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Some fixes for audin redirection (#7231)

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* Fixed research for native PCM format

* Fixed compatibility detection for resampling

* dsp: Makes channel mix and encoding process use separated buffers

* Give FramesPerPacket to freerdp_dsp_context_reset function

* dsp: Fix DVI_ADPCM codec by respecting the required packet size

* Forbid mono channel in macos

* Fix audin command line channel option

* Disable ADPCM codec as it does not work

* Refactor valid native codec research

* Native codec research: Try different channels number

* Fix dsp channel mix

* Fix dsp resample
This commit is contained in:
Simon Nivault 2021-08-23 14:49:28 +02:00 committed by GitHub
parent a7cdd03809
commit e42696bb2e
No known key found for this signature in database
GPG Key ID: 4AEE18F83AFDEB23
8 changed files with 85 additions and 42 deletions
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28
channels/audin/client/audin_main.c
View File

@ -416,22 +416,24 @@ static BOOL audin_open_device(AUDIN_PLUGIN* audin, AUDIN_CHANNEL_CALLBACK* callb
if (!supported)
{
/* Default sample rates supported by most backends. */
const UINT32 samplerates[] = { 96000, 48000, 44100, 22050 };
const UINT32 samplerates[] = { format.nSamplesPerSec, 96000, 48000, 44100, 22050 };
BOOL test = FALSE;
size_t x;
format.wFormatTag = WAVE_FORMAT_PCM;
format.wBitsPerSample = 16;
test = IFCALLRESULT(FALSE, audin->device->FormatSupported, audin->device, &format);
if (!test)
format.cbSize = 0;
for (x = 0; x < ARRAYSIZE(samplerates); x++)
{
size_t x;
for (x = 0; x < ARRAYSIZE(samplerates); x++)
{
format.nSamplesPerSec = samplerates[x];
test = IFCALLRESULT(FALSE, audin->device->FormatSupported, audin->device, &format);
if (test)
break;
}
format.nSamplesPerSec = samplerates[x];
format.nChannels = audin->format->nChannels;
test = IFCALLRESULT(FALSE, audin->device->FormatSupported, audin->device, &format);
if (test)
break;
format.nChannels = 3 - format.nChannels;
test = IFCALLRESULT(FALSE, audin->device->FormatSupported, audin->device, &format);
if (test)
break;
}
if (!test)
return FALSE;
@ -447,7 +449,7 @@ static BOOL audin_open_device(AUDIN_PLUGIN* audin, AUDIN_CHANNEL_CALLBACK* callb
if (!supported)
{
if (!freerdp_dsp_context_reset(audin->dsp_context, audin->format))
if (!freerdp_dsp_context_reset(audin->dsp_context, audin->format, audin->FramesPerPacket))
return FALSE;
}
@ -943,7 +945,7 @@ BOOL audin_process_addin_args(AUDIN_PLUGIN* audin, const ADDIN_ARGV* args)
{
unsigned long val = strtoul(arg->Value, NULL, 0);
if ((errno != 0) || (val > UINT16_MAX))
if ((errno != 0) || (val < UINT16_MAX))
audin->fixed_format->nChannels = val;
}
CommandLineSwitchDefault(arg)

3
channels/audin/client/mac/audin_mac.m
View File

@ -120,6 +120,9 @@ static BOOL audin_mac_format_supported(IAudinDevice *device, const AUDIO_FORMAT
if (device == NULL || format == NULL)
return FALSE;
if (format->nChannels != 2)
return FALSE;
req_fmt = audin_mac_get_format(format);
if (req_fmt == 0)

2
channels/audin/server/audin.c
View File

@ -86,7 +86,7 @@ static UINT audin_server_select_format(audin_server_context* context, size_t cli
context->selected_client_format = (SSIZE_T)client_format_index;
if (!freerdp_dsp_context_reset(audin->dsp_context,
&audin->context.client_formats[client_format_index]))
&audin->context.client_formats[client_format_index], 0u))
{
WLog_ERR(TAG, "Failed to reset dsp context format!");
return ERROR_INTERNAL_ERROR;

3
channels/rdpsnd/client/mac/rdpsnd_mac.m
View File

@ -237,6 +237,9 @@ static BOOL rdpsnd_mac_format_supported(rdpsndDevicePlugin *device, const AUDIO_
case WAVE_FORMAT_PCM:
if (format->wBitsPerSample != 16)
return FALSE;
if (format->nChannels != 2)
return FALSE;
return TRUE;
default:

2
channels/rdpsnd/client/rdpsnd_main.c
View File

@ -407,7 +407,7 @@ static BOOL rdpsnd_ensure_device_is_open(rdpsndPlugin* rdpsnd, UINT32 wFormatNo,
if (!supported)
{
if (!freerdp_dsp_context_reset(rdpsnd->dsp_context, format))
if (!freerdp_dsp_context_reset(rdpsnd->dsp_context, format, 0u))
return FALSE;
}

2
channels/rdpsnd/server/rdpsnd_main.c
View File

@ -365,7 +365,7 @@ static UINT rdpsnd_server_select_format(RdpsndServerContext* context, UINT16 cli
context->priv->out_buffer_size = out_buffer_size;
}
freerdp_dsp_context_reset(context->priv->dsp_context, format);
freerdp_dsp_context_reset(context->priv->dsp_context, format, 0u);
out:
LeaveCriticalSection(&context->priv->lock);
return error;

3
include/freerdp/codec/dsp.h
View File

@ -40,7 +40,8 @@ extern "C"
const BYTE* data, size_t length, wStream* out);
FREERDP_API void freerdp_dsp_context_free(FREERDP_DSP_CONTEXT* context);
FREERDP_API BOOL freerdp_dsp_context_reset(FREERDP_DSP_CONTEXT* context,
const AUDIO_FORMAT* targetFormat);
const AUDIO_FORMAT* targetFormat,
UINT32 FramesPerPacket);
#ifdef __cplusplus
}

84
libfreerdp/codec/dsp.c
View File

@ -64,6 +64,7 @@
union _ADPCM {
struct
{
size_t packet_size;
INT16 last_sample[2];
INT16 last_step[2];
} ima;
@ -84,8 +85,9 @@ struct _FREERDP_DSP_CONTEXT
ADPCM adpcm;
AUDIO_FORMAT format;
wStream* buffer;
wStream* channelmix;
wStream* resample;
wStream* buffer;
#if defined(WITH_GSM)
gsm gsm;
@ -136,7 +138,7 @@ static BOOL freerdp_dsp_channel_mix(FREERDP_DSP_CONTEXT* context, const BYTE* sr
return FALSE;
bpp = srcFormat->wBitsPerSample > 8 ? 2 : 1;
samples = size / bpp;
samples = size / bpp / srcFormat->nChannels;
if (context->format.nChannels == srcFormat->nChannels)
{
@ -145,7 +147,7 @@ static BOOL freerdp_dsp_channel_mix(FREERDP_DSP_CONTEXT* context, const BYTE* sr
return TRUE;
}
Stream_SetPosition(context->buffer, 0);
Stream_SetPosition(context->channelmix, 0);
/* Destination has more channels than source */
if (context->format.nChannels > srcFormat->nChannels)
@ -153,21 +155,21 @@ static BOOL freerdp_dsp_channel_mix(FREERDP_DSP_CONTEXT* context, const BYTE* sr
switch (srcFormat->nChannels)
{
case 1:
if (!Stream_EnsureCapacity(context->buffer, size * 2))
if (!Stream_EnsureCapacity(context->channelmix, size * 2))
return FALSE;
for (x = 0; x < samples; x++)
{
for (y = 0; y < bpp; y++)
Stream_Write_UINT8(context->buffer, src[x * bpp + y]);
Stream_Write_UINT8(context->channelmix, src[x * bpp + y]);
for (y = 0; y < bpp; y++)
Stream_Write_UINT8(context->buffer, src[x * bpp + y]);
Stream_Write_UINT8(context->channelmix, src[x * bpp + y]);
}
Stream_SealLength(context->buffer);
*data = Stream_Buffer(context->buffer);
*length = Stream_Length(context->buffer);
Stream_SealLength(context->channelmix);
*data = Stream_Buffer(context->channelmix);
*length = Stream_Length(context->channelmix);
return TRUE;
case 2: /* We only support stereo, so we can not handle this case. */
@ -180,7 +182,7 @@ static BOOL freerdp_dsp_channel_mix(FREERDP_DSP_CONTEXT* context, const BYTE* sr
switch (srcFormat->nChannels)
{
case 2:
if (!Stream_EnsureCapacity(context->buffer, size / 2))
if (!Stream_EnsureCapacity(context->channelmix, size / 2))
return FALSE;
/* Simply drop second channel.
@ -188,12 +190,12 @@ static BOOL freerdp_dsp_channel_mix(FREERDP_DSP_CONTEXT* context, const BYTE* sr
for (x = 0; x < samples; x++)
{
for (y = 0; y < bpp; y++)
Stream_Write_UINT8(context->buffer, src[2 * x * bpp + y]);
Stream_Write_UINT8(context->channelmix, src[2 * x * bpp + y]);
}
Stream_SealLength(context->buffer);
*data = Stream_Buffer(context->buffer);
*length = Stream_Length(context->buffer);
Stream_SealLength(context->channelmix);
*data = Stream_Buffer(context->channelmix);
*length = Stream_Length(context->channelmix);
return TRUE;
case 1: /* Invalid, do we want to use a 0 channel sound? */
@ -238,9 +240,14 @@ static BOOL freerdp_dsp_resample(FREERDP_DSP_CONTEXT* context, const BYTE* src,
/* We want to ignore differences of source and destination format. */
format = *srcFormat;
format.wFormatTag = WAVE_FORMAT_UNKNOWN;
format.wBitsPerSample = 0;
if (audio_format_compatible(&format, &context->format))
{
*data = src;
*length = size;
return TRUE;
}
#if defined(WITH_SOXR)
sbytes = srcChannels * srcBytesPerFrame;
@ -261,10 +268,6 @@ static BOOL freerdp_dsp_resample(FREERDP_DSP_CONTEXT* context, const BYTE* src,
*length = Stream_Length(context->resample);
return (error == 0) ? TRUE : FALSE;
#else
WINPR_UNUSED(src);
WINPR_UNUSED(size);
WINPR_UNUSED(data);
WINPR_UNUSED(length);
WLog_ERR(TAG, "Missing resample support, recompile -DWITH_SOXR=ON or -DWITH_DSP_FFMPEG=ON");
return FALSE;
#endif
@ -720,10 +723,11 @@ static BOOL freerdp_dsp_encode_ima_adpcm(FREERDP_DSP_CONTEXT* context, const BYT
if (!Stream_EnsureRemainingCapacity(out, size))
return FALSE;
start = dst = Stream_Pointer(out);
start = Stream_Buffer(context->buffer);
dst = Stream_Pointer(context->buffer);
align = (context->format.nChannels > 1) ? 32 : 4;
while (size > align)
while (size >= align)
{
if ((dst - start) % context->format.nBlockAlign == 0)
{
@ -768,9 +772,15 @@ static BOOL freerdp_dsp_encode_ima_adpcm(FREERDP_DSP_CONTEXT* context, const BYT
*dst++ = encoded;
size -= 4;
}
if (dst - start == context->adpcm.ima.packet_size)
{
Stream_Write(out, start, context->adpcm.ima.packet_size);
dst = Stream_Buffer(context->buffer);
}
}
Stream_SetPointer(out, dst);
Stream_SetPointer(context->buffer, dst);
return TRUE;
}
@ -1025,9 +1035,9 @@ FREERDP_DSP_CONTEXT* freerdp_dsp_context_new(BOOL encoder)
if (!context)
return NULL;
context->buffer = Stream_New(NULL, 4096);
context->channelmix = Stream_New(NULL, 4096);
if (!context->buffer)
if (!context->channelmix)
goto fail;
context->resample = Stream_New(NULL, 4096);
@ -1035,6 +1045,11 @@ FREERDP_DSP_CONTEXT* freerdp_dsp_context_new(BOOL encoder)
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();
@ -1095,8 +1110,9 @@ void freerdp_dsp_context_free(FREERDP_DSP_CONTEXT* context)
if (context)
{
Stream_Free(context->buffer, TRUE);
Stream_Free(context->channelmix, TRUE);
Stream_Free(context->resample, TRUE);
Stream_Free(context->buffer, TRUE);
#if defined(WITH_GSM)
gsm_destroy(context->gsm);
#endif
@ -1254,6 +1270,7 @@ BOOL freerdp_dsp_supports_format(const AUDIO_FORMAT* format, BOOL encode)
#if defined(WITH_DSP_EXPERIMENTAL)
case WAVE_FORMAT_ADPCM:
return FALSE;
case WAVE_FORMAT_DVI_ADPCM:
return TRUE;
#endif
@ -1297,7 +1314,8 @@ BOOL freerdp_dsp_supports_format(const AUDIO_FORMAT* format, BOOL encode)
#endif
}
BOOL freerdp_dsp_context_reset(FREERDP_DSP_CONTEXT* context, const AUDIO_FORMAT* targetFormat)
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);
@ -1307,6 +1325,22 @@ BOOL freerdp_dsp_context_reset(FREERDP_DSP_CONTEXT* context, const AUDIO_FORMAT*
return FALSE;
context->format = *targetFormat;
if (context->format.wFormatTag == WAVE_FORMAT_DVI_ADPCM)
{
size_t min_frame_data =
context->format.wBitsPerSample * context->format.nChannels * FramesPerPacket * 1ULL;
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_FAAD2)
context->faadSetup = FALSE;
#endif