---
libavfilter/af_amix.c | 299 +++++++++----------------------------------------
1 file changed, 51 insertions(+), 248 deletions(-)
diff --git a/libavfilter/af_amix.c b/libavfilter/af_amix.c
index 6bc7458..8fe770d 100644
--- a/libavfilter/af_amix.c
+++ b/libavfilter/af_amix.c
@@ -28,7 +28,6 @@
* output.
*/
-#include "libavutil/audio_fifo.h"
#include "libavutil/avassert.h"
#include "libavutil/avstring.h"
#include "libavutil/channel_layout.h"
@@ -52,106 +51,6 @@
#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 */
AVFloatDSPContext fdsp;
@@ -164,12 +63,11 @@ typedef struct MixContext {
int nb_channels; /**< number of channels */
int sample_rate; /**< sample rate */
int planar;
- AVAudioFifo **fifos; /**< audio fifo for each input */
+ AVFrame **input_frames; /**< current input frame 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)
@@ -224,7 +122,6 @@ static int config_output(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
MixContext *s = ctx->priv;
- int i;
char buf[64];
s->planar = av_sample_fmt_is_planar(outlink->format);
@@ -232,20 +129,7 @@ static int config_output(AVFilterLink *outlink)
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)
@@ -269,13 +153,14 @@ static int config_output(AVFilterLink *outlink)
}
/**
- * Read samples from the input FIFOs, mix, and write to the output link.
+ * Read samples from the input frames, mix, and write to the output link.
*/
static int output_frame(AVFilterLink *outlink, int nb_samples)
{
AVFilterContext *ctx = outlink->src;
MixContext *s = ctx->priv;
- AVFrame *out_buf, *in_buf;
+ AVFrame *out_buf;
+ int64_t pts = AV_NOPTS_VALUE;
int i;
calculate_scales(s, nb_samples);
@@ -284,18 +169,16 @@ static int output_frame(AVFilterLink *outlink, int
nb_samples)
if (!out_buf)
return AVERROR(ENOMEM);
- in_buf = ff_get_audio_buffer(outlink, nb_samples);
- if (!in_buf) {
- av_frame_free(&out_buf);
- return AVERROR(ENOMEM);
- }
-
for (i = 0; i < s->nb_inputs; i++) {
if (s->input_state[i] == INPUT_ON) {
int planes, plane_size, p;
- av_audio_fifo_read(s->fifos[i], (void **)in_buf->extended_data,
- nb_samples);
+ if (pts == AV_NOPTS_VALUE &&
+ s->input_frames[i]->pts != AV_NOPTS_VALUE) {
+ pts = av_rescale_q(s->input_frames[i]->pts,
+ ctx->inputs[i]->time_base,
+ outlink->time_base);
+ }
planes = s->planar ? s->nb_channels : 1;
plane_size = nb_samples * (s->planar ? 1 : s->nb_channels);
@@ -303,14 +186,18 @@ static int output_frame(AVFilterLink *outlink, int
nb_samples)
for (p = 0; p < planes; p++) {
s->fdsp.vector_fmac_scalar((float *)out_buf->extended_data[p],
- (float *) in_buf->extended_data[p],
+ (float
*)s->input_frames[i]->extended_data[p],
s->input_scale[i], plane_size);
}
+ av_frame_free(&s->input_frames[i]);
}
}
- av_frame_free(&in_buf);
- out_buf->pts = s->next_pts;
+ if (pts != AV_NOPTS_VALUE)
+ out_buf->pts = s->next_pts = pts;
+ else
+ out_buf->pts = s->next_pts;
+
if (s->next_pts != AV_NOPTS_VALUE)
s->next_pts += nb_samples;
@@ -318,56 +205,6 @@ static int output_frame(AVFilterLink *outlink, int
nb_samples)
}
/**
- * 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 = ff_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 < 0)
- return ret;
- }
- return 0;
-}
-
-/**
* Calculates the number of active inputs and determines EOF based on the
* duration option.
*
@@ -392,74 +229,45 @@ static int request_frame(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
MixContext *s = ctx->priv;
- int ret;
- int wanted_samples, available_samples;
+ int ret, i, j;
+ int nb_samples = 0;
ret = calc_active_inputs(s);
if (ret < 0)
return ret;
- 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 AVERROR(EAGAIN);
-
- return output_frame(outlink, available_samples);
- }
-
- if (s->frame_list->nb_frames == 0) {
- ret = ff_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 < 0)
- return ret;
- }
- av_assert0(s->frame_list->nb_frames > 0);
-
- wanted_samples = frame_list_next_frame_size(s->frame_list);
-
- if (s->active_inputs > 1) {
- ret = request_samples(ctx, wanted_samples);
- if (ret < 0)
- return ret;
+ for (i = 0; i < s->nb_inputs; i++) {
+ if (!s->input_frames[i]) {
+ ret = ff_request_frame(ctx->inputs[i]);
+ if (ret == AVERROR_EOF) {
+ s->input_state[i] = INPUT_OFF;
+ continue;
+ } else if (ret < 0)
+ return ret;
+ }
- ret = calc_active_inputs(s);
- if (ret < 0)
- return ret;
+ if (!nb_samples) {
+ nb_samples = s->input_frames[i]->nb_samples;
+ for (j = i + 1; j < s->nb_inputs; j++)
+ ctx->inputs[j]->request_samples = nb_samples;
+ }
}
- if (s->active_inputs > 1) {
- available_samples = get_available_samples(s);
- if (!available_samples)
- return AVERROR(EAGAIN);
- available_samples = FFMIN(available_samples, wanted_samples);
- } else {
- available_samples = wanted_samples;
- }
+ /* reset requested samples count */
+ for (i = 0; i < s->nb_inputs; i++)
+ ctx->inputs[i]->request_samples = 0;
- s->next_pts = frame_list_next_pts(s->frame_list);
- frame_list_remove_samples(s->frame_list, available_samples);
+ ret = calc_active_inputs(s);
+ if (ret < 0)
+ return ret;
- return output_frame(outlink, available_samples);
+ return output_frame(outlink, nb_samples);
}
static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
{
AVFilterContext *ctx = inlink->dst;
MixContext *s = ctx->priv;
- AVFilterLink *outlink = ctx->outputs[0];
int i, ret = 0;
for (i = 0; i < ctx->nb_inputs; i++)
@@ -471,16 +279,10 @@ static int filter_frame(AVFilterLink *inlink, AVFrame
*buf)
goto fail;
}
- if (i == 0) {
- int64_t pts = av_rescale_q(buf->pts, inlink->time_base,
- outlink->time_base);
- ret = frame_list_add_frame(s->frame_list, buf->nb_samples, pts);
- if (ret < 0)
- goto fail;
- }
+ av_assert0(!s->input_frames[i]);
+ s->input_frames[i] = buf;
- ret = av_audio_fifo_write(s->fifos[i], (void **)buf->extended_data,
- buf->nb_samples);
+ return 0;
fail:
av_frame_free(&buf);
@@ -501,12 +303,17 @@ static int init(AVFilterContext *ctx)
pad.type = AVMEDIA_TYPE_AUDIO;
pad.name = av_strdup(name);
pad.filter_frame = filter_frame;
+ pad.needs_fifo = 1;
ff_insert_inpad(ctx, i, &pad);
}
avpriv_float_dsp_init(&s->fdsp, 0);
+ s->input_frames = av_mallocz(sizeof(*s->input_frames) * s->nb_inputs);
+ if (!s->input_frames)
+ return AVERROR(ENOMEM);
+
return 0;
}
@@ -515,13 +322,9 @@ 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);
+ for (i = 0; i < s->nb_inputs; i++)
+ av_frame_free(&s->input_frames[i]);
+ av_freep(&s->input_frames);
av_freep(&s->input_state);
av_freep(&s->input_scale);
--
1.7.10.4
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