diff options
Diffstat (limited to 'libavfilter/audio.c')
-rw-r--r-- | libavfilter/audio.c | 77 |
1 files changed, 68 insertions, 9 deletions
diff --git a/libavfilter/audio.c b/libavfilter/audio.c index 66010c18c5..aa87488f5f 100644 --- a/libavfilter/audio.c +++ b/libavfilter/audio.c @@ -1,21 +1,25 @@ /* - * This file is part of Libav. + * Copyright (c) Stefano Sabatini | stefasab at gmail.com + * Copyright (c) S.N. Hemanth Meenakshisundaram | smeenaks at ucsd.edu * - * Libav is free software; you can redistribute it and/or + * This file is part of FFmpeg. + * + * 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/avassert.h" #include "libavutil/audioconvert.h" #include "libavutil/common.h" @@ -97,9 +101,9 @@ AVFilterBufferRef* avfilter_get_audio_buffer_ref_from_arrays(uint8_t **data, samplesref->audio->nb_samples = nb_samples; samplesref->audio->channel_layout = channel_layout; - samplesref->audio->planar = av_sample_fmt_is_planar(sample_fmt); - planes = samplesref->audio->planar ? av_get_channel_layout_nb_channels(channel_layout) : 1; + planes = av_sample_fmt_is_planar(sample_fmt) ? + av_get_channel_layout_nb_channels(channel_layout) : 1; /* make sure the buffer gets read permission or it's useless for output */ samplesref->perms = perms | AV_PERM_READ; @@ -153,17 +157,23 @@ static int default_filter_samples(AVFilterLink *link, return ff_filter_samples(link->dst->outputs[0], samplesref); } -int ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref) +int ff_filter_samples_framed(AVFilterLink *link, AVFilterBufferRef *samplesref) { int (*filter_samples)(AVFilterLink *, AVFilterBufferRef *); + AVFilterPad *src = link->srcpad; AVFilterPad *dst = link->dstpad; + int64_t pts; AVFilterBufferRef *buf_out; + int ret; - FF_DPRINTF_START(NULL, filter_samples); ff_dlog_link(NULL, link, 1); + FF_TPRINTF_START(NULL, filter_samples); ff_tlog_link(NULL, link, 1); if (!(filter_samples = dst->filter_samples)) filter_samples = default_filter_samples; + av_assert1((samplesref->perms & src->min_perms) == src->min_perms); + samplesref->perms &= ~ src->rej_perms; + /* prepare to copy the samples if the buffer has insufficient permissions */ if ((dst->min_perms & samplesref->perms) != dst->min_perms || dst->rej_perms & samplesref->perms) { @@ -190,6 +200,55 @@ int ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref) } else buf_out = samplesref; - return filter_samples(link, buf_out); + link->cur_buf = buf_out; + pts = buf_out->pts; + ret = filter_samples(link, buf_out); + ff_update_link_current_pts(link, pts); + return ret; } +int ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref) +{ + int insamples = samplesref->audio->nb_samples, inpos = 0, nb_samples; + AVFilterBufferRef *pbuf = link->partial_buf; + int nb_channels = av_get_channel_layout_nb_channels(link->channel_layout); + int ret = 0; + + if (!link->min_samples || + (!pbuf && + insamples >= link->min_samples && insamples <= link->max_samples)) { + return ff_filter_samples_framed(link, samplesref); + } + /* Handle framing (min_samples, max_samples) */ + while (insamples) { + if (!pbuf) { + AVRational samples_tb = { 1, link->sample_rate }; + int perms = link->dstpad->min_perms | AV_PERM_WRITE; + pbuf = ff_get_audio_buffer(link, perms, link->partial_buf_size); + if (!pbuf) { + av_log(link->dst, AV_LOG_WARNING, + "Samples dropped due to memory allocation failure.\n"); + return 0; + } + avfilter_copy_buffer_ref_props(pbuf, samplesref); + pbuf->pts = samplesref->pts + + av_rescale_q(inpos, samples_tb, link->time_base); + pbuf->audio->nb_samples = 0; + } + nb_samples = FFMIN(insamples, + link->partial_buf_size - pbuf->audio->nb_samples); + av_samples_copy(pbuf->extended_data, samplesref->extended_data, + pbuf->audio->nb_samples, inpos, + nb_samples, nb_channels, link->format); + inpos += nb_samples; + insamples -= nb_samples; + pbuf->audio->nb_samples += nb_samples; + if (pbuf->audio->nb_samples >= link->min_samples) { + ret = ff_filter_samples_framed(link, pbuf); + pbuf = NULL; + } + } + avfilter_unref_buffer(samplesref); + link->partial_buf = pbuf; + return ret; +} |