mirror of https://github.com/FFmpeg/FFmpeg.git
Originally based on code by Stefano Sabatini and S. N. Hemanth. Signed-off-by: Stefano Sabatini <stefano.sabatini-lala@poste.it>pull/2/head
Mina Nagy Zaki
14 years ago
committed by
Stefano Sabatini
parent
f138c7f993
commit
587c8ab912
7 changed files with 495 additions and 1 deletions
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/*
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* Copyright (c) 2010 S.N. Hemanth Meenakshisundaram |
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* Copyright (c) 2011 Mina Nagy Zaki |
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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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/**
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* @file |
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* memory buffer source for audio |
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*/ |
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#include "libavutil/audioconvert.h" |
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#include "libavutil/fifo.h" |
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#include "asrc_abuffer.h" |
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#include "internal.h" |
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typedef struct { |
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// Audio format of incoming buffers
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int sample_rate; |
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unsigned int sample_format; |
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int64_t channel_layout; |
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int packing_format; |
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// FIFO buffer of audio buffer ref pointers
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AVFifoBuffer *fifo; |
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// Normalization filters
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AVFilterContext *aconvert; |
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AVFilterContext *aresample; |
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} ABufferSourceContext; |
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#define FIFO_SIZE 8 |
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static void buf_free(AVFilterBuffer *ptr) |
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{ |
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av_free(ptr); |
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return; |
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} |
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static void set_link_source(AVFilterContext *src, AVFilterLink *link) |
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{ |
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link->src = src; |
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link->srcpad = &(src->output_pads[0]); |
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src->outputs[0] = link; |
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} |
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static int reconfigure_filter(ABufferSourceContext *abuffer, AVFilterContext *filt_ctx) |
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{ |
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int ret; |
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AVFilterLink * const inlink = filt_ctx->inputs[0]; |
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AVFilterLink * const outlink = filt_ctx->outputs[0]; |
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inlink->format = abuffer->sample_format; |
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inlink->channel_layout = abuffer->channel_layout; |
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inlink->planar = abuffer->packing_format; |
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inlink->sample_rate = abuffer->sample_rate; |
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filt_ctx->filter->uninit(filt_ctx); |
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memset(filt_ctx->priv, 0, filt_ctx->filter->priv_size); |
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if ((ret = filt_ctx->filter->init(filt_ctx, NULL , NULL)) < 0) |
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return ret; |
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if ((ret = inlink->srcpad->config_props(inlink)) < 0) |
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return ret; |
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return outlink->srcpad->config_props(outlink); |
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} |
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static int insert_filter(ABufferSourceContext *abuffer, |
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AVFilterLink *link, AVFilterContext **filt_ctx, |
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const char *filt_name) |
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{ |
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int ret; |
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if ((ret = avfilter_open(filt_ctx, avfilter_get_by_name(filt_name), NULL)) < 0) |
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return ret; |
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link->src->outputs[0] = NULL; |
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if ((ret = avfilter_link(link->src, 0, *filt_ctx, 0)) < 0) { |
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link->src->outputs[0] = link; |
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return ret; |
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} |
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set_link_source(*filt_ctx, link); |
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if ((ret = reconfigure_filter(abuffer, *filt_ctx)) < 0) { |
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avfilter_free(*filt_ctx); |
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return ret; |
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} |
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return 0; |
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} |
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static void remove_filter(AVFilterContext **filt_ctx) |
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{ |
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AVFilterLink *outlink = (*filt_ctx)->outputs[0]; |
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AVFilterContext *src = (*filt_ctx)->inputs[0]->src; |
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(*filt_ctx)->outputs[0] = NULL; |
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avfilter_free(*filt_ctx); |
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*filt_ctx = NULL; |
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set_link_source(src, outlink); |
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} |
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static inline void log_input_change(void *ctx, AVFilterLink *link, AVFilterBufferRef *ref) |
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{ |
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char old_layout_str[16], new_layout_str[16]; |
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av_get_channel_layout_string(old_layout_str, sizeof(old_layout_str), |
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-1, link->channel_layout); |
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av_get_channel_layout_string(new_layout_str, sizeof(new_layout_str), |
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-1, ref->audio->channel_layout); |
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av_log(ctx, AV_LOG_INFO, |
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"Audio input format changed: " |
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"%s:%s:%"PRId64" -> %s:%s:%u, normalizing\n", |
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av_get_sample_fmt_name(link->format), |
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old_layout_str, link->sample_rate, |
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av_get_sample_fmt_name(ref->format), |
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new_layout_str, ref->audio->sample_rate); |
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} |
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int av_asrc_buffer_add_audio_buffer_ref(AVFilterContext *ctx, |
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AVFilterBufferRef *samplesref, |
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int av_unused flags) |
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{ |
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ABufferSourceContext *abuffer = ctx->priv; |
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AVFilterLink *link; |
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int ret, logged = 0; |
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if (av_fifo_space(abuffer->fifo) < sizeof(samplesref)) { |
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av_log(ctx, AV_LOG_ERROR, |
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"Buffering limit reached. Please consume some available frames " |
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"before adding new ones.\n"); |
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return AVERROR(EINVAL); |
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} |
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// Normalize input
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link = ctx->outputs[0]; |
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if (samplesref->audio->sample_rate != link->sample_rate) { |
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log_input_change(ctx, link, samplesref); |
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logged = 1; |
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abuffer->sample_rate = samplesref->audio->sample_rate; |
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if (!abuffer->aresample) { |
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ret = insert_filter(abuffer, link, &abuffer->aresample, "aresample"); |
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if (ret < 0) return ret; |
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} else { |
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link = abuffer->aresample->outputs[0]; |
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if (samplesref->audio->sample_rate == link->sample_rate) |
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remove_filter(&abuffer->aresample); |
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else |
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if ((ret = reconfigure_filter(abuffer, abuffer->aresample)) < 0) |
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return ret; |
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} |
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} |
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link = ctx->outputs[0]; |
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if (samplesref->format != link->format || |
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samplesref->audio->channel_layout != link->channel_layout || |
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samplesref->audio->planar != link->planar) { |
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if (!logged) log_input_change(ctx, link, samplesref); |
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abuffer->sample_format = samplesref->format; |
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abuffer->channel_layout = samplesref->audio->channel_layout; |
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abuffer->packing_format = samplesref->audio->planar; |
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if (!abuffer->aconvert) { |
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ret = insert_filter(abuffer, link, &abuffer->aconvert, "aconvert"); |
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if (ret < 0) return ret; |
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} else { |
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link = abuffer->aconvert->outputs[0]; |
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if (samplesref->format == link->format && |
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samplesref->audio->channel_layout == link->channel_layout && |
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samplesref->audio->planar == link->planar |
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) |
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remove_filter(&abuffer->aconvert); |
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else |
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if ((ret = reconfigure_filter(abuffer, abuffer->aconvert)) < 0) |
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return ret; |
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} |
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} |
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if (sizeof(samplesref) != av_fifo_generic_write(abuffer->fifo, &samplesref, |
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sizeof(samplesref), NULL)) { |
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av_log(ctx, AV_LOG_ERROR, "Error while writing to FIFO\n"); |
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return AVERROR(EINVAL); |
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} |
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return 0; |
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} |
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int av_asrc_buffer_add_samples(AVFilterContext *ctx, |
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uint8_t *data[8], int linesize[8], |
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int nb_samples, int sample_rate, |
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int sample_fmt, int64_t channel_layout, int planar, |
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int64_t pts, int av_unused flags) |
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{ |
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AVFilterBufferRef *samplesref; |
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samplesref = avfilter_get_audio_buffer_ref_from_arrays( |
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data, linesize, AV_PERM_WRITE, |
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nb_samples, |
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sample_fmt, channel_layout, planar); |
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if (!samplesref) |
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return AVERROR(ENOMEM); |
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samplesref->buf->free = buf_free; |
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samplesref->pts = pts; |
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samplesref->audio->sample_rate = sample_rate; |
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return av_asrc_buffer_add_audio_buffer_ref(ctx, samplesref, 0); |
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} |
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int av_asrc_buffer_add_buffer(AVFilterContext *ctx, |
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uint8_t *buf, int buf_size, int sample_rate, |
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int sample_fmt, int64_t channel_layout, int planar, |
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int64_t pts, int av_unused flags) |
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{ |
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uint8_t *data[8]; |
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int linesize[8]; |
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int nb_channels = av_get_channel_layout_nb_channels(channel_layout), |
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nb_samples = buf_size / nb_channels / av_get_bytes_per_sample(sample_fmt); |
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av_samples_fill_arrays(data, linesize, |
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buf, nb_channels, nb_samples, |
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sample_fmt, planar, 16); |
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return av_asrc_buffer_add_samples(ctx, |
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data, linesize, nb_samples, |
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sample_rate, |
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sample_fmt, channel_layout, planar, |
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pts, flags); |
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} |
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static av_cold int init(AVFilterContext *ctx, const char *args, void *opaque) |
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{ |
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ABufferSourceContext *abuffer = ctx->priv; |
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char *arg = NULL, *ptr, chlayout_str[16]; |
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int ret; |
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arg = strtok_r(args, ":", &ptr); |
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#define ADD_FORMAT(fmt_name) \ |
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if (!arg) \
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goto arg_fail; \
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if ((ret = ff_parse_##fmt_name(&abuffer->fmt_name, arg, ctx)) < 0) \
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return ret; \
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if (*args) \
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arg = strtok_r(NULL, ":", &ptr) |
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ADD_FORMAT(sample_rate); |
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ADD_FORMAT(sample_format); |
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ADD_FORMAT(channel_layout); |
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ADD_FORMAT(packing_format); |
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abuffer->fifo = av_fifo_alloc(FIFO_SIZE*sizeof(AVFilterBufferRef*)); |
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if (!abuffer->fifo) { |
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av_log(ctx, AV_LOG_ERROR, "Failed to allocate fifo, filter init failed.\n"); |
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return AVERROR(ENOMEM); |
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} |
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av_get_channel_layout_string(chlayout_str, sizeof(chlayout_str), |
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-1, abuffer->channel_layout); |
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av_log(ctx, AV_LOG_INFO, "format:%s layout:%s rate:%d\n", |
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av_get_sample_fmt_name(abuffer->sample_format), chlayout_str, |
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abuffer->sample_rate); |
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return 0; |
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arg_fail: |
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av_log(ctx, AV_LOG_ERROR, "Invalid arguments, must be of the form " |
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"sample_rate:sample_fmt:channel_layout:packing\n"); |
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return AVERROR(EINVAL); |
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} |
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static av_cold void uninit(AVFilterContext *ctx) |
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{ |
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ABufferSourceContext *abuffer = ctx->priv; |
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av_fifo_free(abuffer->fifo); |
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} |
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static int query_formats(AVFilterContext *ctx) |
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{ |
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ABufferSourceContext *abuffer = ctx->priv; |
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AVFilterFormats *formats; |
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formats = NULL; |
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avfilter_add_format(&formats, abuffer->sample_format); |
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avfilter_set_common_sample_formats(ctx, formats); |
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formats = NULL; |
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avfilter_add_format(&formats, abuffer->channel_layout); |
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avfilter_set_common_channel_layouts(ctx, formats); |
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formats = NULL; |
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avfilter_add_format(&formats, abuffer->packing_format); |
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avfilter_set_common_packing_formats(ctx, formats); |
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return 0; |
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} |
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static int config_output(AVFilterLink *outlink) |
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{ |
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ABufferSourceContext *abuffer = outlink->src->priv; |
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outlink->sample_rate = abuffer->sample_rate; |
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return 0; |
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} |
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static int request_frame(AVFilterLink *outlink) |
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{ |
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ABufferSourceContext *abuffer = outlink->src->priv; |
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AVFilterBufferRef *samplesref; |
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if (!av_fifo_size(abuffer->fifo)) { |
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av_log(outlink->src, AV_LOG_ERROR, |
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"request_frame() called with no available frames!\n"); |
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return AVERROR(EINVAL); |
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} |
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av_fifo_generic_read(abuffer->fifo, &samplesref, sizeof(samplesref), NULL); |
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avfilter_filter_samples(outlink, avfilter_ref_buffer(samplesref, ~0)); |
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avfilter_unref_buffer(samplesref); |
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return 0; |
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} |
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static int poll_frame(AVFilterLink *outlink) |
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{ |
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ABufferSourceContext *abuffer = outlink->src->priv; |
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return av_fifo_size(abuffer->fifo)/sizeof(AVFilterBufferRef*); |
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} |
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AVFilter avfilter_asrc_abuffer = { |
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.name = "abuffer", |
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.description = NULL_IF_CONFIG_SMALL("Buffer audio frames, and make them accessible to the filterchain."), |
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.priv_size = sizeof(ABufferSourceContext), |
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.query_formats = query_formats, |
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.init = init, |
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.uninit = uninit, |
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.inputs = (AVFilterPad[]) {{ .name = NULL }}, |
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.outputs = (AVFilterPad[]) {{ .name = "default", |
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.type = AVMEDIA_TYPE_AUDIO, |
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.request_frame = request_frame, |
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.poll_frame = poll_frame, |
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.config_props = config_output, }, |
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{ .name = NULL}}, |
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}; |
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@ -0,0 +1,80 @@ |
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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_ASRC_ABUFFER_H |
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#define AVFILTER_ASRC_ABUFFER_H |
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#include "avfilter.h" |
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/**
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* @file |
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* memory buffer source for audio |
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*/ |
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|
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/**
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* Queue an audio buffer to the audio buffer source. |
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* |
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* @param abuffersrc audio source buffer context |
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* @param data pointers to the samples planes |
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* @param linesize linesizes of each audio buffer plane |
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* @param nb_samples number of samples per channel |
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* @param sample_fmt sample format of the audio data |
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* @param ch_layout channel layout of the audio data |
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* @param planar flag to indicate if audio data is planar or packed |
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* @param pts presentation timestamp of the audio buffer |
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* @param flags unused |
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*/ |
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int av_asrc_buffer_add_samples(AVFilterContext *abuffersrc, |
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uint8_t *data[8], int linesize[8], |
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int nb_samples, int sample_rate, |
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int sample_fmt, int64_t ch_layout, int planar, |
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|
int64_t pts, int av_unused flags); |
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|
/**
|
||||||
|
* Queue an audio buffer to the audio buffer source. |
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* |
||||||
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* This is similar to av_asrc_buffer_add_samples(), but the samples |
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|
* are stored in a buffer with known size. |
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|
* |
||||||
|
* @param abuffersrc audio source buffer context |
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* @param buf pointer to the samples data, packed is assumed |
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|
* @param size the size in bytes of the buffer, it must contain an |
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|
* integer number of samples |
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|
* @param sample_fmt sample format of the audio data |
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|
* @param ch_layout channel layout of the audio data |
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|
* @param pts presentation timestamp of the audio buffer |
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|
* @param flags unused |
||||||
|
*/ |
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|
int av_asrc_buffer_add_buffer(AVFilterContext *abuffersrc, |
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|
uint8_t *buf, int buf_size, |
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|
int sample_rate, |
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|
int sample_fmt, int64_t ch_layout, int planar, |
||||||
|
int64_t pts, int av_unused flags); |
||||||
|
|
||||||
|
/**
|
||||||
|
* Queue an audio buffer to the audio buffer source. |
||||||
|
* |
||||||
|
* @param abuffersrc audio source buffer context |
||||||
|
* @param samplesref buffer ref to queue |
||||||
|
* @param flags unused |
||||||
|
*/ |
||||||
|
int av_asrc_buffer_add_audio_buffer_ref(AVFilterContext *abuffersrc, |
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|
AVFilterBufferRef *samplesref, |
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|
int av_unused flags); |
||||||
|
|
||||||
|
#endif /* AVFILTER_ASRC_ABUFFER_H */ |
||||||
Loading…
Reference in new issue