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author | Justin Ruggles <justin.ruggles@gmail.com> | 2012-02-03 17:53:41 -0500 |
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committer | Justin Ruggles <justin.ruggles@gmail.com> | 2012-02-11 12:49:22 -0500 |
commit | b6e8ff72ea055f40ee272a97bde3ff21b3ea6c27 (patch) | |
tree | 9ffd9d9ac69a04327853834e453b7472c08c1ca4 /libavcodec | |
parent | b590f3a7bf9103ac7a7a61c48568676201d6824b (diff) | |
download | ffmpeg-streaming-b6e8ff72ea055f40ee272a97bde3ff21b3ea6c27.zip ffmpeg-streaming-b6e8ff72ea055f40ee272a97bde3ff21b3ea6c27.tar.gz |
alacenc: consolidate bitstream writing into a single function.
Simplifies use of verbatim mode.
Diffstat (limited to 'libavcodec')
-rw-r--r-- | libavcodec/alacenc.c | 71 |
1 files changed, 33 insertions, 38 deletions
diff --git a/libavcodec/alacenc.c b/libavcodec/alacenc.c index d2a24b1..332356d 100644 --- a/libavcodec/alacenc.c +++ b/libavcodec/alacenc.c @@ -59,6 +59,7 @@ typedef struct AlacLPCContext { typedef struct AlacEncodeContext { int frame_size; /**< current frame size */ + int verbatim; /**< current frame verbatim mode flag */ int compression_level; int min_prediction_order; int max_prediction_order; @@ -118,7 +119,7 @@ static void encode_scalar(AlacEncodeContext *s, int x, } } -static void write_frame_header(AlacEncodeContext *s, int is_verbatim) +static void write_frame_header(AlacEncodeContext *s) { int encode_fs = 0; @@ -129,7 +130,7 @@ static void write_frame_header(AlacEncodeContext *s, int is_verbatim) put_bits(&s->pbctx, 16, 0); // Seems to be zero put_bits(&s->pbctx, 1, encode_fs); // Sample count is in the header put_bits(&s->pbctx, 2, 0); // FIXME: Wasted bytes field - put_bits(&s->pbctx, 1, is_verbatim); // Audio block is verbatim + put_bits(&s->pbctx, 1, s->verbatim); // Audio block is verbatim if (encode_fs) put_bits32(&s->pbctx, s->frame_size); // No. of samples in the frame } @@ -345,27 +346,39 @@ static void alac_entropy_coder(AlacEncodeContext *s) } } -static void write_compressed_frame(AlacEncodeContext *s) +static int write_frame(AlacEncodeContext *s, uint8_t *data, int size, + const int16_t *samples) { int i, j; int prediction_type = 0; + PutBitContext *pb = &s->pbctx; + + init_put_bits(pb, data, size); + + if (s->verbatim) { + write_frame_header(s); + for (i = 0; i < s->frame_size * s->avctx->channels; i++) + put_sbits(pb, 16, *samples++); + } else { + init_sample_buffers(s, samples); + write_frame_header(s); if (s->avctx->channels == 2) alac_stereo_decorrelation(s); - put_bits(&s->pbctx, 8, s->interlacing_shift); - put_bits(&s->pbctx, 8, s->interlacing_leftweight); + put_bits(pb, 8, s->interlacing_shift); + put_bits(pb, 8, s->interlacing_leftweight); for (i = 0; i < s->avctx->channels; i++) { calc_predictor_params(s, i); - put_bits(&s->pbctx, 4, prediction_type); - put_bits(&s->pbctx, 4, s->lpc[i].lpc_quant); + put_bits(pb, 4, prediction_type); + put_bits(pb, 4, s->lpc[i].lpc_quant); - put_bits(&s->pbctx, 3, s->rc.rice_modifier); - put_bits(&s->pbctx, 5, s->lpc[i].lpc_order); + put_bits(pb, 3, s->rc.rice_modifier); + put_bits(pb, 5, s->lpc[i].lpc_order); // predictor coeff. table for (j = 0; j < s->lpc[i].lpc_order; j++) - put_sbits(&s->pbctx, 16, s->lpc[i].lpc_coeff[j]); + put_sbits(pb, 16, s->lpc[i].lpc_coeff[j]); } // apply lpc and entropy coding to audio samples @@ -382,6 +395,10 @@ static void write_compressed_frame(AlacEncodeContext *s) alac_entropy_coder(s); } + } + put_bits(pb, 3, 7); + flush_put_bits(pb); + return put_bits_count(pb) >> 3; } static av_always_inline int get_max_frame_size(int frame_size, int ch, int bps) @@ -523,9 +540,7 @@ static int alac_encode_frame(AVCodecContext *avctx, uint8_t *frame, int buf_size, void *data) { AlacEncodeContext *s = avctx->priv_data; - PutBitContext *pb = &s->pbctx; - int i, out_bytes, verbatim_flag = 0; - int max_frame_size; + int out_bytes, max_frame_size; s->frame_size = avctx->frame_size; @@ -540,35 +555,15 @@ static int alac_encode_frame(AVCodecContext *avctx, uint8_t *frame, return AVERROR(EINVAL); } -verbatim: - init_put_bits(pb, frame, buf_size); + /* use verbatim mode for compression_level 0 */ + s->verbatim = !s->compression_level; - if (s->compression_level == 0 || verbatim_flag) { - // Verbatim mode - const int16_t *samples = data; - write_frame_header(s, 1); - for (i = 0; i < s->frame_size * avctx->channels; i++) { - put_sbits(pb, 16, *samples++); - } - } else { - init_sample_buffers(s, data); - write_frame_header(s, 0); - write_compressed_frame(s); - } - - put_bits(pb, 3, 7); - flush_put_bits(pb); - out_bytes = put_bits_count(pb) >> 3; + out_bytes = write_frame(s, frame, buf_size, data); if (out_bytes > max_frame_size) { /* frame too large. use verbatim mode */ - if (verbatim_flag || s->compression_level == 0) { - /* still too large. must be an error. */ - av_log(avctx, AV_LOG_ERROR, "error encoding frame\n"); - return AVERROR_BUG; - } - verbatim_flag = 1; - goto verbatim; + s->verbatim = 1; + out_bytes = write_frame(s, frame, buf_size, data); } return out_bytes; |