1508 lines
37 KiB
C
1508 lines
37 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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#include <freerdp/config.h>
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#include <winpr/assert.h>
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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/log.h>
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#include <freerdp/codec/dsp.h>
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#if !defined(WITH_DSP_FFMPEG)
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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_OPUS)
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#include <opus/opus.h>
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#define OPUS_MAX_FRAMES 5760
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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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#if defined(WITH_SOXR)
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#include <soxr.h>
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#endif
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#else
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#include "dsp_ffmpeg.h"
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#endif
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#define TAG FREERDP_TAG("dsp")
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#if !defined(WITH_DSP_FFMPEG)
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typedef union
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{
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struct
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{
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size_t packet_size;
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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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} ADPCM;
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struct S_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* channelmix;
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wStream* resample;
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wStream* buffer;
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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_OPUS)
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OpusDecoder* opus_decoder;
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OpusEncoder* opus_encoder;
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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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#if defined(WITH_SOXR)
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soxr_t sox;
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#endif
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};
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#if defined(WITH_OPUS)
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static BOOL opus_is_valid_samplerate(const AUDIO_FORMAT* format)
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{
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WINPR_ASSERT(format);
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switch (format->nSamplesPerSec)
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{
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case 8000:
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case 12000:
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case 16000:
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case 24000:
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case 48000:
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return TRUE;
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default:
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return FALSE;
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}
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}
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#endif
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static INT16 read_int16(const BYTE* src)
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{
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return (INT16)(src[0] | (src[1] << 8));
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}
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static BOOL freerdp_dsp_channel_mix(FREERDP_DSP_CONTEXT* context, const BYTE* src, size_t size,
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const AUDIO_FORMAT* srcFormat, const BYTE** data,
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size_t* length)
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{
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UINT32 bpp;
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size_t samples;
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if (!context || !data || !length)
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return FALSE;
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if (srcFormat->wFormatTag != WAVE_FORMAT_PCM)
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return FALSE;
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bpp = srcFormat->wBitsPerSample > 8 ? 2 : 1;
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samples = size / bpp / srcFormat->nChannels;
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if (context->format.nChannels == srcFormat->nChannels)
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{
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*data = src;
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*length = size;
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return TRUE;
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}
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Stream_SetPosition(context->channelmix, 0);
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/* Destination has more channels than source */
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if (context->format.nChannels > srcFormat->nChannels)
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{
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switch (srcFormat->nChannels)
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{
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case 1:
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if (!Stream_EnsureCapacity(context->channelmix, size * 2))
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return FALSE;
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for (size_t x = 0; x < samples; x++)
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{
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for (size_t y = 0; y < bpp; y++)
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Stream_Write_UINT8(context->channelmix, src[x * bpp + y]);
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for (size_t y = 0; y < bpp; y++)
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Stream_Write_UINT8(context->channelmix, src[x * bpp + y]);
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}
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Stream_SealLength(context->channelmix);
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*data = Stream_Buffer(context->channelmix);
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*length = Stream_Length(context->channelmix);
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return TRUE;
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case 2: /* We only support stereo, so we can not handle this case. */
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default: /* Unsupported number of channels */
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return FALSE;
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}
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}
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/* Destination has less channels than source */
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switch (srcFormat->nChannels)
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{
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case 2:
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if (!Stream_EnsureCapacity(context->channelmix, size / 2))
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return FALSE;
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/* Simply drop second channel.
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* TODO: Calculate average */
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for (size_t x = 0; x < samples; x++)
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{
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for (size_t y = 0; y < bpp; y++)
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Stream_Write_UINT8(context->channelmix, src[2 * x * bpp + y]);
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}
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Stream_SealLength(context->channelmix);
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*data = Stream_Buffer(context->channelmix);
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*length = Stream_Length(context->channelmix);
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return TRUE;
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case 1: /* Invalid, do we want to use a 0 channel sound? */
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default: /* Unsupported number of channels */
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return FALSE;
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}
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return FALSE;
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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, const BYTE* src, size_t size,
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const AUDIO_FORMAT* srcFormat, const BYTE** data, size_t* length)
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{
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#if defined(WITH_SOXR)
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soxr_error_t error;
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size_t idone, odone;
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size_t sframes, rframes;
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size_t rsize;
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size_t sbytes, rbytes;
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size_t dstChannels;
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size_t srcChannels;
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size_t srcBytesPerFrame, dstBytesPerFrame;
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#endif
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AUDIO_FORMAT format;
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if (srcFormat->wFormatTag != WAVE_FORMAT_PCM)
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{
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WLog_ERR(TAG, "requires %s for sample input, got %s",
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audio_format_get_tag_string(WAVE_FORMAT_PCM),
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audio_format_get_tag_string(srcFormat->wFormatTag));
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return FALSE;
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}
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/* We want to ignore differences of source and destination format. */
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format = *srcFormat;
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format.wFormatTag = WAVE_FORMAT_UNKNOWN;
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format.wBitsPerSample = 0;
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if (audio_format_compatible(&format, &context->format))
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{
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*data = src;
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*length = size;
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return TRUE;
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}
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#if defined(WITH_SOXR)
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srcBytesPerFrame = (srcFormat->wBitsPerSample > 8) ? 2 : 1;
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dstBytesPerFrame = (context->format.wBitsPerSample > 8) ? 2 : 1;
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srcChannels = srcFormat->nChannels;
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dstChannels = context->format.nChannels;
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sbytes = srcChannels * srcBytesPerFrame;
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sframes = size / sbytes;
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rbytes = dstBytesPerFrame * dstChannels;
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/* Integer rounding correct division */
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rframes = (sframes * context->format.nSamplesPerSec + (srcFormat->nSamplesPerSec + 1) / 2) /
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srcFormat->nSamplesPerSec;
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rsize = rframes * rbytes;
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if (!Stream_EnsureCapacity(context->resample, rsize))
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return FALSE;
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error = soxr_process(context->sox, src, sframes, &idone, Stream_Buffer(context->resample),
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Stream_Capacity(context->resample) / rbytes, &odone);
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Stream_SetLength(context->resample, odone * rbytes);
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*data = Stream_Buffer(context->resample);
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*length = Stream_Length(context->resample);
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return (error == 0) ? TRUE : FALSE;
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#else
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WLog_ERR(TAG, "Missing resample support, recompile -DWITH_SOXR=ON or -DWITH_DSP_FFMPEG=ON");
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return FALSE;
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#endif
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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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-1, -1, -1, -1, 2, 4, 6, 8, -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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7, 8, 9, 10, 11, 12, 13, 14, 16, 17, 19, 21, 23,
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25, 28, 31, 34, 37, 41, 45, 50, 55, 60, 66, 73, 80,
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88, 97, 107, 118, 130, 143, 157, 173, 190, 209, 230, 253, 279,
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307, 337, 371, 408, 449, 494, 544, 598, 658, 724, 796, 876, 963,
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1060, 1166, 1282, 1411, 1552, 1707, 1878, 2066, 2272, 2499, 2749, 3024, 3327,
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3660, 4026, 4428, 4871, 5358, 5894, 6484, 7132, 7845, 8630, 9493, 10442, 11487,
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12635, 13899, 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, 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, const BYTE* src, size_t size,
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wStream* out)
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{
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BYTE sample;
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UINT16 decoded;
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size_t 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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if (!Stream_EnsureCapacity(out, out_size))
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return FALSE;
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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] =
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(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] =
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(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 (size_t i = 0; i < 8; i++)
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{
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BYTE* dst = Stream_Pointer(out);
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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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if (!Stream_SafeSeek(out, 32))
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return FALSE;
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size -= 8;
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}
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else
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{
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BYTE* dst = Stream_Pointer(out);
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if (!Stream_SafeSeek(out, 4))
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return FALSE;
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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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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, const BYTE* src, size_t size,
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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*)/* API does not modify */ &src[offset],
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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, const BYTE* src, size_t size,
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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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const gsm_signal* signal = (const 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, (gsm_signal*)/* API does not modify */ signal,
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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, const BYTE* src, size_t size,
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wStream* out)
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{
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int rc;
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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, size,
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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, (size_t)rc * context->format.nChannels * 2))
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return FALSE;
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for (size_t x = 0; x < rc; x++)
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{
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Stream_Write_UINT16(out, (UINT16)pcm_l[x]);
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Stream_Write_UINT16(out, (UINT16)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, const BYTE* src, size_t size,
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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;
|
|
|
|
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;
|
|
unsigned int bpp;
|
|
size_t nrSamples;
|
|
int rc;
|
|
|
|
if (!context || !src || !out)
|
|
return FALSE;
|
|
|
|
bpp = context->format.wBitsPerSample / 8;
|
|
nrSamples = size / bpp;
|
|
|
|
if (!Stream_EnsureRemainingCapacity(context->buffer, nrSamples * sizeof(int16_t)))
|
|
return FALSE;
|
|
|
|
for (size_t x = 0; x < nrSamples; x++)
|
|
{
|
|
Stream_Write_INT16(context->buffer, inSamples[x]);
|
|
if (Stream_GetPosition(context->buffer) / bpp >= context->faacInputSamples)
|
|
{
|
|
if (!Stream_EnsureRemainingCapacity(out, context->faacMaxOutputBytes))
|
|
return FALSE;
|
|
rc = faacEncEncode(context->faac, (int32_t*)Stream_Buffer(context->buffer),
|
|
context->faacInputSamples, Stream_Pointer(out),
|
|
Stream_GetRemainingCapacity(out));
|
|
if (rc < 0)
|
|
return FALSE;
|
|
if (rc > 0)
|
|
Stream_Seek(out, (size_t)rc);
|
|
Stream_SetPosition(context->buffer, 0);
|
|
}
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
#endif
|
|
|
|
#if defined(WITH_OPUS)
|
|
static BOOL freerdp_dsp_decode_opus(FREERDP_DSP_CONTEXT* context, const BYTE* src, size_t size,
|
|
wStream* out)
|
|
{
|
|
size_t max_size = 5760;
|
|
int frames;
|
|
|
|
if (!context || !src || !out)
|
|
return FALSE;
|
|
|
|
/* Max packet duration is 120ms (5760 at 48KHz) */
|
|
max_size = OPUS_MAX_FRAMES * context->format.nChannels * sizeof(int16_t);
|
|
if (!Stream_EnsureRemainingCapacity(context->buffer, max_size))
|
|
return FALSE;
|
|
|
|
frames = opus_decode(context->opus_decoder, src, size, Stream_Pointer(out), OPUS_MAX_FRAMES, 0);
|
|
if (frames < 0)
|
|
return FALSE;
|
|
|
|
Stream_Seek(out, frames * context->format.nChannels * sizeof(int16_t));
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static BOOL freerdp_dsp_encode_opus(FREERDP_DSP_CONTEXT* context, const BYTE* src, size_t size,
|
|
wStream* out)
|
|
{
|
|
if (!context || !src || !out)
|
|
return FALSE;
|
|
|
|
/* Max packet duration is 120ms (5760 at 48KHz) */
|
|
const size_t max_size = OPUS_MAX_FRAMES * context->format.nChannels * sizeof(int16_t);
|
|
if (!Stream_EnsureRemainingCapacity(context->buffer, max_size))
|
|
return FALSE;
|
|
|
|
const int src_frames = size / sizeof(opus_int16) / context->format.nChannels;
|
|
const opus_int16* src_data = (const opus_int16*)src;
|
|
const int frames =
|
|
opus_encode(context->opus_encoder, src_data, src_frames, Stream_Pointer(out), max_size);
|
|
if (frames < 0)
|
|
return FALSE;
|
|
return Stream_SafeSeek(out, frames * context->format.nChannels * sizeof(int16_t));
|
|
}
|
|
#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;
|
|
size_t offset = 0;
|
|
|
|
if (!context || !src || !out)
|
|
return FALSE;
|
|
|
|
if (!context->faadSetup)
|
|
{
|
|
union
|
|
{
|
|
const void* cpv;
|
|
void* pv;
|
|
} cnv;
|
|
unsigned long samplerate;
|
|
unsigned char channels;
|
|
long err;
|
|
cnv.cpv = src;
|
|
err = NeAACDecInit(context->faad, /* API is not modifying content */ cnv.pv, 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)
|
|
{
|
|
union
|
|
{
|
|
const void* cpv;
|
|
void* pv;
|
|
} cnv;
|
|
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);
|
|
|
|
cnv.cpv = &src[offset];
|
|
NeAACDecDecode2(context->faad, &info, cnv.pv, 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)
|
|
{
|
|
INT16 sample;
|
|
BYTE encoded;
|
|
size_t align;
|
|
|
|
if (!Stream_EnsureRemainingCapacity(out, size))
|
|
return FALSE;
|
|
|
|
align = (context->format.nChannels > 1) ? 32 : 4;
|
|
|
|
while (size >= align)
|
|
{
|
|
if (Stream_GetPosition(context->buffer) % context->format.nBlockAlign == 0)
|
|
{
|
|
Stream_Write_UINT8(context->buffer, context->adpcm.ima.last_sample[0] & 0xFF);
|
|
Stream_Write_UINT8(context->buffer, (context->adpcm.ima.last_sample[0] >> 8) & 0xFF);
|
|
Stream_Write_UINT8(context->buffer, (BYTE)context->adpcm.ima.last_step[0]);
|
|
Stream_Write_UINT8(context->buffer, 0);
|
|
|
|
if (context->format.nChannels > 1)
|
|
{
|
|
Stream_Write_UINT8(context->buffer, context->adpcm.ima.last_sample[1] & 0xFF);
|
|
Stream_Write_UINT8(context->buffer,
|
|
(context->adpcm.ima.last_sample[1] >> 8) & 0xFF);
|
|
Stream_Write_UINT8(context->buffer, (BYTE)context->adpcm.ima.last_step[1]);
|
|
Stream_Write_UINT8(context->buffer, 0);
|
|
}
|
|
}
|
|
|
|
if (context->format.nChannels > 1)
|
|
{
|
|
BYTE* dst = Stream_Pointer(context->buffer);
|
|
ZeroMemory(dst, 8);
|
|
|
|
for (size_t 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;
|
|
}
|
|
|
|
if (!Stream_SafeSeek(context->buffer, 8))
|
|
return FALSE;
|
|
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;
|
|
Stream_Write_UINT8(context->buffer, encoded);
|
|
size -= 4;
|
|
}
|
|
|
|
if (Stream_GetPosition(context->buffer) >= context->adpcm.ima.packet_size)
|
|
{
|
|
BYTE* bsrc = Stream_Buffer(context->buffer);
|
|
Stream_Write(out, bsrc, context->adpcm.ima.packet_size);
|
|
Stream_SetPosition(context->buffer, 0);
|
|
}
|
|
}
|
|
|
|
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 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;
|
|
|
|
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;
|
|
Stream_Write_INT16(out, (INT16)context->adpcm.ms.sample2[0]);
|
|
Stream_Write_INT16(out, (INT16)context->adpcm.ms.sample2[1]);
|
|
Stream_Write_INT16(out, (INT16)context->adpcm.ms.sample1[0]);
|
|
Stream_Write_INT16(out, (INT16)context->adpcm.ms.sample1[1]);
|
|
}
|
|
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;
|
|
Stream_Write_INT16(out, (INT16)context->adpcm.ms.sample2[0]);
|
|
Stream_Write_INT16(out, (INT16)context->adpcm.ms.sample1[0]);
|
|
}
|
|
}
|
|
|
|
if (channels > 1)
|
|
{
|
|
sample = *src++;
|
|
size--;
|
|
Stream_Write_INT16(out,
|
|
freerdp_dsp_decode_ms_adpcm_sample(&context->adpcm, sample >> 4, 0));
|
|
Stream_Write_INT16(
|
|
out, freerdp_dsp_decode_ms_adpcm_sample(&context->adpcm, sample & 0x0F, 1));
|
|
sample = *src++;
|
|
size--;
|
|
Stream_Write_INT16(out,
|
|
freerdp_dsp_decode_ms_adpcm_sample(&context->adpcm, sample >> 4, 0));
|
|
Stream_Write_INT16(
|
|
out, freerdp_dsp_decode_ms_adpcm_sample(&context->adpcm, sample & 0x0F, 1));
|
|
}
|
|
else
|
|
{
|
|
sample = *src++;
|
|
size--;
|
|
Stream_Write_INT16(out,
|
|
freerdp_dsp_decode_ms_adpcm_sample(&context->adpcm, sample >> 4, 0));
|
|
Stream_Write_INT16(
|
|
out, freerdp_dsp_decode_ms_adpcm_sample(&context->adpcm, sample & 0x0F, 0));
|
|
}
|
|
}
|
|
|
|
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)
|
|
{
|
|
size_t start;
|
|
INT32 sample;
|
|
const size_t step = 8 + ((context->format.nChannels > 1) ? 4 : 0);
|
|
|
|
if (!Stream_EnsureRemainingCapacity(out, size))
|
|
return FALSE;
|
|
|
|
start = Stream_GetPosition(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)
|
|
{
|
|
BYTE val;
|
|
if ((Stream_GetPosition(out) - start) % context->format.nBlockAlign == 0)
|
|
{
|
|
if (context->format.nChannels > 1)
|
|
{
|
|
Stream_Write_UINT8(out, context->adpcm.ms.predictor[0]);
|
|
Stream_Write_UINT8(out, context->adpcm.ms.predictor[1]);
|
|
Stream_Write_UINT8(out, (context->adpcm.ms.delta[0] & 0xFF));
|
|
Stream_Write_UINT8(out, ((context->adpcm.ms.delta[0] >> 8) & 0xFF));
|
|
Stream_Write_UINT8(out, (context->adpcm.ms.delta[1] & 0xFF));
|
|
Stream_Write_UINT8(out, ((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);
|
|
|
|
Stream_Write_INT16(out, (INT16)context->adpcm.ms.sample1[0]);
|
|
Stream_Write_INT16(out, (INT16)context->adpcm.ms.sample1[1]);
|
|
Stream_Write_INT16(out, (INT16)context->adpcm.ms.sample2[0]);
|
|
Stream_Write_INT16(out, (INT16)context->adpcm.ms.sample2[1]);
|
|
|
|
src += 8;
|
|
size -= 8;
|
|
}
|
|
else
|
|
{
|
|
Stream_Write_UINT8(out, context->adpcm.ms.predictor[0]);
|
|
Stream_Write_UINT8(out, (BYTE)(context->adpcm.ms.delta[0] & 0xFF));
|
|
Stream_Write_UINT8(out, (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);
|
|
|
|
Stream_Write_INT16(out, (INT16)context->adpcm.ms.sample1[0]);
|
|
Stream_Write_INT16(out, (INT16)context->adpcm.ms.sample2[0]);
|
|
src += 4;
|
|
size -= 4;
|
|
}
|
|
}
|
|
|
|
sample = read_int16(src);
|
|
src += 2;
|
|
Stream_Write_UINT8(
|
|
out, (freerdp_dsp_encode_ms_adpcm_sample(&context->adpcm, sample, 0) << 4) & 0xFF);
|
|
sample = read_int16(src);
|
|
src += 2;
|
|
|
|
Stream_Read_UINT8(out, val);
|
|
val += freerdp_dsp_encode_ms_adpcm_sample(&context->adpcm, sample,
|
|
context->format.nChannels > 1 ? 1 : 0);
|
|
Stream_Write_UINT8(out, val);
|
|
size -= 4;
|
|
}
|
|
|
|
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->channelmix = Stream_New(NULL, 4096);
|
|
|
|
if (!context->channelmix)
|
|
goto fail;
|
|
|
|
context->resample = Stream_New(NULL, 4096);
|
|
|
|
if (!context->resample)
|
|
goto fail;
|
|
|
|
context->buffer = Stream_New(NULL, 4096);
|
|
|
|
if (!context->buffer)
|
|
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->channelmix, TRUE);
|
|
Stream_Free(context->resample, TRUE);
|
|
Stream_Free(context->buffer, 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_OPUS)
|
|
|
|
if (context->opus_decoder)
|
|
opus_decoder_destroy(context->opus_decoder);
|
|
if (context->opus_encoder)
|
|
opus_encoder_destroy(context->opus_encoder);
|
|
|
|
#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
|
|
#if defined(WITH_OPUS)
|
|
|
|
case WAVE_FORMAT_OPUS:
|
|
return freerdp_dsp_encode_opus(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
|
|
|
|
#if defined(WITH_OPUS)
|
|
case WAVE_FORMAT_OPUS:
|
|
return freerdp_dsp_decode_opus(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:
|
|
return FALSE;
|
|
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)
|
|
|
|
if (encode)
|
|
return TRUE;
|
|
|
|
#endif
|
|
#if defined(WITH_OPUS)
|
|
WINPR_FALLTHROUGH
|
|
case WAVE_FORMAT_OPUS:
|
|
return opus_is_valid_samplerate(format);
|
|
#endif
|
|
WINPR_FALLTHROUGH
|
|
default:
|
|
return FALSE;
|
|
}
|
|
|
|
return FALSE;
|
|
#endif
|
|
}
|
|
|
|
BOOL freerdp_dsp_context_reset(FREERDP_DSP_CONTEXT* context, const AUDIO_FORMAT* targetFormat,
|
|
UINT32 FramesPerPacket)
|
|
{
|
|
#if defined(WITH_DSP_FFMPEG)
|
|
return freerdp_dsp_ffmpeg_context_reset(context, targetFormat);
|
|
#else
|
|
|
|
if (!context || !targetFormat)
|
|
return FALSE;
|
|
|
|
context->format = *targetFormat;
|
|
|
|
if (context->format.wFormatTag == WAVE_FORMAT_DVI_ADPCM)
|
|
{
|
|
size_t min_frame_data =
|
|
1ull * context->format.wBitsPerSample * context->format.nChannels * FramesPerPacket;
|
|
size_t data_per_block = (context->format.nBlockAlign - 4 * context->format.nChannels) * 8;
|
|
size_t nb_block_per_packet = min_frame_data / data_per_block;
|
|
|
|
if (min_frame_data % data_per_block)
|
|
nb_block_per_packet++;
|
|
|
|
context->adpcm.ima.packet_size = nb_block_per_packet * context->format.nBlockAlign;
|
|
Stream_EnsureCapacity(context->buffer, context->adpcm.ima.packet_size);
|
|
Stream_SetPosition(context->buffer, 0);
|
|
}
|
|
|
|
#if defined(WITH_OPUS)
|
|
|
|
if (opus_is_valid_samplerate(&context->format))
|
|
{
|
|
if (!context->encoder)
|
|
{
|
|
int opus_error = OPUS_OK;
|
|
|
|
context->opus_decoder = opus_decoder_create(context->format.nSamplesPerSec,
|
|
context->format.nChannels, &opus_error);
|
|
if (opus_error != OPUS_OK)
|
|
return FALSE;
|
|
}
|
|
else
|
|
{
|
|
int opus_error = OPUS_OK;
|
|
|
|
context->opus_encoder =
|
|
opus_encoder_create(context->format.nSamplesPerSec, context->format.nChannels,
|
|
OPUS_APPLICATION_VOIP, &opus_error);
|
|
if (opus_error != OPUS_OK)
|
|
return FALSE;
|
|
|
|
opus_error = opus_encoder_ctl(context->opus_encoder,
|
|
OPUS_SET_BITRATE(context->format.nAvgBytesPerSec * 8));
|
|
if (opus_error != OPUS_OK)
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
#endif
|
|
#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->inputFormat = FAAC_INPUT_16BIT;
|
|
cfg->outputFormat = 0;
|
|
cfg->mpegVersion = MPEG4;
|
|
cfg->useTns = 1;
|
|
cfg->bandWidth = targetFormat->nAvgBytesPerSec;
|
|
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
|
|
}
|