diff options
Diffstat (limited to 'libavfilter/audio.c')
| -rw-r--r-- | libavfilter/audio.c | 96 |
1 files changed, 89 insertions, 7 deletions
diff --git a/libavfilter/audio.c b/libavfilter/audio.c index 3e12c697ce..31f6796437 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 "audio.h" @@ -29,6 +33,38 @@ AVFilterBufferRef *ff_null_get_audio_buffer(AVFilterLink *link, int perms, } AVFilterBufferRef *ff_default_get_audio_buffer(AVFilterLink *link, int perms, + int nb_samples) +{ + AVFilterBufferRef *samplesref = NULL; + int linesize[8] = {0}; + uint8_t *data[8] = {0}; + int ch, nb_channels = av_get_channel_layout_nb_channels(link->channel_layout); + + /* right now we don't support more than 8 channels */ + av_assert0(nb_channels <= 8); + + /* Calculate total buffer size, round to multiple of 16 to be SIMD friendly */ + if (av_samples_alloc(data, linesize, + nb_channels, nb_samples, + av_get_alt_sample_fmt(link->format, link->planar), + 16) < 0) + return NULL; + + for (ch = 1; link->planar && ch < nb_channels; ch++) + linesize[ch] = linesize[0]; + samplesref = + avfilter_get_audio_buffer_ref_from_arrays(data, linesize, perms, + nb_samples, link->format, + link->channel_layout, link->planar); + if (!samplesref) { + av_free(data[0]); + return NULL; + } + + return samplesref; +} + +static AVFilterBufferRef *ff_default_get_audio_buffer_alt(AVFilterLink *link, int perms, int nb_samples) { AVFilterBufferRef *samplesref = NULL; @@ -44,7 +80,7 @@ AVFilterBufferRef *ff_default_get_audio_buffer(AVFilterLink *link, int perms, if (av_samples_alloc(data, &linesize, nb_channels, nb_samples, link->format, 0) < 0) goto fail; - samplesref = avfilter_get_audio_buffer_ref_from_arrays(data, linesize, perms, + samplesref = avfilter_get_audio_buffer_ref_from_arrays_alt(data, linesize, perms, nb_samples, link->format, link->channel_layout); if (!samplesref) @@ -76,7 +112,49 @@ AVFilterBufferRef *ff_get_audio_buffer(AVFilterLink *link, int perms, return ret; } -AVFilterBufferRef* avfilter_get_audio_buffer_ref_from_arrays(uint8_t **data, +AVFilterBufferRef * +avfilter_get_audio_buffer_ref_from_arrays(uint8_t *data[8], int linesize[8], int perms, + int nb_samples, enum AVSampleFormat sample_fmt, + uint64_t channel_layout, int planar) +{ + AVFilterBuffer *samples = av_mallocz(sizeof(AVFilterBuffer)); + AVFilterBufferRef *samplesref = av_mallocz(sizeof(AVFilterBufferRef)); + + if (!samples || !samplesref) + goto fail; + + samplesref->buf = samples; + samplesref->buf->free = ff_avfilter_default_free_buffer; + if (!(samplesref->audio = av_mallocz(sizeof(AVFilterBufferRefAudioProps)))) + goto fail; + + samplesref->audio->nb_samples = nb_samples; + samplesref->audio->channel_layout = channel_layout; + samplesref->audio->planar = planar; + + /* make sure the buffer gets read permission or it's useless for output */ + samplesref->perms = perms | AV_PERM_READ; + + samples->refcount = 1; + samplesref->type = AVMEDIA_TYPE_AUDIO; + samplesref->format = sample_fmt; + + memcpy(samples->data, data, sizeof(samples->data)); + memcpy(samples->linesize, linesize, sizeof(samples->linesize)); + memcpy(samplesref->data, data, sizeof(samplesref->data)); + memcpy(samplesref->linesize, linesize, sizeof(samplesref->linesize)); + + return samplesref; + +fail: + if (samplesref && samplesref->audio) + av_freep(&samplesref->audio); + av_freep(&samplesref); + av_freep(&samples); + return NULL; +} + +AVFilterBufferRef* avfilter_get_audio_buffer_ref_from_arrays_alt(uint8_t **data, int linesize,int perms, int nb_samples, enum AVSampleFormat sample_fmt, @@ -174,6 +252,7 @@ void ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref) { void (*filter_samples)(AVFilterLink *, AVFilterBufferRef *); AVFilterPad *dst = link->dstpad; + int64_t pts; FF_DPRINTF_START(NULL, filter_samples); ff_dlog_link(NULL, link, 1); @@ -197,6 +276,8 @@ void ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref) link->cur_buf->audio->sample_rate = samplesref->audio->sample_rate; /* Copy actual data into new samples buffer */ + for (i = 0; samplesref->data[i] && i < 8; i++) + memcpy(link->cur_buf->data[i], samplesref->data[i], samplesref->linesize[0]); for (i = 0; i < planes; i++) memcpy(link->cur_buf->extended_data[i], samplesref->extended_data[i], samplesref->linesize[0]); @@ -204,6 +285,7 @@ void ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref) } else link->cur_buf = samplesref; + pts = link->cur_buf->pts; filter_samples(link, link->cur_buf); + ff_update_link_current_pts(link, pts); } - |