#include <vo-aacenc/cmnMemory.h>
#include "avcodec.h"
+#include "audio_frame_queue.h"
+#include "internal.h"
#include "mpeg4audio.h"
+#define FRAME_SIZE 1024
+#define ENC_DELAY 1600
+
typedef struct AACContext {
VO_AUDIO_CODECAPI codec_api;
VO_HANDLE handle;
VO_MEM_OPERATOR mem_operator;
VO_CODEC_INIT_USERDATA user_data;
+ VO_PBYTE end_buffer;
+ AudioFrameQueue afq;
+ int last_frame;
+ int last_samples;
} AACContext;
+
+static int aac_encode_close(AVCodecContext *avctx)
+{
+ AACContext *s = avctx->priv_data;
+
+ s->codec_api.Uninit(s->handle);
+ av_freep(&avctx->extradata);
+ ff_af_queue_close(&s->afq);
+ av_freep(&s->end_buffer);
+
+ return 0;
+}
+
static av_cold int aac_encode_init(AVCodecContext *avctx)
{
AACContext *s = avctx->priv_data;
AACENC_PARAM params = { 0 };
- int index;
+ int index, ret;
- avctx->coded_frame = avcodec_alloc_frame();
- if (!avctx->coded_frame)
- return AVERROR(ENOMEM);
- avctx->frame_size = 1024;
+ avctx->frame_size = FRAME_SIZE;
+ avctx->initial_padding = ENC_DELAY;
+ s->last_frame = 2;
+ ff_af_queue_init(avctx, &s->afq);
+
+ s->end_buffer = av_mallocz(avctx->frame_size * avctx->channels * 2);
+ if (!s->end_buffer) {
+ ret = AVERROR(ENOMEM);
+ goto error;
+ }
voGetAACEncAPI(&s->codec_api);
params.sampleRate = avctx->sample_rate;
params.bitRate = avctx->bit_rate;
params.nChannels = avctx->channels;
- params.adtsUsed = !(avctx->flags & CODEC_FLAG_GLOBAL_HEADER);
+ params.adtsUsed = !(avctx->flags & AV_CODEC_FLAG_GLOBAL_HEADER);
if (s->codec_api.SetParam(s->handle, VO_PID_AAC_ENCPARAM, ¶ms)
!= VO_ERR_NONE) {
av_log(avctx, AV_LOG_ERROR, "Unable to set encoding parameters\n");
- return AVERROR(EINVAL);
+ ret = AVERROR(EINVAL);
+ goto error;
}
for (index = 0; index < 16; index++)
if (index == 16) {
av_log(avctx, AV_LOG_ERROR, "Unsupported sample rate %d\n",
avctx->sample_rate);
- return AVERROR(ENOSYS);
+ ret = AVERROR(ENOSYS);
+ goto error;
}
- if (avctx->flags & CODEC_FLAG_GLOBAL_HEADER) {
+ if (avctx->flags & AV_CODEC_FLAG_GLOBAL_HEADER) {
avctx->extradata_size = 2;
avctx->extradata = av_mallocz(avctx->extradata_size +
- FF_INPUT_BUFFER_PADDING_SIZE);
- if (!avctx->extradata)
- return AVERROR(ENOMEM);
+ AV_INPUT_BUFFER_PADDING_SIZE);
+ if (!avctx->extradata) {
+ ret = AVERROR(ENOMEM);
+ goto error;
+ }
avctx->extradata[0] = 0x02 << 3 | index >> 1;
avctx->extradata[1] = (index & 0x01) << 7 | avctx->channels << 3;
}
return 0;
+error:
+ aac_encode_close(avctx);
+ return ret;
}
-static int aac_encode_close(AVCodecContext *avctx)
-{
- AACContext *s = avctx->priv_data;
-
- s->codec_api.Uninit(s->handle);
- av_freep(&avctx->coded_frame);
-
- return 0;
-}
-
-static int aac_encode_frame(AVCodecContext *avctx,
- unsigned char *frame/*out*/,
- int buf_size, void *data/*in*/)
+static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
+ const AVFrame *frame, int *got_packet_ptr)
{
AACContext *s = avctx->priv_data;
VO_CODECBUFFER input = { 0 }, output = { 0 };
VO_AUDIO_OUTPUTINFO output_info = { { 0 } };
+ VO_PBYTE samples;
+ int ret;
+
+ /* handle end-of-stream small frame and flushing */
+ if (!frame) {
+ if (s->last_frame <= 0)
+ return 0;
+ if (s->last_samples > 0 && s->last_samples < ENC_DELAY - FRAME_SIZE) {
+ s->last_samples = 0;
+ s->last_frame--;
+ }
+ s->last_frame--;
+ memset(s->end_buffer, 0, 2 * avctx->channels * avctx->frame_size);
+ samples = s->end_buffer;
+ } else {
+ if (frame->nb_samples < avctx->frame_size) {
+ s->last_samples = frame->nb_samples;
+ memcpy(s->end_buffer, frame->data[0], 2 * avctx->channels * frame->nb_samples);
+ samples = s->end_buffer;
+ } else {
+ samples = (VO_PBYTE)frame->data[0];
+ }
+ /* add current frame to the queue */
+ if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
+ return ret;
+ }
+
+ if ((ret = ff_alloc_packet(avpkt, FFMAX(8192, 768 * avctx->channels)))) {
+ av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
+ return ret;
+ }
- input.Buffer = data;
- input.Length = 2 * avctx->channels * avctx->frame_size;
- output.Buffer = frame;
- output.Length = buf_size;
+ input.Buffer = samples;
+ input.Length = 2 * avctx->channels * avctx->frame_size;
+ output.Buffer = avpkt->data;
+ output.Length = avpkt->size;
s->codec_api.SetInputData(s->handle, &input);
if (s->codec_api.GetOutputData(s->handle, &output, &output_info)
av_log(avctx, AV_LOG_ERROR, "Unable to encode frame\n");
return AVERROR(EINVAL);
}
- return output.Length;
+
+ /* Get the next frame pts/duration */
+ ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
+ &avpkt->duration);
+
+ avpkt->size = output.Length;
+ *got_packet_ptr = 1;
+ return 0;
}
AVCodec ff_libvo_aacenc_encoder = {
.name = "libvo_aacenc",
+ .long_name = NULL_IF_CONFIG_SMALL("Android VisualOn AAC (Advanced Audio Coding)"),
.type = AVMEDIA_TYPE_AUDIO,
- .id = CODEC_ID_AAC,
+ .id = AV_CODEC_ID_AAC,
.priv_data_size = sizeof(AACContext),
.init = aac_encode_init,
- .encode = aac_encode_frame,
+ .encode2 = aac_encode_frame,
.close = aac_encode_close,
- .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
- .long_name = NULL_IF_CONFIG_SMALL("Android VisualOn AAC"),
+ .capabilities = AV_CODEC_CAP_SMALL_LAST_FRAME | AV_CODEC_CAP_DELAY,
+ .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
+ AV_SAMPLE_FMT_NONE },
};
projects / ffmpeg / blobdiff