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https://gitee.com/openharmony/third_party_ffmpeg
synced 2024-11-23 19:30:05 +00:00
cosmetics: indentation and spacing
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@ -568,8 +568,8 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
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// -1024 - Compensate wrong IMDCT method.
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// 60 - Required to scale values to the correct range [-32768,32767]
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// for float to int16 conversion. (1 << (60 / 4)) == 32768
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ac->sf_scale = 1. / -1024.;
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ac->sf_offset = 60;
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ac->sf_scale = 1. / -1024.;
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ac->sf_offset = 60;
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ff_aac_tableinit();
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@ -195,8 +195,8 @@ static av_cold int ac3_decode_init(AVCodecContext *avctx)
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dsputil_init(&s->dsp, avctx);
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av_lfg_init(&s->dith_state, 0);
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/* set bias values for float to int16 conversion */
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s->mul_bias = 32767.0f;
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/* set scale value for float to int16 conversion */
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s->mul_bias = 32767.0f;
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/* allow downmixing to stereo or mono */
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if (avctx->channels > 0 && avctx->request_channels > 0 &&
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@ -1838,14 +1838,13 @@ static av_cold int dca_decode_init(AVCodecContext * avctx)
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s->samples_chanptr[i] = s->samples + i * 256;
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avctx->sample_fmt = AV_SAMPLE_FMT_S16;
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s->scale_bias = 1.0;
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/* allow downmixing to stereo */
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if (avctx->channels > 0 && avctx->request_channels < avctx->channels &&
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avctx->request_channels == 2) {
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avctx->channels = avctx->request_channels;
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}
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s->scale_bias = 1.0;
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/* allow downmixing to stereo */
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if (avctx->channels > 0 && avctx->request_channels < avctx->channels &&
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avctx->request_channels == 2) {
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avctx->channels = avctx->request_channels;
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}
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return 0;
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}
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@ -135,7 +135,7 @@ static av_cold int decode_init(AVCodecContext * avctx) {
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dsputil_init(&s->dsp, avctx);
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s->scale_bias = 1.0/(1*8);
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s->scale_bias = 1.0/(1*8);
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/* Generate overlap window */
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if (!ff_sine_128[127])
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@ -789,13 +789,13 @@ static int wma_decode_frame(WMACodecContext *s, int16_t *samples)
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/* convert frame to integer */
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n = s->frame_len;
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incr = s->nb_channels;
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for (ch = 0; ch < MAX_CHANNELS; ch++)
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output[ch] = s->frame_out[ch];
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s->dsp.float_to_int16_interleave(samples, output, n, incr);
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for(ch = 0; ch < incr; ch++) {
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/* prepare for next block */
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memmove(&s->frame_out[ch][0], &s->frame_out[ch][n], n * sizeof(float));
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}
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for (ch = 0; ch < MAX_CHANNELS; ch++)
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output[ch] = s->frame_out[ch];
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s->dsp.float_to_int16_interleave(samples, output, n, incr);
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for (ch = 0; ch < incr; ch++) {
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/* prepare for next block */
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memmove(&s->frame_out[ch][0], &s->frame_out[ch][n], n * sizeof(float));
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}
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#ifdef TRACE
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dump_shorts(s, "samples", samples, n * s->nb_channels);
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