diff options
Diffstat (limited to 'ffmpeg/libavfilter/audio.c')
| -rw-r--r-- | ffmpeg/libavfilter/audio.c | 184 |
1 files changed, 184 insertions, 0 deletions
diff --git a/ffmpeg/libavfilter/audio.c b/ffmpeg/libavfilter/audio.c new file mode 100644 index 0000000..1075217 --- /dev/null +++ b/ffmpeg/libavfilter/audio.c @@ -0,0 +1,184 @@ +/* + * Copyright (c) Stefano Sabatini | stefasab at gmail.com + * Copyright (c) S.N. Hemanth Meenakshisundaram | smeenaks at ucsd.edu + * + * 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. + * + * 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 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/channel_layout.h" +#include "libavutil/common.h" +#include "libavcodec/avcodec.h" + +#include "audio.h" +#include "avfilter.h" +#include "internal.h" + +int avfilter_ref_get_channels(AVFilterBufferRef *ref) +{ + return ref->audio ? ref->audio->channels : 0; +} + +AVFrame *ff_null_get_audio_buffer(AVFilterLink *link, int nb_samples) +{ + return ff_get_audio_buffer(link->dst->outputs[0], nb_samples); +} + +AVFrame *ff_default_get_audio_buffer(AVFilterLink *link, int nb_samples) +{ + AVFrame *frame = av_frame_alloc(); + int channels = link->channels; + int buf_size, ret; + + av_assert0(channels == av_get_channel_layout_nb_channels(link->channel_layout) || !av_get_channel_layout_nb_channels(link->channel_layout)); + + if (!frame) + return NULL; + + buf_size = av_samples_get_buffer_size(NULL, channels, nb_samples, + link->format, 0); + if (buf_size < 0) + goto fail; + + frame->buf[0] = av_buffer_alloc(buf_size); + if (!frame->buf[0]) + goto fail; + + frame->nb_samples = nb_samples; + ret = avcodec_fill_audio_frame(frame, channels, link->format, + frame->buf[0]->data, buf_size, 0); + if (ret < 0) + goto fail; + + av_samples_set_silence(frame->extended_data, 0, nb_samples, channels, + link->format); + + frame->nb_samples = nb_samples; + frame->format = link->format; + av_frame_set_channels(frame, link->channels); + frame->channel_layout = link->channel_layout; + frame->sample_rate = link->sample_rate; + + return frame; + +fail: + av_buffer_unref(&frame->buf[0]); + av_frame_free(&frame); + return NULL; +} + +AVFrame *ff_get_audio_buffer(AVFilterLink *link, int nb_samples) +{ + AVFrame *ret = NULL; + + if (link->dstpad->get_audio_buffer) + ret = link->dstpad->get_audio_buffer(link, nb_samples); + + if (!ret) + ret = ff_default_get_audio_buffer(link, nb_samples); + + return ret; +} + +#if FF_API_AVFILTERBUFFER +AVFilterBufferRef* avfilter_get_audio_buffer_ref_from_arrays_channels(uint8_t **data, + int linesize,int perms, + int nb_samples, + enum AVSampleFormat sample_fmt, + int channels, + uint64_t channel_layout) +{ + int planes; + AVFilterBuffer *samples = av_mallocz(sizeof(*samples)); + AVFilterBufferRef *samplesref = av_mallocz(sizeof(*samplesref)); + + if (!samples || !samplesref) + goto fail; + + av_assert0(channels); + av_assert0(channel_layout == 0 || + channels == av_get_channel_layout_nb_channels(channel_layout)); + + samplesref->buf = samples; + samplesref->buf->free = ff_avfilter_default_free_buffer; + if (!(samplesref->audio = av_mallocz(sizeof(*samplesref->audio)))) + goto fail; + + samplesref->audio->nb_samples = nb_samples; + samplesref->audio->channel_layout = channel_layout; + samplesref->audio->channels = channels; + + planes = av_sample_fmt_is_planar(sample_fmt) ? channels : 1; + + /* 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, + FFMIN(FF_ARRAY_ELEMS(samples->data), planes)*sizeof(samples->data[0])); + memcpy(samplesref->data, samples->data, sizeof(samples->data)); + + samples->linesize[0] = samplesref->linesize[0] = linesize; + + if (planes > FF_ARRAY_ELEMS(samples->data)) { + samples-> extended_data = av_mallocz(sizeof(*samples->extended_data) * + planes); + samplesref->extended_data = av_mallocz(sizeof(*samplesref->extended_data) * + planes); + + if (!samples->extended_data || !samplesref->extended_data) + goto fail; + + memcpy(samples-> extended_data, data, sizeof(*data)*planes); + memcpy(samplesref->extended_data, data, sizeof(*data)*planes); + } else { + samples->extended_data = samples->data; + samplesref->extended_data = samplesref->data; + } + + samplesref->pts = AV_NOPTS_VALUE; + + return samplesref; + +fail: + if (samples && samples->extended_data != samples->data) + av_freep(&samples->extended_data); + if (samplesref) { + av_freep(&samplesref->audio); + if (samplesref->extended_data != samplesref->data) + av_freep(&samplesref->extended_data); + } + av_freep(&samplesref); + av_freep(&samples); + return NULL; +} + +AVFilterBufferRef* avfilter_get_audio_buffer_ref_from_arrays(uint8_t **data, + int linesize,int perms, + int nb_samples, + enum AVSampleFormat sample_fmt, + uint64_t channel_layout) +{ + int channels = av_get_channel_layout_nb_channels(channel_layout); + return avfilter_get_audio_buffer_ref_from_arrays_channels(data, linesize, perms, + nb_samples, sample_fmt, + channels, channel_layout); +} +#endif |