FreeRDP/libfreerdp/codec/audio.c
akallabeth afff514ca7
[compiler] add WINPR_ATTR_MALLOC
Add WINPR_ATTR_MALLOC checks to allocating functions to help compilers
find memory leaks or allocation mismatches
2024-09-12 20:09:22 +02:00

300 lines
6.9 KiB
C

/**
* FreeRDP: A Remote Desktop Protocol Implementation
* Audio Formats
*
* Copyright 2013 Marc-Andre Moreau <marcandre.moreau@gmail.com>
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
* You may obtain a copy of the License at
*
* http://www.apache.org/licenses/LICENSE-2.0
*
* Unless required by applicable law or agreed to in writing, software
* distributed under the License is distributed on an "AS IS" BASIS,
* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
* See the License for the specific language governing permissions and
* limitations under the License.
*/
#include <freerdp/config.h>
#include <winpr/crt.h>
#include <freerdp/log.h>
#include <freerdp/codec/audio.h>
#define TAG FREERDP_TAG("codec")
UINT32 audio_format_compute_time_length(const AUDIO_FORMAT* format, size_t size)
{
UINT32 mstime = 0;
UINT32 wSamples = 0;
/**
* [MSDN-AUDIOFORMAT]:
* http://msdn.microsoft.com/en-us/library/ms713497.aspx
*/
if (format->wBitsPerSample)
{
const size_t samples = (size * 8) / format->wBitsPerSample;
WINPR_ASSERT(samples <= UINT32_MAX);
wSamples = (UINT32)samples;
mstime = (((wSamples * 1000) / format->nSamplesPerSec) / format->nChannels);
}
else
{
mstime = 0;
if (format->wFormatTag == WAVE_FORMAT_GSM610)
{
UINT16 nSamplesPerBlock = 0;
if ((format->cbSize == 2) && (format->data))
{
nSamplesPerBlock = *((UINT16*)format->data);
const size_t samples = (size / format->nBlockAlign) * nSamplesPerBlock;
WINPR_ASSERT(samples <= UINT32_MAX);
wSamples = (UINT32)samples;
mstime = (((wSamples * 1000) / format->nSamplesPerSec) / format->nChannels);
}
else
{
WLog_ERR(TAG,
"audio_format_compute_time_length: invalid WAVE_FORMAT_GSM610 format");
}
}
else
{
WLog_ERR(TAG, "audio_format_compute_time_length: unknown format %" PRIu16 "",
format->wFormatTag);
}
}
return mstime;
}
const char* audio_format_get_tag_string(UINT16 wFormatTag)
{
switch (wFormatTag)
{
case WAVE_FORMAT_PCM:
return "WAVE_FORMAT_PCM";
case WAVE_FORMAT_ADPCM:
return "WAVE_FORMAT_ADPCM";
case WAVE_FORMAT_ALAW:
return "WAVE_FORMAT_ALAW";
case WAVE_FORMAT_MULAW:
return "WAVE_FORMAT_MULAW";
case WAVE_FORMAT_DVI_ADPCM:
return "WAVE_FORMAT_DVI_ADPCM";
case WAVE_FORMAT_GSM610:
return "WAVE_FORMAT_GSM610";
case WAVE_FORMAT_MSG723:
return "WAVE_FORMAT_MSG723";
case WAVE_FORMAT_DSPGROUP_TRUESPEECH:
return "WAVE_FORMAT_DSPGROUP_TRUESPEECH ";
case WAVE_FORMAT_MPEGLAYER3:
return "WAVE_FORMAT_MPEGLAYER3";
case WAVE_FORMAT_WMAUDIO2:
return "WAVE_FORMAT_WMAUDIO2";
case WAVE_FORMAT_AAC_MS:
return "WAVE_FORMAT_AAC_MS";
}
return "WAVE_FORMAT_UNKNOWN";
}
void audio_format_print(wLog* log, DWORD level, const AUDIO_FORMAT* format)
{
WLog_Print(log, level,
"%s:\t wFormatTag: 0x%04" PRIX16 " nChannels: %" PRIu16 " nSamplesPerSec: %" PRIu32
" "
"nAvgBytesPerSec: %" PRIu32 " nBlockAlign: %" PRIu16 " wBitsPerSample: %" PRIu16
" cbSize: %" PRIu16 "",
audio_format_get_tag_string(format->wFormatTag), format->wFormatTag,
format->nChannels, format->nSamplesPerSec, format->nAvgBytesPerSec,
format->nBlockAlign, format->wBitsPerSample, format->cbSize);
}
void audio_formats_print(wLog* log, DWORD level, const AUDIO_FORMAT* formats, UINT16 count)
{
if (formats)
{
WLog_Print(log, level, "AUDIO_FORMATS (%" PRIu16 ") ={", count);
for (UINT32 index = 0; index < count; index++)
{
const AUDIO_FORMAT* format = &formats[index];
WLog_Print(log, level, "\t");
audio_format_print(log, level, format);
}
WLog_Print(log, level, "}");
}
}
BOOL audio_format_read(wStream* s, AUDIO_FORMAT* format)
{
if (!s || !format)
return FALSE;
if (!Stream_CheckAndLogRequiredLength(TAG, s, 18))
return FALSE;
Stream_Read_UINT16(s, format->wFormatTag);
Stream_Read_UINT16(s, format->nChannels);
Stream_Read_UINT32(s, format->nSamplesPerSec);
Stream_Read_UINT32(s, format->nAvgBytesPerSec);
Stream_Read_UINT16(s, format->nBlockAlign);
Stream_Read_UINT16(s, format->wBitsPerSample);
Stream_Read_UINT16(s, format->cbSize);
if (!Stream_CheckAndLogRequiredLength(TAG, s, format->cbSize))
return FALSE;
format->data = NULL;
if (format->cbSize > 0)
{
format->data = malloc(format->cbSize);
if (!format->data)
return FALSE;
Stream_Read(s, format->data, format->cbSize);
}
return TRUE;
}
BOOL audio_format_write(wStream* s, const AUDIO_FORMAT* format)
{
if (!s || !format)
return FALSE;
if (!Stream_EnsureRemainingCapacity(s, 18 + format->cbSize))
return FALSE;
Stream_Write_UINT16(s, format->wFormatTag); /* wFormatTag (WAVE_FORMAT_PCM) */
Stream_Write_UINT16(s, format->nChannels); /* nChannels */
Stream_Write_UINT32(s, format->nSamplesPerSec); /* nSamplesPerSec */
Stream_Write_UINT32(s, format->nAvgBytesPerSec); /* nAvgBytesPerSec */
Stream_Write_UINT16(s, format->nBlockAlign); /* nBlockAlign */
Stream_Write_UINT16(s, format->wBitsPerSample); /* wBitsPerSample */
Stream_Write_UINT16(s, format->cbSize); /* cbSize */
if (format->cbSize > 0)
Stream_Write(s, format->data, format->cbSize);
return TRUE;
}
BOOL audio_format_copy(const AUDIO_FORMAT* WINPR_RESTRICT srcFormat,
AUDIO_FORMAT* WINPR_RESTRICT dstFormat)
{
if (!srcFormat || !dstFormat)
return FALSE;
*dstFormat = *srcFormat;
if (srcFormat->cbSize > 0)
{
dstFormat->data = malloc(srcFormat->cbSize);
if (!dstFormat->data)
return FALSE;
memcpy(dstFormat->data, srcFormat->data, dstFormat->cbSize);
}
return TRUE;
}
BOOL audio_format_compatible(const AUDIO_FORMAT* with, const AUDIO_FORMAT* what)
{
if (!with || !what)
return FALSE;
if (with->wFormatTag != WAVE_FORMAT_UNKNOWN)
{
if (with->wFormatTag != what->wFormatTag)
return FALSE;
}
if (with->nChannels != 0)
{
if (with->nChannels != what->nChannels)
return FALSE;
}
if (with->nSamplesPerSec != 0)
{
if (with->nSamplesPerSec != what->nSamplesPerSec)
return FALSE;
}
if (with->wBitsPerSample != 0)
{
if (with->wBitsPerSample != what->wBitsPerSample)
return FALSE;
}
return TRUE;
}
static BOOL audio_format_valid(const AUDIO_FORMAT* format)
{
if (!format)
return FALSE;
if (format->nChannels == 0)
return FALSE;
if (format->nSamplesPerSec == 0)
return FALSE;
return TRUE;
}
AUDIO_FORMAT* audio_format_new(void)
{
return audio_formats_new(1);
}
AUDIO_FORMAT* audio_formats_new(size_t count)
{
return calloc(count, sizeof(AUDIO_FORMAT));
}
void audio_format_free(AUDIO_FORMAT* format)
{
if (format)
free(format->data);
}
void audio_formats_free(AUDIO_FORMAT* formats, size_t count)
{
if (formats)
{
for (size_t index = 0; index < count; index++)
{
AUDIO_FORMAT* format = &formats[index];
audio_format_free(format);
}
free(formats);
}
}