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authorRamiro Polla <ramiro.polla@gmail.com>2008-08-17 22:47:40 +0000
committerRamiro Polla <ramiro.polla@gmail.com>2008-08-17 22:47:40 +0000
commitca048266273a8964d7c7c168bd0255236f460961 (patch)
tree3455516020de4e4d34ee7dc48ae12d8261e70ee3 /libavcodec/alacenc.c
parent46dd2738ae321adcc36cc9137b6a978e2e007733 (diff)
downloadffmpeg-ca048266273a8964d7c7c168bd0255236f460961.tar.gz
Import more ok'd parts of ALAC encoder from GSoC repo.
Originally committed as revision 14820 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavcodec/alacenc.c')
-rw-r--r--libavcodec/alacenc.c140
1 files changed, 125 insertions, 15 deletions
diff --git a/libavcodec/alacenc.c b/libavcodec/alacenc.c
index 98a3451597..0a802f1525 100644
--- a/libavcodec/alacenc.c
+++ b/libavcodec/alacenc.c
@@ -33,15 +33,52 @@
#define ALAC_ESCAPE_CODE 0x1FF
#define ALAC_MAX_LPC_ORDER 30
-
+#define DEFAULT_MAX_PRED_ORDER 6
+#define DEFAULT_MIN_PRED_ORDER 4
+#define ALAC_MAX_LPC_PRECISION 9
+#define ALAC_MAX_LPC_SHIFT 9
+
+typedef struct RiceContext {
+ int history_mult;
+ int initial_history;
+ int k_modifier;
+ int rice_modifier;
+} RiceContext;
+
+typedef struct LPCContext {
+ int lpc_order;
+ int lpc_coeff[ALAC_MAX_LPC_ORDER+1];
+ int lpc_quant;
+} LPCContext;
+
+typedef struct AlacEncodeContext {
+ int compression_level;
+ int max_coded_frame_size;
+ int write_sample_size;
+ int32_t sample_buf[MAX_CHANNELS][DEFAULT_FRAME_SIZE];
int interlacing_shift;
int interlacing_leftweight;
PutBitContext pbctx;
+ RiceContext rc;
+ LPCContext lpc[MAX_CHANNELS];
DSPContext dspctx;
AVCodecContext *avctx;
} AlacEncodeContext;
+static void init_sample_buffers(AlacEncodeContext *s, int16_t *input_samples)
+{
+ int ch, i;
+
+ for(ch=0;ch<s->avctx->channels;ch++) {
+ int16_t *sptr = input_samples + ch;
+ for(i=0;i<s->avctx->frame_size;i++) {
+ s->sample_buf[ch][i] = *sptr;
+ sptr += s->avctx->channels;
+ }
+ }
+}
+
static void encode_scalar(AlacEncodeContext *s, int x, int k, int write_sample_size)
{
int divisor, q, r;
@@ -71,7 +108,7 @@ static void encode_scalar(AlacEncodeContext *s, int x, int k, int write_sample_s
static void write_frame_header(AlacEncodeContext *s, int is_verbatim)
{
- put_bits(&s->pbctx, 3, s->channels-1); // No. of channels -1
+ put_bits(&s->pbctx, 3, s->avctx->channels-1); // No. of channels -1
put_bits(&s->pbctx, 16, 0); // Seems to be zero
put_bits(&s->pbctx, 1, 1); // Sample count is in the header
put_bits(&s->pbctx, 2, 0); // FIXME: Wasted bytes field
@@ -79,6 +116,38 @@ static void write_frame_header(AlacEncodeContext *s, int is_verbatim)
put_bits(&s->pbctx, 32, s->avctx->frame_size); // No. of samples in the frame
}
+static int estimate_stereo_mode(int32_t *left_ch, int32_t *right_ch, int n)
+{
+ int i, best;
+ int32_t lt, rt;
+ uint64_t sum[4];
+ uint64_t score[4];
+
+ /* calculate sum of 2nd order residual for each channel */
+ sum[0] = sum[1] = sum[2] = sum[3] = 0;
+ for(i=2; i<n; i++) {
+ lt = left_ch[i] - 2*left_ch[i-1] + left_ch[i-2];
+ rt = right_ch[i] - 2*right_ch[i-1] + right_ch[i-2];
+ sum[2] += FFABS((lt + rt) >> 1);
+ sum[3] += FFABS(lt - rt);
+ sum[0] += FFABS(lt);
+ sum[1] += FFABS(rt);
+ }
+
+ /* calculate score for each mode */
+ score[0] = sum[0] + sum[1];
+ score[1] = sum[0] + sum[3];
+ score[2] = sum[1] + sum[3];
+ score[3] = sum[2] + sum[3];
+
+ /* return mode with lowest score */
+ best = 0;
+ for(i=1; i<4; i++) {
+ if(score[i] < score[best]) {
+ best = i;
+ }
+ }
+
static void write_compressed_frame(AlacEncodeContext *s)
{
int i, j;
@@ -88,7 +157,7 @@ static void write_compressed_frame(AlacEncodeContext *s)
put_bits(&s->pbctx, 8, s->interlacing_shift);
put_bits(&s->pbctx, 8, s->interlacing_leftweight);
- for(i=0;i<s->channels;i++) {
+ for(i=0;i<s->avctx->channels;i++) {
calc_predictor_params(s, i);
@@ -105,7 +174,7 @@ static void write_compressed_frame(AlacEncodeContext *s)
// apply lpc and entropy coding to audio samples
- for(i=0;i<s->channels;i++) {
+ for(i=0;i<s->avctx->channels;i++) {
alac_linear_predictor(s, i);
alac_entropy_coder(s);
}
@@ -118,8 +187,6 @@ static av_cold int alac_encode_init(AVCodecContext *avctx)
avctx->frame_size = DEFAULT_FRAME_SIZE;
avctx->bits_per_sample = DEFAULT_SAMPLE_SIZE;
- s->channels = avctx->channels;
- s->samplerate = avctx->sample_rate;
if(avctx->sample_fmt != SAMPLE_FMT_S16) {
av_log(avctx, AV_LOG_ERROR, "only pcm_s16 input samples are supported\n");
@@ -139,18 +206,18 @@ static av_cold int alac_encode_init(AVCodecContext *avctx)
s->rc.rice_modifier = 4;
s->max_coded_frame_size = (ALAC_FRAME_HEADER_SIZE + ALAC_FRAME_FOOTER_SIZE +
- avctx->frame_size*s->channels*avctx->bits_per_sample)>>3;
+ avctx->frame_size*avctx->channels*avctx->bits_per_sample)>>3;
- s->write_sample_size = avctx->bits_per_sample + s->channels - 1; // FIXME: consider wasted_bytes
+ s->write_sample_size = avctx->bits_per_sample + avctx->channels - 1; // FIXME: consider wasted_bytes
AV_WB32(alac_extradata, ALAC_EXTRADATA_SIZE);
AV_WB32(alac_extradata+4, MKBETAG('a','l','a','c'));
AV_WB32(alac_extradata+12, avctx->frame_size);
AV_WB8 (alac_extradata+17, avctx->bits_per_sample);
- AV_WB8 (alac_extradata+21, s->channels);
+ AV_WB8 (alac_extradata+21, avctx->channels);
AV_WB32(alac_extradata+24, s->max_coded_frame_size);
- AV_WB32(alac_extradata+28, s->samplerate*s->channels*avctx->bits_per_sample); // average bitrate
- AV_WB32(alac_extradata+32, s->samplerate);
+ AV_WB32(alac_extradata+28, avctx->sample_rate*avctx->channels*avctx->bits_per_sample); // average bitrate
+ AV_WB32(alac_extradata+32, avctx->sample_rate);
// Set relevant extradata fields
if(s->compression_level > 0) {
@@ -168,19 +235,62 @@ static av_cold int alac_encode_init(AVCodecContext *avctx)
s->avctx = avctx;
dsputil_init(&s->dspctx, avctx);
- allocate_sample_buffers(s);
-
return 0;
}
-static av_cold int alac_encode_close(AVCodecContext *avctx)
+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;
+ if(avctx->frame_size > DEFAULT_FRAME_SIZE) {
+ av_log(avctx, AV_LOG_ERROR, "input frame size exceeded\n");
+ return -1;
+ }
+
+ if(buf_size < 2*s->max_coded_frame_size) {
+ av_log(avctx, AV_LOG_ERROR, "buffer size is too small\n");
+ return -1;
+ }
+
+ if((s->compression_level == 0) || verbatim_flag) {
+ // Verbatim mode
+ int16_t *samples = data;
+ write_frame_header(s, 1);
+ for(i=0; i<avctx->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;
+
+ if(out_bytes > s->max_coded_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 -1;
+ }
+ verbatim_flag = 1;
+ goto verbatim;
+ }
+
+ return out_bytes;
+}
+
+static av_cold int alac_encode_close(AVCodecContext *avctx)
+{
av_freep(&avctx->extradata);
avctx->extradata_size = 0;
av_freep(&avctx->coded_frame);
- free_sample_buffers(s);
return 0;
}