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ac3dec: allow selecting float output at runtime.

This commit is contained in:
Reimar Döffinger 2011-04-25 11:59:28 +02:00
parent 5e9de76f54
commit 4c7ad768e1
3 changed files with 27 additions and 27 deletions
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45
libavcodec/ac3dec.c
View File

@ -185,14 +185,6 @@ static av_cold int ac3_decode_init(AVCodecContext *avctx)
ff_fmt_convert_init(&s->fmt_conv, avctx);
av_lfg_init(&s->dith_state, 0);
/* ffdshow custom code */
#if CONFIG_AUDIO_FLOAT
s->mul_bias = 1.0f;
#else
/* set scale value for float to int16 conversion */
s->mul_bias = 32767.0f;
#endif
/* allow downmixing to stereo or mono */
if (avctx->channels > 0 && avctx->request_channels > 0 &&
avctx->request_channels < avctx->channels &&
@ -201,12 +193,14 @@ static av_cold int ac3_decode_init(AVCodecContext *avctx)
}
s->downmixed = 1;
/* ffdshow custom code */
#if CONFIG_AUDIO_FLOAT
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
#else
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
#endif
if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) {
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
s->mul_bias = 1.0f;
} else {
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
/* set scale value for float to int16 conversion */
s->mul_bias = 32767.0f;
}
return 0;
}
@ -1301,12 +1295,8 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size,
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
AC3DecodeContext *s = avctx->priv_data;
/* ffdshow custom code */
#if CONFIG_AUDIO_FLOAT
float *out_samples = (float *)data;
#else
float *out_samples_flt = (float *)data;
int16_t *out_samples = (int16_t *)data;
#endif
int blk, ch, err;
const uint8_t *channel_map;
const float *output[AC3_MAX_CHANNELS];
@ -1412,15 +1402,16 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size,
av_log(avctx, AV_LOG_ERROR, "error decoding the audio block\n");
err = 1;
}
/* ffdshow custom code */
#if CONFIG_AUDIO_FLOAT
float_interleave_noscale(out_samples, output, 256, s->out_channels);
#else
s->fmt_conv.float_to_int16_interleave(out_samples, output, 256, s->out_channels);
#endif
out_samples += 256 * s->out_channels;
if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {
float_interleave_noscale(out_samples_flt, output, 256, s->out_channels);
out_samples_flt += 256 * s->out_channels;
} else {
s->fmt_conv.float_to_int16_interleave(out_samples, output, 256, s->out_channels);
out_samples += 256 * s->out_channels;
}
}
*data_size = s->num_blocks * 256 * avctx->channels * sizeof (out_samples[0]); /* ffdshow custom code */
*data_size = s->num_blocks * 256 * avctx->channels;
*data_size *= avctx->sample_fmt == AV_SAMPLE_FMT_FLT ? sizeof(*out_samples_flt) : sizeof(*out_samples);
return FFMIN(buf_size, s->frame_size);
}

8
libavcodec/avcodec.h
View File

@ -2877,6 +2877,14 @@ typedef struct AVCodecContext {
int64_t pts_correction_last_pts; /// PTS of the last frame
int64_t pts_correction_last_dts; /// DTS of the last frame
/**
* desired sample format
* - encoding: Not used.
* - decoding: Set by user.
* Decoder will decode to this format if it can.
*/
enum AVSampleFormat request_sample_fmt;
} AVCodecContext;
/**

1
libavcodec/options.c
View File

@ -447,6 +447,7 @@ static const AVOption options[]={
{"em", "Emergency", 0, FF_OPT_TYPE_CONST, AV_AUDIO_SERVICE_TYPE_EMERGENCY, INT_MIN, INT_MAX, A|E, "audio_service_type"},
{"vo", "Voice Over", 0, FF_OPT_TYPE_CONST, AV_AUDIO_SERVICE_TYPE_VOICE_OVER, INT_MIN, INT_MAX, A|E, "audio_service_type"},
{"ka", "Karaoke", 0, FF_OPT_TYPE_CONST, AV_AUDIO_SERVICE_TYPE_KARAOKE, INT_MIN, INT_MAX, A|E, "audio_service_type"},
{"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},
{NULL},
};