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@ -103,6 +103,9 @@ typedef struct ATempoContext { |
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// 1: output sample position
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int64_t position[2]; |
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// first input timestamp, all other timestamps are offset by this one
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int64_t start_pts; |
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// sample format:
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enum AVSampleFormat format; |
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@ -186,6 +189,7 @@ static void yae_clear(ATempoContext *atempo) |
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atempo->nfrag = 0; |
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atempo->state = YAE_LOAD_FRAGMENT; |
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atempo->start_pts = AV_NOPTS_VALUE; |
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atempo->position[0] = 0; |
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atempo->position[1] = 0; |
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@ -1068,7 +1072,7 @@ static int push_samples(ATempoContext *atempo, |
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atempo->dst_buffer->nb_samples = n_out; |
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// adjust the PTS:
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atempo->dst_buffer->pts = |
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atempo->dst_buffer->pts = atempo->start_pts + |
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av_rescale_q(atempo->nsamples_out, |
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(AVRational){ 1, outlink->sample_rate }, |
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outlink->time_base); |
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@ -1097,6 +1101,11 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *src_buffer) |
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const uint8_t *src = src_buffer->data[0]; |
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const uint8_t *src_end = src + n_in * atempo->stride; |
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if (atempo->start_pts == AV_NOPTS_VALUE) |
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atempo->start_pts = av_rescale_q(src_buffer->pts, |
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inlink->time_base, |
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outlink->time_base); |
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while (src < src_end) { |
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if (!atempo->dst_buffer) { |
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atempo->dst_buffer = ff_get_audio_buffer(outlink, n_out); |
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