mirror of https://github.com/FFmpeg/FFmpeg.git
* qatar/master: (25 commits) rv40dsp x86: MMX/MMX2/3DNow/SSE2/SSSE3 implementations of MC ape: Use unsigned integer maths arm: dsputil: fix overreads in put/avg_pixels functions h264: K&R formatting cosmetics for header files (part II/II) h264: K&R formatting cosmetics for header files (part I/II) rtmp: Implement check bandwidth notification. rtmp: Support 'rtmp_swfurl', an option which specifies the URL of the SWF player. rtmp: Support 'rtmp_flashver', an option which overrides the version of the Flash plugin. rtmp: Support 'rtmp_tcurl', an option which overrides the URL of the target stream. cmdutils: Add fallback case to switch in check_stream_specifier(). sctp: be consistent with socket option level configure: Add _XOPEN_SOURCE=600 to Solaris preprocessor flags. vcr1enc: drop pointless empty encode_init() wrapper function vcr1: drop pointless write-only AVCodecContext member from VCR1Context vcr1: group encoder code together to save #ifdefs vcr1: cosmetics: K&R prettyprinting, typos, parentheses, dead code, comments mov: make one comment slightly more specific lavr: replace the SSE version of ff_conv_fltp_to_flt_6ch() with SSE4 and AVX lavfi: move audio-related functions to a separate file. lavfi: remove some audio-related function from public API. ... Conflicts: cmdutils.c libavcodec/h264.h libavcodec/h264_mvpred.h libavcodec/vcr1.c libavfilter/avfilter.c libavfilter/avfilter.h libavfilter/defaults.c libavfilter/internal.h Merged-by: Michael Niedermayer <michaelni@gmx.at>pull/30/merge
Michael Niedermayer
13 years ago
commit
015903294c
44 changed files with 2256 additions and 1429 deletions
File diff suppressed because it is too large
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@ -0,0 +1,291 @@ |
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/*
|
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* Copyright (c) Stefano Sabatini | stefasab at gmail.com |
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* Copyright (c) S.N. Hemanth Meenakshisundaram | smeenaks at ucsd.edu |
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* |
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* This file is part of FFmpeg. |
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* |
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* FFmpeg is free software; you can redistribute it and/or |
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* modify it under the terms of the GNU Lesser General Public |
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* License as published by the Free Software Foundation; either |
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* version 2.1 of the License, or (at your option) any later version. |
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* |
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* FFmpeg is distributed in the hope that it will be useful, |
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* but WITHOUT ANY WARRANTY; without even the implied warranty of |
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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* Lesser General Public License for more details. |
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* |
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* You should have received a copy of the GNU Lesser General Public |
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* License along with FFmpeg; if not, write to the Free Software |
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
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*/ |
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#include "libavutil/avassert.h" |
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#include "libavutil/audioconvert.h" |
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#include "audio.h" |
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#include "avfilter.h" |
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#include "internal.h" |
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AVFilterBufferRef *ff_null_get_audio_buffer(AVFilterLink *link, int perms, |
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int nb_samples) |
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{ |
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return ff_get_audio_buffer(link->dst->outputs[0], perms, nb_samples); |
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} |
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AVFilterBufferRef *ff_default_get_audio_buffer(AVFilterLink *link, int perms, |
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int nb_samples) |
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{ |
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AVFilterBufferRef *samplesref = NULL; |
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int linesize[8] = {0}; |
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uint8_t *data[8] = {0}; |
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int ch, nb_channels = av_get_channel_layout_nb_channels(link->channel_layout); |
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/* right now we don't support more than 8 channels */ |
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av_assert0(nb_channels <= 8); |
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/* Calculate total buffer size, round to multiple of 16 to be SIMD friendly */ |
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if (av_samples_alloc(data, linesize, |
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nb_channels, nb_samples, |
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av_get_alt_sample_fmt(link->format, link->planar), |
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16) < 0) |
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return NULL; |
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for (ch = 1; link->planar && ch < nb_channels; ch++) |
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linesize[ch] = linesize[0]; |
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samplesref = |
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avfilter_get_audio_buffer_ref_from_arrays(data, linesize, perms, |
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nb_samples, link->format, |
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link->channel_layout, link->planar); |
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if (!samplesref) { |
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av_free(data[0]); |
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return NULL; |
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} |
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return samplesref; |
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} |
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static AVFilterBufferRef *ff_default_get_audio_buffer_alt(AVFilterLink *link, int perms, |
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int nb_samples) |
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{ |
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AVFilterBufferRef *samplesref = NULL; |
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uint8_t **data; |
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int planar = av_sample_fmt_is_planar(link->format); |
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int nb_channels = av_get_channel_layout_nb_channels(link->channel_layout); |
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int planes = planar ? nb_channels : 1; |
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int linesize; |
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if (!(data = av_mallocz(sizeof(*data) * planes))) |
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goto fail; |
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if (av_samples_alloc(data, &linesize, nb_channels, nb_samples, link->format, 0) < 0) |
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goto fail; |
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samplesref = avfilter_get_audio_buffer_ref_from_arrays_alt(data, linesize, perms, |
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nb_samples, link->format, |
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link->channel_layout); |
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if (!samplesref) |
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goto fail; |
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av_freep(&data); |
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fail: |
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if (data) |
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av_freep(&data[0]); |
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av_freep(&data); |
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return samplesref; |
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} |
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AVFilterBufferRef *ff_get_audio_buffer(AVFilterLink *link, int perms, |
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int nb_samples) |
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{ |
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AVFilterBufferRef *ret = NULL; |
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if (link->dstpad->get_audio_buffer) |
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ret = link->dstpad->get_audio_buffer(link, perms, nb_samples); |
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if (!ret) |
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ret = ff_default_get_audio_buffer(link, perms, nb_samples); |
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if (ret) |
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ret->type = AVMEDIA_TYPE_AUDIO; |
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return ret; |
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} |
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AVFilterBufferRef * |
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avfilter_get_audio_buffer_ref_from_arrays(uint8_t *data[8], int linesize[8], int perms, |
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int nb_samples, enum AVSampleFormat sample_fmt, |
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uint64_t channel_layout, int planar) |
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{ |
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AVFilterBuffer *samples = av_mallocz(sizeof(AVFilterBuffer)); |
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AVFilterBufferRef *samplesref = av_mallocz(sizeof(AVFilterBufferRef)); |
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if (!samples || !samplesref) |
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goto fail; |
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samplesref->buf = samples; |
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samplesref->buf->free = ff_avfilter_default_free_buffer; |
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if (!(samplesref->audio = av_mallocz(sizeof(AVFilterBufferRefAudioProps)))) |
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goto fail; |
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samplesref->audio->nb_samples = nb_samples; |
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samplesref->audio->channel_layout = channel_layout; |
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samplesref->audio->planar = planar; |
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/* make sure the buffer gets read permission or it's useless for output */ |
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samplesref->perms = perms | AV_PERM_READ; |
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samples->refcount = 1; |
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samplesref->type = AVMEDIA_TYPE_AUDIO; |
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samplesref->format = sample_fmt; |
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memcpy(samples->data, data, sizeof(samples->data)); |
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memcpy(samples->linesize, linesize, sizeof(samples->linesize)); |
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memcpy(samplesref->data, data, sizeof(samplesref->data)); |
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memcpy(samplesref->linesize, linesize, sizeof(samplesref->linesize)); |
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return samplesref; |
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fail: |
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if (samplesref && samplesref->audio) |
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av_freep(&samplesref->audio); |
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av_freep(&samplesref); |
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av_freep(&samples); |
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return NULL; |
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} |
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AVFilterBufferRef* avfilter_get_audio_buffer_ref_from_arrays_alt(uint8_t **data, |
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int linesize,int perms, |
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int nb_samples, |
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enum AVSampleFormat sample_fmt, |
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uint64_t channel_layout) |
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{ |
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int planes; |
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AVFilterBuffer *samples = av_mallocz(sizeof(*samples)); |
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AVFilterBufferRef *samplesref = av_mallocz(sizeof(*samplesref)); |
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if (!samples || !samplesref) |
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goto fail; |
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samplesref->buf = samples; |
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samplesref->buf->free = ff_avfilter_default_free_buffer; |
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if (!(samplesref->audio = av_mallocz(sizeof(*samplesref->audio)))) |
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goto fail; |
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samplesref->audio->nb_samples = nb_samples; |
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samplesref->audio->channel_layout = channel_layout; |
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samplesref->audio->planar = av_sample_fmt_is_planar(sample_fmt); |
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planes = samplesref->audio->planar ? av_get_channel_layout_nb_channels(channel_layout) : 1; |
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/* make sure the buffer gets read permission or it's useless for output */ |
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samplesref->perms = perms | AV_PERM_READ; |
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samples->refcount = 1; |
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samplesref->type = AVMEDIA_TYPE_AUDIO; |
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samplesref->format = sample_fmt; |
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memcpy(samples->data, data, |
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FFMIN(FF_ARRAY_ELEMS(samples->data), planes)*sizeof(samples->data[0])); |
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memcpy(samplesref->data, samples->data, sizeof(samples->data)); |
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samples->linesize[0] = samplesref->linesize[0] = linesize; |
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if (planes > FF_ARRAY_ELEMS(samples->data)) { |
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samples-> extended_data = av_mallocz(sizeof(*samples->extended_data) * |
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planes); |
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samplesref->extended_data = av_mallocz(sizeof(*samplesref->extended_data) * |
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planes); |
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if (!samples->extended_data || !samplesref->extended_data) |
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goto fail; |
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memcpy(samples-> extended_data, data, sizeof(*data)*planes); |
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memcpy(samplesref->extended_data, data, sizeof(*data)*planes); |
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} else { |
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samples->extended_data = samples->data; |
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samplesref->extended_data = samplesref->data; |
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} |
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return samplesref; |
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fail: |
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if (samples && samples->extended_data != samples->data) |
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av_freep(&samples->extended_data); |
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if (samplesref) { |
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av_freep(&samplesref->audio); |
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if (samplesref->extended_data != samplesref->data) |
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av_freep(&samplesref->extended_data); |
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} |
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av_freep(&samplesref); |
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av_freep(&samples); |
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return NULL; |
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} |
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void ff_null_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref) |
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{ |
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ff_filter_samples(link->dst->outputs[0], samplesref); |
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} |
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/* FIXME: samplesref is same as link->cur_buf. Need to consider removing the redundant parameter. */ |
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void ff_default_filter_samples(AVFilterLink *inlink, AVFilterBufferRef *samplesref) |
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{ |
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AVFilterLink *outlink = NULL; |
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if (inlink->dst->output_count) |
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outlink = inlink->dst->outputs[0]; |
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if (outlink) { |
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outlink->out_buf = ff_default_get_audio_buffer(inlink, AV_PERM_WRITE, |
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samplesref->audio->nb_samples); |
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outlink->out_buf->pts = samplesref->pts; |
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outlink->out_buf->audio->sample_rate = samplesref->audio->sample_rate; |
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ff_filter_samples(outlink, avfilter_ref_buffer(outlink->out_buf, ~0)); |
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avfilter_unref_buffer(outlink->out_buf); |
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outlink->out_buf = NULL; |
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} |
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avfilter_unref_buffer(samplesref); |
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inlink->cur_buf = NULL; |
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} |
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void ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref) |
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{ |
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void (*filter_samples)(AVFilterLink *, AVFilterBufferRef *); |
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AVFilterPad *dst = link->dstpad; |
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int64_t pts; |
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FF_DPRINTF_START(NULL, filter_samples); ff_dlog_link(NULL, link, 1); |
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if (!(filter_samples = dst->filter_samples)) |
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filter_samples = ff_default_filter_samples; |
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/* prepare to copy the samples if the buffer has insufficient permissions */ |
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if ((dst->min_perms & samplesref->perms) != dst->min_perms || |
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dst->rej_perms & samplesref->perms) { |
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int i, planar = av_sample_fmt_is_planar(samplesref->format); |
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int planes = !planar ? 1: |
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av_get_channel_layout_nb_channels(samplesref->audio->channel_layout); |
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av_log(link->dst, AV_LOG_DEBUG, |
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"Copying audio data in avfilter (have perms %x, need %x, reject %x)\n", |
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samplesref->perms, link->dstpad->min_perms, link->dstpad->rej_perms); |
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link->cur_buf = ff_default_get_audio_buffer(link, dst->min_perms, |
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samplesref->audio->nb_samples); |
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link->cur_buf->pts = samplesref->pts; |
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link->cur_buf->audio->sample_rate = samplesref->audio->sample_rate; |
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/* Copy actual data into new samples buffer */ |
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for (i = 0; samplesref->data[i] && i < 8; i++) |
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memcpy(link->cur_buf->data[i], samplesref->data[i], samplesref->linesize[0]); |
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for (i = 0; i < planes; i++) |
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memcpy(link->cur_buf->extended_data[i], samplesref->extended_data[i], samplesref->linesize[0]); |
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avfilter_unref_buffer(samplesref); |
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} else |
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link->cur_buf = samplesref; |
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pts = link->cur_buf->pts; |
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filter_samples(link, link->cur_buf); |
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ff_update_link_current_pts(link, pts); |
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} |
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@ -0,0 +1,65 @@ |
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/*
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* Copyright (c) Stefano Sabatini | stefasab at gmail.com |
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* Copyright (c) S.N. Hemanth Meenakshisundaram | smeenaks at ucsd.edu |
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* |
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* This file is part of FFmpeg. |
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* |
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* FFmpeg is free software; you can redistribute it and/or |
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* modify it under the terms of the GNU Lesser General Public |
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* License as published by the Free Software Foundation; either |
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* version 2.1 of the License, or (at your option) any later version. |
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* |
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* FFmpeg is distributed in the hope that it will be useful, |
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* but WITHOUT ANY WARRANTY; without even the implied warranty of |
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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* Lesser General Public License for more details. |
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* |
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* You should have received a copy of the GNU Lesser General Public |
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* License along with FFmpeg; if not, write to the Free Software |
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
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*/ |
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#ifndef AVFILTER_AUDIO_H |
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#define AVFILTER_AUDIO_H |
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#include "avfilter.h" |
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/** default handler for get_audio_buffer() for audio inputs */ |
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AVFilterBufferRef *ff_default_get_audio_buffer(AVFilterLink *link, int perms, |
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int nb_samples); |
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|
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/** get_audio_buffer() handler for filters which simply pass audio along */ |
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AVFilterBufferRef *ff_null_get_audio_buffer(AVFilterLink *link, int perms, |
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int nb_samples); |
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|
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/**
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* Request an audio samples buffer with a specific set of permissions. |
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* |
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* @param link the output link to the filter from which the buffer will |
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* be requested |
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* @param perms the required access permissions |
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* @param nb_samples the number of samples per channel |
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* @return A reference to the samples. This must be unreferenced with |
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* avfilter_unref_buffer when you are finished with it. |
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*/ |
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AVFilterBufferRef *ff_get_audio_buffer(AVFilterLink *link, int perms, |
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int nb_samples); |
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|
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/** default handler for filter_samples() for audio inputs */ |
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void ff_default_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref); |
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|
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/** filter_samples() handler for filters which simply pass audio along */ |
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void ff_null_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref); |
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/**
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* Send a buffer of audio samples to the next filter. |
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* |
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* @param link the output link over which the audio samples are being sent |
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* @param samplesref a reference to the buffer of audio samples being sent. The |
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* receiving filter will free this reference when it no longer |
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* needs it or pass it on to the next filter. |
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*/ |
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void ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref); |
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#endif /* AVFILTER_AUDIO_H */ |
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