#include <AudioToolbox/AudioToolbox.h>
+#define FF_BUFQUEUE_SIZE 256
+#include "libavfilter/bufferqueue.h"
+
#include "config.h"
#include "audio_frame_queue.h"
#include "avcodec.h"
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;
{
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;
}
};
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)
- return ret;
+
+ in_frame = av_frame_clone(frame);
+ if (!in_frame)
+ return AVERROR(ENOMEM);
+
+ 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;
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 ?
{
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;
}