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.
Unified · Split
| @@ -2,92 +2,70 @@ · 1 unmodified line | |||
| 2 | * Audio Frame Queue | 2 | * Audio Frame Queue |
| 3 | * Copyright (c) 2012 Justin Ruggles | 3 | * Copyright (c) 2012 Justin Ruggles |
| 4 | * | 4 | * |
| 5 | * This file is part of Libav. | 5 | * This file is part of FFmpeg. |
| 6 | * | 6 | * |
| 7 | * Libav is free software; you can redistribute it and/or | 7 | * FFmpeg is free software; you can redistribute it and/or |
| 8 | * modify it under the terms of the GNU Lesser General Public | 8 | * modify it under the terms of the GNU Lesser General Public |
| 9 | * License as published by the Free Software Foundation; either | 9 | * License as published by the Free Software Foundation; either |
| 10 | * version 2.1 of the License, or (at your option) any later version. | 10 | * version 2.1 of the License, or (at your option) any later version. |
| 11 | * | 11 | * |
| 12 | * Libav is distributed in the hope that it will be useful, | 12 | * FFmpeg is distributed in the hope that it will be useful, |
| 13 | * but WITHOUT ANY WARRANTY; without even the implied warranty of | 13 | * but WITHOUT ANY WARRANTY; without even the implied warranty of |
| 14 | * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | 14 | * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
| 15 | * Lesser General Public License for more details. | 15 | * Lesser General Public License for more details. |
| 16 | * | 16 | * |
| 17 | * You should have received a copy of the GNU Lesser General Public | 17 | * You should have received a copy of the GNU Lesser General Public |
| 18 | * License along with Libav; if not, write to the Free Software | 18 | * License along with FFmpeg; if not, write to the Free Software |
| 19 | * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA | 19 | * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
| 20 | */ | 20 | */ |
| 21 | 21 | ||
| 22 | #include "libavutil/mathematics.h" | ||
| 23 | #include "internal.h" | ||
| 24 | #include "audio_frame_queue.h" | 22 | #include "audio_frame_queue.h" |
| 23 | #include "internal.h" | ||
| 24 | #include "libavutil/avassert.h" | ||
| 25 | 25 | ||
| 26 | void ff_af_queue_init(AVCodecContext *avctx, AudioFrameQueue *afq) | 26 | void ff_af_queue_init(AVCodecContext *avctx, AudioFrameQueue *afq) |
| 27 | { | 27 | { |
| 28 | afq->avctx = avctx; | 28 | afq->avctx = avctx; |
| 29 | afq->next_pts = AV_NOPTS_VALUE; | ||
| 30 | afq->remaining_delay = avctx->delay; | 29 | afq->remaining_delay = avctx->delay; |
| 31 | afq->remaining_samples = avctx->delay; | 30 | afq->remaining_samples = avctx->delay; |
| 32 | afq->frame_queue = NULL; | 31 | afq->frame_count = 0; |
| 33 | } | ||
| 34 | |||
| 35 | static void delete_next_frame(AudioFrameQueue *afq) | ||
| 36 | { | ||
| 37 | AudioFrame *f = afq->frame_queue; | ||
| 38 | if (f) { | ||
| 39 | afq->frame_queue = f->next; | ||
| 40 | f->next = NULL; | ||
| 41 | av_freep(&f); | ||
| 42 | } | ||
| 43 | } | 32 | } |
| 44 | 33 | ||
| 45 | void ff_af_queue_close(AudioFrameQueue *afq) | 34 | void ff_af_queue_close(AudioFrameQueue *afq) |
| 46 | { | 35 | { |
| 47 | /* remove/free any remaining frames */ | 36 | if(afq->frame_count) |
| 48 | while (afq->frame_queue) | 37 | av_log(afq->avctx, AV_LOG_WARNING, "%d frames left in que on closing\n", afq->frame_count); |
| 49 | delete_next_frame(afq); | 38 | av_freep(&afq->frames); |
| 50 | memset(afq, 0, sizeof(*afq)); | 39 | memset(afq, 0, sizeof(*afq)); |
| 51 | } | 40 | } |
| 52 | 41 | ||
| 53 | int ff_af_queue_add(AudioFrameQueue *afq, const AVFrame *f) | 42 | int ff_af_queue_add(AudioFrameQueue *afq, const AVFrame *f) |
| 54 | { | 43 | { |
| 55 | AudioFrame *new_frame; | 44 | AudioFrame *new = av_fast_realloc(afq->frames, &afq->frame_alloc, sizeof(*afq->frames)*(afq->frame_count+1)); |
| 56 | AudioFrame *queue_end = afq->frame_queue; | 45 | if(!new) |
| 57 | |||
| 58 | /* find the end of the queue */ | ||
| 59 | while (queue_end && queue_end->next) | ||
| 60 | queue_end = queue_end->next; | ||
| 61 | |||
| 62 | /* allocate new frame queue entry */ | ||
| 63 | if (!(new_frame = av_malloc(sizeof(*new_frame)))) | ||
| 64 | return AVERROR(ENOMEM); | 46 | return AVERROR(ENOMEM); |
| 47 | afq->frames = new; | ||
| 48 | new += afq->frame_count; | ||
| 65 | 49 | ||
| 66 | /* get frame parameters */ | 50 | /* get frame parameters */ |
| 67 | new_frame->next = NULL; | 51 | new->duration = f->nb_samples; |
| 68 | new_frame->duration = f->nb_samples; | 52 | new->duration += afq->remaining_delay; |
| 69 | if (f->pts != AV_NOPTS_VALUE) { | 53 | if (f->pts != AV_NOPTS_VALUE) { |
| 70 | new_frame->pts = av_rescale_q(f->pts, | 54 | new->pts = av_rescale_q(f->pts, |
| 71 | afq->avctx->time_base, | 55 | afq->avctx->time_base, |
| 72 | (AVRational){ 1, afq->avctx->sample_rate }); | 56 | (AVRational){ 1, afq->avctx->sample_rate }); |
| 73 | afq->next_pts = new_frame->pts + new_frame->duration; | 57 | new->pts -= afq->remaining_delay; |
| 58 | if(afq->frame_count && new[-1].pts >= new->pts) | ||
| 59 | av_log(afq->avctx, AV_LOG_WARNING, "Que input is backward in time\n"); | ||
| 74 | } else { | 60 | } else { |
| 75 | new_frame->pts = AV_NOPTS_VALUE; | 61 | new->pts = AV_NOPTS_VALUE; |
| 76 | afq->next_pts = AV_NOPTS_VALUE; | ||
| 77 | } | 62 | } |
| 78 | 63 | afq->remaining_delay = 0; | |
| 79 | /* add new frame to the end of the queue */ | ||
| 80 | if (!queue_end) | ||
| 81 | afq->frame_queue = new_frame; | ||
| 82 | else | ||
| 83 | queue_end->next = new_frame; | ||
| 84 | 64 | ||
| 85 | /* add frame sample count */ | 65 | /* add frame sample count */ |
| 86 | afq->remaining_samples += f->nb_samples; | 66 | afq->remaining_samples += f->nb_samples; |
| 87 | 67 | ||
| 88 | #ifdef DEBUG | 68 | afq->frame_count++; |
| 89 | ff_af_queue_log_state(afq); | ||
| 90 | #endif | ||
| 91 | 69 | ||
| 92 | return 0; | 70 | return 0; |
| 93 | } | 71 | } |
| @@ -97,66 +75,36 @@ void ff_af_queue_remove(AudioFrameQueue *afq, int nb_samples, int64_t *pts, · 3 unmodified lines | |||
| 97 | { | 75 | { |
| 98 | int64_t out_pts = AV_NOPTS_VALUE; | 76 | int64_t out_pts = AV_NOPTS_VALUE; |
| 99 | int removed_samples = 0; | 77 | int removed_samples = 0; |
| 78 | int i; | ||
| 100 | 79 | ||
| 101 | #ifdef DEBUG | 80 | if (afq->frame_count || afq->frame_alloc) { |
| 102 | ff_af_queue_log_state(afq); | 81 | if (afq->frames->pts != AV_NOPTS_VALUE) |
| 103 | #endif | 82 | out_pts = afq->frames->pts; |
| 104 | |||
| 105 | /* get output pts from the next frame or generated pts */ | ||
| 106 | if (afq->frame_queue) { | ||
| 107 | if (afq->frame_queue->pts != AV_NOPTS_VALUE) | ||
| 108 | out_pts = afq->frame_queue->pts - afq->remaining_delay; | ||
| 109 | } else { | ||
| 110 | if (afq->next_pts != AV_NOPTS_VALUE) | ||
| 111 | out_pts = afq->next_pts - afq->remaining_delay; | ||
| 112 | } | 83 | } |
| 113 | if (pts) { | 84 | if(!afq->frame_count) |
| 114 | if (out_pts != AV_NOPTS_VALUE) | 85 | av_log(afq->avctx, AV_LOG_WARNING, "Trying to remove %d samples, but que empty\n", nb_samples); |
| 115 | *pts = ff_samples_to_time_base(afq->avctx, out_pts); | 86 | if (pts) |
| 116 | else | 87 | *pts = ff_samples_to_time_base(afq->avctx, out_pts); |
| 117 | *pts = AV_NOPTS_VALUE; | 88 | |
| 89 | for(i=0; nb_samples && i<afq->frame_count; i++){ | ||
| 90 | int n= FFMIN(afq->frames[i].duration, nb_samples); | ||
| 91 | afq->frames[i].duration -= n; | ||
| 92 | nb_samples -= n; | ||
| 93 | removed_samples += n; | ||
| 94 | if(afq->frames[i].pts != AV_NOPTS_VALUE) | ||
| 95 | afq->frames[i].pts += n; | ||
| 118 | } | 96 | } |
| 119 | 97 | i -= i && afq->frames[i-1].duration; | |
| 120 | /* if the delay is larger than the packet duration, we use up delay samples | 98 | memmove(afq->frames, afq->frames + i, sizeof(*afq->frames) * (afq->frame_count - i)); |
| 121 | for the output packet and leave all frames in the queue */ | 99 | afq->frame_count -= i; |
| 122 | if (afq->remaining_delay >= nb_samples) { | 100 | |
| 123 | removed_samples += nb_samples; | 101 | if(nb_samples){ |
| 124 | afq->remaining_delay -= nb_samples; | 102 | av_assert0(!afq->frame_count); |
| 103 | if(afq->frames[0].pts != AV_NOPTS_VALUE) | ||
| 104 | afq->frames[0].pts += nb_samples; | ||
| 105 | av_log(afq->avctx, AV_LOG_DEBUG, "Trying to remove %d more samples than are in the que\n", nb_samples); | ||
| 125 | } | 106 | } |
| 126 | /* remove frames from the queue until we have enough to cover the | ||
| 127 | requested number of samples or until the queue is empty */ | ||
| 128 | while (removed_samples < nb_samples && afq->frame_queue) { | ||
| 129 | removed_samples += afq->frame_queue->duration; | ||
| 130 | delete_next_frame(afq); | ||
| 131 | } | ||
| 132 | afq->remaining_samples -= removed_samples; | ||
| 133 | |||
| 134 | /* if there are no frames left and we have room for more samples, use | ||
| 135 | any remaining delay samples */ | ||
| 136 | if (removed_samples < nb_samples && afq->remaining_samples > 0) { | ||
| 137 | int add_samples = FFMIN(afq->remaining_samples, | ||
| 138 | nb_samples - removed_samples); | ||
| 139 | removed_samples += add_samples; | ||
| 140 | afq->remaining_samples -= add_samples; | ||
| 141 | } | ||
| 142 | if (removed_samples > nb_samples) | ||
| 143 | av_log(afq->avctx, AV_LOG_WARNING, "frame_size is too large\n"); | ||
| 144 | if (duration) | 107 | if (duration) |
| 145 | *duration = ff_samples_to_time_base(afq->avctx, removed_samples); | 108 | *duration = ff_samples_to_time_base(afq->avctx, removed_samples); |
| 146 | } | 109 | } |
| 147 | 110 | ||
| 148 | void ff_af_queue_log_state(AudioFrameQueue *afq) | ||
| 149 | { | ||
| 150 | AudioFrame *f; | ||
| 151 | av_log(afq->avctx, AV_LOG_DEBUG, "remaining delay = %d\n", | ||
| 152 | afq->remaining_delay); | ||
| 153 | av_log(afq->avctx, AV_LOG_DEBUG, "remaining samples = %d\n", | ||
| 154 | afq->remaining_samples); | ||
| 155 | av_log(afq->avctx, AV_LOG_DEBUG, "frames:\n"); | ||
| 156 | f = afq->frame_queue; | ||
| 157 | while (f) { | ||
| 158 | av_log(afq->avctx, AV_LOG_DEBUG, " [ pts=%9"PRId64" duration=%d ]\n", | ||
| 159 | f->pts, f->duration); | ||
| 160 | f = f->next; | ||
| 161 | } | ||
| 162 | } | ||
| @@ -27,15 +27,15 @@ · 26 unmodified lines | |||
| 27 | typedef struct AudioFrame { | 27 | typedef struct AudioFrame { |
| 28 | int64_t pts; | 28 | int64_t pts; |
| 29 | int duration; | 29 | int duration; |
| 30 | struct AudioFrame *next; | ||
| 31 | } AudioFrame; | 30 | } AudioFrame; |
| 32 | 31 | ||
| 33 | typedef struct AudioFrameQueue { | 32 | typedef struct AudioFrameQueue { |
| 34 | AVCodecContext *avctx; | 33 | AVCodecContext *avctx; |
| 35 | int64_t next_pts; | ||
| 36 | int remaining_delay; | 34 | int remaining_delay; |
| 37 | int remaining_samples; | 35 | int remaining_samples; |
| 38 | AudioFrame *frame_queue; | 36 | AudioFrame *frames; |
| 37 | unsigned frame_count; | ||
| 38 | unsigned frame_alloc; | ||
| 39 | } AudioFrameQueue; | 39 | } AudioFrameQueue; |
| 40 | 40 | ||
| 41 | /** | 41 | /** |