1211 lines
27 KiB
C
1211 lines
27 KiB
C
/**
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* FreeRDP: A Remote Desktop Protocol Implementation
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* Digital Sound Processing
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*
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* Copyright 2010-2011 Vic Lee
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*
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* Licensed under the Apache License, Version 2.0 (the "License");
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* you may not use this file except in compliance with the License.
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* You may obtain a copy of the License at
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*
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* http://www.apache.org/licenses/LICENSE-2.0
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*
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* Unless required by applicable law or agreed to in writing, software
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* distributed under the License is distributed on an "AS IS" BASIS,
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* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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* See the License for the specific language governing permissions and
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* limitations under the License.
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <stdio.h>
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#include <stdlib.h>
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#include <string.h>
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#include <winpr/crt.h>
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#include <freerdp/types.h>
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#include <freerdp/codec/dsp.h>
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#if defined(WITH_GSM)
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#include <gsm/gsm.h>
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#endif
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#if defined(WITH_LAME)
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#include <lame/lame.h>
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#endif
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#if defined(WITH_DSP_FFMPEG)
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#include "dsp_ffmpeg.h"
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#endif
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#if defined(WITH_FAAD2)
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#include <neaacdec.h>
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#endif
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#if defined(WITH_FAAC)
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#include <faac.h>
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#endif
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#define TAG FREERDP_TAG("dsp")
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union _ADPCM
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{
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struct
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{
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INT16 last_sample[2];
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INT16 last_step[2];
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} ima;
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struct
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{
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BYTE predictor[2];
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INT32 delta[2];
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INT32 sample1[2];
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INT32 sample2[2];
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} ms;
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};
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typedef union _ADPCM ADPCM;
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struct _FREERDP_DSP_CONTEXT
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{
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BOOL encoder;
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ADPCM adpcm;
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AUDIO_FORMAT format;
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wStream* buffer;
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wStream* resample;
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#if defined(WITH_GSM)
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gsm gsm;
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#endif
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#if defined(WITH_LAME)
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lame_t lame;
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hip_t hip;
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#endif
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#if defined(WITH_FAAD2)
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NeAACDecHandle faad;
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BOOL faadSetup;
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#endif
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#if defined(WITH_FAAC)
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faacEncHandle faac;
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unsigned long faacInputSamples;
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unsigned long faacMaxOutputBytes;
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#endif
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};
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/**
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* Microsoft Multimedia Standards Update
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* http://download.microsoft.com/download/9/8/6/9863C72A-A3AA-4DDB-B1BA-CA8D17EFD2D4/RIFFNEW.pdf
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*/
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static BOOL freerdp_dsp_resample(FREERDP_DSP_CONTEXT* context,
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const BYTE* src, size_t bytes_per_sample,
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UINT32 schan, UINT32 srate, size_t sframes,
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UINT32 rchan, UINT32 rrate)
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{
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BYTE* p;
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int rframes;
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int rsize;
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int i, j;
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int n1, n2;
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int sbytes, rbytes;
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sbytes = bytes_per_sample * schan;
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rbytes = bytes_per_sample * rchan;
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rframes = sframes * rrate / srate;
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rsize = rbytes * rframes;
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if (!Stream_EnsureCapacity(context->resample, rsize + 1024))
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return FALSE;
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p = Stream_Buffer(context->resample);
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for (i = 0; i < rframes; i++)
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{
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n1 = i * srate / rrate;
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if (n1 >= sframes)
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n1 = sframes - 1;
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n2 = (n1 * rrate == i * srate || n1 == sframes - 1 ? n1 : n1 + 1);
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for (j = 0; j < rbytes; j++)
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{
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/* Nearest Interpolation, probably the easiest, but works */
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*p++ = (i * srate - n1 * rrate > n2 * rrate - i * srate ?
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src[n2 * sbytes + (j % sbytes)] :
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src[n1 * sbytes + (j % sbytes)]);
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}
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}
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Stream_SetPointer(context->resample, p);
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Stream_SealLength(context->resample);
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return TRUE;
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}
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/**
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* Microsoft IMA ADPCM specification:
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*
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* http://wiki.multimedia.cx/index.php?title=Microsoft_IMA_ADPCM
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* http://wiki.multimedia.cx/index.php?title=IMA_ADPCM
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*/
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static const INT16 ima_step_index_table[] =
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{
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-1, -1, -1, -1, 2, 4, 6, 8,
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-1, -1, -1, -1, 2, 4, 6, 8
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};
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static const INT16 ima_step_size_table[] =
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{
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7, 8, 9, 10, 11, 12, 13, 14, 16, 17,
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19, 21, 23, 25, 28, 31, 34, 37, 41, 45,
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50, 55, 60, 66, 73, 80, 88, 97, 107, 118,
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130, 143, 157, 173, 190, 209, 230, 253, 279, 307,
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337, 371, 408, 449, 494, 544, 598, 658, 724, 796,
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876, 963, 1060, 1166, 1282, 1411, 1552, 1707, 1878, 2066,
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2272, 2499, 2749, 3024, 3327, 3660, 4026, 4428, 4871, 5358,
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5894, 6484, 7132, 7845, 8630, 9493, 10442, 11487, 12635, 13899,
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15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794, 32767
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};
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static UINT16 dsp_decode_ima_adpcm_sample(ADPCM* adpcm,
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unsigned int channel, BYTE sample)
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{
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INT32 ss;
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INT32 d;
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ss = ima_step_size_table[adpcm->ima.last_step[channel]];
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d = (ss >> 3);
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if (sample & 1)
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d += (ss >> 2);
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if (sample & 2)
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d += (ss >> 1);
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if (sample & 4)
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d += ss;
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if (sample & 8)
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d = -d;
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d += adpcm->ima.last_sample[channel];
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if (d < -32768)
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d = -32768;
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else if (d > 32767)
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d = 32767;
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adpcm->ima.last_sample[channel] = (INT16) d;
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adpcm->ima.last_step[channel] += ima_step_index_table[sample];
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if (adpcm->ima.last_step[channel] < 0)
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adpcm->ima.last_step[channel] = 0;
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else if (adpcm->ima.last_step[channel] > 88)
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adpcm->ima.last_step[channel] = 88;
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return (UINT16) d;
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}
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static BOOL freerdp_dsp_decode_ima_adpcm(FREERDP_DSP_CONTEXT* context,
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const BYTE* src, size_t size, wStream* out)
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{
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BYTE* dst;
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BYTE sample;
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UINT16 decoded;
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UINT32 out_size = size * 4;
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UINT32 channel;
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const UINT32 block_size = context->format.nBlockAlign;
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const UINT32 channels = context->format.nChannels;
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int i;
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if (!Stream_EnsureCapacity(out, out_size))
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return FALSE;
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dst = Stream_Pointer(out);
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while (size > 0)
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{
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if (size % block_size == 0)
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{
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context->adpcm.ima.last_sample[0] = (INT16)(((UINT16)(*src)) | (((UINT16)(*(src + 1))) << 8));
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context->adpcm.ima.last_step[0] = (INT16)(*(src + 2));
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src += 4;
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size -= 4;
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out_size -= 16;
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if (channels > 1)
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{
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context->adpcm.ima.last_sample[1] = (INT16)(((UINT16)(*src)) | (((UINT16)(*(src + 1))) << 8));
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context->adpcm.ima.last_step[1] = (INT16)(*(src + 2));
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src += 4;
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size -= 4;
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out_size -= 16;
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}
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}
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if (channels > 1)
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{
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for (i = 0; i < 8; i++)
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{
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channel = (i < 4 ? 0 : 1);
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sample = ((*src) & 0x0f);
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decoded = dsp_decode_ima_adpcm_sample(&context->adpcm, channel, sample);
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dst[((i & 3) << 3) + (channel << 1)] = (decoded & 0xFF);
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dst[((i & 3) << 3) + (channel << 1) + 1] = (decoded >> 8);
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sample = ((*src) >> 4);
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decoded = dsp_decode_ima_adpcm_sample(&context->adpcm, channel, sample);
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dst[((i & 3) << 3) + (channel << 1) + 4] = (decoded & 0xFF);
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dst[((i & 3) << 3) + (channel << 1) + 5] = (decoded >> 8);
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src++;
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}
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dst += 32;
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size -= 8;
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}
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else
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{
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sample = ((*src) & 0x0f);
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decoded = dsp_decode_ima_adpcm_sample(&context->adpcm, 0, sample);
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*dst++ = (decoded & 0xFF);
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*dst++ = (decoded >> 8);
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sample = ((*src) >> 4);
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decoded = dsp_decode_ima_adpcm_sample(&context->adpcm, 0, sample);
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*dst++ = (decoded & 0xFF);
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*dst++ = (decoded >> 8);
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src++;
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size--;
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}
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}
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Stream_SetPointer(out, dst);
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return TRUE;
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}
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#if defined(WITH_GSM)
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static BOOL freerdp_dsp_decode_gsm610(FREERDP_DSP_CONTEXT* context,
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const BYTE* src, size_t size, wStream* out)
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{
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size_t offset = 0;
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while (offset < size)
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{
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int rc;
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gsm_signal gsmBlockBuffer[160] = { 0 };
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rc = gsm_decode(context->gsm, (gsm_byte*) &src[offset], gsmBlockBuffer);
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if (rc < 0)
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return FALSE;
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if ((offset % 65) == 0)
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offset += 33;
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else
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offset += 32;
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if (!Stream_EnsureRemainingCapacity(out, sizeof(gsmBlockBuffer)))
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return FALSE;
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Stream_Write(out, (void*) gsmBlockBuffer, sizeof(gsmBlockBuffer));
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}
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return TRUE;
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}
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static BOOL freerdp_dsp_encode_gsm610(FREERDP_DSP_CONTEXT* context,
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const BYTE* src, size_t size, wStream* out)
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{
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size_t offset = 0;
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while (offset < size)
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{
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gsm_signal* signal = (gsm_signal*)&src[offset];
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if (!Stream_EnsureRemainingCapacity(out, sizeof(gsm_frame)))
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return FALSE;
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gsm_encode(context->gsm, signal, Stream_Pointer(out));
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if ((offset % 65) == 0)
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Stream_Seek(out, 33);
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else
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Stream_Seek(out, 32);
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offset += 160;
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}
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return TRUE;
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}
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#endif
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#if defined(WITH_LAME)
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static BOOL freerdp_dsp_decode_mp3(FREERDP_DSP_CONTEXT* context,
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const BYTE* src, size_t size, wStream* out)
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{
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int rc, x;
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short* pcm_l;
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short* pcm_r;
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size_t buffer_size;
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if (!context || !src || !out)
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return FALSE;
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buffer_size = 2 * context->format.nChannels * context->format.nSamplesPerSec;
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if (!Stream_EnsureCapacity(context->buffer, 2 * buffer_size))
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return FALSE;
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pcm_l = (short*)Stream_Buffer(context->buffer);
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pcm_r = (short*)Stream_Buffer(context->buffer) + buffer_size;
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rc = hip_decode(context->hip, (unsigned char*)/* API is not modifying content */src,
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size, pcm_l, pcm_r);
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if (rc <= 0)
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return FALSE;
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if (!Stream_EnsureRemainingCapacity(out, rc * context->format.nChannels * 2))
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return FALSE;
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for (x = 0; x < rc; x++)
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{
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Stream_Write_UINT16(out, pcm_l[x]);
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Stream_Write_UINT16(out, pcm_r[x]);
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}
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return TRUE;
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}
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static BOOL freerdp_dsp_encode_mp3(FREERDP_DSP_CONTEXT* context,
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const BYTE* src, size_t size, wStream* out)
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{
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size_t samples_per_channel;
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int rc;
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if (!context || !src || !out)
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return FALSE;
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samples_per_channel = size / context->format.nChannels / context->format.wBitsPerSample / 8;
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/* Ensure worst case buffer size for mp3 stream taken from LAME header */
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if (!Stream_EnsureRemainingCapacity(out, 1.25 * samples_per_channel + 7200))
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return FALSE;
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samples_per_channel = size / 2 /* size of a sample */ / context->format.nChannels;
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rc = lame_encode_buffer_interleaved(context->lame, (short*)src, samples_per_channel,
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Stream_Pointer(out), Stream_GetRemainingCapacity(out));
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if (rc < 0)
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return FALSE;
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Stream_Seek(out, rc);
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return TRUE;
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}
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#endif
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#if defined(WITH_FAAC)
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static BOOL freerdp_dsp_encode_faac(FREERDP_DSP_CONTEXT* context,
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const BYTE* src, size_t size, wStream* out)
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{
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int16_t* inSamples = (int16_t*)src;
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int32_t* outSamples;
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unsigned int nrSamples, x;
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int rc;
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if (!context || !src || !out)
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return FALSE;
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nrSamples = size / context->format.nChannels / context->format.wBitsPerSample / 8;
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if (!Stream_EnsureCapacity(context->buffer,
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context->faacInputSamples * sizeof(int32_t) * context->format.nChannels))
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return FALSE;
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if (!Stream_EnsureRemainingCapacity(out, context->faacMaxOutputBytes))
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return FALSE;
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outSamples = Stream_Buffer(context->buffer);
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for (x = 0; x < nrSamples * context->format.nChannels; x++)
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outSamples[x] = inSamples[x];
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rc = faacEncEncode(context->faac, outSamples, nrSamples * context->format.nChannels,
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Stream_Pointer(out), Stream_GetRemainingCapacity(out));
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if (rc < 0)
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return FALSE;
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else if (rc > 0)
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Stream_Seek(out, rc);
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return TRUE;
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}
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#endif
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#if defined(WITH_FAAD2)
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static BOOL freerdp_dsp_decode_faad(FREERDP_DSP_CONTEXT* context,
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const BYTE* src, size_t size, wStream* out)
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{
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NeAACDecFrameInfo info;
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void* output;
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size_t offset = 0;
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if (!context || !src || !out)
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return FALSE;
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if (!context->faadSetup)
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{
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unsigned long samplerate;
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unsigned char channels;
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char err = NeAACDecInit(context->faad, /* API is not modifying content */(unsigned char*)src,
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size, &samplerate, &channels);
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if (err != 0)
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return FALSE;
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if (channels != context->format.nChannels)
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return FALSE;
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if (samplerate != context->format.nSamplesPerSec)
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return FALSE;
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context->faadSetup = TRUE;
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}
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while (offset < size)
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{
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size_t outSize;
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void* sample_buffer;
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outSize = context->format.nSamplesPerSec * context->format.nChannels *
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context->format.wBitsPerSample / 8;
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if (!Stream_EnsureRemainingCapacity(out, outSize))
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return FALSE;
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sample_buffer = Stream_Pointer(out);
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output = NeAACDecDecode2(context->faad, &info, (unsigned char*)&src[offset], size - offset,
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&sample_buffer, Stream_GetRemainingCapacity(out));
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if (info.error != 0)
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return FALSE;
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offset += info.bytesconsumed;
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if (info.samples == 0)
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continue;
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Stream_Seek(out, info.samples * context->format.wBitsPerSample / 8);
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}
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return TRUE;
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}
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#endif
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/**
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* 0 1 2 3
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* 2 0 6 4 10 8 14 12 <left>
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*
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* 4 5 6 7
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* 3 1 7 5 11 9 15 13 <right>
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*/
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static const struct
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{
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BYTE byte_num;
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BYTE byte_shift;
|
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} ima_stereo_encode_map[] =
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{
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{ 0, 0 },
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{ 4, 0 },
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{ 0, 4 },
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{ 4, 4 },
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{ 1, 0 },
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{ 5, 0 },
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{ 1, 4 },
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{ 5, 4 },
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{ 2, 0 },
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{ 6, 0 },
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{ 2, 4 },
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{ 6, 4 },
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{ 3, 0 },
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{ 7, 0 },
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{ 3, 4 },
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{ 7, 4 }
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};
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|
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static BYTE dsp_encode_ima_adpcm_sample(ADPCM* adpcm, int channel, INT16 sample)
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{
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INT32 e;
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INT32 d;
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INT32 ss;
|
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BYTE enc;
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INT32 diff;
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ss = ima_step_size_table[adpcm->ima.last_step[channel]];
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d = e = sample - adpcm->ima.last_sample[channel];
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|
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;
|
|
UINT32 out_size;
|
|
out_size = size / 2;
|
|
|
|
if (!Stream_EnsureRemainingCapacity(out, size))
|
|
return FALSE;
|
|
|
|
start = dst = Stream_Pointer(out);
|
|
|
|
while (size > 0)
|
|
{
|
|
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 : 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 UINT32 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] = *((INT16*) src);
|
|
src += 2;
|
|
context->adpcm.ms.delta[1] = *((INT16*) src);
|
|
src += 2;
|
|
context->adpcm.ms.sample1[0] = *((INT16*) src);
|
|
src += 2;
|
|
context->adpcm.ms.sample1[1] = *((INT16*) src);
|
|
src += 2;
|
|
context->adpcm.ms.sample2[0] = *((INT16*) src);
|
|
src += 2;
|
|
context->adpcm.ms.sample2[1] = *((INT16*) src);
|
|
src += 2;
|
|
size -= 14;
|
|
*((INT16*) dst) = context->adpcm.ms.sample2[0];
|
|
dst += 2;
|
|
*((INT16*) dst) = context->adpcm.ms.sample2[1];
|
|
dst += 2;
|
|
*((INT16*) dst) = context->adpcm.ms.sample1[0];
|
|
dst += 2;
|
|
*((INT16*) dst) = context->adpcm.ms.sample1[1];
|
|
dst += 2;
|
|
}
|
|
else
|
|
{
|
|
context->adpcm.ms.predictor[0] = *src++;
|
|
context->adpcm.ms.delta[0] = *((INT16*) src);
|
|
src += 2;
|
|
context->adpcm.ms.sample1[0] = *((INT16*) src);
|
|
src += 2;
|
|
context->adpcm.ms.sample2[0] = *((INT16*) src);
|
|
src += 2;
|
|
size -= 7;
|
|
*((INT16*) dst) = context->adpcm.ms.sample2[0];
|
|
dst += 2;
|
|
*((INT16*) dst) = context->adpcm.ms.sample1[0];
|
|
dst += 2;
|
|
}
|
|
}
|
|
|
|
if (channels > 1)
|
|
{
|
|
sample = *src++;
|
|
size--;
|
|
*((INT16*) dst) = freerdp_dsp_decode_ms_adpcm_sample(&context->adpcm, sample >> 4, 0);
|
|
dst += 2;
|
|
*((INT16*) dst) = freerdp_dsp_decode_ms_adpcm_sample(&context->adpcm, sample & 0x0F, 1);
|
|
dst += 2;
|
|
sample = *src++;
|
|
size--;
|
|
*((INT16*) dst) = freerdp_dsp_decode_ms_adpcm_sample(&context->adpcm, sample >> 4, 0);
|
|
dst += 2;
|
|
*((INT16*) dst) = freerdp_dsp_decode_ms_adpcm_sample(&context->adpcm, sample & 0x0F, 1);
|
|
dst += 2;
|
|
}
|
|
else
|
|
{
|
|
sample = *src++;
|
|
size--;
|
|
*((INT16*) dst) = freerdp_dsp_decode_ms_adpcm_sample(&context->adpcm, sample >> 4, 0);
|
|
dst += 2;
|
|
*((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;
|
|
UINT32 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] = *((INT16*)(src + 4));
|
|
context->adpcm.ms.sample1[1] = *((INT16*)(src + 6));
|
|
context->adpcm.ms.sample2[0] = *((INT16*)(src + 0));
|
|
context->adpcm.ms.sample2[1] = *((INT16*)(src + 2));
|
|
*((INT16*)(dst + 0)) = (INT16) context->adpcm.ms.sample1[0];
|
|
*((INT16*)(dst + 2)) = (INT16) context->adpcm.ms.sample1[1];
|
|
*((INT16*)(dst + 4)) = (INT16) context->adpcm.ms.sample2[0];
|
|
*((INT16*)(dst + 6)) = (INT16) 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] = *((INT16*)(src + 2));
|
|
context->adpcm.ms.sample2[0] = *((INT16*)(src + 0));
|
|
*((INT16*)(dst + 0)) = (INT16) context->adpcm.ms.sample1[0];
|
|
*((INT16*)(dst + 2)) = (INT16) context->adpcm.ms.sample2[0];
|
|
dst += 4;
|
|
src += 4;
|
|
size -= 4;
|
|
}
|
|
}
|
|
|
|
sample = *((INT16*) src);
|
|
src += 2;
|
|
*dst = freerdp_dsp_encode_ms_adpcm_sample(&context->adpcm, sample, 0) << 4;
|
|
sample = *((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;
|
|
}
|
|
|
|
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
|
|
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
|
|
|
|
if (!context || !context->encoder || !srcFormat || !data || !out)
|
|
return FALSE;
|
|
|
|
// TODO: Resample
|
|
|
|
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
|
|
|
|
switch (format->wFormatTag)
|
|
{
|
|
case WAVE_FORMAT_PCM:
|
|
case WAVE_FORMAT_ADPCM:
|
|
case WAVE_FORMAT_DVI_ADPCM:
|
|
return TRUE;
|
|
#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
|
|
return TRUE;
|
|
#endif
|
|
}
|