[FFmpeg-devel] [PATCH 1/2] audio_frame_que: simplify
Michael Niedermayer
michaelni at gmx.at
Wed Apr 11 20:41:16 CEST 2012
Also update libav->ffmpeg as theres pretty much no code left from libav.
The new code is faster, requires fewer mallocs and less memory. Its
also half the number of lines of code.
This code is not 100% identical in behavior to the previous, but the
differences appear to be rather limitations of the previous design
than intended though i could be wrong of course.
Signed-off-by: Michael Niedermayer <michaelni at gmx.at>
---
libavcodec/audio_frame_queue.c | 150 +++++++++++++---------------------------
libavcodec/audio_frame_queue.h | 6 +-
2 files changed, 52 insertions(+), 104 deletions(-)
diff --git a/libavcodec/audio_frame_queue.c b/libavcodec/audio_frame_queue.c
index 156c3a1..31cab29 100644
--- a/libavcodec/audio_frame_queue.c
+++ b/libavcodec/audio_frame_queue.c
@@ -2,92 +2,70 @@
* Audio Frame Queue
* Copyright (c) 2012 Justin Ruggles
*
- * This file is part of Libav.
+ * This file is part of FFmpeg.
*
- * Libav is free software; you can redistribute it and/or
+ * FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
- * Libav is distributed in the hope that it will be useful,
+ * FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
- * License along with Libav; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
-#include "libavutil/mathematics.h"
-#include "internal.h"
#include "audio_frame_queue.h"
+#include "internal.h"
+#include "libavutil/avassert.h"
void ff_af_queue_init(AVCodecContext *avctx, AudioFrameQueue *afq)
{
- afq->avctx = avctx;
- afq->next_pts = AV_NOPTS_VALUE;
+ afq->avctx = avctx;
afq->remaining_delay = avctx->delay;
afq->remaining_samples = avctx->delay;
- afq->frame_queue = NULL;
-}
-
-static void delete_next_frame(AudioFrameQueue *afq)
-{
- AudioFrame *f = afq->frame_queue;
- if (f) {
- afq->frame_queue = f->next;
- f->next = NULL;
- av_freep(&f);
- }
+ afq->frame_count = 0;
}
void ff_af_queue_close(AudioFrameQueue *afq)
{
- /* remove/free any remaining frames */
- while (afq->frame_queue)
- delete_next_frame(afq);
+ if(afq->frame_count)
+ av_log(afq->avctx, AV_LOG_WARNING, "%d frames left in que on closing\n", afq->frame_count);
+ av_freep(&afq->frames);
memset(afq, 0, sizeof(*afq));
}
int ff_af_queue_add(AudioFrameQueue *afq, const AVFrame *f)
{
- AudioFrame *new_frame;
- AudioFrame *queue_end = afq->frame_queue;
-
- /* find the end of the queue */
- while (queue_end && queue_end->next)
- queue_end = queue_end->next;
-
- /* allocate new frame queue entry */
- if (!(new_frame = av_malloc(sizeof(*new_frame))))
+ AudioFrame *new = av_fast_realloc(afq->frames, &afq->frame_alloc, sizeof(*afq->frames)*(afq->frame_count+1));
+ if(!new)
return AVERROR(ENOMEM);
+ afq->frames = new;
+ new += afq->frame_count;
/* get frame parameters */
- new_frame->next = NULL;
- new_frame->duration = f->nb_samples;
+ new->duration = f->nb_samples;
+ new->duration += afq->remaining_delay;
if (f->pts != AV_NOPTS_VALUE) {
- new_frame->pts = av_rescale_q(f->pts,
+ new->pts = av_rescale_q(f->pts,
afq->avctx->time_base,
(AVRational){ 1, afq->avctx->sample_rate });
- afq->next_pts = new_frame->pts + new_frame->duration;
+ new->pts -= afq->remaining_delay;
+ if(afq->frame_count && new[-1].pts >= new->pts)
+ av_log(afq->avctx, AV_LOG_WARNING, "Que input is backward in time\n");
} else {
- new_frame->pts = AV_NOPTS_VALUE;
- afq->next_pts = AV_NOPTS_VALUE;
+ new->pts = AV_NOPTS_VALUE;
}
-
- /* add new frame to the end of the queue */
- if (!queue_end)
- afq->frame_queue = new_frame;
- else
- queue_end->next = new_frame;
+ afq->remaining_delay = 0;
/* add frame sample count */
afq->remaining_samples += f->nb_samples;
-#ifdef DEBUG
- ff_af_queue_log_state(afq);
-#endif
+ afq->frame_count++;
return 0;
}
@@ -97,66 +75,36 @@ void ff_af_queue_remove(AudioFrameQueue *afq, int nb_samples, int64_t *pts,
{
int64_t out_pts = AV_NOPTS_VALUE;
int removed_samples = 0;
+ int i;
-#ifdef DEBUG
- ff_af_queue_log_state(afq);
-#endif
-
- /* get output pts from the next frame or generated pts */
- if (afq->frame_queue) {
- if (afq->frame_queue->pts != AV_NOPTS_VALUE)
- out_pts = afq->frame_queue->pts - afq->remaining_delay;
- } else {
- if (afq->next_pts != AV_NOPTS_VALUE)
- out_pts = afq->next_pts - afq->remaining_delay;
+ if (afq->frame_count || afq->frame_alloc) {
+ if (afq->frames->pts != AV_NOPTS_VALUE)
+ out_pts = afq->frames->pts;
}
- if (pts) {
- if (out_pts != AV_NOPTS_VALUE)
- *pts = ff_samples_to_time_base(afq->avctx, out_pts);
- else
- *pts = AV_NOPTS_VALUE;
+ if(!afq->frame_count)
+ av_log(afq->avctx, AV_LOG_WARNING, "Trying to remove %d samples, but que empty\n", nb_samples);
+ if (pts)
+ *pts = ff_samples_to_time_base(afq->avctx, out_pts);
+
+ for(i=0; nb_samples && i<afq->frame_count; i++){
+ int n= FFMIN(afq->frames[i].duration, nb_samples);
+ afq->frames[i].duration -= n;
+ nb_samples -= n;
+ removed_samples += n;
+ if(afq->frames[i].pts != AV_NOPTS_VALUE)
+ afq->frames[i].pts += n;
}
-
- /* if the delay is larger than the packet duration, we use up delay samples
- for the output packet and leave all frames in the queue */
- if (afq->remaining_delay >= nb_samples) {
- removed_samples += nb_samples;
- afq->remaining_delay -= nb_samples;
+ i -= i && afq->frames[i-1].duration;
+ memmove(afq->frames, afq->frames + i, sizeof(*afq->frames) * (afq->frame_count - i));
+ afq->frame_count -= i;
+
+ if(nb_samples){
+ av_assert0(!afq->frame_count);
+ if(afq->frames[0].pts != AV_NOPTS_VALUE)
+ afq->frames[0].pts += nb_samples;
+ av_log(afq->avctx, AV_LOG_DEBUG, "Trying to remove %d more samples than are in the que\n", nb_samples);
}
- /* remove frames from the queue until we have enough to cover the
- requested number of samples or until the queue is empty */
- while (removed_samples < nb_samples && afq->frame_queue) {
- removed_samples += afq->frame_queue->duration;
- delete_next_frame(afq);
- }
- afq->remaining_samples -= removed_samples;
-
- /* if there are no frames left and we have room for more samples, use
- any remaining delay samples */
- if (removed_samples < nb_samples && afq->remaining_samples > 0) {
- int add_samples = FFMIN(afq->remaining_samples,
- nb_samples - removed_samples);
- removed_samples += add_samples;
- afq->remaining_samples -= add_samples;
- }
- if (removed_samples > nb_samples)
- av_log(afq->avctx, AV_LOG_WARNING, "frame_size is too large\n");
if (duration)
*duration = ff_samples_to_time_base(afq->avctx, removed_samples);
}
-void ff_af_queue_log_state(AudioFrameQueue *afq)
-{
- AudioFrame *f;
- av_log(afq->avctx, AV_LOG_DEBUG, "remaining delay = %d\n",
- afq->remaining_delay);
- av_log(afq->avctx, AV_LOG_DEBUG, "remaining samples = %d\n",
- afq->remaining_samples);
- av_log(afq->avctx, AV_LOG_DEBUG, "frames:\n");
- f = afq->frame_queue;
- while (f) {
- av_log(afq->avctx, AV_LOG_DEBUG, " [ pts=%9"PRId64" duration=%d ]\n",
- f->pts, f->duration);
- f = f->next;
- }
-}
diff --git a/libavcodec/audio_frame_queue.h b/libavcodec/audio_frame_queue.h
index cfcc6a0..edb2fd2 100644
--- a/libavcodec/audio_frame_queue.h
+++ b/libavcodec/audio_frame_queue.h
@@ -27,15 +27,15 @@
typedef struct AudioFrame {
int64_t pts;
int duration;
- struct AudioFrame *next;
} AudioFrame;
typedef struct AudioFrameQueue {
AVCodecContext *avctx;
- int64_t next_pts;
int remaining_delay;
int remaining_samples;
- AudioFrame *frame_queue;
+ AudioFrame *frames;
+ unsigned frame_count;
+ unsigned frame_alloc;
} AudioFrameQueue;
/**
--
1.7.5.4
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