Files
shairport-sync/alac.c
T
Mike Brady b405c0b896 Version 5.0 Major Release.
New Features:
Multi-Channel and High-Resolution Audio Support
48,000 frames per second ("48k") operation.
48k lossless stereo support.
5.1 and 7.1 surround sound support.
Multi-channel and multi-rate operation on ALSA, PipeWire, PulseAudio, FreeBSD, stdout and Unix pipe output backends.
Automatic Audio Format Selection
Flexible and controllable output format selection.
Automatic rate, sample format, and channel count selection.
Full FFmpeg Integration
Support for transcoding.
Advanced resampling capabilities.
New audio format support.
Enhanced Resampling
New vernier resampling and interpolation method optimized for low-power CPUs.
Better performance on resource-constrained devices.

Convolution and Loudness Enhancements:
Convolution system is now multithreaded and works on stereo and multichannel audio at 48k and 44.1k.
Multiple impulse response (IR) files can now be provided via convolution_ir_files setting.
New convolution_thread_pool_size setting for multithreaded processing (defaults to 1).
Loudness processing now works with stereo and multichannel audio at 48k and 44.1k.
Updated to the most recent HiFi-LoFi FFT convolver.

MQTT Enhancements:
Added new publish_retain boolean option. When enabled, published MQTT messages have the retain flag set, so the MQTT broker stores the last message per topic and new subscribers receive the most recent value immediately. Thanks to lululombard for PR #2142.

D-Bus Enhancements:
Added new dbus_default_message_bus command-line argument (can be system or session) to set the default message bus for both D-Bus native service and MPRIS service.

Performance Improvements:
Enhanced compatibility with AirPlay 2 AutoMix and Smart Tracklists resulting in less unexplained track skipping.
Better operation on low-power devices down to Raspberry Pi B.
Improved efficiency on embedded systems.
Enhanced timestamp handling for better synchronization.
Improved sync error calculation.
Rebuilt buffered audio processor for cleaner handling of immediate and deferred flush requests.

Docker Enhancements:
Reduced Docker image sizes with slimmed-down FFmpeg library.
Removed dhclient from Docker images for smaller footprint.

Bug Fixes:
Fixed MQTT warning on service startup: "Could not establish a mqtt connection". The startup script now correctly states that the mosquitto service is required. Thanks to Hugo Villeneuve for PR #2137.
Fixed compatibility with mbedtls library version 3.4+ (present on recent Linux versions). Thanks to Christian Beier for finding and fixing the bug.
Fixed PulseAudio backend so that PA_ERR_NODATA returns "No latency data yet". Thanks to Vladimir Shakov for the report and fix.
Ensured old flush requests are deleted when a new play session starts. Thanks to saujanyashah for the report.
Fixed format warnings on 64-bit and 32-bit systems
Removed compilation warnings on 32-bit builds
Improved argument checking for debug(), inform(), warn() and die() functions
Fixed "daemon" typos throughout codebase. Thanks to Chris Boot for PR #1981.
Added warning if a convolution impulse response file cannot be read due to bad path or permissions

Build System Improvements:
Unified service file with variable substitution for Avahi support, making it easier to add future service dependencies. Thanks to Hugo Villeneuve.
Network interface selection now only considers interfaces that are up, running and not loopback interfaces. Thanks to Carl Johnson for the suggestion.
Configuration File Changes and Deprecations

New settings: convolution_ir_files (replaces convolution_ir_file), convolution_enabled (replaces convolution), convolution_max_length_in_seconds (replaces convolution_max_length), loudness_enabled (replaces loudness).
New convolution_thread_pool_size setting (defaults to 1).
Deprecated settings: convolution_ir_file, convolution, convolution_max_length, loudness.
Corresponding D-Bus methods and properties have been updated.

Deprecation Notice:
The Jack Audio and soundio backends are deprecated and will be removed in a future release. Consider using the updated PipeWire backend instead.

Documentation Updates
Updated BUILD.md with latest build instructions.
Updated AIRPLAY2.md with feature information.
Enhanced convolution and loudness documentation.

Maintenance:
Fixed FFmpeg deprecation warnings.
Bumped actions/checkout from 6.0.1 to 6.0.2.
Bumped docker/login-action from 3.6.0 to 3.7.0.
Bumped docker/build-push-action from 6.13.0 to 6.15.0.
Bumped docker/setup-qemu-action from 3.4.0 to 3.6.0.
Bumped docker/setup-buildx-action from 3.9.0 to 3.10.0.
2026-02-13 15:17:40 +00:00

1011 lines
31 KiB
C

/*
* ALAC (Apple Lossless Audio Codec) decoder
* Copyright (c) 2005 David Hammerton
* All rights reserved.
*
* This is the actual decoder.
*
* http://crazney.net/programs/itunes/alac.html
*
* Permission is hereby granted, free of charge, to any person
* obtaining a copy of this software and associated documentation
* files (the "Software"), to deal in the Software without
* restriction, including without limitation the rights to use,
* copy, modify, merge, publish, distribute, sublicense, and/or
* sell copies of the Software, and to permit persons to whom the
* Software is furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES
* OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND
* NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT
* HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY,
* WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
* FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR
* OTHER DEALINGS IN THE SOFTWARE.
*
*/
static const int host_bigendian = 0;
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#ifdef _WIN32
#include "stdint_win.h"
#else
#include <stdint.h>
#endif
#include "alac.h"
#define _Swap32(v) \
do { \
v = (((v)&0x000000FF) << 0x18) | (((v)&0x0000FF00) << 0x08) | (((v)&0x00FF0000) >> 0x08) | \
(((v)&0xFF000000) >> 0x18); \
} while (0)
#define _Swap16(v) \
do { \
v = (((v)&0x00FF) << 0x08) | (((v)&0xFF00) >> 0x08); \
} while (0)
struct {
signed int x : 24;
} se_struct_24;
#define SignExtend24(val) (se_struct_24.x = val)
void alac_free(alac_file *alac) {
if (alac->predicterror_buffer_a)
free(alac->predicterror_buffer_a);
if (alac->predicterror_buffer_b)
free(alac->predicterror_buffer_b);
if (alac->outputsamples_buffer_a)
free(alac->outputsamples_buffer_a);
if (alac->outputsamples_buffer_b)
free(alac->outputsamples_buffer_b);
if (alac->uncompressed_bytes_buffer_a)
free(alac->uncompressed_bytes_buffer_a);
if (alac->uncompressed_bytes_buffer_b)
free(alac->uncompressed_bytes_buffer_b);
free(alac);
}
void alac_allocate_buffers(alac_file *alac) {
alac->predicterror_buffer_a = malloc(alac->setinfo_max_samples_per_frame * 4);
alac->predicterror_buffer_b = malloc(alac->setinfo_max_samples_per_frame * 4);
alac->outputsamples_buffer_a = malloc(alac->setinfo_max_samples_per_frame * 4);
alac->outputsamples_buffer_b = malloc(alac->setinfo_max_samples_per_frame * 4);
alac->uncompressed_bytes_buffer_a = malloc(alac->setinfo_max_samples_per_frame * 4);
alac->uncompressed_bytes_buffer_b = malloc(alac->setinfo_max_samples_per_frame * 4);
}
void alac_set_info(alac_file *alac, char *inputbuffer) {
char *ptr = inputbuffer;
ptr += 4; /* size */
ptr += 4; /* frma */
ptr += 4; /* alac */
ptr += 4; /* size */
ptr += 4; /* alac */
ptr += 4; /* 0 ? */
alac->setinfo_max_samples_per_frame = *(uint32_t *)ptr; /* buffer size / 2 ? */
if (!host_bigendian)
_Swap32(alac->setinfo_max_samples_per_frame);
ptr += 4;
alac->setinfo_7a = *(uint8_t *)ptr;
ptr += 1;
alac->setinfo_sample_size = *(uint8_t *)ptr;
ptr += 1;
alac->setinfo_rice_historymult = *(uint8_t *)ptr;
ptr += 1;
alac->setinfo_rice_initialhistory = *(uint8_t *)ptr;
ptr += 1;
alac->setinfo_rice_kmodifier = *(uint8_t *)ptr;
ptr += 1;
alac->setinfo_7f = *(uint8_t *)ptr;
ptr += 1;
alac->setinfo_80 = *(uint16_t *)ptr;
if (!host_bigendian)
_Swap16(alac->setinfo_80);
ptr += 2;
alac->setinfo_82 = *(uint32_t *)ptr;
if (!host_bigendian)
_Swap32(alac->setinfo_82);
ptr += 4;
alac->setinfo_86 = *(uint32_t *)ptr;
if (!host_bigendian)
_Swap32(alac->setinfo_86);
ptr += 4;
alac->setinfo_8a_rate = *(uint32_t *)ptr;
if (!host_bigendian)
_Swap32(alac->setinfo_8a_rate);
alac_allocate_buffers(alac);
}
/* stream reading */
/* supports reading 1 to 16 bits, in big endian format */
static uint32_t readbits_16(alac_file *alac, int bits) {
uint32_t result;
int new_accumulator;
result = (alac->input_buffer[0] << 16) | (alac->input_buffer[1] << 8) | (alac->input_buffer[2]);
/* shift left by the number of bits we've already read,
* so that the top 'n' bits of the 24 bits we read will
* be the return bits */
result = result << alac->input_buffer_bitaccumulator;
result = result & 0x00ffffff;
/* and then only want the top 'n' bits from that, where
* n is 'bits' */
result = result >> (24 - bits);
new_accumulator = (alac->input_buffer_bitaccumulator + bits);
/* increase the buffer pointer if we've read over n bytes. */
alac->input_buffer += (new_accumulator >> 3);
/* and the remainder goes back into the bit accumulator */
alac->input_buffer_bitaccumulator = (new_accumulator & 7);
return result;
}
/* supports reading 1 to 32 bits, in big endian format */
static uint32_t readbits(alac_file *alac, int bits) {
int32_t result = 0;
if (bits > 16) {
bits -= 16;
result = readbits_16(alac, 16) << bits;
}
result |= readbits_16(alac, bits);
return result;
}
/* reads a single bit */
static int readbit(alac_file *alac) {
int result;
int new_accumulator;
result = alac->input_buffer[0];
result = result << alac->input_buffer_bitaccumulator;
result = result >> 7 & 1;
new_accumulator = (alac->input_buffer_bitaccumulator + 1);
alac->input_buffer += (new_accumulator / 8);
alac->input_buffer_bitaccumulator = (new_accumulator % 8);
return result;
}
static void unreadbits(alac_file *alac, int bits) {
int new_accumulator = (alac->input_buffer_bitaccumulator - bits);
alac->input_buffer += (new_accumulator >> 3);
alac->input_buffer_bitaccumulator = (new_accumulator & 7);
if (alac->input_buffer_bitaccumulator < 0)
alac->input_buffer_bitaccumulator *= -1;
}
/* various implementations of count_leading_zero:
* the first one is the original one, the simplest and most
* obvious for what it's doing. never use this.
* then there are the asm ones. fill in as necessary
* and finally an unrolled and optimised c version
* to fall back to
*/
#if 0
/* hideously inefficient. could use a bitmask search,
* alternatively bsr on x86,
*/
static int count_leading_zeros(int32_t input)
{
int i = 0;
while (!(0x80000000 & input) && i < 32)
{
i++;
input = input << 1;
}
return i;
}
#elif defined(__GNUC__)
/* for some reason the unrolled version (below) is
* actually faster than this. yay intel!
*/
static int count_leading_zeros(int input) { return __builtin_clz(input); }
#elif defined(_MSC_VER) && defined(_M_IX86)
static int count_leading_zeros(int input) {
int output = 0;
if (!input)
return 32;
__asm
{
mov eax, input;
mov edx, 0x1f;
bsr ecx, eax;
sub edx, ecx;
mov output, edx;
}
return output;
}
#else
#warning using generic count leading zeroes. You may wish to write one for your CPU / compiler
static int count_leading_zeros(int input) {
int output = 0;
int curbyte = 0;
curbyte = input >> 24;
if (curbyte)
goto found;
output += 8;
curbyte = input >> 16;
if (curbyte & 0xff)
goto found;
output += 8;
curbyte = input >> 8;
if (curbyte & 0xff)
goto found;
output += 8;
curbyte = input;
if (curbyte & 0xff)
goto found;
output += 8;
return output;
found:
if (!(curbyte & 0xf0)) {
output += 4;
} else
curbyte >>= 4;
if (curbyte & 0x8)
return output;
if (curbyte & 0x4)
return output + 1;
if (curbyte & 0x2)
return output + 2;
if (curbyte & 0x1)
return output + 3;
/* shouldn't get here: */
return output + 4;
}
#endif
#define RICE_THRESHOLD 8 // maximum number of bits for a rice prefix.
static int32_t entropy_decode_value(alac_file *alac, int readSampleSize, int k,
int rice_kmodifier_mask) {
int32_t x = 0; // decoded value
// read x, number of 1s before 0 represent the rice value.
while (x <= RICE_THRESHOLD && readbit(alac)) {
x++;
}
if (x > RICE_THRESHOLD) {
// read the number from the bit stream (raw value)
int32_t value;
value = readbits(alac, readSampleSize);
// mask value
value &= (((uint32_t)0xffffffff) >> (32 - readSampleSize));
x = value;
} else {
if (k != 1) {
int extraBits = readbits(alac, k);
// x = x * (2^k - 1)
x *= (((1 << k) - 1) & rice_kmodifier_mask);
if (extraBits > 1)
x += extraBits - 1;
else
unreadbits(alac, 1);
}
}
return x;
}
static void entropy_rice_decode(alac_file *alac, int32_t *outputBuffer, int outputSize,
int readSampleSize, int rice_initialhistory, int rice_kmodifier,
int rice_historymult, int rice_kmodifier_mask) {
int outputCount;
int history = rice_initialhistory;
int signModifier = 0;
for (outputCount = 0; outputCount < outputSize; outputCount++) {
int32_t decodedValue;
int32_t finalValue;
int32_t k;
k = 31 - rice_kmodifier - count_leading_zeros((history >> 9) + 3);
if (k < 0)
k += rice_kmodifier;
else
k = rice_kmodifier;
// note: don't use rice_kmodifier_mask here (set mask to 0xFFFFFFFF)
decodedValue = entropy_decode_value(alac, readSampleSize, k, 0xFFFFFFFF);
decodedValue += signModifier;
finalValue = (decodedValue + 1) / 2; // inc by 1 and shift out sign bit
if (decodedValue & 1) // the sign is stored in the low bit
finalValue *= -1;
outputBuffer[outputCount] = finalValue;
signModifier = 0;
// update history
history += (decodedValue * rice_historymult) - ((history * rice_historymult) >> 9);
if (decodedValue > 0xFFFF)
history = 0xFFFF;
// special case, for compressed blocks of 0
if ((history < 128) && (outputCount + 1 < outputSize)) {
int32_t blockSize;
signModifier = 1;
k = count_leading_zeros(history) + ((history + 16) / 64) - 24;
// note: blockSize is always 16bit
blockSize = entropy_decode_value(alac, 16, k, rice_kmodifier_mask);
// got blockSize 0s
if (blockSize > 0) {
memset(&outputBuffer[outputCount + 1], 0, blockSize * sizeof(*outputBuffer));
outputCount += blockSize;
}
if (blockSize > 0xFFFF)
signModifier = 0;
history = 0;
}
}
}
#define SIGN_EXTENDED32(val, bits) ((val << (32 - bits)) >> (32 - bits))
#define SIGN_ONLY(v) ((v < 0) ? (-1) : ((v > 0) ? (1) : (0)))
static void predictor_decompress_fir_adapt(int32_t *error_buffer, int32_t *buffer_out,
int output_size, int readsamplesize,
int16_t *predictor_coef_table, int predictor_coef_num,
int predictor_quantitization) {
int i;
/* first sample always copies */
*buffer_out = *error_buffer;
if (!predictor_coef_num) {
if (output_size <= 1)
return;
memcpy(buffer_out + 1, error_buffer + 1, (output_size - 1) * 4);
return;
}
if (predictor_coef_num == 0x1f) /* 11111 - max value of predictor_coef_num */
{ /* second-best case scenario for fir decompression,
* error describes a small difference from the previous sample only
*/
if (output_size <= 1)
return;
for (i = 0; i < output_size - 1; i++) {
int32_t prev_value;
int32_t error_value;
prev_value = buffer_out[i];
error_value = error_buffer[i + 1];
buffer_out[i + 1] = SIGN_EXTENDED32((prev_value + error_value), readsamplesize);
}
return;
}
/* read warm-up samples */
if (predictor_coef_num > 0) {
int li;
for (li = 0; li < predictor_coef_num; li++) {
int32_t val;
val = buffer_out[li] + error_buffer[li + 1];
val = SIGN_EXTENDED32(val, readsamplesize);
buffer_out[li + 1] = val;
}
}
#if 0
/* 4 and 8 are very common cases (the only ones i've seen). these
* should be unrolled and optimised
*/
if (predictor_coef_num == 4)
{
/* FIXME: optimised general case */
return;
}
if (predictor_coef_table == 8)
{
/* FIXME: optimised general case */
return;
}
#endif
/* general case */
if (predictor_coef_num > 0) {
for (i = predictor_coef_num + 1; i < output_size; i++) {
int j;
int sum = 0;
int outval;
int error_val = error_buffer[i];
for (j = 0; j < predictor_coef_num; j++) {
sum += (buffer_out[predictor_coef_num - j] - buffer_out[0]) * predictor_coef_table[j];
}
outval = (1 << (predictor_quantitization - 1)) + sum;
outval = outval >> predictor_quantitization;
outval = outval + buffer_out[0] + error_val;
outval = SIGN_EXTENDED32(outval, readsamplesize);
buffer_out[predictor_coef_num + 1] = outval;
if (error_val > 0) {
int predictor_num = predictor_coef_num - 1;
while (predictor_num >= 0 && error_val > 0) {
int val = buffer_out[0] - buffer_out[predictor_coef_num - predictor_num];
int sign = SIGN_ONLY(val);
predictor_coef_table[predictor_num] -= sign;
val *= sign; /* absolute value */
error_val -= ((val >> predictor_quantitization) * (predictor_coef_num - predictor_num));
predictor_num--;
}
} else if (error_val < 0) {
int predictor_num = predictor_coef_num - 1;
while (predictor_num >= 0 && error_val < 0) {
int val = buffer_out[0] - buffer_out[predictor_coef_num - predictor_num];
int sign = -SIGN_ONLY(val);
predictor_coef_table[predictor_num] -= sign;
val *= sign; /* neg value */
error_val -= ((val >> predictor_quantitization) * (predictor_coef_num - predictor_num));
predictor_num--;
}
}
buffer_out++;
}
}
}
static void deinterlace_16(int32_t *buffer_a, int32_t *buffer_b, int16_t *buffer_out,
int numchannels, int numsamples, uint8_t interlacing_shift,
uint8_t interlacing_leftweight) {
int i;
if (numsamples <= 0)
return;
/* weighted interlacing */
if (interlacing_leftweight) {
for (i = 0; i < numsamples; i++) {
int32_t difference, midright;
int16_t left;
int16_t right;
midright = buffer_a[i];
difference = buffer_b[i];
right = midright - ((difference * interlacing_leftweight) >> interlacing_shift);
left = right + difference;
/* output is always little endian */
if (host_bigendian) {
_Swap16(left);
_Swap16(right);
}
buffer_out[i * numchannels] = left;
buffer_out[i * numchannels + 1] = right;
}
return;
}
/* otherwise basic interlacing took place */
for (i = 0; i < numsamples; i++) {
int16_t left, right;
left = buffer_a[i];
right = buffer_b[i];
/* output is always little endian */
if (host_bigendian) {
_Swap16(left);
_Swap16(right);
}
buffer_out[i * numchannels] = left;
buffer_out[i * numchannels + 1] = right;
}
}
static void deinterlace_24(int32_t *buffer_a, int32_t *buffer_b, int uncompressed_bytes,
int32_t *uncompressed_bytes_buffer_a,
int32_t *uncompressed_bytes_buffer_b, void *buffer_out, int numchannels,
int numsamples, uint8_t interlacing_shift,
uint8_t interlacing_leftweight) {
int i;
if (numsamples <= 0)
return;
/* weighted interlacing */
if (interlacing_leftweight) {
for (i = 0; i < numsamples; i++) {
int32_t difference, midright;
int32_t left;
int32_t right;
midright = buffer_a[i];
difference = buffer_b[i];
right = midright - ((difference * interlacing_leftweight) >> interlacing_shift);
left = right + difference;
if (uncompressed_bytes) {
uint32_t mask = ~(0xFFFFFFFF << (uncompressed_bytes * 8));
left <<= (uncompressed_bytes * 8);
right <<= (uncompressed_bytes * 8);
left |= uncompressed_bytes_buffer_a[i] & mask;
right |= uncompressed_bytes_buffer_b[i] & mask;
}
((uint8_t *)buffer_out)[i * numchannels * 3] = (left)&0xFF;
((uint8_t *)buffer_out)[i * numchannels * 3 + 1] = (left >> 8) & 0xFF;
((uint8_t *)buffer_out)[i * numchannels * 3 + 2] = (left >> 16) & 0xFF;
((uint8_t *)buffer_out)[i * numchannels * 3 + 3] = (right)&0xFF;
((uint8_t *)buffer_out)[i * numchannels * 3 + 4] = (right >> 8) & 0xFF;
((uint8_t *)buffer_out)[i * numchannels * 3 + 5] = (right >> 16) & 0xFF;
}
return;
}
/* otherwise basic interlacing took place */
for (i = 0; i < numsamples; i++) {
int32_t left, right;
left = buffer_a[i];
right = buffer_b[i];
if (uncompressed_bytes) {
uint32_t mask = ~(0xFFFFFFFF << (uncompressed_bytes * 8));
left <<= (uncompressed_bytes * 8);
right <<= (uncompressed_bytes * 8);
left |= uncompressed_bytes_buffer_a[i] & mask;
right |= uncompressed_bytes_buffer_b[i] & mask;
}
((uint8_t *)buffer_out)[i * numchannels * 3] = (left)&0xFF;
((uint8_t *)buffer_out)[i * numchannels * 3 + 1] = (left >> 8) & 0xFF;
((uint8_t *)buffer_out)[i * numchannels * 3 + 2] = (left >> 16) & 0xFF;
((uint8_t *)buffer_out)[i * numchannels * 3 + 3] = (right)&0xFF;
((uint8_t *)buffer_out)[i * numchannels * 3 + 4] = (right >> 8) & 0xFF;
((uint8_t *)buffer_out)[i * numchannels * 3 + 5] = (right >> 16) & 0xFF;
}
}
void alac_decode_frame(alac_file *alac, unsigned char *inbuffer, void *outbuffer, int *outputsize) {
int outbuffer_allocation_size = *outputsize; // initial value
int channels;
int32_t outputsamples = alac->setinfo_max_samples_per_frame;
/* setup the stream */
alac->input_buffer = inbuffer;
alac->input_buffer_bitaccumulator = 0;
channels = readbits(alac, 3);
*outputsize = outputsamples * alac->bytespersample;
if (*outputsize > outbuffer_allocation_size) {
fprintf(stderr, "FIXME: Not enough space if the output buffer for audio frame - E1.\n");
*outputsize = 0;
return;
}
switch (channels) {
case 0: /* 1 channel */
{
int hassize;
int isnotcompressed;
int readsamplesize;
int uncompressed_bytes;
int ricemodifier;
/* 2^result = something to do with output waiting.
* perhaps matters if we read > 1 frame in a pass?
*/
readbits(alac, 4);
readbits(alac, 12); /* unknown, skip 12 bits */
hassize = readbits(alac, 1); /* the output sample size is stored soon */
uncompressed_bytes =
readbits(alac, 2); /* number of bytes in the (compressed) stream that are not compressed */
isnotcompressed = readbits(alac, 1); /* whether the frame is compressed */
if (hassize) {
/* now read the number of samples,
* as a 32bit integer */
outputsamples = readbits(alac, 32);
*outputsize = outputsamples * alac->bytespersample;
if (*outputsize > outbuffer_allocation_size) {
fprintf(stderr, "FIXME: Not enough space if the output buffer for audio frame - E2.\n");
*outputsize = 0;
return;
}
}
readsamplesize = alac->setinfo_sample_size - (uncompressed_bytes * 8);
if (!isnotcompressed) { /* so it is compressed */
int16_t predictor_coef_table[32];
int predictor_coef_num;
int prediction_type;
int prediction_quantitization;
int i;
/* skip 16 bits, not sure what they are. seem to be used in
* two channel case */
readbits(alac, 8);
readbits(alac, 8);
prediction_type = readbits(alac, 4);
prediction_quantitization = readbits(alac, 4);
ricemodifier = readbits(alac, 3);
predictor_coef_num = readbits(alac, 5);
/* read the predictor table */
for (i = 0; i < predictor_coef_num; i++) {
predictor_coef_table[i] = (int16_t)readbits(alac, 16);
}
if (uncompressed_bytes) {
int li;
for (li = 0; li < outputsamples; li++) {
alac->uncompressed_bytes_buffer_a[li] = readbits(alac, uncompressed_bytes * 8);
}
}
entropy_rice_decode(alac, alac->predicterror_buffer_a, outputsamples, readsamplesize,
alac->setinfo_rice_initialhistory, alac->setinfo_rice_kmodifier,
ricemodifier * alac->setinfo_rice_historymult / 4,
(1 << alac->setinfo_rice_kmodifier) - 1);
if (prediction_type == 0) { /* adaptive fir */
predictor_decompress_fir_adapt(alac->predicterror_buffer_a, alac->outputsamples_buffer_a,
outputsamples, readsamplesize, predictor_coef_table,
predictor_coef_num, prediction_quantitization);
} else {
fprintf(stderr, "FIXME: unhandled prediction type for compressed case: %i\n",
prediction_type);
/* i think the only other prediction type (or perhaps this is just a
* boolean?) runs adaptive fir twice.. like:
* predictor_decompress_fir_adapt(predictor_error, tempout, ...)
* predictor_decompress_fir_adapt(predictor_error, outputsamples ...)
* little strange..
*/
}
} else { /* not compressed, easy case */
if (alac->setinfo_sample_size <= 16) {
int i;
for (i = 0; i < outputsamples; i++) {
int32_t audiobits = readbits(alac, alac->setinfo_sample_size);
audiobits = SIGN_EXTENDED32(audiobits, alac->setinfo_sample_size);
alac->outputsamples_buffer_a[i] = audiobits;
}
} else {
int i;
for (i = 0; i < outputsamples; i++) {
int32_t audiobits;
audiobits = readbits(alac, 16);
/* special case of sign extension..
* as we'll be ORing the low 16bits into this */
audiobits = audiobits << (alac->setinfo_sample_size - 16);
audiobits |= readbits(alac, alac->setinfo_sample_size - 16);
audiobits = SignExtend24(audiobits);
alac->outputsamples_buffer_a[i] = audiobits;
}
}
uncompressed_bytes = 0; // always 0 for uncompressed
}
switch (alac->setinfo_sample_size) {
case 16: {
int i;
for (i = 0; i < outputsamples; i++) {
int16_t sample = alac->outputsamples_buffer_a[i];
if (host_bigendian)
_Swap16(sample);
((int16_t *)outbuffer)[i * alac->numchannels] = sample;
}
break;
}
case 24: {
int i;
for (i = 0; i < outputsamples; i++) {
int32_t sample = alac->outputsamples_buffer_a[i];
if (uncompressed_bytes) {
uint32_t mask;
sample = sample << (uncompressed_bytes * 8);
mask = ~(0xFFFFFFFF << (uncompressed_bytes * 8));
sample |= alac->uncompressed_bytes_buffer_a[i] & mask;
}
((uint8_t *)outbuffer)[i * alac->numchannels * 3] = (sample)&0xFF;
((uint8_t *)outbuffer)[i * alac->numchannels * 3 + 1] = (sample >> 8) & 0xFF;
((uint8_t *)outbuffer)[i * alac->numchannels * 3 + 2] = (sample >> 16) & 0xFF;
}
break;
}
case 20:
case 32:
fprintf(stderr, "FIXME: unimplemented sample size %i\n", alac->setinfo_sample_size);
break;
default:
break;
}
break;
}
case 1: /* 2 channels */
{
int hassize;
int isnotcompressed;
int readsamplesize;
int uncompressed_bytes;
uint8_t interlacing_shift;
uint8_t interlacing_leftweight;
/* 2^result = something to do with output waiting.
* perhaps matters if we read > 1 frame in a pass?
*/
readbits(alac, 4);
readbits(alac, 12); /* unknown, skip 12 bits */
hassize = readbits(alac, 1); /* the output sample size is stored soon */
uncompressed_bytes = readbits(
alac, 2); /* the number of bytes in the (compressed) stream that are not compressed */
isnotcompressed = readbits(alac, 1); /* whether the frame is compressed */
if (hassize) {
/* now read the number of samples,
* as a 32bit integer */
outputsamples = readbits(alac, 32);
*outputsize = outputsamples * alac->bytespersample;
if (*outputsize > outbuffer_allocation_size) {
fprintf(stderr, "FIXME: Not enough space if the output buffer for audio frame - E3.\n");
*outputsize = 0;
return;
}
}
readsamplesize = alac->setinfo_sample_size - (uncompressed_bytes * 8) + 1;
if (!isnotcompressed) { /* compressed */
int16_t predictor_coef_table_a[32];
int predictor_coef_num_a;
int prediction_type_a;
int prediction_quantitization_a;
int ricemodifier_a;
int16_t predictor_coef_table_b[32];
int predictor_coef_num_b;
int prediction_type_b;
int prediction_quantitization_b;
int ricemodifier_b;
int i;
interlacing_shift = readbits(alac, 8);
interlacing_leftweight = readbits(alac, 8);
/******** channel 1 ***********/
prediction_type_a = readbits(alac, 4);
prediction_quantitization_a = readbits(alac, 4);
ricemodifier_a = readbits(alac, 3);
predictor_coef_num_a = readbits(alac, 5);
/* read the predictor table */
for (i = 0; i < predictor_coef_num_a; i++) {
predictor_coef_table_a[i] = (int16_t)readbits(alac, 16);
}
/******** channel 2 *********/
prediction_type_b = readbits(alac, 4);
prediction_quantitization_b = readbits(alac, 4);
ricemodifier_b = readbits(alac, 3);
predictor_coef_num_b = readbits(alac, 5);
/* read the predictor table */
for (i = 0; i < predictor_coef_num_b; i++) {
predictor_coef_table_b[i] = (int16_t)readbits(alac, 16);
}
/*********************/
if (uncompressed_bytes) { /* see mono case */
int li;
for (li = 0; li < outputsamples; li++) {
alac->uncompressed_bytes_buffer_a[li] = readbits(alac, uncompressed_bytes * 8);
alac->uncompressed_bytes_buffer_b[li] = readbits(alac, uncompressed_bytes * 8);
}
}
/* channel 1 */
entropy_rice_decode(alac, alac->predicterror_buffer_a, outputsamples, readsamplesize,
alac->setinfo_rice_initialhistory, alac->setinfo_rice_kmodifier,
ricemodifier_a * alac->setinfo_rice_historymult / 4,
(1 << alac->setinfo_rice_kmodifier) - 1);
if (prediction_type_a == 0) { /* adaptive fir */
predictor_decompress_fir_adapt(alac->predicterror_buffer_a, alac->outputsamples_buffer_a,
outputsamples, readsamplesize, predictor_coef_table_a,
predictor_coef_num_a, prediction_quantitization_a);
} else { /* see mono case */
fprintf(stderr, "FIXME: unhandled prediction type on channel 1: %i\n", prediction_type_a);
}
/* channel 2 */
entropy_rice_decode(alac, alac->predicterror_buffer_b, outputsamples, readsamplesize,
alac->setinfo_rice_initialhistory, alac->setinfo_rice_kmodifier,
ricemodifier_b * alac->setinfo_rice_historymult / 4,
(1 << alac->setinfo_rice_kmodifier) - 1);
if (prediction_type_b == 0) { /* adaptive fir */
predictor_decompress_fir_adapt(alac->predicterror_buffer_b, alac->outputsamples_buffer_b,
outputsamples, readsamplesize, predictor_coef_table_b,
predictor_coef_num_b, prediction_quantitization_b);
} else {
fprintf(stderr, "FIXME: unhandled prediction type on channel 2: %i\n", prediction_type_b);
}
} else { /* not compressed, easy case */
if (alac->setinfo_sample_size <= 16) {
int i;
for (i = 0; i < outputsamples; i++) {
int32_t audiobits_a, audiobits_b;
audiobits_a = readbits(alac, alac->setinfo_sample_size);
audiobits_b = readbits(alac, alac->setinfo_sample_size);
audiobits_a = SIGN_EXTENDED32(audiobits_a, alac->setinfo_sample_size);
audiobits_b = SIGN_EXTENDED32(audiobits_b, alac->setinfo_sample_size);
alac->outputsamples_buffer_a[i] = audiobits_a;
alac->outputsamples_buffer_b[i] = audiobits_b;
}
} else {
int i;
for (i = 0; i < outputsamples; i++) {
int32_t audiobits_a, audiobits_b;
audiobits_a = readbits(alac, 16);
audiobits_a = audiobits_a << (alac->setinfo_sample_size - 16);
audiobits_a |= readbits(alac, alac->setinfo_sample_size - 16);
audiobits_a = SignExtend24(audiobits_a);
audiobits_b = readbits(alac, 16);
audiobits_b = audiobits_b << (alac->setinfo_sample_size - 16);
audiobits_b |= readbits(alac, alac->setinfo_sample_size - 16);
audiobits_b = SignExtend24(audiobits_b);
alac->outputsamples_buffer_a[i] = audiobits_a;
alac->outputsamples_buffer_b[i] = audiobits_b;
}
}
uncompressed_bytes = 0; // always 0 for uncompressed
interlacing_shift = 0;
interlacing_leftweight = 0;
}
switch (alac->setinfo_sample_size) {
case 16: {
deinterlace_16(alac->outputsamples_buffer_a, alac->outputsamples_buffer_b,
(int16_t *)outbuffer, alac->numchannels, outputsamples, interlacing_shift,
interlacing_leftweight);
break;
}
case 24: {
deinterlace_24(alac->outputsamples_buffer_a, alac->outputsamples_buffer_b, uncompressed_bytes,
alac->uncompressed_bytes_buffer_a, alac->uncompressed_bytes_buffer_b,
(int16_t *)outbuffer, alac->numchannels, outputsamples, interlacing_shift,
interlacing_leftweight);
break;
}
case 20:
case 32:
fprintf(stderr, "FIXME: unimplemented sample size %i\n", alac->setinfo_sample_size);
break;
default:
break;
}
break;
}
}
}
alac_file *alac_create(int samplesize, int numchannels) {
alac_file *newfile = malloc(sizeof(alac_file));
if (newfile) {
memset(newfile, 0, sizeof(alac_file));
newfile->samplesize = samplesize;
newfile->numchannels = numchannels;
newfile->bytespersample = (samplesize / 8) * numchannels;
} else {
fprintf(stderr, "FIXME: can not allocate memory for a new file in alac_cxreate.");
}
return newfile;
}