mirror of https://git.ffmpeg.org/ffmpeg.git
aacdec: Allow selecting float output at runtime.
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4c7ad768e1
commit
26d5a4b6b4
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@ -557,12 +557,8 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
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return -1;
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}
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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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avctx->sample_fmt = avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT ?
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AV_SAMPLE_FMT_FLT : AV_SAMPLE_FMT_S16;
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AAC_INIT_VLC_STATIC( 0, 304);
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AAC_INIT_VLC_STATIC( 1, 270);
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@ -2179,12 +2175,8 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
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avctx->frame_size = samples;
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}
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/* ffdshow custom code */
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#if CONFIG_AUDIO_FLOAT
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data_size_tmp = samples * avctx->channels * sizeof(float);
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#else
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data_size_tmp = samples * avctx->channels * sizeof(int16_t);
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#endif
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data_size_tmp = samples * avctx->channels;
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data_size_tmp *= avctx->sample_fmt == AV_SAMPLE_FMT_FLT ? sizeof(float) : sizeof(int16_t);
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if (*data_size < data_size_tmp) {
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av_log(avctx, AV_LOG_ERROR,
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"Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n",
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@ -2194,12 +2186,10 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
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*data_size = data_size_tmp;
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if (samples) {
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/* ffdshow custom code */
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#if CONFIG_AUDIO_FLOAT
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float_interleave(data, (const float **)ac->output_data, samples, avctx->channels);
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#else
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ac->fmt_conv.float_to_int16_interleave(data, (const float **)ac->output_data, samples, avctx->channels);
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#endif
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if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {
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float_interleave(data, (const float **)ac->output_data, samples, avctx->channels);
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} else
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ac->fmt_conv.float_to_int16_interleave(data, (const float **)ac->output_data, samples, avctx->channels);
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}
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if (ac->output_configured)
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@ -2518,11 +2508,7 @@ AVCodec ff_aac_decoder = {
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aac_decode_frame,
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.long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
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.sample_fmts = (const enum AVSampleFormat[]) {
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#if CONFIG_AUDIO_FLOAT
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AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE
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#else
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AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE
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#endif
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AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE
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},
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.channel_layouts = aac_channel_layout,
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};
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@ -2542,11 +2528,7 @@ AVCodec ff_aac_latm_decoder = {
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.decode = latm_decode_frame,
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.long_name = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Codec LATM syntax)"),
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.sample_fmts = (const enum AVSampleFormat[]) {
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#if CONFIG_AUDIO_FLOAT
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AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE
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#else
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AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE
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#endif
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AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE
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},
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.channel_layouts = aac_channel_layout,
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};
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