diff options
author | Mina Nagy Zaki <mnzaki@gmail.com> | 2011-08-01 11:33:26 +0300 |
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committer | Stefano Sabatini <stefano.sabatini-lala@poste.it> | 2011-08-21 11:37:57 +0200 |
commit | 587c8ab9128455ccf2580c5350992e4a402dc8fd (patch) | |
tree | 90e8b2389003a3fde082987cbfdd3cb6e3bcdd56 | |
parent | f138c7f993e1aaf5223c546da5292993a467ee8d (diff) | |
download | ffmpeg-587c8ab9128455ccf2580c5350992e4a402dc8fd.tar.gz |
lavfi: add asrc_abuffer - audio buffer source
Originally based on code by Stefano Sabatini and S. N. Hemanth.
Signed-off-by: Stefano Sabatini <stefano.sabatini-lala@poste.it>
-rwxr-xr-x | configure | 1 | ||||
-rw-r--r-- | doc/filters.texi | 45 | ||||
-rw-r--r-- | libavfilter/Makefile | 1 | ||||
-rw-r--r-- | libavfilter/allfilters.c | 1 | ||||
-rw-r--r-- | libavfilter/asrc_abuffer.c | 366 | ||||
-rw-r--r-- | libavfilter/asrc_abuffer.h | 80 | ||||
-rw-r--r-- | libavfilter/avfilter.h | 2 |
7 files changed, 495 insertions, 1 deletions
@@ -1501,6 +1501,7 @@ tcp_protocol_deps="network" udp_protocol_deps="network" # filters +abuffer="strtok_r" aformat_filter_deps="strtok_r" blackframe_filter_deps="gpl" boxblur_filter_deps="gpl" diff --git a/doc/filters.texi b/doc/filters.texi index dd99c73175..69ba4b1698 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -194,6 +194,51 @@ Adler-32 checksum for each input frame plane, expressed in the form Below is a description of the currently available audio sources. +@section abuffer + +Buffer audio frames, and make them available to the filter chain. + +This source is mainly intended for a programmatic use, in particular +through the interface defined in @file{libavfilter/asrc_abuffer.h}. + +It accepts the following mandatory parameters: +@var{sample_rate}:@var{sample_fmt}:@var{channel_layout}:@var{packing} + +@table @option + +@item sample_rate +The sample rate of the incoming audio buffers. + +@item sample_fmt +The sample format of the incoming audio buffers. +Either a sample format name or its corresponging integer representation from +the enum AVSampleFormat in @file{libavutil/samplefmt.h} + +@item channel_layout +The channel layout of the incoming audio buffers. +Either a channel layout name from channel_layout_map in +@file{libavutil/audioconvert.c} or its corresponding integer representation +from the AV_CH_LAYOUT_* macros in @file{libavutil/audioconvert.h} + +@item packing +Either "packed" or "planar", or their integer representation: 0 or 1 +respectively. + +@end table + +For example: +@example +abuffer=44100:s16:stereo:planar +@end example + +will instruct the source to accept planar 16bit signed stereo at 44100Hz. +Since the sample format with name "s16" corresponds to the number +1 and the "stereo" channel layout corresponds to the value 3, this is +equivalent to: +@example +abuffer=44100:1:3:1 +@end example + @section anullsrc Null audio source, never return audio frames. It is mainly useful as a diff --git a/libavfilter/Makefile b/libavfilter/Makefile index a3faf27fb4..5ed7e997ed 100644 --- a/libavfilter/Makefile +++ b/libavfilter/Makefile @@ -24,6 +24,7 @@ OBJS-$(CONFIG_ANULL_FILTER) += af_anull.o OBJS-$(CONFIG_ARESAMPLE_FILTER) += af_aresample.o OBJS-$(CONFIG_ASHOWINFO_FILTER) += af_ashowinfo.o +OBJS-$(CONFIG_ABUFFER_FILTER) += asrc_abuffer.o OBJS-$(CONFIG_ANULLSRC_FILTER) += asrc_anullsrc.o OBJS-$(CONFIG_ABUFFERSINK_FILTER) += asink_abuffer.o diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c index 5cf330c395..3675561062 100644 --- a/libavfilter/allfilters.c +++ b/libavfilter/allfilters.c @@ -39,6 +39,7 @@ void avfilter_register_all(void) REGISTER_FILTER (ARESAMPLE, aresample, af); REGISTER_FILTER (ASHOWINFO, ashowinfo, af); + REGISTER_FILTER (ABUFFER, abuffer, asrc); REGISTER_FILTER (ANULLSRC, anullsrc, asrc); REGISTER_FILTER (ABUFFERSINK, abuffersink, asink); diff --git a/libavfilter/asrc_abuffer.c b/libavfilter/asrc_abuffer.c new file mode 100644 index 0000000000..badc2d8adf --- /dev/null +++ b/libavfilter/asrc_abuffer.c @@ -0,0 +1,366 @@ +/* + * Copyright (c) 2010 S.N. Hemanth Meenakshisundaram + * Copyright (c) 2011 Mina Nagy Zaki + * + * 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 + */ + +/** + * @file + * memory buffer source for audio + */ + +#include "libavutil/audioconvert.h" +#include "libavutil/fifo.h" +#include "asrc_abuffer.h" +#include "internal.h" + +typedef struct { + // Audio format of incoming buffers + int sample_rate; + unsigned int sample_format; + int64_t channel_layout; + int packing_format; + + // FIFO buffer of audio buffer ref pointers + AVFifoBuffer *fifo; + + // Normalization filters + AVFilterContext *aconvert; + AVFilterContext *aresample; +} ABufferSourceContext; + +#define FIFO_SIZE 8 + +static void buf_free(AVFilterBuffer *ptr) +{ + av_free(ptr); + return; +} + +static void set_link_source(AVFilterContext *src, AVFilterLink *link) +{ + link->src = src; + link->srcpad = &(src->output_pads[0]); + src->outputs[0] = link; +} + +static int reconfigure_filter(ABufferSourceContext *abuffer, AVFilterContext *filt_ctx) +{ + int ret; + AVFilterLink * const inlink = filt_ctx->inputs[0]; + AVFilterLink * const outlink = filt_ctx->outputs[0]; + + inlink->format = abuffer->sample_format; + inlink->channel_layout = abuffer->channel_layout; + inlink->planar = abuffer->packing_format; + inlink->sample_rate = abuffer->sample_rate; + + filt_ctx->filter->uninit(filt_ctx); + memset(filt_ctx->priv, 0, filt_ctx->filter->priv_size); + if ((ret = filt_ctx->filter->init(filt_ctx, NULL , NULL)) < 0) + return ret; + if ((ret = inlink->srcpad->config_props(inlink)) < 0) + return ret; + return outlink->srcpad->config_props(outlink); +} + +static int insert_filter(ABufferSourceContext *abuffer, + AVFilterLink *link, AVFilterContext **filt_ctx, + const char *filt_name) +{ + int ret; + + if ((ret = avfilter_open(filt_ctx, avfilter_get_by_name(filt_name), NULL)) < 0) + return ret; + + link->src->outputs[0] = NULL; + if ((ret = avfilter_link(link->src, 0, *filt_ctx, 0)) < 0) { + link->src->outputs[0] = link; + return ret; + } + + set_link_source(*filt_ctx, link); + + if ((ret = reconfigure_filter(abuffer, *filt_ctx)) < 0) { + avfilter_free(*filt_ctx); + return ret; + } + + return 0; +} + +static void remove_filter(AVFilterContext **filt_ctx) +{ + AVFilterLink *outlink = (*filt_ctx)->outputs[0]; + AVFilterContext *src = (*filt_ctx)->inputs[0]->src; + + (*filt_ctx)->outputs[0] = NULL; + avfilter_free(*filt_ctx); + *filt_ctx = NULL; + + set_link_source(src, outlink); +} + +static inline void log_input_change(void *ctx, AVFilterLink *link, AVFilterBufferRef *ref) +{ + char old_layout_str[16], new_layout_str[16]; + av_get_channel_layout_string(old_layout_str, sizeof(old_layout_str), + -1, link->channel_layout); + av_get_channel_layout_string(new_layout_str, sizeof(new_layout_str), + -1, ref->audio->channel_layout); + av_log(ctx, AV_LOG_INFO, + "Audio input format changed: " + "%s:%s:%"PRId64" -> %s:%s:%u, normalizing\n", + av_get_sample_fmt_name(link->format), + old_layout_str, link->sample_rate, + av_get_sample_fmt_name(ref->format), + new_layout_str, ref->audio->sample_rate); +} + +int av_asrc_buffer_add_audio_buffer_ref(AVFilterContext *ctx, + AVFilterBufferRef *samplesref, + int av_unused flags) +{ + ABufferSourceContext *abuffer = ctx->priv; + AVFilterLink *link; + int ret, logged = 0; + + if (av_fifo_space(abuffer->fifo) < sizeof(samplesref)) { + av_log(ctx, AV_LOG_ERROR, + "Buffering limit reached. Please consume some available frames " + "before adding new ones.\n"); + return AVERROR(EINVAL); + } + + // Normalize input + + link = ctx->outputs[0]; + if (samplesref->audio->sample_rate != link->sample_rate) { + + log_input_change(ctx, link, samplesref); + logged = 1; + + abuffer->sample_rate = samplesref->audio->sample_rate; + + if (!abuffer->aresample) { + ret = insert_filter(abuffer, link, &abuffer->aresample, "aresample"); + if (ret < 0) return ret; + } else { + link = abuffer->aresample->outputs[0]; + if (samplesref->audio->sample_rate == link->sample_rate) + remove_filter(&abuffer->aresample); + else + if ((ret = reconfigure_filter(abuffer, abuffer->aresample)) < 0) + return ret; + } + } + + link = ctx->outputs[0]; + if (samplesref->format != link->format || + samplesref->audio->channel_layout != link->channel_layout || + samplesref->audio->planar != link->planar) { + + if (!logged) log_input_change(ctx, link, samplesref); + + abuffer->sample_format = samplesref->format; + abuffer->channel_layout = samplesref->audio->channel_layout; + abuffer->packing_format = samplesref->audio->planar; + + if (!abuffer->aconvert) { + ret = insert_filter(abuffer, link, &abuffer->aconvert, "aconvert"); + if (ret < 0) return ret; + } else { + link = abuffer->aconvert->outputs[0]; + if (samplesref->format == link->format && + samplesref->audio->channel_layout == link->channel_layout && + samplesref->audio->planar == link->planar + ) + remove_filter(&abuffer->aconvert); + else + if ((ret = reconfigure_filter(abuffer, abuffer->aconvert)) < 0) + return ret; + } + } + + if (sizeof(samplesref) != av_fifo_generic_write(abuffer->fifo, &samplesref, + sizeof(samplesref), NULL)) { + av_log(ctx, AV_LOG_ERROR, "Error while writing to FIFO\n"); + return AVERROR(EINVAL); + } + + return 0; +} + +int av_asrc_buffer_add_samples(AVFilterContext *ctx, + uint8_t *data[8], int linesize[8], + int nb_samples, int sample_rate, + int sample_fmt, int64_t channel_layout, int planar, + int64_t pts, int av_unused flags) +{ + AVFilterBufferRef *samplesref; + + samplesref = avfilter_get_audio_buffer_ref_from_arrays( + data, linesize, AV_PERM_WRITE, + nb_samples, + sample_fmt, channel_layout, planar); + if (!samplesref) + return AVERROR(ENOMEM); + + samplesref->buf->free = buf_free; + samplesref->pts = pts; + samplesref->audio->sample_rate = sample_rate; + + return av_asrc_buffer_add_audio_buffer_ref(ctx, samplesref, 0); +} + +int av_asrc_buffer_add_buffer(AVFilterContext *ctx, + uint8_t *buf, int buf_size, int sample_rate, + int sample_fmt, int64_t channel_layout, int planar, + int64_t pts, int av_unused flags) +{ + uint8_t *data[8]; + int linesize[8]; + int nb_channels = av_get_channel_layout_nb_channels(channel_layout), + nb_samples = buf_size / nb_channels / av_get_bytes_per_sample(sample_fmt); + + av_samples_fill_arrays(data, linesize, + buf, nb_channels, nb_samples, + sample_fmt, planar, 16); + + return av_asrc_buffer_add_samples(ctx, + data, linesize, nb_samples, + sample_rate, + sample_fmt, channel_layout, planar, + pts, flags); +} + +static av_cold int init(AVFilterContext *ctx, const char *args, void *opaque) +{ + ABufferSourceContext *abuffer = ctx->priv; + char *arg = NULL, *ptr, chlayout_str[16]; + int ret; + + arg = strtok_r(args, ":", &ptr); + +#define ADD_FORMAT(fmt_name) \ + if (!arg) \ + goto arg_fail; \ + if ((ret = ff_parse_##fmt_name(&abuffer->fmt_name, arg, ctx)) < 0) \ + return ret; \ + if (*args) \ + arg = strtok_r(NULL, ":", &ptr) + + ADD_FORMAT(sample_rate); + ADD_FORMAT(sample_format); + ADD_FORMAT(channel_layout); + ADD_FORMAT(packing_format); + + abuffer->fifo = av_fifo_alloc(FIFO_SIZE*sizeof(AVFilterBufferRef*)); + if (!abuffer->fifo) { + av_log(ctx, AV_LOG_ERROR, "Failed to allocate fifo, filter init failed.\n"); + return AVERROR(ENOMEM); + } + + av_get_channel_layout_string(chlayout_str, sizeof(chlayout_str), + -1, abuffer->channel_layout); + av_log(ctx, AV_LOG_INFO, "format:%s layout:%s rate:%d\n", + av_get_sample_fmt_name(abuffer->sample_format), chlayout_str, + abuffer->sample_rate); + + return 0; + +arg_fail: + av_log(ctx, AV_LOG_ERROR, "Invalid arguments, must be of the form " + "sample_rate:sample_fmt:channel_layout:packing\n"); + return AVERROR(EINVAL); +} + +static av_cold void uninit(AVFilterContext *ctx) +{ + ABufferSourceContext *abuffer = ctx->priv; + av_fifo_free(abuffer->fifo); +} + +static int query_formats(AVFilterContext *ctx) +{ + ABufferSourceContext *abuffer = ctx->priv; + AVFilterFormats *formats; + + formats = NULL; + avfilter_add_format(&formats, abuffer->sample_format); + avfilter_set_common_sample_formats(ctx, formats); + + formats = NULL; + avfilter_add_format(&formats, abuffer->channel_layout); + avfilter_set_common_channel_layouts(ctx, formats); + + formats = NULL; + avfilter_add_format(&formats, abuffer->packing_format); + avfilter_set_common_packing_formats(ctx, formats); + + return 0; +} + +static int config_output(AVFilterLink *outlink) +{ + ABufferSourceContext *abuffer = outlink->src->priv; + outlink->sample_rate = abuffer->sample_rate; + return 0; +} + +static int request_frame(AVFilterLink *outlink) +{ + ABufferSourceContext *abuffer = outlink->src->priv; + AVFilterBufferRef *samplesref; + + if (!av_fifo_size(abuffer->fifo)) { + av_log(outlink->src, AV_LOG_ERROR, + "request_frame() called with no available frames!\n"); + return AVERROR(EINVAL); + } + + av_fifo_generic_read(abuffer->fifo, &samplesref, sizeof(samplesref), NULL); + avfilter_filter_samples(outlink, avfilter_ref_buffer(samplesref, ~0)); + avfilter_unref_buffer(samplesref); + + return 0; +} + +static int poll_frame(AVFilterLink *outlink) +{ + ABufferSourceContext *abuffer = outlink->src->priv; + return av_fifo_size(abuffer->fifo)/sizeof(AVFilterBufferRef*); +} + +AVFilter avfilter_asrc_abuffer = { + .name = "abuffer", + .description = NULL_IF_CONFIG_SMALL("Buffer audio frames, and make them accessible to the filterchain."), + .priv_size = sizeof(ABufferSourceContext), + .query_formats = query_formats, + + .init = init, + .uninit = uninit, + + .inputs = (AVFilterPad[]) {{ .name = NULL }}, + .outputs = (AVFilterPad[]) {{ .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + .request_frame = request_frame, + .poll_frame = poll_frame, + .config_props = config_output, }, + { .name = NULL}}, +}; diff --git a/libavfilter/asrc_abuffer.h b/libavfilter/asrc_abuffer.h new file mode 100644 index 0000000000..4352c74646 --- /dev/null +++ b/libavfilter/asrc_abuffer.h @@ -0,0 +1,80 @@ +/* + * 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 + */ + +#ifndef AVFILTER_ASRC_ABUFFER_H +#define AVFILTER_ASRC_ABUFFER_H + +#include "avfilter.h" + +/** + * @file + * memory buffer source for audio + */ + +/** + * Queue an audio buffer to the audio buffer source. + * + * @param abuffersrc audio source buffer context + * @param data pointers to the samples planes + * @param linesize linesizes of each audio buffer plane + * @param nb_samples number of samples per channel + * @param sample_fmt sample format of the audio data + * @param ch_layout channel layout of the audio data + * @param planar flag to indicate if audio data is planar or packed + * @param pts presentation timestamp of the audio buffer + * @param flags unused + */ +int av_asrc_buffer_add_samples(AVFilterContext *abuffersrc, + uint8_t *data[8], int linesize[8], + int nb_samples, int sample_rate, + 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. + * + * This is similar to av_asrc_buffer_add_samples(), but the samples + * are stored in a buffer with known size. + * + * @param abuffersrc audio source buffer context + * @param buf pointer to the samples data, packed is assumed + * @param size the size in bytes of the buffer, it must contain an + * integer number of samples + * @param sample_fmt sample format of the audio data + * @param ch_layout channel layout of the audio data + * @param pts presentation timestamp of the audio buffer + * @param flags unused + */ +int av_asrc_buffer_add_buffer(AVFilterContext *abuffersrc, + uint8_t *buf, int buf_size, + int sample_rate, + 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, + AVFilterBufferRef *samplesref, + int av_unused flags); + +#endif /* AVFILTER_ASRC_ABUFFER_H */ diff --git a/libavfilter/avfilter.h b/libavfilter/avfilter.h index af2d3cd3e3..68b5d23689 100644 --- a/libavfilter/avfilter.h +++ b/libavfilter/avfilter.h @@ -29,7 +29,7 @@ #include "libavutil/rational.h" #define LIBAVFILTER_VERSION_MAJOR 2 -#define LIBAVFILTER_VERSION_MINOR 33 +#define LIBAVFILTER_VERSION_MINOR 34 #define LIBAVFILTER_VERSION_MICRO 0 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \ |