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Milestone 5: deliver embedded RDP sessions and lifecycle hardening

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This commit is contained in:
Keith Smith
2026-03-03 18:59:26 -07:00
parent 230a401386
commit 36006bd4aa
2941 changed files with 724359 additions and 77 deletions
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30
third_party/FreeRDP/channels/tsmf/client/alsa/CMakeLists.txt vendored Normal file
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# FreeRDP: A Remote Desktop Protocol Implementation
# FreeRDP cmake build script
#
# Copyright 2012 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.
define_channel_client_subsystem("tsmf" "alsa" "audio")
find_package(ALSA REQUIRED)
freerdp_client_pc_add_requires_private("alsa")
set(${MODULE_PREFIX}_SRCS tsmf_alsa.c)
set(${MODULE_PREFIX}_LIBS winpr freerdp ${ALSA_LIBRARIES})
include_directories(..)
include_directories(SYSTEM ${ALSA_INCLUDE_DIRS})
add_channel_client_subsystem_library(${MODULE_PREFIX} ${MODULE_NAME} ${CHANNEL_NAME} "" TRUE "")

238
third_party/FreeRDP/channels/tsmf/client/alsa/tsmf_alsa.c vendored Normal file
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/**
* FreeRDP: A Remote Desktop Protocol Implementation
* Video Redirection Virtual Channel - ALSA Audio Device
*
* Copyright 2010-2011 Vic Lee
*
* 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 <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <unistd.h>
#include <winpr/crt.h>
#include <alsa/asoundlib.h>
#include <freerdp/types.h>
#include <freerdp/codec/dsp.h>
#include "tsmf_audio.h"
typedef struct
{
ITSMFAudioDevice iface;
char device[32];
snd_pcm_t* out_handle;
UINT32 source_rate;
UINT32 actual_rate;
UINT32 source_channels;
UINT32 actual_channels;
UINT32 bytes_per_sample;
} TSMFAlsaAudioDevice;
static BOOL tsmf_alsa_open_device(TSMFAlsaAudioDevice* alsa)
{
int error = 0;
error = snd_pcm_open(&alsa->out_handle, alsa->device, SND_PCM_STREAM_PLAYBACK, 0);
if (error < 0)
{
WLog_ERR(TAG, "failed to open device %s", alsa->device);
return FALSE;
}
DEBUG_TSMF("open device %s", alsa->device);
return TRUE;
}
static BOOL tsmf_alsa_open(ITSMFAudioDevice* audio, const char* device)
{
TSMFAlsaAudioDevice* alsa = (TSMFAlsaAudioDevice*)audio;
if (!device)
{
strncpy(alsa->device, "default", sizeof(alsa->device));
}
else
{
strncpy(alsa->device, device, sizeof(alsa->device) - 1);
}
return tsmf_alsa_open_device(alsa);
}
static BOOL tsmf_alsa_set_format(ITSMFAudioDevice* audio, UINT32 sample_rate, UINT32 channels,
UINT32 bits_per_sample)
{
int error = 0;
snd_pcm_uframes_t frames = 0;
snd_pcm_hw_params_t* hw_params = nullptr;
snd_pcm_sw_params_t* sw_params = nullptr;
TSMFAlsaAudioDevice* alsa = (TSMFAlsaAudioDevice*)audio;
if (!alsa->out_handle)
return FALSE;
snd_pcm_drop(alsa->out_handle);
alsa->actual_rate = alsa->source_rate = sample_rate;
alsa->actual_channels = alsa->source_channels = channels;
alsa->bytes_per_sample = bits_per_sample / 8;
error = snd_pcm_hw_params_malloc(&hw_params);
if (error < 0)
{
WLog_ERR(TAG, "snd_pcm_hw_params_malloc failed");
return FALSE;
}
snd_pcm_hw_params_any(alsa->out_handle, hw_params);
snd_pcm_hw_params_set_access(alsa->out_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
snd_pcm_hw_params_set_format(alsa->out_handle, hw_params, SND_PCM_FORMAT_S16_LE);
snd_pcm_hw_params_set_rate_near(alsa->out_handle, hw_params, &alsa->actual_rate, nullptr);
snd_pcm_hw_params_set_channels_near(alsa->out_handle, hw_params, &alsa->actual_channels);
frames = sample_rate;
snd_pcm_hw_params_set_buffer_size_near(alsa->out_handle, hw_params, &frames);
snd_pcm_hw_params(alsa->out_handle, hw_params);
snd_pcm_hw_params_free(hw_params);
error = snd_pcm_sw_params_malloc(&sw_params);
if (error < 0)
{
WLog_ERR(TAG, "snd_pcm_sw_params_malloc");
return FALSE;
}
snd_pcm_sw_params_current(alsa->out_handle, sw_params);
snd_pcm_sw_params_set_start_threshold(alsa->out_handle, sw_params, frames / 2);
snd_pcm_sw_params(alsa->out_handle, sw_params);
snd_pcm_sw_params_free(sw_params);
snd_pcm_prepare(alsa->out_handle);
DEBUG_TSMF("sample_rate %" PRIu32 " channels %" PRIu32 " bits_per_sample %" PRIu32 "",
sample_rate, channels, bits_per_sample);
DEBUG_TSMF("hardware buffer %lu frames", frames);
if ((alsa->actual_rate != alsa->source_rate) ||
(alsa->actual_channels != alsa->source_channels))
{
DEBUG_TSMF("actual rate %" PRIu32 " / channel %" PRIu32 " is different "
"from source rate %" PRIu32 " / channel %" PRIu32 ", resampling required.",
alsa->actual_rate, alsa->actual_channels, alsa->source_rate,
alsa->source_channels);
}
return TRUE;
}
static BOOL tsmf_alsa_play(ITSMFAudioDevice* audio, const BYTE* src, UINT32 data_size)
{
const BYTE* pindex = nullptr;
TSMFAlsaAudioDevice* alsa = (TSMFAlsaAudioDevice*)audio;
DEBUG_TSMF("data_size %" PRIu32 "", data_size);
if (alsa->out_handle)
{
const size_t rbytes_per_frame = 1ULL * alsa->actual_channels * alsa->bytes_per_sample;
pindex = src;
const BYTE* end = pindex + data_size;
while (pindex < end)
{
const size_t len = (size_t)(end - pindex);
const size_t frames = len / rbytes_per_frame;
snd_pcm_sframes_t error = snd_pcm_writei(alsa->out_handle, pindex, frames);
if (error == -EPIPE)
{
snd_pcm_recover(alsa->out_handle, -EPIPE, 0);
error = 0;
}
else if (error < 0)
{
DEBUG_TSMF("error len %ld", error);
snd_pcm_close(alsa->out_handle);
alsa->out_handle = 0;
tsmf_alsa_open_device(alsa);
break;
}
DEBUG_TSMF("%d frames played.", error);
if (error == 0)
break;
pindex += (size_t)error * rbytes_per_frame;
}
}
return TRUE;
}
static UINT64 tsmf_alsa_get_latency(ITSMFAudioDevice* audio)
{
UINT64 latency = 0;
snd_pcm_sframes_t frames = 0;
TSMFAlsaAudioDevice* alsa = (TSMFAlsaAudioDevice*)audio;
if (alsa->out_handle && alsa->actual_rate > 0 &&
snd_pcm_delay(alsa->out_handle, &frames) == 0 && frames > 0)
{
latency = ((UINT64)frames) * 10000000LL / (UINT64)alsa->actual_rate;
}
return latency;
}
static BOOL tsmf_alsa_flush(WINPR_ATTR_UNUSED ITSMFAudioDevice* audio)
{
return TRUE;
}
static void tsmf_alsa_free(ITSMFAudioDevice* audio)
{
TSMFAlsaAudioDevice* alsa = (TSMFAlsaAudioDevice*)audio;
DEBUG_TSMF("");
if (alsa->out_handle)
{
snd_pcm_drain(alsa->out_handle);
snd_pcm_close(alsa->out_handle);
}
free(alsa);
}
FREERDP_ENTRY_POINT(UINT VCAPITYPE alsa_freerdp_tsmf_client_audio_subsystem_entry(void* ptr))
{
ITSMFAudioDevice** sptr = (ITSMFAudioDevice**)ptr;
WINPR_ASSERT(sptr);
*sptr = nullptr;
TSMFAlsaAudioDevice* alsa = calloc(1, sizeof(TSMFAlsaAudioDevice));
if (!alsa)
return ERROR_OUTOFMEMORY;
alsa->iface.Open = tsmf_alsa_open;
alsa->iface.SetFormat = tsmf_alsa_set_format;
alsa->iface.Play = tsmf_alsa_play;
alsa->iface.GetLatency = tsmf_alsa_get_latency;
alsa->iface.Flush = tsmf_alsa_flush;
alsa->iface.Free = tsmf_alsa_free;
*sptr = &alsa->iface;
return CHANNEL_RC_OK;
}