Author: ayoung
Date: Tue Feb 8 03:08:53 2011
New Revision: 9305
URL: http://svn.slimdevices.com/jive?rev=9305&view=rev
Log:
bug 16908: ALAC fixed at 44100/2/16 & poor error handling
Parse number of channels and sample-rate from metadata.
Improve error-handling
Add 24-bit support.
Modified:
7.6/trunk/squeezeplay/src/squeezeplay/src/audio/alac/alac.c
7.6/trunk/squeezeplay/src/squeezeplay/src/audio/alac/alac.h
7.6/trunk/squeezeplay/src/squeezeplay/src/audio/decode/decode_alac.c
Modified: 7.6/trunk/squeezeplay/src/squeezeplay/src/audio/alac/alac.c
URL:
http://svn.slimdevices.com/jive/7.6/trunk/squeezeplay/src/squeezeplay/src/audio/alac/alac.c?rev=9305&r1=9304&r2=9305&view=diff
==============================================================================
--- 7.6/trunk/squeezeplay/src/squeezeplay/src/audio/alac/alac.c (original)
+++ 7.6/trunk/squeezeplay/src/squeezeplay/src/audio/alac/alac.c Tue Feb 8
03:08:53 2011
@@ -104,13 +104,15 @@
* set this to 1 */
int context_initialized;
- int numchannels;
- int bytespersample;
+ unsigned int numchannels;
+ unsigned int bytespersample;
/* buffers */
int32_t *predicterror_buffer[MAX_CHANNELS];
int32_t *outputsamples_buffer[MAX_CHANNELS];
+
+ int32_t *wasted_bits_buffer[MAX_CHANNELS];
/* stuff from setinfo */
uint32_t setinfo_max_samples_per_frame; /* 0x1000 = 4096 */ /* max
samples per frame? */
@@ -120,6 +122,7 @@
uint8_t setinfo_rice_kmodifier; /* 0x0e */
/* end setinfo stuff */
+ int wasted_bits;
} ALACContext;
static void allocate_buffers(ALACContext *alac)
@@ -131,6 +134,8 @@
alac->outputsamples_buffer[chan] =
av_malloc(alac->setinfo_max_samples_per_frame * 4);
+
+ alac->wasted_bits_buffer[chan] =
av_malloc(alac->setinfo_max_samples_per_frame * 4);
}
}
@@ -151,18 +156,14 @@
alac->setinfo_max_samples_per_frame = bytestream_get_be32(&ptr);
ptr++; /* ??? */
alac->setinfo_sample_size = *ptr++;
- if (alac->setinfo_sample_size > 32) {
- av_log(alac->avctx, AV_LOG_ERROR, "setinfo_sample_size too large\n");
- return -1;
- }
alac->setinfo_rice_historymult = *ptr++;
alac->setinfo_rice_initialhistory = *ptr++;
alac->setinfo_rice_kmodifier = *ptr++;
- ptr++; /* channels? */
+ alac->numchannels = *ptr++; /*
channels? */
bytestream_get_be16(&ptr); /* ??? */
bytestream_get_be32(&ptr); /* max coded frame size */
bytestream_get_be32(&ptr); /* bitrate ? */
- bytestream_get_be32(&ptr); /* samplerate */
+ alac->avctx->samplerate = bytestream_get_be32(&ptr); /*
samplerate */
allocate_buffers(alac);
@@ -198,7 +199,7 @@
static void bastardized_rice_decompress(ALACContext *alac,
int32_t *output_buffer,
- int output_size,
+ unsigned int output_size,
int readsamplesize, /* arg_10 */
int rice_initialhistory, /* arg424->b */
int rice_kmodifier, /* arg424->d */
@@ -206,7 +207,7 @@
int rice_kmodifier_mask /* arg424->e */
)
{
- int output_count;
+ unsigned int output_count;
unsigned int history = rice_initialhistory;
int sign_modifier = 0;
@@ -240,7 +241,7 @@
/* special case: there may be compressed blocks of 0 */
if ((history < 128) && (output_count+1 < output_size)) {
int k;
- int block_size;
+ unsigned int block_size;
sign_modifier = 1;
@@ -395,11 +396,11 @@
static void reconstruct_stereo_16(int32_t *buffer[MAX_CHANNELS],
int16_t *buffer_out,
- int numchannels, int numsamples,
+ unsigned int numchannels, unsigned int
numsamples,
uint8_t interlacing_shift,
uint8_t interlacing_leftweight)
{
- int i;
+ unsigned int i;
if (numsamples <= 0)
return;
@@ -433,40 +434,75 @@
}
}
+static void decorrelate_stereo_24(int32_t *buffer[MAX_CHANNELS],
+ int32_t *buffer_out,
+ int32_t *wasted_bits_buffer[MAX_CHANNELS],
+ int wasted_bits,
+ unsigned int numchannels, unsigned int
numsamples,
+ uint8_t interlacing_shift,
+ uint8_t interlacing_leftweight)
+{
+ unsigned int i;
+
+ if (numsamples <= 0)
+ return;
+
+ /* weighted interlacing */
+ if (interlacing_leftweight) {
+ for (i = 0; i < numsamples; i++) {
+ int32_t a, b;
+
+ a = buffer[0][i];
+ b = buffer[1][i];
+
+ a -= (b * interlacing_leftweight) >> interlacing_shift;
+ b += a;
+
+ if (wasted_bits) {
+ b = (b << wasted_bits) | wasted_bits_buffer[0][i];
+ a = (a << wasted_bits) | wasted_bits_buffer[1][i];
+ }
+
+ buffer_out[i * numchannels] = b << 8;
+ buffer_out[i * numchannels + 1] = a << 8;
+ }
+ } else {
+ for (i = 0; i < numsamples; i++) {
+ int32_t left, right;
+
+ left = buffer[0][i];
+ right = buffer[1][i];
+
+ if (wasted_bits) {
+ left = (left << wasted_bits) | wasted_bits_buffer[0][i];
+ right = (right << wasted_bits) | wasted_bits_buffer[1][i];
+ }
+
+ buffer_out[i * numchannels] = left << 8;
+ buffer_out[i * numchannels + 1] = right << 8;
+ }
+ }
+}
+
int alac_decode_frame(AVCodecContext *avctx,
- void *outbuffer, int *outputsize,
+ void *outbuffer, unsigned int *outputsize,
AVPacket *avpkt)
{
const uint8_t *inbuffer = avpkt->data;
int input_buffer_size = avpkt->size;
ALACContext *alac = avctx->priv_data;
- int channels;
- int outputsamples;
+ unsigned int channels;
+ unsigned int outputsamples;
int hassize;
unsigned int readsamplesize;
- int wasted_bytes;
int isnotcompressed;
uint8_t interlacing_shift;
uint8_t interlacing_leftweight;
/* short-circuit null buffers */
if (!inbuffer || !input_buffer_size)
- return input_buffer_size;
-
- /* initialize from the extradata */
- if (!alac->context_initialized) {
- if (alac->avctx->extradata_size != ALAC_EXTRADATA_SIZE) {
- av_log(avctx, AV_LOG_ERROR, "alac: expected %d extradata bytes\n",
- ALAC_EXTRADATA_SIZE);
- return input_buffer_size;
- }
- if (alac_set_info(alac)) {
- av_log(avctx, AV_LOG_ERROR, "alac: set_info failed\n");
- return input_buffer_size;
- }
- alac->context_initialized = 1;
- }
+ return -1;
init_get_bits(&alac->gb, inbuffer, input_buffer_size * 8);
@@ -474,7 +510,7 @@
if (channels > MAX_CHANNELS) {
av_log(avctx, AV_LOG_ERROR, "channels > %d not supported\n",
MAX_CHANNELS);
- return input_buffer_size;
+ return -1;
}
/* 2^result = something to do with output waiting.
@@ -487,7 +523,7 @@
/* the output sample size is stored soon */
hassize = get_bits1(&alac->gb);
- wasted_bytes = get_bits(&alac->gb, 2); /* unknown ? */
+ alac->wasted_bits = get_bits(&alac->gb, 2) << 3;
/* whether the frame is compressed */
isnotcompressed = get_bits1(&alac->gb);
@@ -495,7 +531,7 @@
if (hassize) {
/* now read the number of samples as a 32bit integer */
outputsamples = get_bits_long(&alac->gb, 32);
- if(outputsamples > (int) alac->setinfo_max_samples_per_frame){
+ if(outputsamples > alac->setinfo_max_samples_per_frame){
av_log(avctx, AV_LOG_ERROR, "outputsamples %d > %d\n",
outputsamples, alac->setinfo_max_samples_per_frame);
return -1;
}
@@ -508,7 +544,7 @@
}
*outputsize = outputsamples * alac->bytespersample;
- readsamplesize = alac->setinfo_sample_size - (wasted_bytes * 8) + channels
- 1;
+ readsamplesize = alac->setinfo_sample_size - (alac->wasted_bits) +
channels - 1;
if (readsamplesize > MIN_CACHE_BITS) {
av_log(avctx, AV_LOG_ERROR, "readsamplesize too big (%d)\n",
readsamplesize);
return -1;
@@ -521,12 +557,14 @@
int prediction_type[channels];
int prediction_quantitization[channels];
int ricemodifier[channels];
- int i, chan;
+ unsigned int i, chan;
interlacing_shift = get_bits(&alac->gb, 8);
interlacing_leftweight = get_bits(&alac->gb, 8);
for (chan = 0; chan < channels; chan++) {
+ int i;
+
prediction_type[chan] = get_bits(&alac->gb, 4);
prediction_quantitization[chan] = get_bits(&alac->gb, 4);
@@ -538,9 +576,12 @@
predictor_coef_table[chan][i] = (int16_t)get_bits(&alac->gb,
16);
}
- if (wasted_bytes)
- av_log(avctx, AV_LOG_ERROR, "FIXME: unimplemented, unhandling of
wasted_bytes\n");
-
+ if (alac->wasted_bits) {
+ for (i = 0; i < outputsamples; i++) {
+ for (chan = 0; chan < channels; chan++)
+ alac->wasted_bits_buffer[chan][i] = get_bits(&alac->gb,
alac->wasted_bits);
+ }
+ }
for (chan = 0; chan < channels; chan++) {
bastardized_rice_decompress(alac,
alac->predicterror_buffer[chan],
@@ -572,7 +613,8 @@
}
} else {
/* not compressed, easy case */
- int i, chan;
+ unsigned int i, chan;
+ if (alac->setinfo_sample_size <= 16) {
for (i = 0; i < outputsamples; i++)
for (chan = 0; chan < channels; chan++) {
int32_t audiobits;
@@ -581,7 +623,17 @@
alac->outputsamples_buffer[chan][i] = audiobits;
}
- /* wasted_bytes = 0; */
+ } else {
+ for (i = 0; i < outputsamples; i++) {
+ for (chan = 0; chan < channels; chan++) {
+ alac->outputsamples_buffer[chan][i] = get_bits(&alac->gb,
+
alac->setinfo_sample_size);
+ alac->outputsamples_buffer[chan][i] =
sign_extend(alac->outputsamples_buffer[chan][i],
+
alac->setinfo_sample_size);
+ }
+ }
+ }
+ alac->wasted_bits = 0;
interlacing_shift = 0;
interlacing_leftweight = 0;
}
@@ -598,17 +650,34 @@
interlacing_shift,
interlacing_leftweight);
} else {
- int i;
+ unsigned int i;
for (i = 0; i < outputsamples; i++) {
- int16_t sample = alac->outputsamples_buffer[0][i];
- ((int16_t*)outbuffer)[i * alac->numchannels] = sample;
+ int16_t s = alac->outputsamples_buffer[0][i];
+ ((int16_t*)outbuffer)[i*alac->numchannels] = s;
+ ((int16_t*)outbuffer)[i*alac->numchannels + 1] = s;
}
}
break;
- case 20:
case 24:
- // It is not clear if there exist any encoder that creates 24 bit ALAC
- // files. iTunes convert 24 bit raw files to 16 bit before encoding.
+ if (channels == 2) {
+ decorrelate_stereo_24(alac->outputsamples_buffer,
+ outbuffer,
+ alac->wasted_bits_buffer,
+ alac->wasted_bits,
+ alac->numchannels,
+ outputsamples,
+ interlacing_shift,
+ interlacing_leftweight);
+ } else {
+ unsigned int i;
+ for (i = 0; i < outputsamples; i++) {
+ int32_t s = alac->outputsamples_buffer[0][i] << 8;
+ ((int32_t *)outbuffer)[i*alac->numchannels] = s;
+ ((int32_t *)outbuffer)[i*alac->numchannels + 1] = s;
+ }
+ }
+ break;
+ case 20:
case 32:
av_log(avctx, AV_LOG_ERROR, "FIXME: unimplemented sample size %i\n",
alac->setinfo_sample_size);
break;
@@ -628,9 +697,42 @@
alac->avctx = avctx;
alac->context_initialized = 0;
- alac->numchannels = alac->avctx->channels;
- alac->bytespersample = 2 * alac->numchannels;
- avctx->sample_fmt = SAMPLE_FMT_S16;
+ /* initialize from the extradata */
+ if (!alac->context_initialized) {
+ if (alac->avctx->extradata_size != ALAC_EXTRADATA_SIZE) {
+ av_log(avctx, AV_LOG_ERROR, "alac: expected %d extradata bytes\n",
+ ALAC_EXTRADATA_SIZE);
+ return -1;
+ }
+ if (alac_set_info(alac)) {
+ av_log(avctx, AV_LOG_ERROR, "alac: set_info failed\n");
+ return -1;
+ }
+ alac->context_initialized = 1;
+ }
+
+ if (alac->setinfo_sample_size > 32) {
+ av_log(alac->avctx, AV_LOG_ERROR, "setinfo_sample_size too large\n");
+ return -1;
+ }
+
+ if (alac->numchannels > MAX_CHANNELS) {
+ av_log(avctx, AV_LOG_ERROR, "alac: unsupported number of channels:
%d\n", alac->numchannels);
+ return -1;
+ }
+ alac->avctx->channels = alac->numchannels = 2;
+
+ switch (alac->setinfo_sample_size) {
+ case 16: alac->bytespersample = alac->numchannels << 1;
+ avctx->sample_fmt = SAMPLE_FMT_S16;
+ break;
+ case 24: alac->bytespersample = alac->numchannels << 2;
+ avctx->sample_fmt = SAMPLE_FMT_S32;
+ break;
+ default: av_log(avctx, AV_LOG_ERROR, "Sample depth %d is not supported.\n",
+ alac->setinfo_sample_size);
+ return -1;
+ }
return 0;
}
@@ -641,12 +743,15 @@
int chan;
for (chan = 0; chan < MAX_CHANNELS; chan++) {
- if (alac->predicterror_buffer[chan]) {
- av_free(alac->predicterror_buffer[chan]);
- }
- if (alac->outputsamples_buffer[chan]) {
- av_free(alac->outputsamples_buffer[chan]);
- }
+ if (alac->predicterror_buffer[chan]) {
+ av_free(alac->predicterror_buffer[chan]);
+ }
+ if (alac->outputsamples_buffer[chan]) {
+ av_free(alac->outputsamples_buffer[chan]);
+ }
+ if (alac->wasted_bits_buffer[chan]) {
+ av_free(alac->wasted_bits_buffer[chan]);
+ }
}
return 0;
@@ -657,7 +762,7 @@
#else // SQUEEZEPLAY
AVCodec alac_decoder = {
"alac",
- CODEC_TYPE_AUDIO,
+ AVMEDIA_TYPE_AUDIO,
CODEC_ID_ALAC,
sizeof(ALACContext),
alac_decode_init,
Modified: 7.6/trunk/squeezeplay/src/squeezeplay/src/audio/alac/alac.h
URL:
http://svn.slimdevices.com/jive/7.6/trunk/squeezeplay/src/squeezeplay/src/audio/alac/alac.h?rev=9305&r1=9304&r2=9305&view=diff
==============================================================================
--- 7.6/trunk/squeezeplay/src/squeezeplay/src/audio/alac/alac.h (original)
+++ 7.6/trunk/squeezeplay/src/squeezeplay/src/audio/alac/alac.h Tue Feb 8
03:08:53 2011
@@ -12,6 +12,7 @@
#define INT_BIT 32
#define SAMPLE_FMT_S16 16
+#define SAMPLE_FMT_S32 32
#define av_malloc(X) malloc(X)
#define av_free(X) free(X)
@@ -28,6 +29,7 @@
int channels;
int sample_fmt;
+ int samplerate;
} AVCodecContext;
typedef struct {
@@ -37,7 +39,7 @@
int alac_decode_frame(AVCodecContext *avctx,
- void *outbuffer, int *outputsize,
+ void *outbuffer, unsigned int *outputsize,
AVPacket *avpkt);
int alac_decode_init(AVCodecContext *avctx);
Modified: 7.6/trunk/squeezeplay/src/squeezeplay/src/audio/decode/decode_alac.c
URL:
http://svn.slimdevices.com/jive/7.6/trunk/squeezeplay/src/squeezeplay/src/audio/decode/decode_alac.c?rev=9305&r1=9304&r2=9305&view=diff
==============================================================================
--- 7.6/trunk/squeezeplay/src/squeezeplay/src/audio/decode/decode_alac.c
(original)
+++ 7.6/trunk/squeezeplay/src/squeezeplay/src/audio/decode/decode_alac.c Tue
Feb 8 03:08:53 2011
@@ -42,8 +42,8 @@
struct decode_alac *self = (struct decode_alac *) data;
bool_t streaming;
AVPacket avpkt;
- int outputsize, num;
- s16_t *rptr;
+ unsigned int outputsize;
+ int num;
sample_t *wptr, s;
size_t len;
int i, frames;
@@ -59,26 +59,32 @@
if (status == 2) {
return TRUE; /* need to wait for some more
data */
} else if (status != 1) {
- LOG_DEBUG(log_audio_codec, "mp4_open() failed");
+ LOG_WARN(log_audio_codec, "mp4_open() failed");
current_decoder_state |= DECODE_STATE_ERROR |
DECODE_STATE_NOT_SUPPORTED;
return FALSE;
}
mp4_track_conf(&self->mp4, 0, &conf, &conf_size);
if (!conf) {
- LOG_DEBUG(log_audio_codec, "mp4_track_conf() failed");
+ LOG_WARN(log_audio_codec, "mp4_track_conf() failed");
current_decoder_state |= DECODE_STATE_ERROR |
DECODE_STATE_NOT_SUPPORTED;
return FALSE;
}
-
- self->num_channels = 2; // XXXX
- self->sample_rate = 44100; // XXXX
self->alacdec.channels = self->num_channels;
self->alacdec.extradata = conf + 28;
self->alacdec.extradata_size = conf_size - 28;
- alac_decode_init(&self->alacdec);
+ if (alac_decode_init(&self->alacdec) < 0) {
+ LOG_WARN(log_audio_codec, "alac_decode_init() failed");
+ current_decoder_state |= DECODE_STATE_ERROR;
+ return FALSE;
+ }
+
+ self->sample_rate = self->alacdec.samplerate;
+ self->num_channels = self->alacdec.channels;
+
+ LOG_INFO(log_audio_codec, "sample_rate=%d channels=%d",
sample_rate, num_channels);
self->init = TRUE;
}
@@ -93,34 +99,68 @@
}
}
- outputsize = OUTPUT_BUFFER_SIZE / 2;
+ outputsize = OUTPUT_BUFFER_SIZE;
+ if (self->alacdec.sample_fmt == SAMPLE_FMT_S16) outputsize /= 2;
num = alac_decode_frame(&self->alacdec,
self->output_buffer, &outputsize,
&avpkt);
+ if (num < 0) {
+ LOG_WARN(log_audio_codec, "alac_decode_frame() failed");
+ current_decoder_state |= DECODE_STATE_ERROR;
+ return FALSE;
+ }
+
frames = outputsize / sizeof(u16_t) / self->num_channels;
wptr = ((sample_t *)(void *)self->output_buffer) + (frames * 2);
- if (self->num_channels == 1) {
- /* mono */
- rptr = ((s16_t *)(void *)self->output_buffer) + (frames * 1);
-
- for (i = 0; i < frames; i++) {
- s = (*--rptr) << 16;
- *--wptr = s;
- *--wptr = s;
- }
- }
- else if (self->num_channels == 2) {
- /* stereo */
- rptr = ((s16_t *)(void *)self->output_buffer) + (frames * 2);
-
- for (i = 0; i < frames; i++) {
- *--wptr = (*--rptr) << 16;
- *--wptr = (*--rptr) << 16;
- }
+ switch (self->alacdec.sample_fmt) {
+ case SAMPLE_FMT_S16:
+ if (self->num_channels == 1) {
+ /* mono */
+ s16_t *rptr = ((s16_t *)(void *)self->output_buffer) +
(frames * 1);
+
+ for (i = 0; i < frames; i++) {
+ s = (*--rptr) << 16;
+ *--wptr = s;
+ *--wptr = s;
+ }
+ }
+ else if (self->num_channels == 2) {
+ /* stereo */
+ s16_t *rptr = ((s16_t *)(void *)self->output_buffer) +
(frames * 2);
+
+ for (i = 0; i < frames; i++) {
+ *--wptr = (*--rptr) << 16;
+ *--wptr = (*--rptr) << 16;
+ }
+ }
+ break;
+
+ case SAMPLE_FMT_S32:
+ if (self->num_channels == 1) {
+ /* mono */
+ s32_t *rptr = ((s32_t *)(void *)self->output_buffer) +
(frames * 1);
+
+ for (i = 0; i < frames; i++) {
+ s = *--rptr;
+ *--wptr = s;
+ *--wptr = s;
+ }
+ }
+ else if (self->num_channels == 2) {
+ /* stereo */
+
+ /* nothing to do */
+ }
+ break;
+
+ default:
+ LOG_WARN(log_audio_codec, "unsupported sample format: %d",
self->alacdec.sample_fmt);
+ current_decoder_state |= DECODE_STATE_ERROR |
DECODE_STATE_NOT_SUPPORTED;
+ return FALSE;
}
decode_output_samples(self->output_buffer,
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