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author | Justin Ruggles <justin.ruggles@gmail.com> | 2011-09-06 12:17:45 -0400 |
---|---|---|
committer | Justin Ruggles <justin.ruggles@gmail.com> | 2011-12-02 17:40:40 -0500 |
commit | 0eea212943544d40f99b05571aa7159d78667154 (patch) | |
tree | 1e6b0271a633bf8a3f92c78bdfbaca275498ee26 /libavcodec/s302m.c | |
parent | 560f773c7ddd17f66e2621222980c1359a9027be (diff) | |
download | ffmpeg-0eea212943544d40f99b05571aa7159d78667154.tar.gz |
Add avcodec_decode_audio4().
Deprecate avcodec_decode_audio3().
Implement audio support in avcodec_default_get_buffer().
Implement the new audio decoder API in all audio decoders.
Diffstat (limited to 'libavcodec/s302m.c')
-rw-r--r-- | libavcodec/s302m.c | 46 |
1 files changed, 36 insertions, 10 deletions
diff --git a/libavcodec/s302m.c b/libavcodec/s302m.c index f6f096d89f..34018aeb46 100644 --- a/libavcodec/s302m.c +++ b/libavcodec/s302m.c @@ -25,6 +25,10 @@ #define AES3_HEADER_LEN 4 +typedef struct S302MDecodeContext { + AVFrame frame; +} S302MDecodeContext; + static int s302m_parse_frame_header(AVCodecContext *avctx, const uint8_t *buf, int buf_size) { @@ -73,10 +77,12 @@ static int s302m_parse_frame_header(AVCodecContext *avctx, const uint8_t *buf, } static int s302m_decode_frame(AVCodecContext *avctx, void *data, - int *data_size, AVPacket *avpkt) + int *got_frame_ptr, AVPacket *avpkt) { + S302MDecodeContext *s = avctx->priv_data; const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; + int block_size, ret; int frame_size = s302m_parse_frame_header(avctx, buf, buf_size); if (frame_size < 0) @@ -85,11 +91,18 @@ static int s302m_decode_frame(AVCodecContext *avctx, void *data, buf_size -= AES3_HEADER_LEN; buf += AES3_HEADER_LEN; - if (*data_size < 4 * buf_size * 8 / (avctx->bits_per_coded_sample + 4)) - return -1; + /* get output buffer */ + block_size = (avctx->bits_per_coded_sample + 4) / 4; + s->frame.nb_samples = 2 * (buf_size / block_size) / avctx->channels; + if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; + } + + buf_size = (s->frame.nb_samples * avctx->channels / 2) * block_size; if (avctx->bits_per_coded_sample == 24) { - uint32_t *o = data; + uint32_t *o = (uint32_t *)s->frame.data[0]; for (; buf_size > 6; buf_size -= 7) { *o++ = (av_reverse[buf[2]] << 24) | (av_reverse[buf[1]] << 16) | @@ -100,9 +113,8 @@ static int s302m_decode_frame(AVCodecContext *avctx, void *data, (av_reverse[buf[3] & 0x0f] << 4); buf += 7; } - *data_size = (uint8_t*) o - (uint8_t*) data; } else if (avctx->bits_per_coded_sample == 20) { - uint32_t *o = data; + uint32_t *o = (uint32_t *)s->frame.data[0]; for (; buf_size > 5; buf_size -= 6) { *o++ = (av_reverse[buf[2] & 0xf0] << 28) | (av_reverse[buf[1]] << 20) | @@ -112,9 +124,8 @@ static int s302m_decode_frame(AVCodecContext *avctx, void *data, (av_reverse[buf[3]] << 12); buf += 6; } - *data_size = (uint8_t*) o - (uint8_t*) data; } else { - uint16_t *o = data; + uint16_t *o = (uint16_t *)s->frame.data[0]; for (; buf_size > 4; buf_size -= 5) { *o++ = (av_reverse[buf[1]] << 8) | av_reverse[buf[0]]; @@ -123,10 +134,22 @@ static int s302m_decode_frame(AVCodecContext *avctx, void *data, (av_reverse[buf[2]] >> 4); buf += 5; } - *data_size = (uint8_t*) o - (uint8_t*) data; } - return buf - avpkt->data; + *got_frame_ptr = 1; + *(AVFrame *)data = s->frame; + + return avpkt->size; +} + +static int s302m_decode_init(AVCodecContext *avctx) +{ + S302MDecodeContext *s = avctx->priv_data; + + avcodec_get_frame_defaults(&s->frame); + avctx->coded_frame = &s->frame; + + return 0; } @@ -134,6 +157,9 @@ AVCodec ff_s302m_decoder = { .name = "s302m", .type = AVMEDIA_TYPE_AUDIO, .id = CODEC_ID_S302M, + .priv_data_size = sizeof(S302MDecodeContext), + .init = s302m_decode_init, .decode = s302m_decode_frame, + .capabilities = CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("SMPTE 302M"), }; |