afff514ca7
Add WINPR_ATTR_MALLOC checks to allocating functions to help compilers find memory leaks or allocation mismatches
300 lines
6.9 KiB
C
300 lines
6.9 KiB
C
/**
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* FreeRDP: A Remote Desktop Protocol Implementation
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* Audio Formats
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*
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* Copyright 2013 Marc-Andre Moreau <marcandre.moreau@gmail.com>
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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/crt.h>
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#include <freerdp/log.h>
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#include <freerdp/codec/audio.h>
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#define TAG FREERDP_TAG("codec")
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UINT32 audio_format_compute_time_length(const AUDIO_FORMAT* format, size_t size)
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{
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UINT32 mstime = 0;
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UINT32 wSamples = 0;
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/**
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* [MSDN-AUDIOFORMAT]:
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* http://msdn.microsoft.com/en-us/library/ms713497.aspx
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*/
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if (format->wBitsPerSample)
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{
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const size_t samples = (size * 8) / format->wBitsPerSample;
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WINPR_ASSERT(samples <= UINT32_MAX);
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wSamples = (UINT32)samples;
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mstime = (((wSamples * 1000) / format->nSamplesPerSec) / format->nChannels);
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}
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else
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{
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mstime = 0;
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if (format->wFormatTag == WAVE_FORMAT_GSM610)
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{
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UINT16 nSamplesPerBlock = 0;
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if ((format->cbSize == 2) && (format->data))
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{
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nSamplesPerBlock = *((UINT16*)format->data);
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const size_t samples = (size / format->nBlockAlign) * nSamplesPerBlock;
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WINPR_ASSERT(samples <= UINT32_MAX);
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wSamples = (UINT32)samples;
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mstime = (((wSamples * 1000) / format->nSamplesPerSec) / format->nChannels);
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}
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else
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{
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WLog_ERR(TAG,
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"audio_format_compute_time_length: invalid WAVE_FORMAT_GSM610 format");
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}
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}
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else
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{
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WLog_ERR(TAG, "audio_format_compute_time_length: unknown format %" PRIu16 "",
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format->wFormatTag);
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}
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}
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return mstime;
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}
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const char* audio_format_get_tag_string(UINT16 wFormatTag)
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{
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switch (wFormatTag)
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{
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case WAVE_FORMAT_PCM:
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return "WAVE_FORMAT_PCM";
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case WAVE_FORMAT_ADPCM:
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return "WAVE_FORMAT_ADPCM";
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case WAVE_FORMAT_ALAW:
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return "WAVE_FORMAT_ALAW";
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case WAVE_FORMAT_MULAW:
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return "WAVE_FORMAT_MULAW";
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case WAVE_FORMAT_DVI_ADPCM:
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return "WAVE_FORMAT_DVI_ADPCM";
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case WAVE_FORMAT_GSM610:
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return "WAVE_FORMAT_GSM610";
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case WAVE_FORMAT_MSG723:
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return "WAVE_FORMAT_MSG723";
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case WAVE_FORMAT_DSPGROUP_TRUESPEECH:
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return "WAVE_FORMAT_DSPGROUP_TRUESPEECH ";
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case WAVE_FORMAT_MPEGLAYER3:
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return "WAVE_FORMAT_MPEGLAYER3";
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case WAVE_FORMAT_WMAUDIO2:
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return "WAVE_FORMAT_WMAUDIO2";
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case WAVE_FORMAT_AAC_MS:
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return "WAVE_FORMAT_AAC_MS";
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}
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return "WAVE_FORMAT_UNKNOWN";
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}
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void audio_format_print(wLog* log, DWORD level, const AUDIO_FORMAT* format)
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{
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WLog_Print(log, level,
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"%s:\t wFormatTag: 0x%04" PRIX16 " nChannels: %" PRIu16 " nSamplesPerSec: %" PRIu32
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" "
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"nAvgBytesPerSec: %" PRIu32 " nBlockAlign: %" PRIu16 " wBitsPerSample: %" PRIu16
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" cbSize: %" PRIu16 "",
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audio_format_get_tag_string(format->wFormatTag), format->wFormatTag,
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format->nChannels, format->nSamplesPerSec, format->nAvgBytesPerSec,
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format->nBlockAlign, format->wBitsPerSample, format->cbSize);
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}
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void audio_formats_print(wLog* log, DWORD level, const AUDIO_FORMAT* formats, UINT16 count)
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{
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if (formats)
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{
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WLog_Print(log, level, "AUDIO_FORMATS (%" PRIu16 ") ={", count);
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for (UINT32 index = 0; index < count; index++)
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{
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const AUDIO_FORMAT* format = &formats[index];
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WLog_Print(log, level, "\t");
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audio_format_print(log, level, format);
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}
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WLog_Print(log, level, "}");
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}
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}
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BOOL audio_format_read(wStream* s, AUDIO_FORMAT* format)
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{
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if (!s || !format)
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return FALSE;
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if (!Stream_CheckAndLogRequiredLength(TAG, s, 18))
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return FALSE;
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Stream_Read_UINT16(s, format->wFormatTag);
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Stream_Read_UINT16(s, format->nChannels);
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Stream_Read_UINT32(s, format->nSamplesPerSec);
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Stream_Read_UINT32(s, format->nAvgBytesPerSec);
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Stream_Read_UINT16(s, format->nBlockAlign);
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Stream_Read_UINT16(s, format->wBitsPerSample);
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Stream_Read_UINT16(s, format->cbSize);
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if (!Stream_CheckAndLogRequiredLength(TAG, s, format->cbSize))
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return FALSE;
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format->data = NULL;
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if (format->cbSize > 0)
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{
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format->data = malloc(format->cbSize);
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if (!format->data)
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return FALSE;
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Stream_Read(s, format->data, format->cbSize);
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}
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return TRUE;
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}
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BOOL audio_format_write(wStream* s, const AUDIO_FORMAT* format)
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{
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if (!s || !format)
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return FALSE;
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if (!Stream_EnsureRemainingCapacity(s, 18 + format->cbSize))
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return FALSE;
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Stream_Write_UINT16(s, format->wFormatTag); /* wFormatTag (WAVE_FORMAT_PCM) */
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Stream_Write_UINT16(s, format->nChannels); /* nChannels */
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Stream_Write_UINT32(s, format->nSamplesPerSec); /* nSamplesPerSec */
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Stream_Write_UINT32(s, format->nAvgBytesPerSec); /* nAvgBytesPerSec */
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Stream_Write_UINT16(s, format->nBlockAlign); /* nBlockAlign */
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Stream_Write_UINT16(s, format->wBitsPerSample); /* wBitsPerSample */
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Stream_Write_UINT16(s, format->cbSize); /* cbSize */
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if (format->cbSize > 0)
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Stream_Write(s, format->data, format->cbSize);
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return TRUE;
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}
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BOOL audio_format_copy(const AUDIO_FORMAT* WINPR_RESTRICT srcFormat,
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AUDIO_FORMAT* WINPR_RESTRICT dstFormat)
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{
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if (!srcFormat || !dstFormat)
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return FALSE;
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*dstFormat = *srcFormat;
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if (srcFormat->cbSize > 0)
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{
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dstFormat->data = malloc(srcFormat->cbSize);
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if (!dstFormat->data)
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return FALSE;
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memcpy(dstFormat->data, srcFormat->data, dstFormat->cbSize);
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}
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return TRUE;
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}
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BOOL audio_format_compatible(const AUDIO_FORMAT* with, const AUDIO_FORMAT* what)
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{
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if (!with || !what)
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return FALSE;
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if (with->wFormatTag != WAVE_FORMAT_UNKNOWN)
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{
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if (with->wFormatTag != what->wFormatTag)
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return FALSE;
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}
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if (with->nChannels != 0)
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{
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if (with->nChannels != what->nChannels)
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return FALSE;
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}
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if (with->nSamplesPerSec != 0)
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{
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if (with->nSamplesPerSec != what->nSamplesPerSec)
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return FALSE;
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}
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if (with->wBitsPerSample != 0)
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{
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if (with->wBitsPerSample != what->wBitsPerSample)
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return FALSE;
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}
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return TRUE;
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}
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static BOOL audio_format_valid(const AUDIO_FORMAT* format)
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{
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if (!format)
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return FALSE;
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if (format->nChannels == 0)
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return FALSE;
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if (format->nSamplesPerSec == 0)
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return FALSE;
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return TRUE;
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}
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AUDIO_FORMAT* audio_format_new(void)
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{
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return audio_formats_new(1);
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}
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AUDIO_FORMAT* audio_formats_new(size_t count)
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{
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return calloc(count, sizeof(AUDIO_FORMAT));
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}
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void audio_format_free(AUDIO_FORMAT* format)
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{
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if (format)
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free(format->data);
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}
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void audio_formats_free(AUDIO_FORMAT* formats, size_t count)
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{
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if (formats)
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{
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for (size_t index = 0; index < count; index++)
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{
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AUDIO_FORMAT* format = &formats[index];
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audio_format_free(format);
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}
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free(formats);
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}
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}
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