diff options
Diffstat (limited to 'libavfilter')
29 files changed, 210 insertions, 113 deletions
diff --git a/libavfilter/af_aconvert.c b/libavfilter/af_aconvert.c index 51167f4327..dc9cf455a3 100644 --- a/libavfilter/af_aconvert.c +++ b/libavfilter/af_aconvert.c @@ -135,12 +135,13 @@ static int config_output(AVFilterLink *outlink) return 0; } -static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref) +static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref) { AConvertContext *aconvert = inlink->dst->priv; const int n = insamplesref->audio->nb_samples; AVFilterLink *const outlink = inlink->dst->outputs[0]; AVFilterBufferRef *outsamplesref = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n); + int ret; swr_convert(aconvert->swr, outsamplesref->data, n, (void *)insamplesref->data, n); @@ -148,8 +149,9 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref avfilter_copy_buffer_ref_props(outsamplesref, insamplesref); outsamplesref->audio->channel_layout = outlink->channel_layout; - ff_filter_samples(outlink, outsamplesref); + ret = ff_filter_samples(outlink, outsamplesref); avfilter_unref_buffer(insamplesref); + return ret; } AVFilter avfilter_af_aconvert = { diff --git a/libavfilter/af_amerge.c b/libavfilter/af_amerge.c index 802188d028..660ae5dc9d 100644 --- a/libavfilter/af_amerge.c +++ b/libavfilter/af_amerge.c @@ -212,7 +212,7 @@ static inline void copy_samples(int nb_inputs, struct amerge_input in[], } } -static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples) +static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples) { AVFilterContext *ctx = inlink->dst; AMergeContext *am = ctx->priv; @@ -232,7 +232,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples) for (i = 1; i < am->nb_inputs; i++) nb_samples = FFMIN(nb_samples, am->in[i].nb_samples); if (!nb_samples) - return; + return 0; outbuf = ff_get_audio_buffer(ctx->outputs[0], AV_PERM_WRITE, nb_samples); outs = outbuf->data[0]; @@ -285,7 +285,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples) } } } - ff_filter_samples(ctx->outputs[0], outbuf); + return ff_filter_samples(ctx->outputs[0], outbuf); } static av_cold int init(AVFilterContext *ctx, const char *args) diff --git a/libavfilter/af_amix.c b/libavfilter/af_amix.c index 6dad3db0d0..7f83750fa1 100644 --- a/libavfilter/af_amix.c +++ b/libavfilter/af_amix.c @@ -305,9 +305,7 @@ static int output_frame(AVFilterLink *outlink, int nb_samples) if (s->next_pts != AV_NOPTS_VALUE) s->next_pts += nb_samples; - ff_filter_samples(outlink, out_buf); - - return 0; + return ff_filter_samples(outlink, out_buf); } /** @@ -448,31 +446,37 @@ static int request_frame(AVFilterLink *outlink) return output_frame(outlink, available_samples); } -static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf) +static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf) { AVFilterContext *ctx = inlink->dst; MixContext *s = ctx->priv; AVFilterLink *outlink = ctx->outputs[0]; - int i; + int i, ret = 0; for (i = 0; i < ctx->nb_inputs; i++) if (ctx->inputs[i] == inlink) break; if (i >= ctx->nb_inputs) { av_log(ctx, AV_LOG_ERROR, "unknown input link\n"); - return; + ret = AVERROR(EINVAL); + goto fail; } 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); + ret = frame_list_add_frame(s->frame_list, buf->audio->nb_samples, pts); + if (ret < 0) + goto fail; } - av_audio_fifo_write(s->fifos[i], (void **)buf->extended_data, - buf->audio->nb_samples); + ret = av_audio_fifo_write(s->fifos[i], (void **)buf->extended_data, + buf->audio->nb_samples); +fail: avfilter_unref_buffer(buf); + + return ret; } static int init(AVFilterContext *ctx, const char *args) diff --git a/libavfilter/af_aresample.c b/libavfilter/af_aresample.c index 095a2b50e1..cf1d8df0ce 100644 --- a/libavfilter/af_aresample.c +++ b/libavfilter/af_aresample.c @@ -168,13 +168,14 @@ static int config_output(AVFilterLink *outlink) return 0; } -static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref) +static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref) { AResampleContext *aresample = inlink->dst->priv; const int n_in = insamplesref->audio->nb_samples; int n_out = n_in * aresample->ratio * 2 ; AVFilterLink *const outlink = inlink->dst->outputs[0]; AVFilterBufferRef *outsamplesref = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n_out); + int ret; avfilter_copy_buffer_ref_props(outsamplesref, insamplesref); @@ -193,15 +194,16 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref if (n_out <= 0) { avfilter_unref_buffer(outsamplesref); avfilter_unref_buffer(insamplesref); - return; + return 0; } outsamplesref->audio->sample_rate = outlink->sample_rate; outsamplesref->audio->nb_samples = n_out; - ff_filter_samples(outlink, outsamplesref); + ret = ff_filter_samples(outlink, outsamplesref); aresample->req_fullfilled= 1; avfilter_unref_buffer(insamplesref); + return ret; } static int request_frame(AVFilterLink *outlink) diff --git a/libavfilter/af_asetnsamples.c b/libavfilter/af_asetnsamples.c index 7a6c381853..95fd507d4d 100644 --- a/libavfilter/af_asetnsamples.c +++ b/libavfilter/af_asetnsamples.c @@ -131,7 +131,7 @@ static int push_samples(AVFilterLink *outlink) return nb_out_samples; } -static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples) +static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples) { AVFilterContext *ctx = inlink->dst; ASNSContext *asns = ctx->priv; @@ -145,7 +145,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples) if (ret < 0) { av_log(ctx, AV_LOG_ERROR, "Stretching audio fifo failed, discarded %d samples\n", nb_samples); - return; + return -1; } } av_audio_fifo_write(asns->fifo, (void **)insamples->extended_data, nb_samples); @@ -155,6 +155,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples) if (av_audio_fifo_size(asns->fifo) >= asns->nb_out_samples) push_samples(outlink); + return 0; } static int request_frame(AVFilterLink *outlink) diff --git a/libavfilter/af_ashowinfo.c b/libavfilter/af_ashowinfo.c index d774ec72a1..0d4bbb23f6 100644 --- a/libavfilter/af_ashowinfo.c +++ b/libavfilter/af_ashowinfo.c @@ -40,7 +40,7 @@ static av_cold int init(AVFilterContext *ctx, const char *args) return 0; } -static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *samplesref) +static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *samplesref) { AVFilterContext *ctx = inlink->dst; ShowInfoContext *showinfo = ctx->priv; @@ -83,7 +83,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *samplesref) av_log(ctx, AV_LOG_INFO, "]\n"); showinfo->frame++; - ff_filter_samples(inlink->dst->outputs[0], samplesref); + return ff_filter_samples(inlink->dst->outputs[0], samplesref); } AVFilter avfilter_af_ashowinfo = { diff --git a/libavfilter/af_astreamsync.c b/libavfilter/af_astreamsync.c index 8cf3f39b52..587d9a7662 100644 --- a/libavfilter/af_astreamsync.c +++ b/libavfilter/af_astreamsync.c @@ -107,11 +107,12 @@ static int config_output(AVFilterLink *outlink) return 0; } -static void send_out(AVFilterContext *ctx, int out_id) +static int send_out(AVFilterContext *ctx, int out_id) { AStreamSyncContext *as = ctx->priv; struct buf_queue *queue = &as->queue[out_id]; AVFilterBufferRef *buf = queue->buf[queue->tail]; + int ret; queue->buf[queue->tail] = NULL; as->var_values[VAR_B1 + out_id]++; @@ -121,11 +122,12 @@ static void send_out(AVFilterContext *ctx, int out_id) av_q2d(ctx->outputs[out_id]->time_base) * buf->pts; as->var_values[VAR_T1 + out_id] += buf->audio->nb_samples / (double)ctx->inputs[out_id]->sample_rate; - ff_filter_samples(ctx->outputs[out_id], buf); + ret = ff_filter_samples(ctx->outputs[out_id], buf); queue->nb--; queue->tail = (queue->tail + 1) % QUEUE_SIZE; if (as->req[out_id]) as->req[out_id]--; + return ret; } static void send_next(AVFilterContext *ctx) @@ -165,7 +167,7 @@ static int request_frame(AVFilterLink *outlink) return 0; } -static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples) +static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples) { AVFilterContext *ctx = inlink->dst; AStreamSyncContext *as = ctx->priv; @@ -175,6 +177,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples) insamples; as->eof &= ~(1 << id); send_next(ctx); + return 0; } AVFilter avfilter_af_astreamsync = { diff --git a/libavfilter/af_asyncts.c b/libavfilter/af_asyncts.c index 2435cca26e..7459610af8 100644 --- a/libavfilter/af_asyncts.c +++ b/libavfilter/af_asyncts.c @@ -37,6 +37,9 @@ typedef struct ASyncContext { int resample; float min_delta_sec; int max_comp; + + /* set by filter_samples() to signal an output frame to request_frame() */ + int got_output; } ASyncContext; #define OFFSET(x) offsetof(ASyncContext, x) @@ -112,9 +115,13 @@ static int request_frame(AVFilterLink *link) { AVFilterContext *ctx = link->src; ASyncContext *s = ctx->priv; - int ret = ff_request_frame(ctx->inputs[0]); + int ret = 0; int nb_samples; + s->got_output = 0; + while (ret >= 0 && !s->got_output) + ret = ff_request_frame(ctx->inputs[0]); + /* flush the fifo */ if (ret == AVERROR_EOF && (nb_samples = avresample_get_delay(s->avr))) { AVFilterBufferRef *buf = ff_get_audio_buffer(link, AV_PERM_WRITE, @@ -124,18 +131,18 @@ static int request_frame(AVFilterLink *link) avresample_convert(s->avr, (void**)buf->extended_data, buf->linesize[0], nb_samples, NULL, 0, 0); buf->pts = s->pts; - ff_filter_samples(link, buf); - return 0; + return ff_filter_samples(link, buf); } return ret; } -static void write_to_fifo(ASyncContext *s, AVFilterBufferRef *buf) +static int write_to_fifo(ASyncContext *s, AVFilterBufferRef *buf) { - avresample_convert(s->avr, NULL, 0, 0, (void**)buf->extended_data, - buf->linesize[0], buf->audio->nb_samples); + int ret = avresample_convert(s->avr, NULL, 0, 0, (void**)buf->extended_data, + buf->linesize[0], buf->audio->nb_samples); avfilter_unref_buffer(buf); + return ret; } /* get amount of data currently buffered, in samples */ @@ -144,7 +151,7 @@ static int64_t get_delay(ASyncContext *s) return avresample_available(s->avr) + avresample_get_delay(s->avr); } -static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf) +static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf) { AVFilterContext *ctx = inlink->dst; ASyncContext *s = ctx->priv; @@ -152,7 +159,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf) int nb_channels = av_get_channel_layout_nb_channels(buf->audio->channel_layout); int64_t pts = (buf->pts == AV_NOPTS_VALUE) ? buf->pts : av_rescale_q(buf->pts, inlink->time_base, outlink->time_base); - int out_size; + int out_size, ret; int64_t delta; /* buffer data until we get the first timestamp */ @@ -160,14 +167,12 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf) if (pts != AV_NOPTS_VALUE) { s->pts = pts - get_delay(s); } - write_to_fifo(s, buf); - return; + return write_to_fifo(s, buf); } /* now wait for the next timestamp */ if (pts == AV_NOPTS_VALUE) { - write_to_fifo(s, buf); - return; + return write_to_fifo(s, buf); } /* when we have two timestamps, compute how many samples would we have @@ -190,8 +195,10 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf) if (out_size > 0) { AVFilterBufferRef *buf_out = ff_get_audio_buffer(outlink, AV_PERM_WRITE, out_size); - if (!buf_out) - return; + if (!buf_out) { + ret = AVERROR(ENOMEM); + goto fail; + } avresample_read(s->avr, (void**)buf_out->extended_data, out_size); buf_out->pts = s->pts; @@ -200,7 +207,10 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf) av_samples_set_silence(buf_out->extended_data, out_size - delta, delta, nb_channels, buf->format); } - ff_filter_samples(outlink, buf_out); + ret = ff_filter_samples(outlink, buf_out); + if (ret < 0) + goto fail; + s->got_output = 1; } else { av_log(ctx, AV_LOG_WARNING, "Non-monotonous timestamps, dropping " "whole buffer.\n"); @@ -210,9 +220,13 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf) avresample_read(s->avr, NULL, avresample_available(s->avr)); s->pts = pts - avresample_get_delay(s->avr); - avresample_convert(s->avr, NULL, 0, 0, (void**)buf->extended_data, - buf->linesize[0], buf->audio->nb_samples); + ret = avresample_convert(s->avr, NULL, 0, 0, (void**)buf->extended_data, + buf->linesize[0], buf->audio->nb_samples); + +fail: avfilter_unref_buffer(buf); + + return ret; } AVFilter avfilter_af_asyncts = { diff --git a/libavfilter/af_atempo.c b/libavfilter/af_atempo.c index 7a08503906..959cacb6ad 100644 --- a/libavfilter/af_atempo.c +++ b/libavfilter/af_atempo.c @@ -1040,7 +1040,7 @@ static void push_samples(ATempoContext *atempo, atempo->nsamples_out += n_out; } -static void filter_samples(AVFilterLink *inlink, +static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *src_buffer) { AVFilterContext *ctx = inlink->dst; @@ -1074,6 +1074,7 @@ static void filter_samples(AVFilterLink *inlink, atempo->nsamples_in += n_in; avfilter_unref_bufferp(&src_buffer); + return 0; } static int request_frame(AVFilterLink *outlink) diff --git a/libavfilter/af_channelmap.c b/libavfilter/af_channelmap.c index 8d908ca737..1c1837c3d4 100644 --- a/libavfilter/af_channelmap.c +++ b/libavfilter/af_channelmap.c @@ -313,7 +313,7 @@ static int channelmap_query_formats(AVFilterContext *ctx) return 0; } -static void channelmap_filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf) +static int channelmap_filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf) { AVFilterContext *ctx = inlink->dst; AVFilterLink *outlink = ctx->outputs[0]; @@ -330,8 +330,10 @@ static void channelmap_filter_samples(AVFilterLink *inlink, AVFilterBufferRef *b if (nch_out > FF_ARRAY_ELEMS(buf->data)) { uint8_t **new_extended_data = av_mallocz(nch_out * sizeof(*buf->extended_data)); - if (!new_extended_data) - return; + if (!new_extended_data) { + avfilter_unref_buffer(buf); + return AVERROR(ENOMEM); + } if (buf->extended_data == buf->data) { buf->extended_data = new_extended_data; } else { @@ -353,7 +355,7 @@ static void channelmap_filter_samples(AVFilterLink *inlink, AVFilterBufferRef *b memcpy(buf->data, buf->extended_data, FFMIN(FF_ARRAY_ELEMS(buf->data), nch_out) * sizeof(buf->data[0])); - ff_filter_samples(outlink, buf); + return ff_filter_samples(outlink, buf); } static int channelmap_config_input(AVFilterLink *inlink) diff --git a/libavfilter/af_channelsplit.c b/libavfilter/af_channelsplit.c index bf0b24dc5e..3db08045c2 100644 --- a/libavfilter/af_channelsplit.c +++ b/libavfilter/af_channelsplit.c @@ -105,24 +105,29 @@ static int query_formats(AVFilterContext *ctx) return 0; } -static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf) +static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf) { AVFilterContext *ctx = inlink->dst; - int i; + int i, ret = 0; for (i = 0; i < ctx->nb_outputs; i++) { AVFilterBufferRef *buf_out = avfilter_ref_buffer(buf, ~AV_PERM_WRITE); - if (!buf_out) - return; + if (!buf_out) { + ret = AVERROR(ENOMEM); + break; + } buf_out->data[0] = buf_out->extended_data[0] = buf_out->extended_data[i]; buf_out->audio->channel_layout = av_channel_layout_extract_channel(buf->audio->channel_layout, i); - ff_filter_samples(ctx->outputs[i], buf_out); + ret = ff_filter_samples(ctx->outputs[i], buf_out); + if (ret < 0) + break; } avfilter_unref_buffer(buf); + return ret; } AVFilter avfilter_af_channelsplit = { diff --git a/libavfilter/af_earwax.c b/libavfilter/af_earwax.c index d86b410a1e..7265c437d3 100644 --- a/libavfilter/af_earwax.c +++ b/libavfilter/af_earwax.c @@ -120,13 +120,15 @@ static inline int16_t *scalarproduct(const int16_t *in, const int16_t *endin, in return out; } -static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples) +static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples) { AVFilterLink *outlink = inlink->dst->outputs[0]; int16_t *taps, *endin, *in, *out; AVFilterBufferRef *outsamples = ff_get_audio_buffer(inlink, AV_PERM_WRITE, insamples->audio->nb_samples); + int ret; + avfilter_copy_buffer_ref_props(outsamples, insamples); taps = ((EarwaxContext *)inlink->dst->priv)->taps; @@ -144,8 +146,9 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples) // save part of input for next round memcpy(taps, endin, NUMTAPS * sizeof(*taps)); - ff_filter_samples(outlink, outsamples); + ret = ff_filter_samples(outlink, outsamples); avfilter_unref_buffer(insamples); + return ret; } AVFilter avfilter_af_earwax = { diff --git a/libavfilter/af_join.c b/libavfilter/af_join.c index e86c556f5b..9ed11a9991 100644 --- a/libavfilter/af_join.c +++ b/libavfilter/af_join.c @@ -92,7 +92,7 @@ static const AVClass join_class = { .version = LIBAVUTIL_VERSION_INT, }; -static void filter_samples(AVFilterLink *link, AVFilterBufferRef *buf) +static int filter_samples(AVFilterLink *link, AVFilterBufferRef *buf) { AVFilterContext *ctx = link->dst; JoinContext *s = ctx->priv; @@ -104,6 +104,8 @@ static void filter_samples(AVFilterLink *link, AVFilterBufferRef *buf) av_assert0(i < ctx->nb_inputs); av_assert0(!s->input_frames[i]); s->input_frames[i] = buf; + + return 0; } static int parse_maps(AVFilterContext *ctx) @@ -468,11 +470,11 @@ static int join_request_frame(AVFilterLink *outlink) priv->nb_in_buffers = ctx->nb_inputs; buf->buf->priv = priv; - ff_filter_samples(outlink, buf); + ret = ff_filter_samples(outlink, buf); memset(s->input_frames, 0, sizeof(*s->input_frames) * ctx->nb_inputs); - return 0; + return ret; fail: avfilter_unref_buffer(buf); diff --git a/libavfilter/af_pan.c b/libavfilter/af_pan.c index f451e0034c..bb96ad0511 100644 --- a/libavfilter/af_pan.c +++ b/libavfilter/af_pan.c @@ -343,8 +343,9 @@ static int config_props(AVFilterLink *link) return 0; } -static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples) +static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples) { + int ret; int n = insamples->audio->nb_samples; AVFilterLink *const outlink = inlink->dst->outputs[0]; AVFilterBufferRef *outsamples = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n); @@ -354,8 +355,9 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples) avfilter_copy_buffer_ref_props(outsamples, insamples); outsamples->audio->channel_layout = outlink->channel_layout; - ff_filter_samples(outlink, outsamples); + ret = ff_filter_samples(outlink, outsamples); avfilter_unref_buffer(insamples); + return ret; } static av_cold void uninit(AVFilterContext *ctx) diff --git a/libavfilter/af_resample.c b/libavfilter/af_resample.c index 8a02cfe976..1360c1ca49 100644 --- a/libavfilter/af_resample.c +++ b/libavfilter/af_resample.c @@ -38,6 +38,9 @@ typedef struct ResampleContext { AVAudioResampleContext *avr; int64_t next_pts; + + /* set by filter_samples() to signal an output frame to request_frame() */ + int got_output; } ResampleContext; static av_cold void uninit(AVFilterContext *ctx) @@ -102,12 +105,6 @@ static int config_output(AVFilterLink *outlink) av_opt_set_int(s->avr, "in_sample_rate", inlink ->sample_rate, 0); av_opt_set_int(s->avr, "out_sample_rate", outlink->sample_rate, 0); - /* if both the input and output formats are s16 or u8, use s16 as - the internal sample format */ - if (av_get_bytes_per_sample(inlink->format) <= 2 && - av_get_bytes_per_sample(outlink->format) <= 2) - av_opt_set_int(s->avr, "internal_sample_fmt", AV_SAMPLE_FMT_S16P, 0); - if ((ret = avresample_open(s->avr)) < 0) return ret; @@ -130,7 +127,11 @@ static int request_frame(AVFilterLink *outlink) { AVFilterContext *ctx = outlink->src; ResampleContext *s = ctx->priv; - int ret = ff_request_frame(ctx->inputs[0]); + int ret = 0; + + s->got_output = 0; + while (ret >= 0 && !s->got_output) + ret = ff_request_frame(ctx->inputs[0]); /* flush the lavr delay buffer */ if (ret == AVERROR_EOF && s->avr) { @@ -156,21 +157,21 @@ static int request_frame(AVFilterLink *outlink) } buf->pts = s->next_pts; - ff_filter_samples(outlink, buf); - return 0; + return ff_filter_samples(outlink, buf); } return ret; } -static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf) +static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf) { AVFilterContext *ctx = inlink->dst; ResampleContext *s = ctx->priv; AVFilterLink *outlink = ctx->outputs[0]; + int ret; if (s->avr) { AVFilterBufferRef *buf_out; - int delay, nb_samples, ret; + int delay, nb_samples; /* maximum possible samples lavr can output */ delay = avresample_get_delay(s->avr); @@ -179,10 +180,19 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf) AV_ROUND_UP); buf_out = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples); + if (!buf_out) { + ret = AVERROR(ENOMEM); + goto fail; + } + ret = avresample_convert(s->avr, (void**)buf_out->extended_data, buf_out->linesize[0], nb_samples, (void**)buf->extended_data, buf->linesize[0], buf->audio->nb_samples); + if (ret < 0) { + avfilter_unref_buffer(buf_out); + goto fail; + } av_assert0(!avresample_available(s->avr)); @@ -208,11 +218,18 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf) s->next_pts = buf_out->pts + buf_out->audio->nb_samples; - ff_filter_samples(outlink, buf_out); + ret = ff_filter_samples(outlink, buf_out); + s->got_output = 1; } + +fail: avfilter_unref_buffer(buf); - } else - ff_filter_samples(outlink, buf); + } else { + ret = ff_filter_samples(outlink, buf); + s->got_output = 1; + } + + return ret; } AVFilter avfilter_af_resample = { diff --git a/libavfilter/af_silencedetect.c b/libavfilter/af_silencedetect.c index 724a92362f..d3b125fc5b 100644 --- a/libavfilter/af_silencedetect.c +++ b/libavfilter/af_silencedetect.c @@ -78,7 +78,7 @@ static av_cold int init(AVFilterContext *ctx, const char *args) return 0; } -static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples) +static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples) { int i; SilenceDetectContext *silence = inlink->dst->priv; @@ -118,7 +118,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples) } } - ff_filter_samples(inlink->dst->outputs[0], insamples); + return ff_filter_samples(inlink->dst->outputs[0], insamples); } static int query_formats(AVFilterContext *ctx) diff --git a/libavfilter/af_volume.c b/libavfilter/af_volume.c index 11da2265c9..09302ee5d9 100644 --- a/libavfilter/af_volume.c +++ b/libavfilter/af_volume.c @@ -110,7 +110,7 @@ static int query_formats(AVFilterContext *ctx) return 0; } -static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples) +static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples) { VolumeContext *vol = inlink->dst->priv; AVFilterLink *outlink = inlink->dst->outputs[0]; @@ -169,7 +169,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples) } } } - ff_filter_samples(outlink, insamples); + return ff_filter_samples(outlink, insamples); } AVFilter avfilter_af_volume = { diff --git a/libavfilter/asink_anullsink.c b/libavfilter/asink_anullsink.c index 4349544b62..d9e3e5a0cd 100644 --- a/libavfilter/asink_anullsink.c +++ b/libavfilter/asink_anullsink.c @@ -21,7 +21,10 @@ #include "avfilter.h" #include "internal.h" -static void null_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref) { } +static int null_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref) +{ + return 0; +} AVFilter avfilter_asink_anullsink = { .name = "anullsink", diff --git a/libavfilter/audio.c b/libavfilter/audio.c index 0ebec3c2d0..f3eebbfdae 100644 --- a/libavfilter/audio.c +++ b/libavfilter/audio.c @@ -150,19 +150,19 @@ fail: return NULL; } -static void default_filter_samples(AVFilterLink *link, - AVFilterBufferRef *samplesref) +static int default_filter_samples(AVFilterLink *link, + AVFilterBufferRef *samplesref) { - ff_filter_samples(link->dst->outputs[0], samplesref); + return ff_filter_samples(link->dst->outputs[0], samplesref); } -void ff_filter_samples_framed(AVFilterLink *link, - AVFilterBufferRef *samplesref) +int ff_filter_samples_framed(AVFilterLink *link, AVFilterBufferRef *samplesref) { - void (*filter_samples)(AVFilterLink *, AVFilterBufferRef *); + int (*filter_samples)(AVFilterLink *, AVFilterBufferRef *); AVFilterPad *dst = link->dstpad; int64_t pts; AVFilterBufferRef *buf_out; + int ret; FF_TPRINTF_START(NULL, filter_samples); ff_tlog_link(NULL, link, 1); @@ -193,21 +193,22 @@ void ff_filter_samples_framed(AVFilterLink *link, link->cur_buf = buf_out; pts = buf_out->pts; - filter_samples(link, buf_out); + ret = filter_samples(link, buf_out); ff_update_link_current_pts(link, pts); + return ret; } -void ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref) +int ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref) { int insamples = samplesref->audio->nb_samples, inpos = 0, nb_samples; AVFilterBufferRef *pbuf = link->partial_buf; int nb_channels = av_get_channel_layout_nb_channels(link->channel_layout); + int ret = 0; if (!link->min_samples || (!pbuf && insamples >= link->min_samples && insamples <= link->max_samples)) { - ff_filter_samples_framed(link, samplesref); - return; + return ff_filter_samples_framed(link, samplesref); } /* Handle framing (min_samples, max_samples) */ while (insamples) { @@ -218,7 +219,7 @@ void ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref) if (!pbuf) { av_log(link->dst, AV_LOG_WARNING, "Samples dropped due to memory allocation failure.\n"); - return; + return 0; } avfilter_copy_buffer_ref_props(pbuf, samplesref); pbuf->pts = samplesref->pts + @@ -234,10 +235,11 @@ void ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref) insamples -= nb_samples; pbuf->audio->nb_samples += nb_samples; if (pbuf->audio->nb_samples >= link->min_samples) { - ff_filter_samples_framed(link, pbuf); + ret = ff_filter_samples_framed(link, pbuf); pbuf = NULL; } } avfilter_unref_buffer(samplesref); link->partial_buf = pbuf; + return ret; } diff --git a/libavfilter/audio.h b/libavfilter/audio.h index cab1a6c722..a84c378ec8 100644 --- a/libavfilter/audio.h +++ b/libavfilter/audio.h @@ -70,14 +70,17 @@ AVFilterBufferRef *ff_get_audio_buffer(AVFilterLink *link, int perms, * @param samplesref a reference to the buffer of audio samples being sent. The * receiving filter will free this reference when it no longer * needs it or pass it on to the next filter. + * + * @return >= 0 on success, a negative AVERROR on error. The receiving filter + * is responsible for unreferencing samplesref in case of error. */ -void ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref); +int ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref); /** * Send a buffer of audio samples to the next link, without checking * min_samples. */ -void ff_filter_samples_framed(AVFilterLink *link, +int ff_filter_samples_framed(AVFilterLink *link, AVFilterBufferRef *samplesref); #endif /* AVFILTER_AUDIO_H */ diff --git a/libavfilter/avf_showwaves.c b/libavfilter/avf_showwaves.c index f0ebbf3c84..9a267a6b71 100644 --- a/libavfilter/avf_showwaves.c +++ b/libavfilter/avf_showwaves.c @@ -180,7 +180,7 @@ static int request_frame(AVFilterLink *outlink) #define MAX_INT16 ((1<<15) -1) -static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples) +static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples) { AVFilterContext *ctx = inlink->dst; AVFilterLink *outlink = ctx->outputs[0]; @@ -225,6 +225,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples) } avfilter_unref_buffer(insamples); + return 0; } AVFilter avfilter_avf_showwaves = { diff --git a/libavfilter/avfilter.h b/libavfilter/avfilter.h index e08a389275..52de7405e3 100644 --- a/libavfilter/avfilter.h +++ b/libavfilter/avfilter.h @@ -301,8 +301,12 @@ struct AVFilterPad { * and should do its processing. * * Input audio pads only. + * + * @return >= 0 on success, a negative AVERROR on error. This function + * must ensure that samplesref is properly unreferenced on error if it + * hasn't been passed on to another filter. */ - void (*filter_samples)(AVFilterLink *link, AVFilterBufferRef *samplesref); + int (*filter_samples)(AVFilterLink *link, AVFilterBufferRef *samplesref); /** * Frame poll callback. This returns the number of immediately available diff --git a/libavfilter/buffersink.c b/libavfilter/buffersink.c index 642350080b..9e908adf6b 100644 --- a/libavfilter/buffersink.c +++ b/libavfilter/buffersink.c @@ -56,6 +56,12 @@ static void start_frame(AVFilterLink *link, AVFilterBufferRef *buf) link->cur_buf = NULL; }; +static int filter_samples(AVFilterLink *link, AVFilterBufferRef *buf) +{ + start_frame(link, buf); + return 0; +} + int av_buffersink_read(AVFilterContext *ctx, AVFilterBufferRef **buf) { BufferSinkContext *s = ctx->priv; @@ -160,7 +166,7 @@ AVFilter avfilter_asink_abuffer = { .inputs = (AVFilterPad[]) {{ .name = "default", .type = AVMEDIA_TYPE_AUDIO, - .filter_samples = start_frame, + .filter_samples = filter_samples, .min_perms = AV_PERM_READ, .needs_fifo = 1 }, { .name = NULL }}, diff --git a/libavfilter/buffersrc.c b/libavfilter/buffersrc.c index dd9eb39b59..2592cfb64a 100644 --- a/libavfilter/buffersrc.c +++ b/libavfilter/buffersrc.c @@ -408,6 +408,7 @@ static int request_frame(AVFilterLink *link) { BufferSourceContext *c = link->src->priv; AVFilterBufferRef *buf; + int ret = 0; if (!av_fifo_size(c->fifo)) { if (c->eof) @@ -424,7 +425,7 @@ static int request_frame(AVFilterLink *link) ff_end_frame(link); break; case AVMEDIA_TYPE_AUDIO: - ff_filter_samples(link, avfilter_ref_buffer(buf, ~0)); + ret = ff_filter_samples(link, avfilter_ref_buffer(buf, ~0)); break; default: return AVERROR(EINVAL); @@ -432,7 +433,7 @@ static int request_frame(AVFilterLink *link) avfilter_unref_buffer(buf); - return 0; + return ret; } static int poll_frame(AVFilterLink *link) diff --git a/libavfilter/f_settb.c b/libavfilter/f_settb.c index 6549a5c26c..3ba35be70e 100644 --- a/libavfilter/f_settb.c +++ b/libavfilter/f_settb.c @@ -117,7 +117,7 @@ static void start_frame(AVFilterLink *inlink, AVFilterBufferRef *picref) ff_start_frame(outlink, picref2); } -static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples) +static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples) { AVFilterContext *ctx = inlink->dst; AVFilterLink *outlink = ctx->outputs[0]; @@ -132,7 +132,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples) avfilter_unref_buffer(insamples); } - ff_filter_samples(outlink, outsamples); + return ff_filter_samples(outlink, outsamples); } #if CONFIG_SETTB_FILTER diff --git a/libavfilter/fifo.c b/libavfilter/fifo.c index bc9c8fa580..34db5ecbee 100644 --- a/libavfilter/fifo.c +++ b/libavfilter/fifo.c @@ -72,13 +72,25 @@ static av_cold void uninit(AVFilterContext *ctx) avfilter_unref_buffer(fifo->buf_out); } -static void add_to_queue(AVFilterLink *inlink, AVFilterBufferRef *buf) +static int add_to_queue(AVFilterLink *inlink, AVFilterBufferRef *buf) { FifoContext *fifo = inlink->dst->priv; fifo->last->next = av_mallocz(sizeof(Buf)); + if (!fifo->last->next) { + avfilter_unref_buffer(buf); + return AVERROR(ENOMEM); + } + fifo->last = fifo->last->next; fifo->last->buf = buf; + + return 0; +} + +static void start_frame(AVFilterLink *inlink, AVFilterBufferRef *buf) +{ + add_to_queue(inlink, buf); } static void queue_pop(FifoContext *s) @@ -210,15 +222,13 @@ static int return_audio_frame(AVFilterContext *ctx) buf_out = s->buf_out; s->buf_out = NULL; } - ff_filter_samples(link, buf_out); - - return 0; + return ff_filter_samples(link, buf_out); } static int request_frame(AVFilterLink *outlink) { FifoContext *fifo = outlink->src->priv; - int ret; + int ret = 0; if (!fifo->root.next) { if ((ret = ff_request_frame(outlink->src->inputs[0])) < 0) @@ -238,7 +248,7 @@ static int request_frame(AVFilterLink *outlink) if (outlink->request_samples) { return return_audio_frame(outlink->src); } else { - ff_filter_samples(outlink, fifo->root.next->buf); + ret = ff_filter_samples(outlink, fifo->root.next->buf); queue_pop(fifo); } break; @@ -246,7 +256,7 @@ static int request_frame(AVFilterLink *outlink) return AVERROR(EINVAL); } - return 0; + return ret; } AVFilter avfilter_vf_fifo = { @@ -261,7 +271,7 @@ AVFilter avfilter_vf_fifo = { .inputs = (const AVFilterPad[]) {{ .name = "default", .type = AVMEDIA_TYPE_VIDEO, .get_video_buffer= ff_null_get_video_buffer, - .start_frame = add_to_queue, + .start_frame = start_frame, .draw_slice = draw_slice, .end_frame = end_frame, .rej_perms = AV_PERM_REUSE2, }, diff --git a/libavfilter/internal.h b/libavfilter/internal.h index 40ffef5721..d1bcb0353c 100644 --- a/libavfilter/internal.h +++ b/libavfilter/internal.h @@ -135,8 +135,12 @@ struct AVFilterPad { * and should do its processing. * * Input audio pads only. + * + * @return >= 0 on success, a negative AVERROR on error. This function + * must ensure that samplesref is properly unreferenced on error if it + * hasn't been passed on to another filter. */ - void (*filter_samples)(AVFilterLink *link, AVFilterBufferRef *samplesref); + int (*filter_samples)(AVFilterLink *link, AVFilterBufferRef *samplesref); /** * Frame poll callback. This returns the number of immediately available diff --git a/libavfilter/sink_buffer.c b/libavfilter/sink_buffer.c index 8275f80965..ceae11203d 100644 --- a/libavfilter/sink_buffer.c +++ b/libavfilter/sink_buffer.c @@ -244,9 +244,10 @@ AVFilter avfilter_vsink_buffersink = { #if CONFIG_ABUFFERSINK_FILTER -static void filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref) +static int filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref) { end_frame(link); + return 0; } static av_cold int asink_init(AVFilterContext *ctx, const char *args) diff --git a/libavfilter/split.c b/libavfilter/split.c index 837dc0da15..98be342bfc 100644 --- a/libavfilter/split.c +++ b/libavfilter/split.c @@ -110,15 +110,19 @@ AVFilter avfilter_vf_split = { .outputs = (AVFilterPad[]) {{ .name = NULL}}, }; -static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *samplesref) +static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *samplesref) { AVFilterContext *ctx = inlink->dst; - int i; + int i, ret = 0; - for (i = 0; i < ctx->nb_outputs; i++) - ff_filter_samples(inlink->dst->outputs[i], - avfilter_ref_buffer(samplesref, ~AV_PERM_WRITE)); + for (i = 0; i < ctx->nb_outputs; i++) { + ret = ff_filter_samples(inlink->dst->outputs[i], + avfilter_ref_buffer(samplesref, ~AV_PERM_WRITE)); + if (ret < 0) + break; + } avfilter_unref_buffer(samplesref); + return ret; } AVFilter avfilter_af_asplit = { |