February 2022 - ffmpeg-cvslog

avfilter/af_dynaudnorm: reduce number of lines by using for (int ...
by Paul B Mahol 28 Feb '22

28 Feb '22

ffmpeg | branch: master | Paul B Mahol <onemda(a)gmail.com> | Mon Feb 28 21:54:28 2022 +0100| [178d8036dc4bb5f20b58e6a02da3d596b2f6f615] | committer: Paul B Mahol avfilter/af_dynaudnorm: reduce number of lines by using for (int ... > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=178d8036dc4bb5f20… --- libavfilter/af_dynaudnorm.c | 73 +++++++++++++++++---------------------------- 1 file changed, 28 insertions(+), 45 deletions(-) diff --git a/libavfilter/af_dynaudnorm.c b/libavfilter/af_dynaudnorm.c index 19f3b528d9..837126ac0d 100644 --- a/libavfilter/af_dynaudnorm.c +++ b/libavfilter/af_dynaudnorm.c @@ -248,7 +248,6 @@ static void init_gaussian_filter(DynamicAudioNormalizerContext *s) double total_weight = 0.0; const double sigma = (((s->filter_size / 2.0) - 1.0) / 3.0) + (1.0 / 3.0); double adjust; - int i; // Pre-compute constants const int offset = s->filter_size / 2; @@ -256,7 +255,7 @@ static void init_gaussian_filter(DynamicAudioNormalizerContext *s) const double c2 = 2.0 * sigma * sigma; // Compute weights - for (i = 0; i < s->filter_size; i++) { + for (int i = 0; i < s->filter_size; i++) { const int x = i - offset; s->weights[i] = c1 * exp(-x * x / c2); @@ -265,7 +264,7 @@ static void init_gaussian_filter(DynamicAudioNormalizerContext *s) // Adjust weights adjust = 1.0 / total_weight; - for (i = 0; i < s->filter_size; i++) { + for (int i = 0; i < s->filter_size; i++) { s->weights[i] *= adjust; } } @@ -273,13 +272,12 @@ static void init_gaussian_filter(DynamicAudioNormalizerContext *s) static av_cold void uninit(AVFilterContext *ctx) { DynamicAudioNormalizerContext *s = ctx->priv; - int c; av_freep(&s->prev_amplification_factor); av_freep(&s->dc_correction_value); av_freep(&s->compress_threshold); - for (c = 0; c < s->channels; c++) { + for (int c = 0; c < s->channels; c++) { if (s->gain_history_original) cqueue_free(s->gain_history_original[c]); if (s->gain_history_minimum) @@ -309,7 +307,6 @@ static int config_input(AVFilterLink *inlink) { AVFilterContext *ctx = inlink->dst; DynamicAudioNormalizerContext *s = ctx->priv; - int c; uninit(ctx); @@ -333,7 +330,7 @@ static int config_input(AVFilterLink *inlink) !s->is_enabled || !s->weights) return AVERROR(ENOMEM); - for (c = 0; c < inlink->channels; c++) { + for (int c = 0; c < inlink->channels; c++) { s->prev_amplification_factor[c] = 1.0; s->gain_history_original[c] = cqueue_create(s->filter_size, MAX_FILTER_SIZE); @@ -378,19 +375,18 @@ static inline double bound(const double threshold, const double val) static double find_peak_magnitude(AVFrame *frame, int channel) { double max = DBL_EPSILON; - int c, i; if (channel == -1) { - for (c = 0; c < frame->channels; c++) { + for (int c = 0; c < frame->channels; c++) { double *data_ptr = (double *)frame->extended_data[c]; - for (i = 0; i < frame->nb_samples; i++) + for (int i = 0; i < frame->nb_samples; i++) max = fmax(max, fabs(data_ptr[i])); } } else { double *data_ptr = (double *)frame->extended_data[channel]; - for (i = 0; i < frame->nb_samples; i++) + for (int i = 0; i < frame->nb_samples; i++) max = fmax(max, fabs(data_ptr[i])); } @@ -400,13 +396,12 @@ static double find_peak_magnitude(AVFrame *frame, int channel) static double compute_frame_rms(AVFrame *frame, int channel) { double rms_value = 0.0; - int c, i; if (channel == -1) { - for (c = 0; c < frame->channels; c++) { + for (int c = 0; c < frame->channels; c++) { const double *data_ptr = (double *)frame->extended_data[c]; - for (i = 0; i < frame->nb_samples; i++) { + for (int i = 0; i < frame->nb_samples; i++) { rms_value += pow_2(data_ptr[i]); } } @@ -414,7 +409,7 @@ static double compute_frame_rms(AVFrame *frame, int channel) rms_value /= frame->nb_samples * frame->channels; } else { const double *data_ptr = (double *)frame->extended_data[channel]; - for (i = 0; i < frame->nb_samples; i++) { + for (int i = 0; i < frame->nb_samples; i++) { rms_value += pow_2(data_ptr[i]); } @@ -441,9 +436,8 @@ static local_gain get_max_local_gain(DynamicAudioNormalizerContext *s, AVFrame * static double minimum_filter(cqueue *q) { double min = DBL_MAX; - int i; - for (i = 0; i < cqueue_size(q); i++) { + for (int i = 0; i < cqueue_size(q); i++) { min = fmin(min, cqueue_peek(q, i)); } @@ -454,9 +448,8 @@ static double gaussian_filter(DynamicAudioNormalizerContext *s, cqueue *q, cqueu { const double *weights = s->weights; double result = 0.0, tsum = 0.0; - int i; - for (i = 0; i < cqueue_size(q); i++) { + for (int i = 0; i < cqueue_size(q); i++) { double tq_item = cqueue_peek(tq, i); double q_item = cqueue_peek(q, i); @@ -540,21 +533,20 @@ static void perform_dc_correction(DynamicAudioNormalizerContext *s, AVFrame *fra { const double diff = 1.0 / frame->nb_samples; int is_first_frame = cqueue_empty(s->gain_history_original[0]); - int c, i; - for (c = 0; c < s->channels; c++) { + for (int c = 0; c < s->channels; c++) { const int bypass = bypass_channel(s, frame, c); double *dst_ptr = (double *)frame->extended_data[c]; double current_average_value = 0.0; double prev_value; - for (i = 0; i < frame->nb_samples; i++) + for (int i = 0; i < frame->nb_samples; i++) current_average_value += dst_ptr[i] * diff; prev_value = is_first_frame ? current_average_value : s->dc_correction_value[c]; s->dc_correction_value[c] = is_first_frame ? current_average_value : update_value(current_average_value, s->dc_correction_value[c], 0.1); - for (i = 0; i < frame->nb_samples && !bypass; i++) { + for (int i = 0; i < frame->nb_samples && !bypass; i++) { dst_ptr[i] -= fade(prev_value, s->dc_correction_value[c], i, frame->nb_samples); } } @@ -586,13 +578,12 @@ static double compute_frame_std_dev(DynamicAudioNormalizerContext *s, AVFrame *frame, int channel) { double variance = 0.0; - int i, c; if (channel == -1) { - for (c = 0; c < s->channels; c++) { + for (int c = 0; c < s->channels; c++) { const double *data_ptr = (double *)frame->extended_data[c]; - for (i = 0; i < frame->nb_samples; i++) { + for (int i = 0; i < frame->nb_samples; i++) { variance += pow_2(data_ptr[i]); // Assume that MEAN is *zero* } } @@ -600,7 +591,7 @@ static double compute_frame_std_dev(DynamicAudioNormalizerContext *s, } else { const double *data_ptr = (double *)frame->extended_data[channel]; - for (i = 0; i < frame->nb_samples; i++) { + for (int i = 0; i < frame->nb_samples; i++) { variance += pow_2(data_ptr[i]); // Assume that MEAN is *zero* } variance /= frame->nb_samples - 1; @@ -612,7 +603,6 @@ static double compute_frame_std_dev(DynamicAudioNormalizerContext *s, static void perform_compression(DynamicAudioNormalizerContext *s, AVFrame *frame) { int is_first_frame = cqueue_empty(s->gain_history_original[0]); - int c, i; if (s->channels_coupled) { const double standard_deviation = compute_frame_std_dev(s, frame, -1); @@ -625,20 +615,20 @@ static void perform_compression(DynamicAudioNormalizerContext *s, AVFrame *frame prev_actual_thresh = setup_compress_thresh(prev_value); curr_actual_thresh = setup_compress_thresh(s->compress_threshold[0]); - for (c = 0; c < s->channels; c++) { + for (int c = 0; c < s->channels; c++) { double *const dst_ptr = (double *)frame->extended_data[c]; const int bypass = bypass_channel(s, frame, c); if (bypass) continue; - for (i = 0; i < frame->nb_samples; i++) { + for (int i = 0; i < frame->nb_samples; i++) { const double localThresh = fade(prev_actual_thresh, curr_actual_thresh, i, frame->nb_samples); dst_ptr[i] = copysign(bound(localThresh, fabs(dst_ptr[i])), dst_ptr[i]); } } } else { - for (c = 0; c < s->channels; c++) { + for (int c = 0; c < s->channels; c++) { const int bypass = bypass_channel(s, frame, c); const double standard_deviation = compute_frame_std_dev(s, frame, c); const double current_threshold = setup_compress_thresh(fmin(1.0, s->compress_factor * standard_deviation)); @@ -652,7 +642,7 @@ static void perform_compression(DynamicAudioNormalizerContext *s, AVFrame *frame curr_actual_thresh = setup_compress_thresh(s->compress_threshold[c]); dst_ptr = (double *)frame->extended_data[c]; - for (i = 0; i < frame->nb_samples && !bypass; i++) { + for (int i = 0; i < frame->nb_samples && !bypass; i++) { const double localThresh = fade(prev_actual_thresh, curr_actual_thresh, i, frame->nb_samples); dst_ptr[i] = copysign(bound(localThresh, fabs(dst_ptr[i])), dst_ptr[i]); } @@ -718,14 +708,10 @@ static int analyze_frame(DynamicAudioNormalizerContext *s, AVFilterLink *outlink if (s->channels_coupled) { const local_gain gain = get_max_local_gain(s, analyze_frame, -1); - int c; - - for (c = 0; c < s->channels; c++) + for (int c = 0; c < s->channels; c++) update_gain_history(s, c, gain); } else { - int c; - - for (c = 0; c < s->channels; c++) + for (int c = 0; c < s->channels; c++) update_gain_history(s, c, get_max_local_gain(s, analyze_frame, c)); } @@ -735,9 +721,7 @@ static int analyze_frame(DynamicAudioNormalizerContext *s, AVFilterLink *outlink static void amplify_frame(DynamicAudioNormalizerContext *s, AVFrame *in, AVFrame *frame, int enabled) { - int c, i; - - for (c = 0; c < s->channels; c++) { + for (int c = 0; c < s->channels; c++) { const int bypass = bypass_channel(s, frame, c); const double *src_ptr = (const double *)in->extended_data[c]; double *dst_ptr = (double *)frame->extended_data[c]; @@ -745,7 +729,7 @@ static void amplify_frame(DynamicAudioNormalizerContext *s, AVFrame *in, cqueue_dequeue(s->gain_history_smoothed[c], &current_amplification_factor); - for (i = 0; i < frame->nb_samples && enabled && !bypass; i++) { + for (int i = 0; i < frame->nb_samples && enabled && !bypass; i++) { const double amplification_factor = fade(s->prev_amplification_factor[c], current_amplification_factor, i, frame->nb_samples); @@ -811,15 +795,14 @@ static int flush_buffer(DynamicAudioNormalizerContext *s, AVFilterLink *inlink, AVFilterLink *outlink) { AVFrame *out = ff_get_audio_buffer(outlink, s->sample_advance); - int c, i; if (!out) return AVERROR(ENOMEM); - for (c = 0; c < s->channels; c++) { + for (int c = 0; c < s->channels; c++) { double *dst_ptr = (double *)out->extended_data[c]; - for (i = 0; i < out->nb_samples; i++) { + for (int i = 0; i < out->nb_samples; i++) { dst_ptr[i] = s->alt_boundary_mode ? DBL_EPSILON : ((s->target_rms > DBL_EPSILON) ? fmin(s->peak_value, s->target_rms) : s->peak_value); if (s->dc_correction) { dst_ptr[i] *= ((i % 2) == 1) ? -1 : 1;

1 0

avfilter/af_dynaudnorm: use fmin/fmax for doubles
by Paul B Mahol 28 Feb '22

28 Feb '22

ffmpeg | branch: master | Paul B Mahol <onemda(a)gmail.com> | Mon Feb 28 10:15:25 2022 +0100| [aa6b9066b9323b4af44eb723db141b5d4dda7c3a] | committer: Paul B Mahol avfilter/af_dynaudnorm: use fmin/fmax for doubles > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=aa6b9066b9323b4af… --- libavfilter/af_dynaudnorm.c | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) diff --git a/libavfilter/af_dynaudnorm.c b/libavfilter/af_dynaudnorm.c index 2adbcf3e10..19f3b528d9 100644 --- a/libavfilter/af_dynaudnorm.c +++ b/libavfilter/af_dynaudnorm.c @@ -385,13 +385,13 @@ static double find_peak_magnitude(AVFrame *frame, int channel) double *data_ptr = (double *)frame->extended_data[c]; for (i = 0; i < frame->nb_samples; i++) - max = FFMAX(max, fabs(data_ptr[i])); + max = fmax(max, fabs(data_ptr[i])); } } else { double *data_ptr = (double *)frame->extended_data[channel]; for (i = 0; i < frame->nb_samples; i++) - max = FFMAX(max, fabs(data_ptr[i])); + max = fmax(max, fabs(data_ptr[i])); } return max; @@ -421,7 +421,7 @@ static double compute_frame_rms(AVFrame *frame, int channel) rms_value /= frame->nb_samples; } - return FFMAX(sqrt(rms_value), DBL_EPSILON); + return fmax(sqrt(rms_value), DBL_EPSILON); } static local_gain get_max_local_gain(DynamicAudioNormalizerContext *s, AVFrame *frame, @@ -433,7 +433,7 @@ static local_gain get_max_local_gain(DynamicAudioNormalizerContext *s, AVFrame * local_gain gain; gain.threshold = peak_magnitude > s->threshold; - gain.max_gain = bound(s->max_amplification, FFMIN(maximum_gain, rms_gain)); + gain.max_gain = bound(s->max_amplification, fmin(maximum_gain, rms_gain)); return gain; } @@ -444,7 +444,7 @@ static double minimum_filter(cqueue *q) int i; for (i = 0; i < cqueue_size(q); i++) { - min = FFMIN(min, cqueue_peek(q, i)); + min = fmin(min, cqueue_peek(q, i)); } return min; @@ -475,7 +475,7 @@ static void update_gain_history(DynamicAudioNormalizerContext *s, int channel, { if (cqueue_empty(s->gain_history_original[channel])) { const int pre_fill_size = s->filter_size / 2; - const double initial_value = s->alt_boundary_mode ? gain.max_gain : FFMIN(1.0, gain.max_gain); + const double initial_value = s->alt_boundary_mode ? gain.max_gain : fmin(1.0, gain.max_gain); s->prev_amplification_factor[channel] = initial_value; @@ -497,7 +497,7 @@ static void update_gain_history(DynamicAudioNormalizerContext *s, int channel, while (cqueue_size(s->gain_history_minimum[channel]) < pre_fill_size) { input++; - initial_value = FFMIN(initial_value, cqueue_peek(s->gain_history_original[channel], input)); + initial_value = fmin(initial_value, cqueue_peek(s->gain_history_original[channel], input)); cqueue_enqueue(s->gain_history_minimum[channel], initial_value); } } @@ -516,7 +516,7 @@ static void update_gain_history(DynamicAudioNormalizerContext *s, int channel, smoothed = gaussian_filter(s, s->gain_history_minimum[channel], s->threshold_history[channel]); limit = cqueue_peek(s->gain_history_original[channel], 0); - smoothed = FFMIN(smoothed, limit); + smoothed = fmin(smoothed, limit); cqueue_enqueue(s->gain_history_smoothed[channel], smoothed); @@ -606,7 +606,7 @@ static double compute_frame_std_dev(DynamicAudioNormalizerContext *s, variance /= frame->nb_samples - 1; } - return FFMAX(sqrt(variance), DBL_EPSILON); + return fmax(sqrt(variance), DBL_EPSILON); } static void perform_compression(DynamicAudioNormalizerContext *s, AVFrame *frame) @@ -616,7 +616,7 @@ static void perform_compression(DynamicAudioNormalizerContext *s, AVFrame *frame if (s->channels_coupled) { const double standard_deviation = compute_frame_std_dev(s, frame, -1); - const double current_threshold = FFMIN(1.0, s->compress_factor * standard_deviation); + const double current_threshold = fmin(1.0, s->compress_factor * standard_deviation); const double prev_value = is_first_frame ? current_threshold : s->compress_threshold[0]; double prev_actual_thresh, curr_actual_thresh; @@ -641,7 +641,7 @@ static void perform_compression(DynamicAudioNormalizerContext *s, AVFrame *frame for (c = 0; c < s->channels; c++) { const int bypass = bypass_channel(s, frame, c); const double standard_deviation = compute_frame_std_dev(s, frame, c); - const double current_threshold = setup_compress_thresh(FFMIN(1.0, s->compress_factor * standard_deviation)); + const double current_threshold = setup_compress_thresh(fmin(1.0, s->compress_factor * standard_deviation)); const double prev_value = is_first_frame ? current_threshold : s->compress_threshold[c]; double prev_actual_thresh, curr_actual_thresh; double *dst_ptr; @@ -820,7 +820,7 @@ static int flush_buffer(DynamicAudioNormalizerContext *s, AVFilterLink *inlink, double *dst_ptr = (double *)out->extended_data[c]; for (i = 0; i < out->nb_samples; i++) { - dst_ptr[i] = s->alt_boundary_mode ? DBL_EPSILON : ((s->target_rms > DBL_EPSILON) ? FFMIN(s->peak_value, s->target_rms) : s->peak_value); + dst_ptr[i] = s->alt_boundary_mode ? DBL_EPSILON : ((s->target_rms > DBL_EPSILON) ? fmin(s->peak_value, s->target_rms) : s->peak_value); if (s->dc_correction) { dst_ptr[i] *= ((i % 2) == 1) ? -1 : 1; dst_ptr[i] += s->dc_correction_value[c];

1 0

avfilter/af_dynaudnorm: add support for overlapping frames
by Paul B Mahol 28 Feb '22

28 Feb '22

ffmpeg | branch: master | Paul B Mahol <onemda(a)gmail.com> | Mon Feb 28 09:19:42 2022 +0100| [456d48c752fb960508b373afa3f56d389c22a8e4] | committer: Paul B Mahol avfilter/af_dynaudnorm: add support for overlapping frames > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=456d48c752fb96050… --- doc/filters.texi | 7 +++++++ libavfilter/af_dynaudnorm.c | 44 +++++++++++++++++++++++++++++++++++++++----- 2 files changed, 46 insertions(+), 5 deletions(-) diff --git a/doc/filters.texi b/doc/filters.texi index b3ae139613..c3623dee89 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -4392,6 +4392,13 @@ This option is mostly useful if digital noise is not wanted to be amplified. @item channels, h Specify which channels to filter, by default all available channels are filtered. + +@item overlap, o +Specify overlap for frames. If set to 0 (default) no frame overlapping is done. +Using >0 and <1 values will make less conservative gain adjustments, like +when framelen option is set to smaller value, if framelen option value is +compensated for non-zero overlap then gain adjustments will be smoother across time +compared to zero overlap case. @end table @subsection Commands diff --git a/libavfilter/af_dynaudnorm.c b/libavfilter/af_dynaudnorm.c index 77899474bf..2adbcf3e10 100644 --- a/libavfilter/af_dynaudnorm.c +++ b/libavfilter/af_dynaudnorm.c @@ -64,6 +64,7 @@ typedef struct DynamicAudioNormalizerContext { int dc_correction; int channels_coupled; int alt_boundary_mode; + double overlap; double peak_value; double max_amplification; @@ -76,6 +77,7 @@ typedef struct DynamicAudioNormalizerContext { double *weights; int channels; + int sample_advance; int eof; uint64_t channels_to_filter; int64_t pts; @@ -86,6 +88,8 @@ typedef struct DynamicAudioNormalizerContext { cqueue **threshold_history; cqueue *is_enabled; + + AVFrame *window; } DynamicAudioNormalizerContext; #define OFFSET(x) offsetof(DynamicAudioNormalizerContext, x) @@ -114,6 +118,8 @@ static const AVOption dynaudnorm_options[] = { { "t", "set the threshold value", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl = 0.0}, 0.0, 1.0, FLAGS }, { "channels", "set channels to filter", OFFSET(channels_to_filter),AV_OPT_TYPE_CHANNEL_LAYOUT, {.i64=-1}, INT64_MIN, INT64_MAX, FLAGS }, { "h", "set channels to filter", OFFSET(channels_to_filter),AV_OPT_TYPE_CHANNEL_LAYOUT, {.i64=-1}, INT64_MIN, INT64_MAX, FLAGS }, + { "overlap", "set the frame overlap", OFFSET(overlap), AV_OPT_TYPE_DOUBLE, {.dbl=.0}, 0.0, 1.0, FLAGS }, + { "o", "set the frame overlap", OFFSET(overlap), AV_OPT_TYPE_DOUBLE, {.dbl=.0}, 0.0, 1.0, FLAGS }, { NULL } }; @@ -295,6 +301,8 @@ static av_cold void uninit(AVFilterContext *ctx) av_freep(&s->weights); ff_bufqueue_discard_all(&s->queue); + + av_frame_free(&s->window); } static int config_input(AVFilterLink *inlink) @@ -340,6 +348,11 @@ static int config_input(AVFilterLink *inlink) init_gaussian_filter(s); + s->window = ff_get_audio_buffer(ctx->outputs[0], s->frame_len * 2); + if (!s->window) + return AVERROR(ENOMEM); + s->sample_advance = FFMAX(1, lrint(s->frame_len * (1. - s->overlap))); + return 0; } @@ -649,6 +662,8 @@ static void perform_compression(DynamicAudioNormalizerContext *s, AVFrame *frame static int analyze_frame(DynamicAudioNormalizerContext *s, AVFilterLink *outlink, AVFrame **frame) { + AVFrame *analyze_frame; + if (s->dc_correction || s->compress_factor > DBL_EPSILON) { int ret; @@ -683,8 +698,26 @@ static int analyze_frame(DynamicAudioNormalizerContext *s, AVFilterLink *outlink if (s->compress_factor > DBL_EPSILON) perform_compression(s, *frame); + if (s->frame_len != s->sample_advance) { + const int offset = s->frame_len - s->sample_advance; + + for (int c = 0; c < s->channels; c++) { + double *src = (double *)s->window->extended_data[c]; + + memmove(src, &src[s->sample_advance], offset * sizeof(double)); + memcpy(&src[offset], (*frame)->extended_data[c], (*frame)->nb_samples * sizeof(double)); + memset(&src[offset + (*frame)->nb_samples], 0, (s->sample_advance - (*frame)->nb_samples) * sizeof(double)); + } + + analyze_frame = s->window; + } else { + av_samples_copy(s->window->extended_data, (*frame)->extended_data, 0, 0, + s->frame_len, (*frame)->channels, (*frame)->format); + analyze_frame = *frame; + } + if (s->channels_coupled) { - const local_gain gain = get_max_local_gain(s, *frame, -1); + const local_gain gain = get_max_local_gain(s, analyze_frame, -1); int c; for (c = 0; c < s->channels; c++) @@ -693,7 +726,7 @@ static int analyze_frame(DynamicAudioNormalizerContext *s, AVFilterLink *outlink int c; for (c = 0; c < s->channels; c++) - update_gain_history(s, c, get_max_local_gain(s, *frame, c)); + update_gain_history(s, c, get_max_local_gain(s, analyze_frame, c)); } return 0; @@ -777,7 +810,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in) static int flush_buffer(DynamicAudioNormalizerContext *s, AVFilterLink *inlink, AVFilterLink *outlink) { - AVFrame *out = ff_get_audio_buffer(outlink, s->frame_len); + AVFrame *out = ff_get_audio_buffer(outlink, s->sample_advance); int c, i; if (!out) @@ -830,7 +863,7 @@ static int activate(AVFilterContext *ctx) FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink); if (!s->eof) { - ret = ff_inlink_consume_samples(inlink, s->frame_len, s->frame_len, &in); + ret = ff_inlink_consume_samples(inlink, s->sample_advance, s->sample_advance, &in); if (ret < 0) return ret; if (ret > 0) { @@ -839,7 +872,7 @@ static int activate(AVFilterContext *ctx) return ret; } - if (ff_inlink_check_available_samples(inlink, s->frame_len) > 0) { + if (ff_inlink_check_available_samples(inlink, s->sample_advance) > 0) { ff_filter_set_ready(ctx, 10); return 0; } @@ -888,6 +921,7 @@ static int process_command(AVFilterContext *ctx, const char *cmd, const char *ar } s->frame_len = frame_size(inlink->sample_rate, s->frame_len_msec); + s->sample_advance = FFMAX(1, lrint(s->frame_len * (1. - s->overlap))); return 0; }

1 0

avfilter/af_dynaudnorm: do not leak frame on error
by Paul B Mahol 28 Feb '22

28 Feb '22

ffmpeg | branch: master | Paul B Mahol <onemda(a)gmail.com> | Mon Feb 28 09:06:20 2022 +0100| [9a86e5338c5d140965e17e258a62c4c9499e7a54] | committer: Paul B Mahol avfilter/af_dynaudnorm: do not leak frame on error > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=9a86e5338c5d14096… --- libavfilter/af_dynaudnorm.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/libavfilter/af_dynaudnorm.c b/libavfilter/af_dynaudnorm.c index 8015c8429c..77899474bf 100644 --- a/libavfilter/af_dynaudnorm.c +++ b/libavfilter/af_dynaudnorm.c @@ -661,11 +661,13 @@ static int analyze_frame(DynamicAudioNormalizerContext *s, AVFilterLink *outlink } ret = av_frame_copy_props(out, *frame); if (ret < 0) { + av_frame_free(frame); av_frame_free(&out); return ret; } ret = av_frame_copy(out, *frame); if (ret < 0) { + av_frame_free(frame); av_frame_free(&out); return ret; }

1 0

avcodec/setts_bsf: add an option to set output time base
by James Almer 28 Feb '22

28 Feb '22

ffmpeg | branch: master | James Almer <jamrial(a)gmail.com> | Mon Feb 28 12:35:27 2022 -0300| [4e33bae9f971ae6ddc914ea5081aab1ea3f85b31] | committer: James Almer avcodec/setts_bsf: add an option to set output time base And a relevant constant to use it in expressions. Signed-off-by: James Almer <jamrial(a)gmail.com> > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=4e33bae9f971ae6dd… --- doc/bitstream_filters.texi | 3 +++ libavcodec/setts_bsf.c | 11 ++++++++++- libavcodec/version.h | 2 +- 3 files changed, 14 insertions(+), 2 deletions(-) diff --git a/doc/bitstream_filters.texi b/doc/bitstream_filters.texi index 87c984185d..b11d357941 100644 --- a/doc/bitstream_filters.texi +++ b/doc/bitstream_filters.texi @@ -800,6 +800,9 @@ The next input duration. @item TB The timebase of stream packet belongs. +@item TB_OUT +The output timebase. + @item SR The sample rate of stream packet belongs. diff --git a/libavcodec/setts_bsf.c b/libavcodec/setts_bsf.c index d574449838..69977bc166 100644 --- a/libavcodec/setts_bsf.c +++ b/libavcodec/setts_bsf.c @@ -47,7 +47,8 @@ static const char *const var_names[] = { "DURATION", ///< original duration in the file of the frame "STARTPTS", ///< PTS at start of movie "STARTDTS", ///< DTS at start of movie - "TB", ///< timebase of the stream + "TB", ///< input timebase of the stream + "TB_OUT", ///< output timebase of the stream "SR", ///< sample rate of the stream "NOPTS", ///< The AV_NOPTS_VALUE constant NULL @@ -72,6 +73,7 @@ enum var_name { VAR_STARTPTS, VAR_STARTDTS, VAR_TB, + VAR_TB_OUT, VAR_SR, VAR_NOPTS, VAR_VARS_NB @@ -85,6 +87,8 @@ typedef struct SetTSContext { char *dts_str; char *duration_str; + AVRational time_base; + int64_t frame_number; int64_t start_pts; @@ -141,11 +145,15 @@ static int setts_init(AVBSFContext *ctx) } } + if (s->time_base.den) + ctx->time_base_out = s->time_base; + s->frame_number= 0; s->start_pts = AV_NOPTS_VALUE; s->start_dts = AV_NOPTS_VALUE; s->var_values[VAR_NOPTS] = AV_NOPTS_VALUE; s->var_values[VAR_TB] = ctx->time_base_in.den ? av_q2d(ctx->time_base_in) : 0; + s->var_values[VAR_TB_OUT]= ctx->time_base_out.den ? av_q2d(ctx->time_base_out) : 0; s->var_values[VAR_SR] = ctx->par_in->sample_rate; return 0; @@ -251,6 +259,7 @@ static const AVOption options[] = { { "pts", "set expression for packet PTS", OFFSET(pts_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, FLAGS }, { "dts", "set expression for packet DTS", OFFSET(dts_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, FLAGS }, { "duration", "set expression for packet duration", OFFSET(duration_str), AV_OPT_TYPE_STRING, {.str="DURATION"}, 0, 0, FLAGS }, + { "time_base", "set output timebase", OFFSET(time_base), AV_OPT_TYPE_RATIONAL, {.dbl=0}, 0, 0, FLAGS }, { NULL }, }; diff --git a/libavcodec/version.h b/libavcodec/version.h index 991b4f69bc..063d61e2fc 100644 --- a/libavcodec/version.h +++ b/libavcodec/version.h @@ -29,7 +29,7 @@ #define LIBAVCODEC_VERSION_MAJOR 59 #define LIBAVCODEC_VERSION_MINOR 21 -#define LIBAVCODEC_VERSION_MICRO 102 +#define LIBAVCODEC_VERSION_MICRO 103 #define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \ LIBAVCODEC_VERSION_MINOR, \

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avcodec/setts_bsf: stop repeatedly setting values that don't change between packets
by James Almer 28 Feb '22

28 Feb '22

ffmpeg | branch: master | James Almer <jamrial(a)gmail.com> | Mon Feb 28 12:30:45 2022 -0300| [9c8a06a7d6ed1647185622fdc03d993a4f8d12d7] | committer: James Almer avcodec/setts_bsf: stop repeatedly setting values that don't change between packets Signed-off-by: James Almer <jamrial(a)gmail.com> > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=9c8a06a7d6ed16471… --- libavcodec/setts_bsf.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/libavcodec/setts_bsf.c b/libavcodec/setts_bsf.c index eadc4e4d34..d574449838 100644 --- a/libavcodec/setts_bsf.c +++ b/libavcodec/setts_bsf.c @@ -145,6 +145,8 @@ static int setts_init(AVBSFContext *ctx) s->start_pts = AV_NOPTS_VALUE; s->start_dts = AV_NOPTS_VALUE; s->var_values[VAR_NOPTS] = AV_NOPTS_VALUE; + s->var_values[VAR_TB] = ctx->time_base_in.den ? av_q2d(ctx->time_base_in) : 0; + s->var_values[VAR_SR] = ctx->par_in->sample_rate; return 0; } @@ -187,8 +189,6 @@ static int setts_filter(AVBSFContext *ctx, AVPacket *pkt) s->var_values[VAR_NEXT_DUR] = pkt->duration; s->var_values[VAR_STARTPTS] = s->start_pts; s->var_values[VAR_STARTDTS] = s->start_dts; - s->var_values[VAR_TB] = ctx->time_base_out.den ? av_q2d(ctx->time_base_out) : 0; - s->var_values[VAR_SR] = ctx->par_in->sample_rate; new_ts = llrint(av_expr_eval(s->ts_expr, s->var_values, NULL)); new_duration = llrint(av_expr_eval(s->duration_expr, s->var_values, NULL));

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doc/bitstream_filters: add missing entry for the duration setts option
by James Almer 28 Feb '22

28 Feb '22

ffmpeg | branch: master | James Almer <jamrial(a)gmail.com> | Mon Feb 28 12:36:45 2022 -0300| [129f5ed87e716788191a073791a2446a6099b72f] | committer: James Almer doc/bitstream_filters: add missing entry for the duration setts option Forgotten in 825fb5f1cb1a1a64f6d2e4066639c88f224bf971. Signed-off-by: James Almer <jamrial(a)gmail.com> > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=129f5ed87e7167881… --- doc/bitstream_filters.texi | 2 ++ 1 file changed, 2 insertions(+) diff --git a/doc/bitstream_filters.texi b/doc/bitstream_filters.texi index 689bb52141..87c984185d 100644 --- a/doc/bitstream_filters.texi +++ b/doc/bitstream_filters.texi @@ -736,6 +736,8 @@ It accepts the following parameters: @item pts @item dts Set expressions for PTS, DTS or both. +@item duration +Set expression for duration. @end table The expressions are evaluated through the eval API and can contain the following

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avcodec/setts_bsf: add constants to modify packet duration
by James Almer 28 Feb '22

28 Feb '22

ffmpeg | branch: master | James Almer <jamrial(a)gmail.com> | Mon Feb 14 11:28:13 2022 -0300| [825fb5f1cb1a1a64f6d2e4066639c88f224bf971] | committer: James Almer avcodec/setts_bsf: add constants to modify packet duration Signed-off-by: James Almer <jamrial(a)gmail.com> > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=825fb5f1cb1a1a64f… --- doc/bitstream_filters.texi | 12 ++++++++++++ libavcodec/setts_bsf.c | 25 ++++++++++++++++++++++++- libavcodec/version.h | 2 +- 3 files changed, 37 insertions(+), 2 deletions(-) diff --git a/doc/bitstream_filters.texi b/doc/bitstream_filters.texi index 46654e9369..689bb52141 100644 --- a/doc/bitstream_filters.texi +++ b/doc/bitstream_filters.texi @@ -759,6 +759,9 @@ The demux timestamp in input. @item PTS The presentation timestamp in input. +@item DURATION +The duration in input. + @item STARTDTS The DTS of the first packet. @@ -771,18 +774,27 @@ The previous input DTS. @item PREV_INPTS The previous input PTS. +@item PREV_INDURATION +The previous input duration. + @item PREV_OUTDTS The previous output DTS. @item PREV_OUTPTS The previous output PTS. +@item PREV_OUTDURATION +The previous output duration. + @item NEXT_DTS The next input DTS. @item NEXT_PTS The next input PTS. +@item NEXT_DURATION +The next input duration. + @item TB The timebase of stream packet belongs. diff --git a/libavcodec/setts_bsf.c b/libavcodec/setts_bsf.c index f0243a1114..eadc4e4d34 100644 --- a/libavcodec/setts_bsf.c +++ b/libavcodec/setts_bsf.c @@ -35,12 +35,16 @@ static const char *const var_names[] = { "POS", ///< original position in the file of the frame "PREV_INPTS", ///< previous input PTS "PREV_INDTS", ///< previous input DTS + "PREV_INDURATION", ///< previous input duration "PREV_OUTPTS", ///< previous output PTS "PREV_OUTDTS", ///< previous output DTS + "PREV_OUTDURATION", ///< previous output duration "NEXT_PTS", ///< next input PTS "NEXT_DTS", ///< next input DTS + "NEXT_DURATION", ///< next input duration "PTS", ///< original PTS in the file of the frame "DTS", ///< original DTS in the file of the frame + "DURATION", ///< original duration in the file of the frame "STARTPTS", ///< PTS at start of movie "STARTDTS", ///< DTS at start of movie "TB", ///< timebase of the stream @@ -55,12 +59,16 @@ enum var_name { VAR_POS, VAR_PREV_INPTS, VAR_PREV_INDTS, + VAR_PREV_INDUR, VAR_PREV_OUTPTS, VAR_PREV_OUTDTS, + VAR_PREV_OUTDUR, VAR_NEXT_PTS, VAR_NEXT_DTS, + VAR_NEXT_DUR, VAR_PTS, VAR_DTS, + VAR_DURATION, VAR_STARTPTS, VAR_STARTDTS, VAR_TB, @@ -75,6 +83,7 @@ typedef struct SetTSContext { char *ts_str; char *pts_str; char *dts_str; + char *duration_str; int64_t frame_number; @@ -86,6 +95,7 @@ typedef struct SetTSContext { AVExpr *ts_expr; AVExpr *pts_expr; AVExpr *dts_expr; + AVExpr *duration_expr; AVPacket *prev_inpkt; AVPacket *prev_outpkt; @@ -109,6 +119,12 @@ static int setts_init(AVBSFContext *ctx) return ret; } + if ((ret = av_expr_parse(&s->duration_expr, s->duration_str, + var_names, NULL, NULL, NULL, NULL, 0, ctx)) < 0) { + av_log(ctx, AV_LOG_ERROR, "Error while parsing duration expression '%s'\n", s->duration_str); + return ret; + } + if (s->pts_str) { if ((ret = av_expr_parse(&s->pts_expr, s->pts_str, var_names, NULL, NULL, NULL, NULL, 0, ctx)) < 0) { @@ -136,7 +152,7 @@ static int setts_init(AVBSFContext *ctx) static int setts_filter(AVBSFContext *ctx, AVPacket *pkt) { SetTSContext *s = ctx->priv_data; - int64_t new_ts, new_pts, new_dts; + int64_t new_ts, new_pts, new_dts, new_duration; int ret; ret = ff_bsf_get_packet_ref(ctx, pkt); @@ -159,18 +175,23 @@ static int setts_filter(AVBSFContext *ctx, AVPacket *pkt) s->var_values[VAR_POS] = s->cur_pkt->pos; s->var_values[VAR_PTS] = s->cur_pkt->pts; s->var_values[VAR_DTS] = s->cur_pkt->dts; + s->var_values[VAR_DURATION] = s->cur_pkt->duration; s->var_values[VAR_PREV_INPTS] = s->prev_inpkt->pts; s->var_values[VAR_PREV_INDTS] = s->prev_inpkt->dts; + s->var_values[VAR_PREV_INDUR] = s->prev_inpkt->duration; s->var_values[VAR_PREV_OUTPTS] = s->prev_outpkt->pts; s->var_values[VAR_PREV_OUTDTS] = s->prev_outpkt->dts; + s->var_values[VAR_PREV_OUTDUR] = s->prev_outpkt->duration; s->var_values[VAR_NEXT_PTS] = pkt->pts; s->var_values[VAR_NEXT_DTS] = pkt->dts; + s->var_values[VAR_NEXT_DUR] = pkt->duration; s->var_values[VAR_STARTPTS] = s->start_pts; s->var_values[VAR_STARTDTS] = s->start_dts; s->var_values[VAR_TB] = ctx->time_base_out.den ? av_q2d(ctx->time_base_out) : 0; s->var_values[VAR_SR] = ctx->par_in->sample_rate; new_ts = llrint(av_expr_eval(s->ts_expr, s->var_values, NULL)); + new_duration = llrint(av_expr_eval(s->duration_expr, s->var_values, NULL)); if (s->pts_str) { s->var_values[VAR_TS] = s->cur_pkt->pts; @@ -197,6 +218,7 @@ static int setts_filter(AVBSFContext *ctx, AVPacket *pkt) pkt->pts = new_pts; pkt->dts = new_dts; + pkt->duration = new_duration; ret = av_packet_ref(s->prev_outpkt, pkt); if (ret < 0) @@ -228,6 +250,7 @@ static const AVOption options[] = { { "ts", "set expression for packet PTS and DTS", OFFSET(ts_str), AV_OPT_TYPE_STRING, {.str="TS"}, 0, 0, FLAGS }, { "pts", "set expression for packet PTS", OFFSET(pts_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, FLAGS }, { "dts", "set expression for packet DTS", OFFSET(dts_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, FLAGS }, + { "duration", "set expression for packet duration", OFFSET(duration_str), AV_OPT_TYPE_STRING, {.str="DURATION"}, 0, 0, FLAGS }, { NULL }, }; diff --git a/libavcodec/version.h b/libavcodec/version.h index bff9745669..991b4f69bc 100644 --- a/libavcodec/version.h +++ b/libavcodec/version.h @@ -29,7 +29,7 @@ #define LIBAVCODEC_VERSION_MAJOR 59 #define LIBAVCODEC_VERSION_MINOR 21 -#define LIBAVCODEC_VERSION_MICRO 101 +#define LIBAVCODEC_VERSION_MICRO 102 #define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \ LIBAVCODEC_VERSION_MINOR, \

1 0

avcodec/setts_bsf: add NEXT_PTS/DTS expression constants
by James Almer 28 Feb '22

28 Feb '22

ffmpeg | branch: master | James Almer <jamrial(a)gmail.com> | Mon Feb 14 11:09:17 2022 -0300| [9ed3139a76f055473083167e95c59afd54360bd0] | committer: James Almer avcodec/setts_bsf: add NEXT_PTS/DTS expression constants They correspond to the relevant fields from the packet that follows the one where the expressions are being applied. Signed-off-by: James Almer <jamrial(a)gmail.com> > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=9ed3139a76f055473… --- doc/bitstream_filters.texi | 6 ++++ libavcodec/setts_bsf.c | 75 ++++++++++++++++++++++++++++++---------------- libavcodec/version.h | 2 +- 3 files changed, 57 insertions(+), 26 deletions(-) diff --git a/doc/bitstream_filters.texi b/doc/bitstream_filters.texi index 6051dc28d2..46654e9369 100644 --- a/doc/bitstream_filters.texi +++ b/doc/bitstream_filters.texi @@ -777,6 +777,12 @@ The previous output DTS. @item PREV_OUTPTS The previous output PTS. +@item NEXT_DTS +The next input DTS. + +@item NEXT_PTS +The next input PTS. + @item TB The timebase of stream packet belongs. diff --git a/libavcodec/setts_bsf.c b/libavcodec/setts_bsf.c index d604f91f71..f0243a1114 100644 --- a/libavcodec/setts_bsf.c +++ b/libavcodec/setts_bsf.c @@ -37,6 +37,8 @@ static const char *const var_names[] = { "PREV_INDTS", ///< previous input DTS "PREV_OUTPTS", ///< previous output PTS "PREV_OUTDTS", ///< previous output DTS + "NEXT_PTS", ///< next input PTS + "NEXT_DTS", ///< next input DTS "PTS", ///< original PTS in the file of the frame "DTS", ///< original DTS in the file of the frame "STARTPTS", ///< PTS at start of movie @@ -55,6 +57,8 @@ enum var_name { VAR_PREV_INDTS, VAR_PREV_OUTPTS, VAR_PREV_OUTDTS, + VAR_NEXT_PTS, + VAR_NEXT_DTS, VAR_PTS, VAR_DTS, VAR_STARTPTS, @@ -76,16 +80,16 @@ typedef struct SetTSContext { int64_t start_pts; int64_t start_dts; - int64_t prev_inpts; - int64_t prev_indts; - int64_t prev_outpts; - int64_t prev_outdts; double var_values[VAR_VARS_NB]; AVExpr *ts_expr; AVExpr *pts_expr; AVExpr *dts_expr; + + AVPacket *prev_inpkt; + AVPacket *prev_outpkt; + AVPacket *cur_pkt; } SetTSContext; static int setts_init(AVBSFContext *ctx) @@ -93,6 +97,12 @@ static int setts_init(AVBSFContext *ctx) SetTSContext *s = ctx->priv_data; int ret; + s->prev_inpkt = av_packet_alloc(); + s->prev_outpkt = av_packet_alloc(); + s->cur_pkt = av_packet_alloc(); + if (!s->prev_inpkt || !s->prev_outpkt || !s->cur_pkt) + return AVERROR(ENOMEM); + if ((ret = av_expr_parse(&s->ts_expr, s->ts_str, var_names, NULL, NULL, NULL, NULL, 0, ctx)) < 0) { av_log(ctx, AV_LOG_ERROR, "Error while parsing ts expression '%s'\n", s->ts_str); @@ -118,10 +128,6 @@ static int setts_init(AVBSFContext *ctx) s->frame_number= 0; s->start_pts = AV_NOPTS_VALUE; s->start_dts = AV_NOPTS_VALUE; - s->prev_inpts = AV_NOPTS_VALUE; - s->prev_indts = AV_NOPTS_VALUE; - s->prev_outpts = AV_NOPTS_VALUE; - s->prev_outdts = AV_NOPTS_VALUE; s->var_values[VAR_NOPTS] = AV_NOPTS_VALUE; return 0; @@ -134,24 +140,31 @@ static int setts_filter(AVBSFContext *ctx, AVPacket *pkt) int ret; ret = ff_bsf_get_packet_ref(ctx, pkt); - if (ret < 0) + if (ret < 0 && (ret != AVERROR_EOF || !s->cur_pkt->data)) return ret; + if (!s->cur_pkt->data) { + av_packet_move_ref(s->cur_pkt, pkt); + return AVERROR(EAGAIN); + } + if (s->start_pts == AV_NOPTS_VALUE) - s->start_pts = pkt->pts; + s->start_pts = s->cur_pkt->pts; if (s->start_dts == AV_NOPTS_VALUE) - s->start_dts = pkt->dts; + s->start_dts = s->cur_pkt->dts; s->var_values[VAR_N] = s->frame_number++; - s->var_values[VAR_TS] = pkt->dts; - s->var_values[VAR_POS] = pkt->pos; - s->var_values[VAR_PTS] = pkt->pts; - s->var_values[VAR_DTS] = pkt->dts; - s->var_values[VAR_PREV_INPTS] = s->prev_inpts; - s->var_values[VAR_PREV_INDTS] = s->prev_indts; - s->var_values[VAR_PREV_OUTPTS] = s->prev_outpts; - s->var_values[VAR_PREV_OUTDTS] = s->prev_outdts; + s->var_values[VAR_TS] = s->cur_pkt->dts; + s->var_values[VAR_POS] = s->cur_pkt->pos; + s->var_values[VAR_PTS] = s->cur_pkt->pts; + s->var_values[VAR_DTS] = s->cur_pkt->dts; + s->var_values[VAR_PREV_INPTS] = s->prev_inpkt->pts; + s->var_values[VAR_PREV_INDTS] = s->prev_inpkt->dts; + s->var_values[VAR_PREV_OUTPTS] = s->prev_outpkt->pts; + s->var_values[VAR_PREV_OUTDTS] = s->prev_outpkt->dts; + s->var_values[VAR_NEXT_PTS] = pkt->pts; + s->var_values[VAR_NEXT_DTS] = pkt->dts; s->var_values[VAR_STARTPTS] = s->start_pts; s->var_values[VAR_STARTDTS] = s->start_dts; s->var_values[VAR_TB] = ctx->time_base_out.den ? av_q2d(ctx->time_base_out) : 0; @@ -160,27 +173,35 @@ static int setts_filter(AVBSFContext *ctx, AVPacket *pkt) new_ts = llrint(av_expr_eval(s->ts_expr, s->var_values, NULL)); if (s->pts_str) { - s->var_values[VAR_TS] = pkt->pts; + s->var_values[VAR_TS] = s->cur_pkt->pts; new_pts = llrint(av_expr_eval(s->pts_expr, s->var_values, NULL)); } else { new_pts = new_ts; } if (s->dts_str) { - s->var_values[VAR_TS] = pkt->dts; + s->var_values[VAR_TS] = s->cur_pkt->dts; new_dts = llrint(av_expr_eval(s->dts_expr, s->var_values, NULL)); } else { new_dts = new_ts; } - s->prev_inpts = pkt->pts; - s->prev_indts = pkt->dts; - s->prev_outpts = new_pts; - s->prev_outdts = new_dts; + av_packet_unref(s->prev_inpkt); + av_packet_unref(s->prev_outpkt); + av_packet_move_ref(s->prev_inpkt, s->cur_pkt); + av_packet_move_ref(s->cur_pkt, pkt); + + ret = av_packet_ref(pkt, s->prev_inpkt); + if (ret < 0) + return ret; pkt->pts = new_pts; pkt->dts = new_dts; + ret = av_packet_ref(s->prev_outpkt, pkt); + if (ret < 0) + av_packet_unref(pkt); + return ret; } @@ -188,6 +209,10 @@ static void setts_close(AVBSFContext *bsf) { SetTSContext *s = bsf->priv_data; + av_packet_free(&s->prev_inpkt); + av_packet_free(&s->prev_outpkt); + av_packet_free(&s->cur_pkt); + av_expr_free(s->ts_expr); s->ts_expr = NULL; av_expr_free(s->pts_expr); diff --git a/libavcodec/version.h b/libavcodec/version.h index d900503954..bff9745669 100644 --- a/libavcodec/version.h +++ b/libavcodec/version.h @@ -29,7 +29,7 @@ #define LIBAVCODEC_VERSION_MAJOR 59 #define LIBAVCODEC_VERSION_MINOR 21 -#define LIBAVCODEC_VERSION_MICRO 100 +#define LIBAVCODEC_VERSION_MICRO 101 #define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \ LIBAVCODEC_VERSION_MINOR, \

1 0

ffmpeg: ensure a keyframe was not seen before skipping packets
by James Almer 28 Feb '22

28 Feb '22

ffmpeg | branch: master | James Almer <jamrial(a)gmail.com> | Tue Feb 15 09:44:21 2022 -0300| [d75e7a9a83b2774900bcb5c58236c108d11af041] | committer: James Almer ffmpeg: ensure a keyframe was not seen before skipping packets A keyframe could be buffered in the bsf and not be output until more packets had been fed to it. Signed-off-by: James Almer <jamrial(a)gmail.com> > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=d75e7a9a83b277490… --- fftools/ffmpeg.c | 4 +++- fftools/ffmpeg.h | 1 + 2 files changed, 4 insertions(+), 1 deletion(-) diff --git a/fftools/ffmpeg.c b/fftools/ffmpeg.c index 66e5ec8c8e..d5cd010a9c 100644 --- a/fftools/ffmpeg.c +++ b/fftools/ffmpeg.c @@ -889,6 +889,8 @@ static void output_packet(OutputFile *of, AVPacket *pkt, /* apply the output bitstream filters */ if (ost->bsf_ctx) { + if (pkt->flags & AV_PKT_FLAG_KEY) + ost->seen_kf = 1; ret = av_bsf_send_packet(ost->bsf_ctx, eof ? NULL : pkt); if (ret < 0) goto finish; @@ -2026,7 +2028,7 @@ static void do_streamcopy(InputStream *ist, OutputStream *ost, const AVPacket *p } if ((!ost->frame_number && !(pkt->flags & AV_PKT_FLAG_KEY)) && - !ost->copy_initial_nonkeyframes) + !ost->copy_initial_nonkeyframes && !ost->seen_kf) return; if (!ost->frame_number && !ost->copy_prior_start) { diff --git a/fftools/ffmpeg.h b/fftools/ffmpeg.h index 1b8bbace3f..cc8f767e5d 100644 --- a/fftools/ffmpeg.h +++ b/fftools/ffmpeg.h @@ -535,6 +535,7 @@ typedef struct OutputStream { int inputs_done; const char *attachment_filename; + int seen_kf; int copy_initial_nonkeyframes; int copy_prior_start; char *disposition;

1 0