[FFmpeg-cvslog] lavfi: add audio mix filter
Justin Ruggles
git at videolan.org
Fri May 25 00:52:46 CEST 2012
ffmpeg | branch: master | Justin Ruggles <justin.ruggles at gmail.com> | Mon May 21 21:27:59 2012 -0400| [c7448c182a701b4c6efc52e0224bcbecc1aa6c3b] | committer: Justin Ruggles
lavfi: add audio mix filter
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=c7448c182a701b4c6efc52e0224bcbecc1aa6c3b
---
Changelog | 1 +
doc/filters.texi | 38 ++++
libavfilter/Makefile | 1 +
libavfilter/af_amix.c | 545 ++++++++++++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 1 +
libavfilter/version.h | 2 +-
6 files changed, 587 insertions(+), 1 deletion(-)
diff --git a/Changelog b/Changelog
index 2da60b1..252cd9e 100644
--- a/Changelog
+++ b/Changelog
@@ -20,6 +20,7 @@ version <next>:
- audio filters support in libavfilter and avconv
- add fps filter
- audio split filter
+- audio mix filter
version 0.8:
diff --git a/doc/filters.texi b/doc/filters.texi
index ac78029..0179682 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -133,6 +133,44 @@ For example to force the output to either unsigned 8-bit or signed 16-bit stereo
aformat=sample_fmts\=u8\,s16:channel_layouts\=stereo
@end example
+ at section amix
+
+Mixes multiple audio inputs into a single output.
+
+For example
+ at example
+avconv -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex amix=inputs=3:duration=first:dropout_transition=3 OUTPUT
+ at end example
+will mix 3 input audio streams to a single output with the same duration as the
+first input and a dropout transition time of 3 seconds.
+
+The filter accepts the following named parameters:
+ at table @option
+
+ at item inputs
+Number of inputs. If unspecified, it defaults to 2.
+
+ at item duration
+How to determine the end-of-stream.
+ at table @option
+
+ at item longest
+Duration of longest input. (default)
+
+ at item shortest
+Duration of shortest input.
+
+ at item first
+Duration of first input.
+
+ at end table
+
+ at item dropout_transition
+Transition time, in seconds, for volume renormalization when an input
+stream ends. The default value is 2 seconds.
+
+ at end table
+
@section anull
Pass the audio source unchanged to the output.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 212e992..914f0c6 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -25,6 +25,7 @@ OBJS = allfilters.o \
video.o \
OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o
+OBJS-$(CONFIG_AMIX_FILTER) += af_amix.o
OBJS-$(CONFIG_ANULL_FILTER) += af_anull.o
OBJS-$(CONFIG_ASPLIT_FILTER) += split.o
OBJS-$(CONFIG_ASYNCTS_FILTER) += af_asyncts.o
diff --git a/libavfilter/af_amix.c b/libavfilter/af_amix.c
new file mode 100644
index 0000000..3399b7c
--- /dev/null
+++ b/libavfilter/af_amix.c
@@ -0,0 +1,545 @@
+/*
+ * Audio Mix Filter
+ * Copyright (c) 2012 Justin Ruggles <justin.ruggles at gmail.com>
+ *
+ * This file is part of Libav.
+ *
+ * Libav 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,
+ * 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
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * Audio Mix Filter
+ *
+ * Mixes audio from multiple sources into a single output. The channel layout,
+ * sample rate, and sample format will be the same for all inputs and the
+ * output.
+ */
+
+#include "libavutil/audioconvert.h"
+#include "libavutil/audio_fifo.h"
+#include "libavutil/avassert.h"
+#include "libavutil/avstring.h"
+#include "libavutil/mathematics.h"
+#include "libavutil/opt.h"
+#include "libavutil/samplefmt.h"
+
+#include "audio.h"
+#include "avfilter.h"
+#include "formats.h"
+#include "internal.h"
+
+#define INPUT_OFF 0 /**< input has reached EOF */
+#define INPUT_ON 1 /**< input is active */
+#define INPUT_INACTIVE 2 /**< input is on, but is currently inactive */
+
+#define DURATION_LONGEST 0
+#define DURATION_SHORTEST 1
+#define DURATION_FIRST 2
+
+
+typedef struct FrameInfo {
+ int nb_samples;
+ int64_t pts;
+ struct FrameInfo *next;
+} FrameInfo;
+
+/**
+ * Linked list used to store timestamps and frame sizes of all frames in the
+ * FIFO for the first input.
+ *
+ * This is needed to keep timestamps synchronized for the case where multiple
+ * input frames are pushed to the filter for processing before a frame is
+ * requested by the output link.
+ */
+typedef struct FrameList {
+ int nb_frames;
+ int nb_samples;
+ FrameInfo *list;
+ FrameInfo *end;
+} FrameList;
+
+static void frame_list_clear(FrameList *frame_list)
+{
+ if (frame_list) {
+ while (frame_list->list) {
+ FrameInfo *info = frame_list->list;
+ frame_list->list = info->next;
+ av_free(info);
+ }
+ frame_list->nb_frames = 0;
+ frame_list->nb_samples = 0;
+ frame_list->end = NULL;
+ }
+}
+
+static int frame_list_next_frame_size(FrameList *frame_list)
+{
+ if (!frame_list->list)
+ return 0;
+ return frame_list->list->nb_samples;
+}
+
+static int64_t frame_list_next_pts(FrameList *frame_list)
+{
+ if (!frame_list->list)
+ return AV_NOPTS_VALUE;
+ return frame_list->list->pts;
+}
+
+static void frame_list_remove_samples(FrameList *frame_list, int nb_samples)
+{
+ if (nb_samples >= frame_list->nb_samples) {
+ frame_list_clear(frame_list);
+ } else {
+ int samples = nb_samples;
+ while (samples > 0) {
+ FrameInfo *info = frame_list->list;
+ av_assert0(info != NULL);
+ if (info->nb_samples <= samples) {
+ samples -= info->nb_samples;
+ frame_list->list = info->next;
+ if (!frame_list->list)
+ frame_list->end = NULL;
+ frame_list->nb_frames--;
+ frame_list->nb_samples -= info->nb_samples;
+ av_free(info);
+ } else {
+ info->nb_samples -= samples;
+ info->pts += samples;
+ frame_list->nb_samples -= samples;
+ samples = 0;
+ }
+ }
+ }
+}
+
+static int frame_list_add_frame(FrameList *frame_list, int nb_samples, int64_t pts)
+{
+ FrameInfo *info = av_malloc(sizeof(*info));
+ if (!info)
+ return AVERROR(ENOMEM);
+ info->nb_samples = nb_samples;
+ info->pts = pts;
+ info->next = NULL;
+
+ if (!frame_list->list) {
+ frame_list->list = info;
+ frame_list->end = info;
+ } else {
+ av_assert0(frame_list->end != NULL);
+ frame_list->end->next = info;
+ frame_list->end = info;
+ }
+ frame_list->nb_frames++;
+ frame_list->nb_samples += nb_samples;
+
+ return 0;
+}
+
+
+typedef struct MixContext {
+ const AVClass *class; /**< class for AVOptions */
+
+ int nb_inputs; /**< number of inputs */
+ int active_inputs; /**< number of input currently active */
+ int duration_mode; /**< mode for determining duration */
+ float dropout_transition; /**< transition time when an input drops out */
+
+ int nb_channels; /**< number of channels */
+ int sample_rate; /**< sample rate */
+ AVAudioFifo **fifos; /**< audio fifo for each input */
+ uint8_t *input_state; /**< current state of each input */
+ float *input_scale; /**< mixing scale factor for each input */
+ float scale_norm; /**< normalization factor for all inputs */
+ int64_t next_pts; /**< calculated pts for next output frame */
+ FrameList *frame_list; /**< list of frame info for the first input */
+} MixContext;
+
+#define OFFSET(x) offsetof(MixContext, x)
+#define A AV_OPT_FLAG_AUDIO_PARAM
+static const AVOption options[] = {
+ { "inputs", "Number of inputs.",
+ OFFSET(nb_inputs), AV_OPT_TYPE_INT, { 2 }, 1, 32, A },
+ { "duration", "How to determine the end-of-stream.",
+ OFFSET(duration_mode), AV_OPT_TYPE_INT, { DURATION_LONGEST }, 0, 2, A, "duration" },
+ { "longest", "Duration of longest input.", 0, AV_OPT_TYPE_CONST, { DURATION_LONGEST }, INT_MIN, INT_MAX, A, "duration" },
+ { "shortest", "Duration of shortest input.", 0, AV_OPT_TYPE_CONST, { DURATION_SHORTEST }, INT_MIN, INT_MAX, A, "duration" },
+ { "first", "Duration of first input.", 0, AV_OPT_TYPE_CONST, { DURATION_FIRST }, INT_MIN, INT_MAX, A, "duration" },
+ { "dropout_transition", "Transition time, in seconds, for volume "
+ "renormalization when an input stream ends.",
+ OFFSET(dropout_transition), AV_OPT_TYPE_FLOAT, { 2.0 }, 0, INT_MAX, A },
+ { NULL },
+};
+
+static const AVClass amix_class = {
+ .class_name = "amix filter",
+ .item_name = av_default_item_name,
+ .option = options,
+ .version = LIBAVUTIL_VERSION_INT,
+};
+
+
+/**
+ * Update the scaling factors to apply to each input during mixing.
+ *
+ * This balances the full volume range between active inputs and handles
+ * volume transitions when EOF is encountered on an input but mixing continues
+ * with the remaining inputs.
+ */
+static void calculate_scales(MixContext *s, int nb_samples)
+{
+ int i;
+
+ if (s->scale_norm > s->active_inputs) {
+ s->scale_norm -= nb_samples / (s->dropout_transition * s->sample_rate);
+ s->scale_norm = FFMAX(s->scale_norm, s->active_inputs);
+ }
+
+ for (i = 0; i < s->nb_inputs; i++) {
+ if (s->input_state[i] == INPUT_ON)
+ s->input_scale[i] = 1.0f / s->scale_norm;
+ else
+ s->input_scale[i] = 0.0f;
+ }
+}
+
+static int config_output(AVFilterLink *outlink)
+{
+ AVFilterContext *ctx = outlink->src;
+ MixContext *s = ctx->priv;
+ int i;
+ char buf[64];
+
+ s->sample_rate = outlink->sample_rate;
+ outlink->time_base = (AVRational){ 1, outlink->sample_rate };
+ s->next_pts = AV_NOPTS_VALUE;
+
+ s->frame_list = av_mallocz(sizeof(*s->frame_list));
+ if (!s->frame_list)
+ return AVERROR(ENOMEM);
+
+ s->fifos = av_mallocz(s->nb_inputs * sizeof(*s->fifos));
+ if (!s->fifos)
+ return AVERROR(ENOMEM);
+
+ s->nb_channels = av_get_channel_layout_nb_channels(outlink->channel_layout);
+ for (i = 0; i < s->nb_inputs; i++) {
+ s->fifos[i] = av_audio_fifo_alloc(outlink->format, s->nb_channels, 1024);
+ if (!s->fifos[i])
+ return AVERROR(ENOMEM);
+ }
+
+ s->input_state = av_malloc(s->nb_inputs);
+ if (!s->input_state)
+ return AVERROR(ENOMEM);
+ memset(s->input_state, INPUT_ON, s->nb_inputs);
+ s->active_inputs = s->nb_inputs;
+
+ s->input_scale = av_mallocz(s->nb_inputs * sizeof(*s->input_scale));
+ if (!s->input_scale)
+ return AVERROR(ENOMEM);
+ s->scale_norm = s->active_inputs;
+ calculate_scales(s, 0);
+
+ av_get_channel_layout_string(buf, sizeof(buf), -1, outlink->channel_layout);
+
+ av_log(ctx, AV_LOG_VERBOSE,
+ "inputs:%d fmt:%s srate:%"PRId64" cl:%s\n", s->nb_inputs,
+ av_get_sample_fmt_name(outlink->format), outlink->sample_rate, buf);
+
+ return 0;
+}
+
+/* TODO: move optimized version from DSPContext to libavutil */
+static void vector_fmac_scalar(float *dst, const float *src, float mul, int len)
+{
+ int i;
+ for (i = 0; i < len; i++)
+ dst[i] += src[i] * mul;
+}
+
+/**
+ * Read samples from the input FIFOs, mix, and write to the output link.
+ */
+static int output_frame(AVFilterLink *outlink, int nb_samples)
+{
+ AVFilterContext *ctx = outlink->src;
+ MixContext *s = ctx->priv;
+ AVFilterBufferRef *out_buf, *in_buf;
+ int i;
+
+ calculate_scales(s, nb_samples);
+
+ out_buf = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
+ if (!out_buf)
+ return AVERROR(ENOMEM);
+
+ in_buf = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
+ if (!in_buf)
+ return AVERROR(ENOMEM);
+
+ for (i = 0; i < s->nb_inputs; i++) {
+ if (s->input_state[i] == INPUT_ON) {
+ av_audio_fifo_read(s->fifos[i], (void **)in_buf->extended_data,
+ nb_samples);
+ vector_fmac_scalar((float *)out_buf->extended_data[0],
+ (float *) in_buf->extended_data[0],
+ s->input_scale[i], nb_samples * s->nb_channels);
+ }
+ }
+ avfilter_unref_buffer(in_buf);
+
+ out_buf->pts = s->next_pts;
+ if (s->next_pts != AV_NOPTS_VALUE)
+ s->next_pts += nb_samples;
+
+ ff_filter_samples(outlink, out_buf);
+
+ return 0;
+}
+
+/**
+ * Returns the smallest number of samples available in the input FIFOs other
+ * than that of the first input.
+ */
+static int get_available_samples(MixContext *s)
+{
+ int i;
+ int available_samples = INT_MAX;
+
+ av_assert0(s->nb_inputs > 1);
+
+ for (i = 1; i < s->nb_inputs; i++) {
+ int nb_samples;
+ if (s->input_state[i] == INPUT_OFF)
+ continue;
+ nb_samples = av_audio_fifo_size(s->fifos[i]);
+ available_samples = FFMIN(available_samples, nb_samples);
+ }
+ if (available_samples == INT_MAX)
+ return 0;
+ return available_samples;
+}
+
+/**
+ * Requests a frame, if needed, from each input link other than the first.
+ */
+static int request_samples(AVFilterContext *ctx, int min_samples)
+{
+ MixContext *s = ctx->priv;
+ int i, ret;
+
+ av_assert0(s->nb_inputs > 1);
+
+ for (i = 1; i < s->nb_inputs; i++) {
+ ret = 0;
+ if (s->input_state[i] == INPUT_OFF)
+ continue;
+ while (!ret && av_audio_fifo_size(s->fifos[i]) < min_samples)
+ ret = avfilter_request_frame(ctx->inputs[i]);
+ if (ret == AVERROR_EOF) {
+ if (av_audio_fifo_size(s->fifos[i]) == 0) {
+ s->input_state[i] = INPUT_OFF;
+ continue;
+ }
+ } else if (ret)
+ return ret;
+ }
+ return 0;
+}
+
+/**
+ * Calculates the number of active inputs and determines EOF based on the
+ * duration option.
+ *
+ * @return 0 if mixing should continue, or AVERROR_EOF if mixing should stop.
+ */
+static int calc_active_inputs(MixContext *s)
+{
+ int i;
+ int active_inputs = 0;
+ for (i = 0; i < s->nb_inputs; i++)
+ active_inputs += !!(s->input_state[i] != INPUT_OFF);
+ s->active_inputs = active_inputs;
+
+ if (!active_inputs ||
+ (s->duration_mode == DURATION_FIRST && s->input_state[0] == INPUT_OFF) ||
+ (s->duration_mode == DURATION_SHORTEST && active_inputs != s->nb_inputs))
+ return AVERROR_EOF;
+ return 0;
+}
+
+static int request_frame(AVFilterLink *outlink)
+{
+ AVFilterContext *ctx = outlink->src;
+ MixContext *s = ctx->priv;
+ int ret;
+ int wanted_samples, available_samples;
+
+ if (s->input_state[0] == INPUT_OFF) {
+ ret = request_samples(ctx, 1);
+ if (ret < 0)
+ return ret;
+
+ ret = calc_active_inputs(s);
+ if (ret < 0)
+ return ret;
+
+ available_samples = get_available_samples(s);
+ if (!available_samples)
+ return 0;
+
+ return output_frame(outlink, available_samples);
+ }
+
+ if (s->frame_list->nb_frames == 0) {
+ ret = avfilter_request_frame(ctx->inputs[0]);
+ if (ret == AVERROR_EOF) {
+ s->input_state[0] = INPUT_OFF;
+ if (s->nb_inputs == 1)
+ return AVERROR_EOF;
+ else
+ return AVERROR(EAGAIN);
+ } else if (ret)
+ return ret;
+ }
+ av_assert0(s->frame_list->nb_frames > 0);
+
+ wanted_samples = frame_list_next_frame_size(s->frame_list);
+ ret = request_samples(ctx, wanted_samples);
+ if (ret < 0)
+ return ret;
+
+ ret = calc_active_inputs(s);
+ if (ret < 0)
+ return ret;
+
+ if (s->active_inputs > 1) {
+ available_samples = get_available_samples(s);
+ if (!available_samples)
+ return 0;
+ available_samples = FFMIN(available_samples, wanted_samples);
+ } else {
+ available_samples = wanted_samples;
+ }
+
+ s->next_pts = frame_list_next_pts(s->frame_list);
+ frame_list_remove_samples(s->frame_list, available_samples);
+
+ return output_frame(outlink, available_samples);
+}
+
+static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
+{
+ AVFilterContext *ctx = inlink->dst;
+ MixContext *s = ctx->priv;
+ AVFilterLink *outlink = ctx->outputs[0];
+ int i;
+
+ for (i = 0; i < ctx->input_count; i++)
+ if (ctx->inputs[i] == inlink)
+ break;
+ if (i >= ctx->input_count) {
+ av_log(ctx, AV_LOG_ERROR, "unknown input link\n");
+ return;
+ }
+
+ if (i == 0) {
+ int64_t pts = av_rescale_q(buf->pts, inlink->time_base,
+ outlink->time_base);
+ frame_list_add_frame(s->frame_list, buf->audio->nb_samples, pts);
+ }
+
+ av_audio_fifo_write(s->fifos[i], (void **)buf->extended_data,
+ buf->audio->nb_samples);
+
+ avfilter_unref_buffer(buf);
+}
+
+static int init(AVFilterContext *ctx, const char *args, void *opaque)
+{
+ MixContext *s = ctx->priv;
+ int i, ret;
+
+ s->class = &amix_class;
+ av_opt_set_defaults(s);
+
+ if ((ret = av_set_options_string(s, args, "=", ":")) < 0) {
+ av_log(ctx, AV_LOG_ERROR, "Error parsing options string '%s'.\n", args);
+ return ret;
+ }
+ av_opt_free(s);
+
+ for (i = 0; i < s->nb_inputs; i++) {
+ char name[32];
+ AVFilterPad pad = { 0 };
+
+ snprintf(name, sizeof(name), "input%d", i);
+ pad.type = AVMEDIA_TYPE_AUDIO;
+ pad.name = av_strdup(name);
+ pad.filter_samples = filter_samples;
+
+ avfilter_insert_inpad(ctx, i, &pad);
+ }
+
+ return 0;
+}
+
+static void uninit(AVFilterContext *ctx)
+{
+ int i;
+ MixContext *s = ctx->priv;
+
+ if (s->fifos) {
+ for (i = 0; i < s->nb_inputs; i++)
+ av_audio_fifo_free(s->fifos[i]);
+ av_freep(&s->fifos);
+ }
+ frame_list_clear(s->frame_list);
+ av_freep(&s->frame_list);
+ av_freep(&s->input_state);
+ av_freep(&s->input_scale);
+
+ for (i = 0; i < ctx->input_count; i++)
+ av_freep(&ctx->input_pads[i].name);
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+ AVFilterFormats *formats = NULL;
+ avfilter_add_format(&formats, AV_SAMPLE_FMT_FLT);
+ avfilter_set_common_formats(ctx, formats);
+ ff_set_common_channel_layouts(ctx, ff_all_channel_layouts());
+ ff_set_common_samplerates(ctx, ff_all_samplerates());
+ return 0;
+}
+
+AVFilter avfilter_af_amix = {
+ .name = "amix",
+ .description = NULL_IF_CONFIG_SMALL("Audio mixing."),
+ .priv_size = sizeof(MixContext),
+
+ .init = init,
+ .uninit = uninit,
+ .query_formats = query_formats,
+
+ .inputs = (const AVFilterPad[]) {{ .name = NULL}},
+ .outputs = (const AVFilterPad[]) {{ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .config_props = config_output,
+ .request_frame = request_frame },
+ { .name = NULL}},
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 316fff1..941ca6a 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -35,6 +35,7 @@ void avfilter_register_all(void)
initialized = 1;
REGISTER_FILTER (AFORMAT, aformat, af);
+ REGISTER_FILTER (AMIX, amix, af);
REGISTER_FILTER (ANULL, anull, af);
REGISTER_FILTER (ASPLIT, asplit, af);
REGISTER_FILTER (ASYNCTS, asyncts, af);
diff --git a/libavfilter/version.h b/libavfilter/version.h
index 6194876..9a64316 100644
--- a/libavfilter/version.h
+++ b/libavfilter/version.h
@@ -29,7 +29,7 @@
#include "libavutil/avutil.h"
#define LIBAVFILTER_VERSION_MAJOR 2
-#define LIBAVFILTER_VERSION_MINOR 19
+#define LIBAVFILTER_VERSION_MINOR 20
#define LIBAVFILTER_VERSION_MICRO 0
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
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