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FreeRDP/libfreerdp/codec/dsp.c

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/**
* FreeRDP: A Remote Desktop Protocol Implementation
* Digital Sound Processing
*
* Copyright 2010-2011 Vic Lee
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
* You may obtain a copy of the License at
*
* http://www.apache.org/licenses/LICENSE-2.0
*
* Unless required by applicable law or agreed to in writing, software
* distributed under the License is distributed on an "AS IS" BASIS,
* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
* See the License for the specific language governing permissions and
* limitations under the License.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <assert.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <winpr/crt.h>
#include <freerdp/types.h>
#include <freerdp/log.h>
#include <freerdp/codec/dsp.h>
#if !defined(WITH_DSP_FFMPEG)
#if defined(WITH_GSM)
#include <gsm/gsm.h>
#endif
#if defined(WITH_LAME)
#include <lame/lame.h>
#endif
#if defined(WITH_FAAD2)
#include <neaacdec.h>
#endif
#if defined(WITH_FAAC)
#include <faac.h>
#endif
#if defined(WITH_SOXR)
#include <soxr.h>
#endif
#else
#include "dsp_ffmpeg.h"
#endif
#define TAG FREERDP_TAG("dsp")
#if !defined(WITH_DSP_FFMPEG)
union _ADPCM
{
struct
{
INT16 last_sample[2];
INT16 last_step[2];
} ima;
struct
{
BYTE predictor[2];
INT32 delta[2];
INT32 sample1[2];
INT32 sample2[2];
} ms;
};
typedef union _ADPCM ADPCM;
struct _FREERDP_DSP_CONTEXT
{
BOOL encoder;
ADPCM adpcm;
AUDIO_FORMAT format;
wStream* buffer;
wStream* resample;
#if defined(WITH_GSM)
gsm gsm;
#endif
#if defined(WITH_LAME)
lame_t lame;
hip_t hip;
#endif
#if defined(WITH_FAAD2)
NeAACDecHandle faad;
BOOL faadSetup;
#endif
#if defined(WITH_FAAC)
faacEncHandle faac;
unsigned long faacInputSamples;
unsigned long faacMaxOutputBytes;
#endif
#if defined(WITH_SOXR)
soxr_t sox;
#endif
};
static INT16 read_int16(const BYTE* src)
{
return (INT16) (src[0] | (src[1] << 8));
}
static void write_int16(BYTE* dst, INT32 val)
{
dst[1] = (val >> 8) & 0xFF;
dst[0] = val & 0xFF;
}
static BOOL freerdp_dsp_channel_mix(FREERDP_DSP_CONTEXT* context,
const BYTE* src, size_t size,
const AUDIO_FORMAT* srcFormat,
const BYTE** data, size_t* length)
{
UINT32 bpp;
size_t samples;
size_t x, y;
if (!context || !data || !length)
return FALSE;
if (srcFormat->wFormatTag != WAVE_FORMAT_PCM)
return FALSE;
bpp = srcFormat->wBitsPerSample > 8 ? 2 : 1;
samples = size / bpp;
if (context->format.nChannels == srcFormat->nChannels)
{
*data = src;
*length = size;
return TRUE;
}
Stream_SetPosition(context->buffer, 0);
/* Destination has more channels than source */
if (context->format.nChannels > srcFormat->nChannels)
{
switch (srcFormat->nChannels)
{
case 1:
if (!Stream_EnsureCapacity(context->buffer, size * 2))
return FALSE;
for (x = 0; x < samples; x++)
{
for (y = 0; y < bpp; y++)
Stream_Write_UINT8(context->buffer, src[x * bpp + y]);
for (y = 0; y < bpp; y++)
Stream_Write_UINT8(context->buffer, src[x * bpp + y]);
}
Stream_SealLength(context->buffer);
*data = Stream_Buffer(context->buffer);
*length = Stream_Length(context->buffer);
return TRUE;
case 2: /* We only support stereo, so we can not handle this case. */
default: /* Unsupported number of channels */
return FALSE;
}
}
/* Destination has less channels than source */
switch (srcFormat->nChannels)
{
case 2:
if (!Stream_EnsureCapacity(context->buffer, size / 2))
return FALSE;
/* Simply drop second channel.
* TODO: Calculate average */
for (x = 0; x < samples; x++)
{
for (y = 0; y < bpp; y++)
Stream_Write_UINT8(context->buffer, src[2 * x * bpp + y]);
}
Stream_SealLength(context->buffer);
*data = Stream_Buffer(context->buffer);
*length = Stream_Length(context->buffer);
return TRUE;
case 1: /* Invalid, do we want to use a 0 channel sound? */
default: /* Unsupported number of channels */
return FALSE;
}
return FALSE;
}
/**
* Microsoft Multimedia Standards Update
* http://download.microsoft.com/download/9/8/6/9863C72A-A3AA-4DDB-B1BA-CA8D17EFD2D4/RIFFNEW.pdf
*/
static BOOL freerdp_dsp_resample(FREERDP_DSP_CONTEXT* context,
const BYTE* src, size_t size,
const AUDIO_FORMAT* srcFormat,
const BYTE** data, size_t* length)
{
#if defined(WITH_SOXR)
soxr_error_t error;
size_t idone, odone;
size_t sframes, rframes;
size_t rsize;
size_t j;
size_t sbytes, rbytes;
#endif
size_t srcBytesPerFrame, dstBytesPerFrame;
size_t srcChannels, dstChannels;
AUDIO_FORMAT format;
if (srcFormat->wFormatTag != WAVE_FORMAT_PCM)
{
WLog_ERR(TAG, "%s requires %s for sample input, got %s", __FUNCTION__,
audio_format_get_tag_string(WAVE_FORMAT_PCM),
audio_format_get_tag_string(srcFormat->wFormatTag));
return FALSE;
}
srcChannels = srcFormat->nChannels;
dstChannels = context->format.nChannels;
srcBytesPerFrame = (srcFormat->wBitsPerSample > 8) ? 2 : 1;
dstBytesPerFrame = (context->format.wBitsPerSample > 8) ? 2 : 1;
/* We want to ignore differences of source and destination format. */
format = *srcFormat;
format.wFormatTag = WAVE_FORMAT_UNKNOWN;
if (audio_format_compatible(&format, &context->format))
return TRUE;
#if defined(WITH_SOXR)
sbytes = srcChannels * srcBytesPerFrame;
sframes = size / sbytes;
rbytes = dstBytesPerFrame * dstChannels;
/* Integer rounding correct division */
rframes = (sframes * context->format.nSamplesPerSec + (srcFormat->nSamplesPerSec + 1) / 2) /
srcFormat->nSamplesPerSec;
rsize = rframes * rbytes;
if (!Stream_EnsureCapacity(context->resample, rsize))
return FALSE;
error = soxr_process(context->sox, src, sframes, &idone,
Stream_Buffer(context->resample),
Stream_Capacity(context->resample) / rbytes, &odone);
Stream_SetLength(context->resample, odone * rbytes);
*data = Stream_Buffer(context->resample);
*length = Stream_Length(context->resample);
return (error == 0) ? TRUE : FALSE;
#else
WINPR_UNUSED(src);
WINPR_UNUSED(size);
WINPR_UNUSED(data);
WINPR_UNUSED(length);
WLog_ERR(TAG, "Missing resample support, recompile -DWITH_SOXR=ON or -DWITH_DSP_FFMPEG=ON");
return FALSE;
#endif
}
/**
* Microsoft IMA ADPCM specification:
*
* http://wiki.multimedia.cx/index.php?title=Microsoft_IMA_ADPCM
* http://wiki.multimedia.cx/index.php?title=IMA_ADPCM
*/
static const INT16 ima_step_index_table[] =
{
-1, -1, -1, -1, 2, 4, 6, 8,
-1, -1, -1, -1, 2, 4, 6, 8
};
static const INT16 ima_step_size_table[] =
{
7, 8, 9, 10, 11, 12, 13, 14, 16, 17,
19, 21, 23, 25, 28, 31, 34, 37, 41, 45,
50, 55, 60, 66, 73, 80, 88, 97, 107, 118,
130, 143, 157, 173, 190, 209, 230, 253, 279, 307,
337, 371, 408, 449, 494, 544, 598, 658, 724, 796,
876, 963, 1060, 1166, 1282, 1411, 1552, 1707, 1878, 2066,
2272, 2499, 2749, 3024, 3327, 3660, 4026, 4428, 4871, 5358,
5894, 6484, 7132, 7845, 8630, 9493, 10442, 11487, 12635, 13899,
15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794, 32767
};
static UINT16 dsp_decode_ima_adpcm_sample(ADPCM* adpcm,
unsigned int channel, BYTE sample)
{
INT32 ss;
INT32 d;
ss = ima_step_size_table[adpcm->ima.last_step[channel]];
d = (ss >> 3);
if (sample & 1)
d += (ss >> 2);
if (sample & 2)
d += (ss >> 1);
if (sample & 4)
d += ss;
if (sample & 8)
d = -d;
d += adpcm->ima.last_sample[channel];
if (d < -32768)
d = -32768;
else if (d > 32767)
d = 32767;
adpcm->ima.last_sample[channel] = (INT16) d;
adpcm->ima.last_step[channel] += ima_step_index_table[sample];
if (adpcm->ima.last_step[channel] < 0)
adpcm->ima.last_step[channel] = 0;
else if (adpcm->ima.last_step[channel] > 88)
adpcm->ima.last_step[channel] = 88;
return (UINT16) d;
}
static BOOL freerdp_dsp_decode_ima_adpcm(FREERDP_DSP_CONTEXT* context,
const BYTE* src, size_t size, wStream* out)
{
BYTE* dst;
BYTE sample;
UINT16 decoded;
size_t out_size = size * 4;
UINT32 channel;
const UINT32 block_size = context->format.nBlockAlign;
const UINT32 channels = context->format.nChannels;
size_t i;
if (!Stream_EnsureCapacity(out, out_size))
return FALSE;
dst = Stream_Pointer(out);
while (size > 0)
{
if (size % block_size == 0)
{
context->adpcm.ima.last_sample[0] = (INT16)(((UINT16)(*src)) | (((UINT16)(*(src + 1))) << 8));
context->adpcm.ima.last_step[0] = (INT16)(*(src + 2));
src += 4;
size -= 4;
out_size -= 16;
if (channels > 1)
{
context->adpcm.ima.last_sample[1] = (INT16)(((UINT16)(*src)) | (((UINT16)(*(src + 1))) << 8));
context->adpcm.ima.last_step[1] = (INT16)(*(src + 2));
src += 4;
size -= 4;
out_size -= 16;
}
}
if (channels > 1)
{
for (i = 0; i < 8; i++)
{
channel = (i < 4 ? 0 : 1);
sample = ((*src) & 0x0f);
decoded = dsp_decode_ima_adpcm_sample(&context->adpcm, channel, sample);
dst[((i & 3) << 3) + (channel << 1)] = (decoded & 0xFF);
dst[((i & 3) << 3) + (channel << 1) + 1] = (decoded >> 8);
sample = ((*src) >> 4);
decoded = dsp_decode_ima_adpcm_sample(&context->adpcm, channel, sample);
dst[((i & 3) << 3) + (channel << 1) + 4] = (decoded & 0xFF);
dst[((i & 3) << 3) + (channel << 1) + 5] = (decoded >> 8);
src++;
}
dst += 32;
size -= 8;
}
else
{
sample = ((*src) & 0x0f);
decoded = dsp_decode_ima_adpcm_sample(&context->adpcm, 0, sample);
*dst++ = (decoded & 0xFF);
*dst++ = (decoded >> 8);
sample = ((*src) >> 4);
decoded = dsp_decode_ima_adpcm_sample(&context->adpcm, 0, sample);
*dst++ = (decoded & 0xFF);
*dst++ = (decoded >> 8);
src++;
size--;
}
}
Stream_SetPointer(out, dst);
return TRUE;
}
#if defined(WITH_GSM)
static BOOL freerdp_dsp_decode_gsm610(FREERDP_DSP_CONTEXT* context,
const BYTE* src, size_t size, wStream* out)
{
size_t offset = 0;
while (offset < size)
{
int rc;
gsm_signal gsmBlockBuffer[160] = { 0 };
rc = gsm_decode(context->gsm, (gsm_byte*) /* API does not modify */ &src[offset], gsmBlockBuffer);
if (rc < 0)
return FALSE;
if ((offset % 65) == 0)
offset += 33;
else
offset += 32;
if (!Stream_EnsureRemainingCapacity(out, sizeof(gsmBlockBuffer)))
return FALSE;
Stream_Write(out, (void*) gsmBlockBuffer, sizeof(gsmBlockBuffer));
}
return TRUE;
}
static BOOL freerdp_dsp_encode_gsm610(FREERDP_DSP_CONTEXT* context,
const BYTE* src, size_t size, wStream* out)
{
size_t offset = 0;
while (offset < size)
{
const gsm_signal* signal = (const gsm_signal*)&src[offset];
if (!Stream_EnsureRemainingCapacity(out, sizeof(gsm_frame)))
return FALSE;
gsm_encode(context->gsm, (gsm_signal*) /* API does not modify */ signal, Stream_Pointer(out));
if ((offset % 65) == 0)
Stream_Seek(out, 33);
else
Stream_Seek(out, 32);
offset += 160;
}
return TRUE;
}
#endif
#if defined(WITH_LAME)
static BOOL freerdp_dsp_decode_mp3(FREERDP_DSP_CONTEXT* context,
const BYTE* src, size_t size, wStream* out)
{
int rc, x;
short* pcm_l;
short* pcm_r;
size_t buffer_size;
if (!context || !src || !out)
return FALSE;
buffer_size = 2 * context->format.nChannels * context->format.nSamplesPerSec;
if (!Stream_EnsureCapacity(context->buffer, 2 * buffer_size))
return FALSE;
pcm_l = (short*)Stream_Buffer(context->buffer);
pcm_r = (short*)Stream_Buffer(context->buffer) + buffer_size;
rc = hip_decode(context->hip, (unsigned char*)/* API is not modifying content */src,
size, pcm_l, pcm_r);
if (rc <= 0)
return FALSE;
if (!Stream_EnsureRemainingCapacity(out, (size_t)rc * context->format.nChannels * 2))
return FALSE;
for (x = 0; x < rc; x++)
{
Stream_Write_UINT16(out, (UINT16)pcm_l[x]);
Stream_Write_UINT16(out, (UINT16)pcm_r[x]);
}
return TRUE;
}
static BOOL freerdp_dsp_encode_mp3(FREERDP_DSP_CONTEXT* context,
const BYTE* src, size_t size, wStream* out)
{
size_t samples_per_channel;
int rc;
if (!context || !src || !out)
return FALSE;
samples_per_channel = size / context->format.nChannels / context->format.wBitsPerSample / 8;
/* Ensure worst case buffer size for mp3 stream taken from LAME header */
if (!Stream_EnsureRemainingCapacity(out, 5 / 4 * samples_per_channel + 7200))
return FALSE;
samples_per_channel = size / 2 /* size of a sample */ / context->format.nChannels;
rc = lame_encode_buffer_interleaved(context->lame, (short*)src, samples_per_channel,
Stream_Pointer(out), Stream_GetRemainingCapacity(out));
if (rc < 0)
return FALSE;
Stream_Seek(out, (size_t)rc);
return TRUE;
}
#endif
#if defined(WITH_FAAC)
static BOOL freerdp_dsp_encode_faac(FREERDP_DSP_CONTEXT* context,
const BYTE* src, size_t size, wStream* out)
{
const int16_t* inSamples = (const int16_t*)src;
int32_t* outSamples;
unsigned int bpp;
size_t nrSamples, x;
int rc;
if (!context || !src || !out)
return FALSE;
bpp = context->format.wBitsPerSample / 8 * context->format.nChannels;
nrSamples = size / bpp;
if (!Stream_EnsureCapacity(context->buffer,
nrSamples * sizeof(int32_t) * context->format.nChannels))
return FALSE;
if (!Stream_EnsureRemainingCapacity(out, context->faacMaxOutputBytes))
return FALSE;
outSamples = (int32_t*)Stream_Buffer(context->buffer);
for (x = 0; x < nrSamples * context->format.nChannels; x++)
outSamples[x] = inSamples[x];
rc = faacEncEncode(context->faac, outSamples, nrSamples * context->format.nChannels,
Stream_Pointer(out), Stream_GetRemainingCapacity(out));
if (rc < 0)
return FALSE;
else if (rc > 0)
Stream_Seek(out, (size_t)rc);
return TRUE;
}
#endif
#if defined(WITH_FAAD2)
static BOOL freerdp_dsp_decode_faad(FREERDP_DSP_CONTEXT* context,
const BYTE* src, size_t size, wStream* out)
{
NeAACDecFrameInfo info;
void* output;
size_t offset = 0;
if (!context || !src || !out)
return FALSE;
if (!context->faadSetup)
{
unsigned long samplerate;
unsigned char channels;
long err = NeAACDecInit(context->faad, /* API is not modifying content */(unsigned char*)src,
size, &samplerate, &channels);
if (err != 0)
return FALSE;
if (channels != context->format.nChannels)
return FALSE;
if (samplerate != context->format.nSamplesPerSec)
return FALSE;
context->faadSetup = TRUE;
}
while (offset < size)
{
size_t outSize;
void* sample_buffer;
outSize = context->format.nSamplesPerSec * context->format.nChannels *
context->format.wBitsPerSample / 8;
if (!Stream_EnsureRemainingCapacity(out, outSize))
return FALSE;
sample_buffer = Stream_Pointer(out);
output = NeAACDecDecode2(context->faad, &info, (unsigned char*)&src[offset], size - offset,
&sample_buffer, Stream_GetRemainingCapacity(out));
if (info.error != 0)
return FALSE;
offset += info.bytesconsumed;
if (info.samples == 0)
continue;
Stream_Seek(out, info.samples * context->format.wBitsPerSample / 8);
}
return TRUE;
}
#endif
/**
* 0 1 2 3
* 2 0 6 4 10 8 14 12 <left>
*
* 4 5 6 7
* 3 1 7 5 11 9 15 13 <right>
*/
static const struct
{
BYTE byte_num;
BYTE byte_shift;
} ima_stereo_encode_map[] =
{
{ 0, 0 },
{ 4, 0 },
{ 0, 4 },
{ 4, 4 },
{ 1, 0 },
{ 5, 0 },
{ 1, 4 },
{ 5, 4 },
{ 2, 0 },
{ 6, 0 },
{ 2, 4 },
{ 6, 4 },
{ 3, 0 },
{ 7, 0 },
{ 3, 4 },
{ 7, 4 }
};
static BYTE dsp_encode_ima_adpcm_sample(ADPCM* adpcm, int channel, INT16 sample)
{
INT32 e;
INT32 d;
INT32 ss;
BYTE enc;
INT32 diff;
ss = ima_step_size_table[adpcm->ima.last_step[channel]];
d = e = sample - adpcm->ima.last_sample[channel];
diff = ss >> 3;
enc = 0;
if (e < 0)
{
enc = 8;
e = -e;
}
if (e >= ss)
{
enc |= 4;
e -= ss;
}
ss >>= 1;
if (e >= ss)
{
enc |= 2;
e -= ss;
}
ss >>= 1;
if (e >= ss)
{
enc |= 1;
e -= ss;
}
if (d < 0)
diff = d + e - diff;
else
diff = d - e + diff;
diff += adpcm->ima.last_sample[channel];
if (diff < -32768)
diff = -32768;
else if (diff > 32767)
diff = 32767;
adpcm->ima.last_sample[channel] = (INT16) diff;
adpcm->ima.last_step[channel] += ima_step_index_table[enc];
if (adpcm->ima.last_step[channel] < 0)
adpcm->ima.last_step[channel] = 0;
else if (adpcm->ima.last_step[channel] > 88)
adpcm->ima.last_step[channel] = 88;
return enc;
}
static BOOL freerdp_dsp_encode_ima_adpcm(FREERDP_DSP_CONTEXT* context,
const BYTE* src, size_t size, wStream* out)
{
int i;
BYTE* dst;
BYTE* start;
INT16 sample;
BYTE encoded;
size_t out_size;
size_t align;
out_size = size / 2;
if (!Stream_EnsureRemainingCapacity(out, size))
return FALSE;
start = dst = Stream_Pointer(out);
align = (context->format.nChannels > 1) ? 32 : 4;
while (size > align)
{
if ((dst - start) % context->format.nBlockAlign == 0)
{
*dst++ = context->adpcm.ima.last_sample[0] & 0xFF;
*dst++ = (context->adpcm.ima.last_sample[0] >> 8) & 0xFF;
*dst++ = (BYTE) context->adpcm.ima.last_step[0];
*dst++ = 0;
if (context->format.nChannels > 1)
{
*dst++ = context->adpcm.ima.last_sample[1] & 0xFF;
*dst++ = (context->adpcm.ima.last_sample[1] >> 8) & 0xFF;
*dst++ = (BYTE) context->adpcm.ima.last_step[1];
*dst++ = 0;
}
}
if (context->format.nChannels > 1)
{
ZeroMemory(dst, 8);
for (i = 0; i < 16; i++)
{
sample = (INT16)(((UINT16)(*src)) | (((UINT16)(*(src + 1))) << 8));
src += 2;
encoded = dsp_encode_ima_adpcm_sample(&context->adpcm, i % 2, sample);
dst[ima_stereo_encode_map[i].byte_num] |= encoded << ima_stereo_encode_map[i].byte_shift;
}
dst += 8;
size -= 32;
}
else
{
sample = (INT16)(((UINT16)(*src)) | (((UINT16)(*(src + 1))) << 8));
src += 2;
encoded = dsp_encode_ima_adpcm_sample(&context->adpcm, 0, sample);
sample = (INT16)(((UINT16)(*src)) | (((UINT16)(*(src + 1))) << 8));
src += 2;
encoded |= dsp_encode_ima_adpcm_sample(&context->adpcm, 0, sample) << 4;
*dst++ = encoded;
size -= 4;
}
}
Stream_SetPointer(out, dst);
return TRUE;
}
/**
* Microsoft ADPCM Specification:
*
* http://wiki.multimedia.cx/index.php?title=Microsoft_ADPCM
*/
static const INT32 ms_adpcm_adaptation_table[] =
{
230, 230, 230, 230, 307, 409, 512, 614,
768, 614, 512, 409, 307, 230, 230, 230
};
static const INT32 ms_adpcm_coeffs1[7] =
{
256, 512, 0, 192, 240, 460, 392
};
static const INT32 ms_adpcm_coeffs2[7] =
{
0, -256, 0, 64, 0, -208, -232
};
static INLINE INT16 freerdp_dsp_decode_ms_adpcm_sample(ADPCM* adpcm, BYTE sample, int channel)
{
INT8 nibble;
INT32 presample;
nibble = (sample & 0x08 ? (INT8) sample - 16 : (INT8)sample);
presample = ((adpcm->ms.sample1[channel] * ms_adpcm_coeffs1[adpcm->ms.predictor[channel]]) +
(adpcm->ms.sample2[channel] * ms_adpcm_coeffs2[adpcm->ms.predictor[channel]])) / 256;
presample += nibble * adpcm->ms.delta[channel];
if (presample > 32767)
presample = 32767;
else if (presample < -32768)
presample = -32768;
adpcm->ms.sample2[channel] = adpcm->ms.sample1[channel];
adpcm->ms.sample1[channel] = presample;
adpcm->ms.delta[channel] = adpcm->ms.delta[channel] * ms_adpcm_adaptation_table[sample] / 256;
if (adpcm->ms.delta[channel] < 16)
adpcm->ms.delta[channel] = 16;
return (INT16) presample;
}
static BOOL freerdp_dsp_decode_ms_adpcm(FREERDP_DSP_CONTEXT* context,
const BYTE* src, size_t size, wStream* out)
{
BYTE* dst;
BYTE sample;
const size_t out_size = size * 4;
const UINT32 channels = context->format.nChannels;
const UINT32 block_size = context->format.nBlockAlign;
if (!Stream_EnsureCapacity(out, out_size))
return FALSE;
dst = Stream_Pointer(out);
while (size > 0)
{
if (size % block_size == 0)
{
if (channels > 1)
{
context->adpcm.ms.predictor[0] = *src++;
context->adpcm.ms.predictor[1] = *src++;
context->adpcm.ms.delta[0] = read_int16(src);
src += 2;
context->adpcm.ms.delta[1] = read_int16(src);
src += 2;
context->adpcm.ms.sample1[0] = read_int16(src);
src += 2;
context->adpcm.ms.sample1[1] = read_int16(src);
src += 2;
context->adpcm.ms.sample2[0] = read_int16(src);
src += 2;
context->adpcm.ms.sample2[1] = read_int16(src);
src += 2;
size -= 14;
write_int16(dst, context->adpcm.ms.sample2[0]);
dst += 2;
write_int16(dst, context->adpcm.ms.sample2[1]);
dst += 2;
write_int16(dst, context->adpcm.ms.sample1[0]);
dst += 2;
write_int16(dst, context->adpcm.ms.sample1[1]);
dst += 2;
}
else
{
context->adpcm.ms.predictor[0] = *src++;
context->adpcm.ms.delta[0] = read_int16(src);
src += 2;
context->adpcm.ms.sample1[0] = read_int16(src);
src += 2;
context->adpcm.ms.sample2[0] = read_int16(src);
src += 2;
size -= 7;
write_int16(dst, context->adpcm.ms.sample2[0]);
dst += 2;
write_int16(dst, context->adpcm.ms.sample1[0]);
dst += 2;
}
}
if (channels > 1)
{
sample = *src++;
size--;
write_int16(dst, freerdp_dsp_decode_ms_adpcm_sample(&context->adpcm, sample >> 4, 0));
dst += 2;
write_int16(dst, freerdp_dsp_decode_ms_adpcm_sample(&context->adpcm, sample & 0x0F, 1));
dst += 2;
sample = *src++;
size--;
write_int16(dst, freerdp_dsp_decode_ms_adpcm_sample(&context->adpcm, sample >> 4, 0));
dst += 2;
write_int16(dst, freerdp_dsp_decode_ms_adpcm_sample(&context->adpcm, sample & 0x0F, 1));
dst += 2;
}
else
{
sample = *src++;
size--;
write_int16(dst, freerdp_dsp_decode_ms_adpcm_sample(&context->adpcm, sample >> 4, 0));
dst += 2;
write_int16(dst, freerdp_dsp_decode_ms_adpcm_sample(&context->adpcm, sample & 0x0F, 0));
dst += 2;
}
}
Stream_SetPointer(out, dst);
return TRUE;
}
static BYTE freerdp_dsp_encode_ms_adpcm_sample(ADPCM* adpcm, INT32 sample, int channel)
{
INT32 presample;
INT32 errordelta;
presample = ((adpcm->ms.sample1[channel] * ms_adpcm_coeffs1[adpcm->ms.predictor[channel]]) +
(adpcm->ms.sample2[channel] * ms_adpcm_coeffs2[adpcm->ms.predictor[channel]])) / 256;
errordelta = (sample - presample) / adpcm->ms.delta[channel];
if ((sample - presample) % adpcm->ms.delta[channel] > adpcm->ms.delta[channel] / 2)
errordelta++;
if (errordelta > 7)
errordelta = 7;
else if (errordelta < -8)
errordelta = -8;
presample += adpcm->ms.delta[channel] * errordelta;
if (presample > 32767)
presample = 32767;
else if (presample < -32768)
presample = -32768;
adpcm->ms.sample2[channel] = adpcm->ms.sample1[channel];
adpcm->ms.sample1[channel] = presample;
adpcm->ms.delta[channel] = adpcm->ms.delta[channel] * ms_adpcm_adaptation_table[(((
BYTE) errordelta) & 0x0F)] / 256;
if (adpcm->ms.delta[channel] < 16)
adpcm->ms.delta[channel] = 16;
return ((BYTE) errordelta) & 0x0F;
}
static BOOL freerdp_dsp_encode_ms_adpcm(FREERDP_DSP_CONTEXT* context, const BYTE* src, size_t size,
wStream* out)
{
BYTE* dst;
BYTE* start;
INT32 sample;
size_t out_size;
const size_t step = 8 + ((context->format.nChannels > 1) ? 4 : 0);
out_size = size / 2;
if (!Stream_EnsureRemainingCapacity(out, size))
return FALSE;
start = dst = Stream_Pointer(out);
if (context->adpcm.ms.delta[0] < 16)
context->adpcm.ms.delta[0] = 16;
if (context->adpcm.ms.delta[1] < 16)
context->adpcm.ms.delta[1] = 16;
while (size >= step)
{
if ((dst - start) % context->format.nBlockAlign == 0)
{
if (context->format.nChannels > 1)
{
*dst++ = context->adpcm.ms.predictor[0];
*dst++ = context->adpcm.ms.predictor[1];
*dst++ = (BYTE)(context->adpcm.ms.delta[0] & 0xFF);
*dst++ = (BYTE)((context->adpcm.ms.delta[0] >> 8) & 0xFF);
*dst++ = (BYTE)(context->adpcm.ms.delta[1] & 0xFF);
*dst++ = (BYTE)((context->adpcm.ms.delta[1] >> 8) & 0xFF);
context->adpcm.ms.sample1[0] = read_int16(src + 4);
context->adpcm.ms.sample1[1] = read_int16(src + 6);
context->adpcm.ms.sample2[0] = read_int16(src + 0);
context->adpcm.ms.sample2[1] = read_int16(src + 2);
write_int16(dst + 0, context->adpcm.ms.sample1[0]);
write_int16(dst + 2, context->adpcm.ms.sample1[1]);
write_int16(dst + 4, context->adpcm.ms.sample2[0]);
write_int16(dst + 6, context->adpcm.ms.sample2[1]);
dst += 8;
src += 8;
size -= 8;
}
else
{
*dst++ = context->adpcm.ms.predictor[0];
*dst++ = (BYTE)(context->adpcm.ms.delta[0] & 0xFF);
*dst++ = (BYTE)((context->adpcm.ms.delta[0] >> 8) & 0xFF);
context->adpcm.ms.sample1[0] = read_int16(src + 2);
context->adpcm.ms.sample2[0] = read_int16(src + 0);
write_int16(dst + 0, context->adpcm.ms.sample1[0]);
write_int16(dst + 2, context->adpcm.ms.sample2[0]);
dst += 4;
src += 4;
size -= 4;
}
}
sample = read_int16(src);
src += 2;
*dst = (freerdp_dsp_encode_ms_adpcm_sample(&context->adpcm, sample, 0) << 4) & 0xFF;
sample = read_int16(src);
src += 2;
*dst += freerdp_dsp_encode_ms_adpcm_sample(&context->adpcm, sample,
context->format.nChannels > 1 ? 1 : 0);
dst++;
size -= 4;
}
Stream_SetPointer(out, dst);
return TRUE;
}
#endif
FREERDP_DSP_CONTEXT* freerdp_dsp_context_new(BOOL encoder)
{
#if defined(WITH_DSP_FFMPEG)
return freerdp_dsp_ffmpeg_context_new(encoder);
#else
FREERDP_DSP_CONTEXT* context = calloc(1, sizeof(FREERDP_DSP_CONTEXT));
if (!context)
return NULL;
context->buffer = Stream_New(NULL, 4096);
if (!context->buffer)
goto fail;
context->resample = Stream_New(NULL, 4096);
if (!context->resample)
goto fail;
context->encoder = encoder;
#if defined(WITH_GSM)
context->gsm = gsm_create();
if (!context->gsm)
goto fail;
{
int rc;
int val = 1;
rc = gsm_option(context->gsm, GSM_OPT_WAV49, &val);
if (rc < 0)
goto fail;
}
#endif
#if defined(WITH_LAME)
if (encoder)
{
context->lame = lame_init();
if (!context->lame)
goto fail;
}
else
{
context->hip = hip_decode_init();
if (!context->hip)
goto fail;
}
#endif
#if defined(WITH_FAAD2)
if (!encoder)
{
context->faad = NeAACDecOpen();
if (!context->faad)
goto fail;
}
#endif
return context;
fail:
freerdp_dsp_context_free(context);
return NULL;
#endif
}
void freerdp_dsp_context_free(FREERDP_DSP_CONTEXT* context)
{
#if defined(WITH_DSP_FFMPEG)
freerdp_dsp_ffmpeg_context_free(context);
#else
if (context)
{
Stream_Free(context->buffer, TRUE);
Stream_Free(context->resample, TRUE);
#if defined(WITH_GSM)
gsm_destroy(context->gsm);
#endif
#if defined(WITH_LAME)
if (context->encoder)
lame_close(context->lame);
else
hip_decode_exit(context->hip);
#endif
#if defined(WITH_FAAD2)
if (!context->encoder)
NeAACDecClose(context->faad);
#endif
#if defined (WITH_FAAC)
if (context->faac)
faacEncClose(context->faac);
#endif
#if defined(WITH_SOXR)
soxr_delete(context->sox);
#endif
free(context);
}
#endif
}
BOOL freerdp_dsp_encode(FREERDP_DSP_CONTEXT* context, const AUDIO_FORMAT* srcFormat,
const BYTE* data, size_t length, wStream* out)
{
#if defined(WITH_DSP_FFMPEG)
return freerdp_dsp_ffmpeg_encode(context, srcFormat, data, length, out);
#else
const BYTE* resampleData;
size_t resampleLength;
AUDIO_FORMAT format;
if (!context || !context->encoder || !srcFormat || !data || !out)
return FALSE;
format = *srcFormat;
if (!freerdp_dsp_channel_mix(context, data, length, srcFormat, &resampleData, &resampleLength))
return FALSE;
format.nChannels = context->format.nChannels;
if (!freerdp_dsp_resample(context, resampleData, resampleLength, &format, &data, &length))
return FALSE;
switch (context->format.wFormatTag)
{
case WAVE_FORMAT_PCM:
if (!Stream_EnsureRemainingCapacity(out, length))
return FALSE;
Stream_Write(out, data, length);
return TRUE;
case WAVE_FORMAT_ADPCM:
return freerdp_dsp_encode_ms_adpcm(context, data, length, out);
case WAVE_FORMAT_DVI_ADPCM:
return freerdp_dsp_encode_ima_adpcm(context, data, length, out);
#if defined(WITH_GSM)
case WAVE_FORMAT_GSM610:
return freerdp_dsp_encode_gsm610(context, data, length, out);
#endif
#if defined(WITH_LAME)
case WAVE_FORMAT_MPEGLAYER3:
return freerdp_dsp_encode_mp3(context, data, length, out);
#endif
#if defined(WITH_FAAC)
case WAVE_FORMAT_AAC_MS:
return freerdp_dsp_encode_faac(context, data, length, out);
#endif
default:
return FALSE;
}
return FALSE;
#endif
}
BOOL freerdp_dsp_decode(FREERDP_DSP_CONTEXT* context, const AUDIO_FORMAT* srcFormat,
const BYTE* data, size_t length, wStream* out)
{
#if defined(WITH_DSP_FFMPEG)
return freerdp_dsp_ffmpeg_decode(context, srcFormat, data, length, out);
#else
if (!context || context->encoder || !srcFormat || !data || !out)
return FALSE;
switch (context->format.wFormatTag)
{
case WAVE_FORMAT_PCM:
if (!Stream_EnsureRemainingCapacity(out, length))
return FALSE;
Stream_Write(out, data, length);
return TRUE;
case WAVE_FORMAT_ADPCM:
return freerdp_dsp_decode_ms_adpcm(context, data, length, out);
case WAVE_FORMAT_DVI_ADPCM:
return freerdp_dsp_decode_ima_adpcm(context, data, length, out);
#if defined(WITH_GSM)
case WAVE_FORMAT_GSM610:
return freerdp_dsp_decode_gsm610(context, data, length, out);
#endif
#if defined(WITH_LAME)
case WAVE_FORMAT_MPEGLAYER3:
return freerdp_dsp_decode_mp3(context, data, length, out);
#endif
#if defined(WITH_FAAD2)
case WAVE_FORMAT_AAC_MS:
return freerdp_dsp_decode_faad(context, data, length, out);
#endif
default:
return FALSE;
}
return FALSE;
#endif
}
BOOL freerdp_dsp_supports_format(const AUDIO_FORMAT* format, BOOL encode)
{
#if defined(WITH_DSP_FFMPEG)
return freerdp_dsp_ffmpeg_supports_format(format, encode);
#else
#if !defined(WITH_DSP_EXPERIMENTAL)
WINPR_UNUSED(encode);
#endif
switch (format->wFormatTag)
{
case WAVE_FORMAT_PCM:
return TRUE;
#if defined(WITH_DSP_EXPERIMENTAL)
case WAVE_FORMAT_ADPCM:
case WAVE_FORMAT_DVI_ADPCM:
return TRUE;
#endif
#if defined(WITH_GSM)
case WAVE_FORMAT_GSM610:
#if defined(WITH_DSP_EXPERIMENTAL)
return TRUE;
#else
return !encode;
#endif
#endif
#if defined(WITH_LAME)
case WAVE_FORMAT_MPEGLAYER3:
#if defined(WITH_DSP_EXPERIMENTAL)
return TRUE;
#else
return !encode;
#endif
#endif
case WAVE_FORMAT_AAC_MS:
#if defined(WITH_FAAD2)
if (!encode)
return TRUE;
#endif
#if defined(WITH_FAAC) && defined(WITH_DSP_EXPERIMENTAL)
if (encode)
return TRUE;
#endif
default:
return FALSE;
}
return FALSE;
#endif
}
BOOL freerdp_dsp_context_reset(FREERDP_DSP_CONTEXT* context, const AUDIO_FORMAT* targetFormat)
{
#if defined(WITH_DSP_FFMPEG)
return freerdp_dsp_ffmpeg_context_reset(context, targetFormat);
#else
if (!context || !targetFormat)
return FALSE;
context->format = *targetFormat;
#if defined(WITH_FAAD2)
context->faadSetup = FALSE;
#endif
#if defined(WITH_FAAC)
if (context->encoder)
{
faacEncConfigurationPtr cfg;
if (context->faac)
faacEncClose(context->faac);
context->faac = faacEncOpen(targetFormat->nSamplesPerSec, targetFormat->nChannels,
&context->faacInputSamples, &context->faacMaxOutputBytes);
if (!context->faac)
return FALSE;
cfg = faacEncGetCurrentConfiguration(context->faac);
cfg->bitRate = 10000;
faacEncSetConfiguration(context->faac, cfg);
}
#endif
#if defined(WITH_SOXR)
{
soxr_io_spec_t iospec = soxr_io_spec(SOXR_INT16, SOXR_INT16);
soxr_error_t error;
soxr_delete(context->sox);
context->sox = soxr_create(context->format.nSamplesPerSec, targetFormat->nSamplesPerSec,
targetFormat->nChannels, &error, &iospec, NULL, NULL);
if (!context->sox || (error != 0))
return FALSE;
}
#endif
return TRUE;
#endif
}
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