diff options
Diffstat (limited to 'libavfilter/af_dynaudnorm.c')
| -rw-r--r-- | libavfilter/af_dynaudnorm.c | 45 |
1 files changed, 27 insertions, 18 deletions
diff --git a/libavfilter/af_dynaudnorm.c b/libavfilter/af_dynaudnorm.c index 837126ac0d..b46420d291 100644 --- a/libavfilter/af_dynaudnorm.c +++ b/libavfilter/af_dynaudnorm.c @@ -79,7 +79,8 @@ typedef struct DynamicAudioNormalizerContext { int channels; int sample_advance; int eof; - uint64_t channels_to_filter; + char *channels_to_filter; + AVChannelLayout ch_layout; int64_t pts; cqueue **gain_history_original; @@ -116,8 +117,8 @@ static const AVOption dynaudnorm_options[] = { { "s", "set the compress factor", OFFSET(compress_factor), AV_OPT_TYPE_DOUBLE, {.dbl = 0.0}, 0.0, 30.0, FLAGS }, { "threshold", "set the threshold value", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl = 0.0}, 0.0, 1.0, FLAGS }, { "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 }, + { "channels", "set channels to filter", OFFSET(channels_to_filter),AV_OPT_TYPE_STRING, {.str="all"}, 0, 0, FLAGS }, + { "h", "set channels to filter", OFFSET(channels_to_filter),AV_OPT_TYPE_STRING, {.str="all"}, 0, 0, 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 } @@ -128,13 +129,17 @@ AVFILTER_DEFINE_CLASS(dynaudnorm); static av_cold int init(AVFilterContext *ctx) { DynamicAudioNormalizerContext *s = ctx->priv; + int ret = 0; if (!(s->filter_size & 1)) { av_log(ctx, AV_LOG_WARNING, "filter size %d is invalid. Changing to an odd value.\n", s->filter_size); s->filter_size |= 1; } - return 0; + if (strcmp(s->channels_to_filter, "all")) + ret = av_channel_layout_from_string(&s->ch_layout, s->channels_to_filter); + + return ret; } static inline int frame_size(int sample_rate, int frame_len_msec) @@ -298,6 +303,8 @@ static av_cold void uninit(AVFilterContext *ctx) av_freep(&s->weights); + av_channel_layout_uninit(&s->ch_layout); + ff_bufqueue_discard_all(&s->queue); av_frame_free(&s->window); @@ -310,17 +317,17 @@ static int config_input(AVFilterLink *inlink) uninit(ctx); - s->channels = inlink->channels; + s->channels = inlink->ch_layout.nb_channels; s->frame_len = frame_size(inlink->sample_rate, s->frame_len_msec); av_log(ctx, AV_LOG_DEBUG, "frame len %d\n", s->frame_len); - s->prev_amplification_factor = av_malloc_array(inlink->channels, sizeof(*s->prev_amplification_factor)); - s->dc_correction_value = av_calloc(inlink->channels, sizeof(*s->dc_correction_value)); - s->compress_threshold = av_calloc(inlink->channels, sizeof(*s->compress_threshold)); - s->gain_history_original = av_calloc(inlink->channels, sizeof(*s->gain_history_original)); - s->gain_history_minimum = av_calloc(inlink->channels, sizeof(*s->gain_history_minimum)); - s->gain_history_smoothed = av_calloc(inlink->channels, sizeof(*s->gain_history_smoothed)); - s->threshold_history = av_calloc(inlink->channels, sizeof(*s->threshold_history)); + s->prev_amplification_factor = av_malloc_array(inlink->ch_layout.nb_channels, sizeof(*s->prev_amplification_factor)); + s->dc_correction_value = av_calloc(inlink->ch_layout.nb_channels, sizeof(*s->dc_correction_value)); + s->compress_threshold = av_calloc(inlink->ch_layout.nb_channels, sizeof(*s->compress_threshold)); + s->gain_history_original = av_calloc(inlink->ch_layout.nb_channels, sizeof(*s->gain_history_original)); + s->gain_history_minimum = av_calloc(inlink->ch_layout.nb_channels, sizeof(*s->gain_history_minimum)); + s->gain_history_smoothed = av_calloc(inlink->ch_layout.nb_channels, sizeof(*s->gain_history_smoothed)); + s->threshold_history = av_calloc(inlink->ch_layout.nb_channels, sizeof(*s->threshold_history)); s->weights = av_malloc_array(MAX_FILTER_SIZE, sizeof(*s->weights)); s->is_enabled = cqueue_create(s->filter_size, MAX_FILTER_SIZE); if (!s->prev_amplification_factor || !s->dc_correction_value || @@ -330,7 +337,7 @@ static int config_input(AVFilterLink *inlink) !s->is_enabled || !s->weights) return AVERROR(ENOMEM); - for (int c = 0; c < inlink->channels; c++) { + for (int c = 0; c < inlink->ch_layout.nb_channels; c++) { s->prev_amplification_factor[c] = 1.0; s->gain_history_original[c] = cqueue_create(s->filter_size, MAX_FILTER_SIZE); @@ -377,7 +384,7 @@ static double find_peak_magnitude(AVFrame *frame, int channel) double max = DBL_EPSILON; if (channel == -1) { - for (int c = 0; c < frame->channels; c++) { + for (int c = 0; c < frame->ch_layout.nb_channels; c++) { double *data_ptr = (double *)frame->extended_data[c]; for (int i = 0; i < frame->nb_samples; i++) @@ -398,7 +405,7 @@ static double compute_frame_rms(AVFrame *frame, int channel) double rms_value = 0.0; if (channel == -1) { - for (int c = 0; c < frame->channels; c++) { + for (int c = 0; c < frame->ch_layout.nb_channels; c++) { const double *data_ptr = (double *)frame->extended_data[c]; for (int i = 0; i < frame->nb_samples; i++) { @@ -406,7 +413,7 @@ static double compute_frame_rms(AVFrame *frame, int channel) } } - rms_value /= frame->nb_samples * frame->channels; + rms_value /= frame->nb_samples * frame->ch_layout.nb_channels; } else { const double *data_ptr = (double *)frame->extended_data[channel]; for (int i = 0; i < frame->nb_samples; i++) { @@ -526,7 +533,9 @@ static inline double update_value(double new, double old, double aggressiveness) static inline int bypass_channel(DynamicAudioNormalizerContext *s, AVFrame *frame, int ch) { - return !(av_channel_layout_extract_channel(frame->channel_layout, ch) & s->channels_to_filter); + enum AVChannel channel = av_channel_layout_channel_from_index(&frame->ch_layout, ch); + + return av_channel_layout_index_from_channel(&s->ch_layout, channel) < 0; } static void perform_dc_correction(DynamicAudioNormalizerContext *s, AVFrame *frame) @@ -702,7 +711,7 @@ static int analyze_frame(DynamicAudioNormalizerContext *s, AVFilterLink *outlink analyze_frame = s->window; } else { av_samples_copy(s->window->extended_data, (*frame)->extended_data, 0, 0, - s->frame_len, (*frame)->channels, (*frame)->format); + s->frame_len, (*frame)->ch_layout.nb_channels, (*frame)->format); analyze_frame = *frame; } |