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aacdec: Allow selecting float output at runtime.

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This commit is contained in:
Reimar Döffinger 2011-04-25 12:16:40 +02:00
parent 4c7ad768e1
commit 26d5a4b6b4
1 changed files with 10 additions and 28 deletions
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38
libavcodec/aacdec.c
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@ -557,12 +557,8 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
return -1; return -1;
} }
/* ffdshow custom code */ avctx->sample_fmt = avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT ?
#if CONFIG_AUDIO_FLOAT AV_SAMPLE_FMT_FLT : AV_SAMPLE_FMT_S16;
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
#else
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
#endif
AAC_INIT_VLC_STATIC( 0, 304); AAC_INIT_VLC_STATIC( 0, 304);
AAC_INIT_VLC_STATIC( 1, 270); AAC_INIT_VLC_STATIC( 1, 270);
@ -2179,12 +2175,8 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
avctx->frame_size = samples; avctx->frame_size = samples;
} }
/* ffdshow custom code */ data_size_tmp = samples * avctx->channels;
#if CONFIG_AUDIO_FLOAT data_size_tmp *= avctx->sample_fmt == AV_SAMPLE_FMT_FLT ? sizeof(float) : sizeof(int16_t);
data_size_tmp = samples * avctx->channels * sizeof(float);
#else
data_size_tmp = samples * avctx->channels * sizeof(int16_t);
#endif
if (*data_size < data_size_tmp) { if (*data_size < data_size_tmp) {
av_log(avctx, AV_LOG_ERROR, av_log(avctx, AV_LOG_ERROR,
"Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n", "Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n",
@ -2194,12 +2186,10 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
*data_size = data_size_tmp; *data_size = data_size_tmp;
if (samples) { if (samples) {
/* ffdshow custom code */ if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {
#if CONFIG_AUDIO_FLOAT float_interleave(data, (const float **)ac->output_data, samples, avctx->channels);
float_interleave(data, (const float **)ac->output_data, samples, avctx->channels); } else
#else ac->fmt_conv.float_to_int16_interleave(data, (const float **)ac->output_data, samples, avctx->channels);
ac->fmt_conv.float_to_int16_interleave(data, (const float **)ac->output_data, samples, avctx->channels);
#endif
} }
if (ac->output_configured) if (ac->output_configured)
@ -2518,11 +2508,7 @@ AVCodec ff_aac_decoder = {
aac_decode_frame, aac_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"), .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
.sample_fmts = (const enum AVSampleFormat[]) { .sample_fmts = (const enum AVSampleFormat[]) {
#if CONFIG_AUDIO_FLOAT AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE
AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE
#else
AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE
#endif
}, },
.channel_layouts = aac_channel_layout, .channel_layouts = aac_channel_layout,
}; };
@ -2542,11 +2528,7 @@ AVCodec ff_aac_latm_decoder = {
.decode = latm_decode_frame, .decode = latm_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Codec LATM syntax)"), .long_name = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Codec LATM syntax)"),
.sample_fmts = (const enum AVSampleFormat[]) { .sample_fmts = (const enum AVSampleFormat[]) {
#if CONFIG_AUDIO_FLOAT AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE
AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE
#else
AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE
#endif
}, },
.channel_layouts = aac_channel_layout, .channel_layouts = aac_channel_layout,
}; };