
> + }
> +
> if ((ret = ff_af_queue_add(&at->afq, frame)) < 0)
> return ret;
> - if ((ret = av_frame_ref(&at->new_in_frame, frame)) < 0)
> +
> + in_frame = av_frame_alloc();
> + if (!in_frame)
> + return AVERROR(ENOMEM);
> +
> + if ((ret = av_frame_ref(in_frame, frame)) < 0)
> return ret;
> +
> + ff_bufqueue_add(avctx, &at->frame_queue, in_frame);
> } else {
> at->eof = 1;
> }
>
> + if ((ret = ff_alloc_packet2(avctx, avpkt, at->pkt_size, 0)) < 0)
> + return ret;
> +
> +
> out_buffers.mBuffers[0].mData = avpkt->data;
>
> *got_packet_ptr = avctx->frame_size / at->frame_size;
> @@ -513,6 +536,9 @@ static int ffat_encode(AVCodecContext *avctx, AVPacket
> *avpkt,
> ret = AudioConverterFillComplexBuffer(at->converter,
> ffat_encode_callback, avctx,
> got_packet_ptr, &out_buffers,
> (avctx->frame_size >
> at->frame_size) ? NULL : &out_pkt_desc);
> +
> + ff_bufqueue_discard_all(&at->used_frame_queue);
> +
> if ((!ret || ret == 1) && *got_packet_ptr) {
> avpkt->size = out_buffers.mBuffers[0].mDataByteSize;
> ff_af_queue_remove(&at->afq, out_pkt_desc.mVariableFramesInPacket ?
> @@ -531,16 +557,16 @@ static av_cold void ffat_encode_flush(AVCodecContext
> *avctx)
> {
> ATDecodeContext *at = avctx->priv_data;
> AudioConverterReset(at->converter);
> - av_frame_unref(&at->new_in_frame);
> - av_frame_unref(&at->in_frame);
> + ff_bufqueue_discard_all(&at->frame_queue);
> + ff_bufqueue_discard_all(&at->used_frame_queue);
> }
>
> static av_cold int ffat_close_encoder(AVCodecContext *avctx)
> {
> ATDecodeContext *at = avctx->priv_data;
> AudioConverterDispose(at->converter);
> - av_frame_unref(&at->new_in_frame);
> - av_frame_unref(&at->in_frame);
> + ff_bufqueue_discard_all(&at->frame_queue);
> + ff_bufqueue_discard_all(&at->used_frame_queue);
> ff_af_queue_close(&at->afq);
> return 0;
> }
> --
> 2.7.4
>
Roger Combs - I see in git that you wrote/maintain this encoder. Would you have
time to review?
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