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roqaudioenc: use AVCodec.encode2()

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The first frame pts must be saved until we have 8 frames since RoQ audio
requires 8 frames in the first packet.
This commit is contained in:
Justin Ruggles 2012-02-27 20:53:09 -05:00
parent b03dcf07f6
commit 32173df3d2
1 changed files with 27 additions and 15 deletions
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42
libavcodec/roqaudioenc.c
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@ -24,6 +24,7 @@
#include "libavutil/intmath.h"
#include "avcodec.h"
#include "bytestream.h"
#include "internal.h"
#define ROQ_FRAME_SIZE 735
#define ROQ_HEADER_SIZE 8
@ -37,6 +38,7 @@ typedef struct
int input_frames;
int buffered_samples;
int16_t *frame_buffer;
int64_t first_pts;
} ROQDPCMContext;
@ -44,7 +46,9 @@ static av_cold int roq_dpcm_encode_close(AVCodecContext *avctx)
{
ROQDPCMContext *context = avctx->priv_data;
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
av_freep(&context->frame_buffer);
return 0;
@ -77,11 +81,13 @@ static av_cold int roq_dpcm_encode_init(AVCodecContext *avctx)
context->lastSample[0] = context->lastSample[1] = 0;
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame= avcodec_alloc_frame();
if (!avctx->coded_frame) {
ret = AVERROR(ENOMEM);
goto error;
}
#endif
return 0;
error:
@ -129,23 +135,25 @@ static unsigned char dpcm_predict(short *previous, short current)
return result;
}
static int roq_dpcm_encode_frame(AVCodecContext *avctx,
unsigned char *frame, int buf_size, void *data)
static int roq_dpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
int i, stereo, data_size;
const int16_t *in = data;
uint8_t *out = frame;
int i, stereo, data_size, ret;
const int16_t *in = frame ? (const int16_t *)frame->data[0] : NULL;
uint8_t *out;
ROQDPCMContext *context = avctx->priv_data;
stereo = (avctx->channels == 2);
if (!data && context->input_frames >= 8)
if (!in && context->input_frames >= 8)
return 0;
if (data && context->input_frames < 8) {
if (in && context->input_frames < 8) {
memcpy(&context->frame_buffer[context->buffered_samples * avctx->channels],
in, avctx->frame_size * avctx->channels * sizeof(*in));
context->buffered_samples += avctx->frame_size;
if (context->input_frames == 0)
context->first_pts = frame->pts;
if (context->input_frames < 7) {
context->input_frames++;
return 0;
@ -158,15 +166,16 @@ static int roq_dpcm_encode_frame(AVCodecContext *avctx,
context->lastSample[1] &= 0xFF00;
}
if (context->input_frames == 7 || !data)
if (context->input_frames == 7 || !in)
data_size = avctx->channels * context->buffered_samples;
else
data_size = avctx->channels * avctx->frame_size;
if (buf_size < ROQ_HEADER_SIZE + data_size) {
av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n");
return AVERROR(EINVAL);
if ((ret = ff_alloc_packet(avpkt, ROQ_HEADER_SIZE + data_size))) {
av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
return ret;
}
out = avpkt->data;
bytestream_put_byte(&out, stereo ? 0x21 : 0x20);
bytestream_put_byte(&out, 0x10);
@ -182,12 +191,15 @@ static int roq_dpcm_encode_frame(AVCodecContext *avctx,
for (i = 0; i < data_size; i++)
*out++ = dpcm_predict(&context->lastSample[i & 1], *in++);
avpkt->pts = context->input_frames <= 7 ? context->first_pts : frame->pts;
avpkt->duration = data_size / avctx->channels;
context->input_frames++;
if (!data)
if (!in)
context->input_frames = FFMAX(context->input_frames, 8);
/* Return the result size */
return ROQ_HEADER_SIZE + data_size;
*got_packet_ptr = 1;
return 0;
}
AVCodec ff_roq_dpcm_encoder = {
@ -196,7 +208,7 @@ AVCodec ff_roq_dpcm_encoder = {
.id = CODEC_ID_ROQ_DPCM,
.priv_data_size = sizeof(ROQDPCMContext),
.init = roq_dpcm_encode_init,
.encode = roq_dpcm_encode_frame,
.encode2 = roq_dpcm_encode_frame,
.close = roq_dpcm_encode_close,
.capabilities = CODEC_CAP_DELAY,
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},