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aacenc: use the PCE comment field for encoder ID

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Also handle extradata of variable size (for bitexact/if PCEs aren't used).

Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
This commit is contained in:
Rostislav Pehlivanov
2017-11-09 04:30:07 +00:00
parent fbf295e2bd
commit 7b7775a604
1 changed files with 16 additions and 6 deletions
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22
libavcodec/aacenc.c
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@@ -55,6 +55,8 @@ static void put_pce(PutBitContext *pb, AVCodecContext *avctx)
int i, j; int i, j;
AACEncContext *s = avctx->priv_data; AACEncContext *s = avctx->priv_data;
AACPCEInfo *pce = &s->pce; AACPCEInfo *pce = &s->pce;
const int bitexact = avctx->flags & AV_CODEC_FLAG_BITEXACT;
const char *aux_data = bitexact ? "Lavc" : LIBAVCODEC_IDENT;
put_bits(pb, 4, 0); put_bits(pb, 4, 0);
@@ -81,20 +83,26 @@ static void put_pce(PutBitContext *pb, AVCodecContext *avctx)
} }
avpriv_align_put_bits(pb); avpriv_align_put_bits(pb);
put_bits(pb, 8, 0); put_bits(pb, 8, strlen(aux_data));
avpriv_put_string(pb, aux_data, 0);
} }
/** /**
* Make AAC audio config object. * Make AAC audio config object.
* @see 1.6.2.1 "Syntax - AudioSpecificConfig" * @see 1.6.2.1 "Syntax - AudioSpecificConfig"
*/ */
static void put_audio_specific_config(AVCodecContext *avctx) static int put_audio_specific_config(AVCodecContext *avctx)
{ {
PutBitContext pb; PutBitContext pb;
AACEncContext *s = avctx->priv_data; AACEncContext *s = avctx->priv_data;
int channels = (!s->needs_pce)*(s->channels - (s->channels == 8 ? 1 : 0)); int channels = (!s->needs_pce)*(s->channels - (s->channels == 8 ? 1 : 0));
const int max_size = 32;
init_put_bits(&pb, avctx->extradata, avctx->extradata_size); avctx->extradata = av_mallocz(max_size);
if (!avctx->extradata)
return AVERROR(ENOMEM);
init_put_bits(&pb, avctx->extradata, max_size);
put_bits(&pb, 5, s->profile+1); //profile put_bits(&pb, 5, s->profile+1); //profile
put_bits(&pb, 4, s->samplerate_index); //sample rate index put_bits(&pb, 4, s->samplerate_index); //sample rate index
put_bits(&pb, 4, channels); put_bits(&pb, 4, channels);
@@ -110,6 +118,9 @@ static void put_audio_specific_config(AVCodecContext *avctx)
put_bits(&pb, 5, AOT_SBR); put_bits(&pb, 5, AOT_SBR);
put_bits(&pb, 1, 0); put_bits(&pb, 1, 0);
flush_put_bits(&pb); flush_put_bits(&pb);
avctx->extradata_size = put_bits_count(&pb) >> 3;
return 0;
} }
void ff_quantize_band_cost_cache_init(struct AACEncContext *s) void ff_quantize_band_cost_cache_init(struct AACEncContext *s)
@@ -931,7 +942,6 @@ static av_cold int alloc_buffers(AVCodecContext *avctx, AACEncContext *s)
int ch; int ch;
FF_ALLOCZ_ARRAY_OR_GOTO(avctx, s->buffer.samples, s->channels, 3 * 1024 * sizeof(s->buffer.samples[0]), alloc_fail); FF_ALLOCZ_ARRAY_OR_GOTO(avctx, s->buffer.samples, s->channels, 3 * 1024 * sizeof(s->buffer.samples[0]), alloc_fail);
FF_ALLOCZ_ARRAY_OR_GOTO(avctx, s->cpe, s->chan_map[0], sizeof(ChannelElement), alloc_fail); FF_ALLOCZ_ARRAY_OR_GOTO(avctx, s->cpe, s->chan_map[0], sizeof(ChannelElement), alloc_fail);
FF_ALLOCZ_OR_GOTO(avctx, avctx->extradata, 5 + AV_INPUT_BUFFER_PADDING_SIZE, alloc_fail);
for(ch = 0; ch < s->channels; ch++) for(ch = 0; ch < s->channels; ch++)
s->planar_samples[ch] = s->buffer.samples + 3 * 1024 * ch; s->planar_samples[ch] = s->buffer.samples + 3 * 1024 * ch;
@@ -956,7 +966,6 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
/* Constants */ /* Constants */
s->last_frame_pb_count = 0; s->last_frame_pb_count = 0;
avctx->extradata_size = 20;
avctx->frame_size = 1024; avctx->frame_size = 1024;
avctx->initial_padding = 1024; avctx->initial_padding = 1024;
s->lambda = avctx->global_quality > 0 ? avctx->global_quality : 120; s->lambda = avctx->global_quality > 0 ? avctx->global_quality : 120;
@@ -1071,7 +1080,8 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
if ((ret = alloc_buffers(avctx, s)) < 0) if ((ret = alloc_buffers(avctx, s)) < 0)
goto fail; goto fail;
put_audio_specific_config(avctx); if ((ret = put_audio_specific_config(avctx)))
goto fail;
sizes[0] = ff_aac_swb_size_1024[s->samplerate_index]; sizes[0] = ff_aac_swb_size_1024[s->samplerate_index];
sizes[1] = ff_aac_swb_size_128[s->samplerate_index]; sizes[1] = ff_aac_swb_size_128[s->samplerate_index];