mirror of https://git.ffmpeg.org/ffmpeg.git
ac3dec: allow selecting float output at runtime.
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@ -185,14 +185,6 @@ static av_cold int ac3_decode_init(AVCodecContext *avctx)
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ff_fmt_convert_init(&s->fmt_conv, avctx);
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av_lfg_init(&s->dith_state, 0);
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/* ffdshow custom code */
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#if CONFIG_AUDIO_FLOAT
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s->mul_bias = 1.0f;
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#else
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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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#endif
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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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avctx->request_channels < avctx->channels &&
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@ -201,12 +193,14 @@ static av_cold int ac3_decode_init(AVCodecContext *avctx)
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}
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s->downmixed = 1;
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/* ffdshow custom code */
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#if CONFIG_AUDIO_FLOAT
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avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
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#else
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avctx->sample_fmt = AV_SAMPLE_FMT_S16;
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#endif
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if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) {
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avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
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s->mul_bias = 1.0f;
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} else {
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avctx->sample_fmt = AV_SAMPLE_FMT_S16;
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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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}
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return 0;
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}
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@ -1301,12 +1295,8 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size,
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const uint8_t *buf = avpkt->data;
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int buf_size = avpkt->size;
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AC3DecodeContext *s = avctx->priv_data;
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/* ffdshow custom code */
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#if CONFIG_AUDIO_FLOAT
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float *out_samples = (float *)data;
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#else
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float *out_samples_flt = (float *)data;
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int16_t *out_samples = (int16_t *)data;
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#endif
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int blk, ch, err;
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const uint8_t *channel_map;
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const float *output[AC3_MAX_CHANNELS];
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@ -1412,15 +1402,16 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size,
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av_log(avctx, AV_LOG_ERROR, "error decoding the audio block\n");
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err = 1;
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}
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/* ffdshow custom code */
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#if CONFIG_AUDIO_FLOAT
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float_interleave_noscale(out_samples, output, 256, s->out_channels);
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#else
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s->fmt_conv.float_to_int16_interleave(out_samples, output, 256, s->out_channels);
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#endif
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out_samples += 256 * s->out_channels;
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if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {
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float_interleave_noscale(out_samples_flt, output, 256, s->out_channels);
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out_samples_flt += 256 * s->out_channels;
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} else {
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s->fmt_conv.float_to_int16_interleave(out_samples, output, 256, s->out_channels);
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out_samples += 256 * s->out_channels;
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}
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}
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*data_size = s->num_blocks * 256 * avctx->channels * sizeof (out_samples[0]); /* ffdshow custom code */
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*data_size = s->num_blocks * 256 * avctx->channels;
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*data_size *= avctx->sample_fmt == AV_SAMPLE_FMT_FLT ? sizeof(*out_samples_flt) : sizeof(*out_samples);
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return FFMIN(buf_size, s->frame_size);
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}
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@ -2877,6 +2877,14 @@ typedef struct AVCodecContext {
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int64_t pts_correction_last_pts; /// PTS of the last frame
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int64_t pts_correction_last_dts; /// DTS of the last frame
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/**
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* desired sample format
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* - encoding: Not used.
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* - decoding: Set by user.
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* Decoder will decode to this format if it can.
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*/
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enum AVSampleFormat request_sample_fmt;
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} AVCodecContext;
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/**
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@ -447,6 +447,7 @@ static const AVOption options[]={
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{"em", "Emergency", 0, FF_OPT_TYPE_CONST, AV_AUDIO_SERVICE_TYPE_EMERGENCY, INT_MIN, INT_MAX, A|E, "audio_service_type"},
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{"vo", "Voice Over", 0, FF_OPT_TYPE_CONST, AV_AUDIO_SERVICE_TYPE_VOICE_OVER, INT_MIN, INT_MAX, A|E, "audio_service_type"},
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{"ka", "Karaoke", 0, FF_OPT_TYPE_CONST, AV_AUDIO_SERVICE_TYPE_KARAOKE, INT_MIN, INT_MAX, A|E, "audio_service_type"},
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{"request_sample_fmt", "sample format audio decoders should prefer", OFFSET(request_sample_fmt), FF_OPT_TYPE_INT, AV_SAMPLE_FMT_NONE, AV_SAMPLE_FMT_NONE, AV_SAMPLE_FMT_NB-1, A|D},
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{NULL},
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};
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