Adding upstream version 3.3.0+dfsg1.upstream/3.3.0+dfsg1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

2 files changed, 608 insertions, 0 deletions

diff --git a/channels/rdpsnd/client/alsa/CMakeLists.txt b/channels/rdpsnd/client/alsa/CMakeLists.txt
new file mode 100644
index 0000000..3041b95
--- /dev/null
+++ b/channels/rdpsnd/client/alsa/CMakeLists.txt
@@ -0,0 +1,34 @@
+# 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("rdpsnd" "alsa" "")
+
+find_package(ALSA REQUIRED)
+
+set(${MODULE_PREFIX}_SRCS
+	rdpsnd_alsa.c)
+
+set(${MODULE_PREFIX}_LIBS
+    winpr
+    freerdp
+    ${ALSA_LIBRARIES}
+)
+
+include_directories(..)
+include_directories(${ALSA_INCLUDE_DIRS})
+
+add_channel_client_subsystem_library(${MODULE_PREFIX} ${MODULE_NAME} ${CHANNEL_NAME} "" TRUE "")
diff --git a/channels/rdpsnd/client/alsa/rdpsnd_alsa.c b/channels/rdpsnd/client/alsa/rdpsnd_alsa.c
new file mode 100644
index 0000000..97f0ba0
--- /dev/null
+++ b/channels/rdpsnd/client/alsa/rdpsnd_alsa.c
@@ -0,0 +1,574 @@
+/**
+ * FreeRDP: A Remote Desktop Protocol Implementation
+ * Audio Output Virtual Channel
+ *
+ * Copyright 2009-2011 Jay Sorg
+ * Copyright 2010-2011 Vic Lee
+ * Copyright 2015 Thincast Technologies GmbH
+ * Copyright 2015 DI (FH) Martin Haimberger <martin.haimberger@thincast.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 <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+
+#include <winpr/crt.h>
+#include <winpr/cmdline.h>
+#include <winpr/sysinfo.h>
+#include <winpr/collections.h>
+
+#include <alsa/asoundlib.h>
+
+#include <freerdp/types.h>
+#include <freerdp/codec/dsp.h>
+#include <freerdp/channels/log.h>
+
+#include "rdpsnd_main.h"
+
+typedef struct
+{
+	rdpsndDevicePlugin device;
+
+	UINT32 latency;
+	AUDIO_FORMAT aformat;
+	char* device_name;
+	snd_pcm_t* pcm_handle;
+	snd_mixer_t* mixer_handle;
+
+	UINT32 actual_rate;
+	snd_pcm_format_t format;
+	UINT32 actual_channels;
+
+	snd_pcm_uframes_t buffer_size;
+	snd_pcm_uframes_t period_size;
+} rdpsndAlsaPlugin;
+
+#define SND_PCM_CHECK(_func, _status)                  \
+	do                                                 \
+	{                                                  \
+		if (_status < 0)                               \
+		{                                              \
+			WLog_ERR(TAG, "%s: %d\n", _func, _status); \
+			return -1;                                 \
+		}                                              \
+	} while (0)
+
+static int rdpsnd_alsa_set_hw_params(rdpsndAlsaPlugin* alsa)
+{
+	int status = 0;
+	snd_pcm_hw_params_t* hw_params = NULL;
+	snd_pcm_uframes_t buffer_size_max = 0;
+	status = snd_pcm_hw_params_malloc(&hw_params);
+	SND_PCM_CHECK("snd_pcm_hw_params_malloc", status);
+	status = snd_pcm_hw_params_any(alsa->pcm_handle, hw_params);
+	SND_PCM_CHECK("snd_pcm_hw_params_any", status);
+	/* Set interleaved read/write access */
+	status =
+	    snd_pcm_hw_params_set_access(alsa->pcm_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
+	SND_PCM_CHECK("snd_pcm_hw_params_set_access", status);
+	/* Set sample format */
+	status = snd_pcm_hw_params_set_format(alsa->pcm_handle, hw_params, alsa->format);
+	SND_PCM_CHECK("snd_pcm_hw_params_set_format", status);
+	/* Set sample rate */
+	status = snd_pcm_hw_params_set_rate_near(alsa->pcm_handle, hw_params, &alsa->actual_rate, NULL);
+	SND_PCM_CHECK("snd_pcm_hw_params_set_rate_near", status);
+	/* Set number of channels */
+	status = snd_pcm_hw_params_set_channels(alsa->pcm_handle, hw_params, alsa->actual_channels);
+	SND_PCM_CHECK("snd_pcm_hw_params_set_channels", status);
+	/* Get maximum buffer size */
+	status = snd_pcm_hw_params_get_buffer_size_max(hw_params, &buffer_size_max);
+	SND_PCM_CHECK("snd_pcm_hw_params_get_buffer_size_max", status);
+	/**
+	 * ALSA Parameters
+	 *
+	 * http://www.alsa-project.org/main/index.php/FramesPeriods
+	 *
+	 * buffer_size = period_size * periods
+	 * period_bytes = period_size * bytes_per_frame
+	 * bytes_per_frame = channels * bytes_per_sample
+	 *
+	 * A frame is equivalent of one sample being played,
+	 * irrespective of the number of channels or the number of bits
+	 *
+	 * A period is the number of frames in between each hardware interrupt.
+	 *
+	 * The buffer size always has to be greater than one period size.
+	 * Commonly this is (2 * period_size), but some hardware can do 8 periods per buffer.
+	 * It is also possible for the buffer size to not be an integer multiple of the period size.
+	 */
+	int interrupts_per_sec_near = 50;
+	int bytes_per_sec =
+	    (alsa->actual_rate * alsa->aformat.wBitsPerSample / 8 * alsa->actual_channels);
+	alsa->buffer_size = buffer_size_max;
+	alsa->period_size = (bytes_per_sec / interrupts_per_sec_near);
+
+	if (alsa->period_size > buffer_size_max)
+	{
+		WLog_ERR(TAG, "Warning: requested sound buffer size %lu, got %lu instead\n",
+		         alsa->buffer_size, buffer_size_max);
+		alsa->period_size = (buffer_size_max / 8);
+	}
+
+	/* Set buffer size */
+	status =
+	    snd_pcm_hw_params_set_buffer_size_near(alsa->pcm_handle, hw_params, &alsa->buffer_size);
+	SND_PCM_CHECK("snd_pcm_hw_params_set_buffer_size_near", status);
+	/* Set period size */
+	status = snd_pcm_hw_params_set_period_size_near(alsa->pcm_handle, hw_params, &alsa->period_size,
+	                                                NULL);
+	SND_PCM_CHECK("snd_pcm_hw_params_set_period_size_near", status);
+	status = snd_pcm_hw_params(alsa->pcm_handle, hw_params);
+	SND_PCM_CHECK("snd_pcm_hw_params", status);
+	snd_pcm_hw_params_free(hw_params);
+	return 0;
+}
+
+static int rdpsnd_alsa_set_sw_params(rdpsndAlsaPlugin* alsa)
+{
+	int status = 0;
+	snd_pcm_sw_params_t* sw_params = NULL;
+	status = snd_pcm_sw_params_malloc(&sw_params);
+	SND_PCM_CHECK("snd_pcm_sw_params_malloc", status);
+	status = snd_pcm_sw_params_current(alsa->pcm_handle, sw_params);
+	SND_PCM_CHECK("snd_pcm_sw_params_current", status);
+	status = snd_pcm_sw_params_set_avail_min(alsa->pcm_handle, sw_params,
+	                                         (alsa->aformat.nChannels * alsa->actual_channels));
+	SND_PCM_CHECK("snd_pcm_sw_params_set_avail_min", status);
+	status = snd_pcm_sw_params_set_start_threshold(alsa->pcm_handle, sw_params,
+	                                               alsa->aformat.nBlockAlign);
+	SND_PCM_CHECK("snd_pcm_sw_params_set_start_threshold", status);
+	status = snd_pcm_sw_params(alsa->pcm_handle, sw_params);
+	SND_PCM_CHECK("snd_pcm_sw_params", status);
+	snd_pcm_sw_params_free(sw_params);
+	status = snd_pcm_prepare(alsa->pcm_handle);
+	SND_PCM_CHECK("snd_pcm_prepare", status);
+	return 0;
+}
+
+static int rdpsnd_alsa_validate_params(rdpsndAlsaPlugin* alsa)
+{
+	int status = 0;
+	snd_pcm_uframes_t buffer_size = 0;
+	snd_pcm_uframes_t period_size = 0;
+	status = snd_pcm_get_params(alsa->pcm_handle, &buffer_size, &period_size);
+	SND_PCM_CHECK("snd_pcm_get_params", status);
+	return 0;
+}
+
+static int rdpsnd_alsa_set_params(rdpsndAlsaPlugin* alsa)
+{
+	snd_pcm_drop(alsa->pcm_handle);
+
+	if (rdpsnd_alsa_set_hw_params(alsa) < 0)
+		return -1;
+
+	if (rdpsnd_alsa_set_sw_params(alsa) < 0)
+		return -1;
+
+	return rdpsnd_alsa_validate_params(alsa);
+}
+
+static BOOL rdpsnd_alsa_set_format(rdpsndDevicePlugin* device, const AUDIO_FORMAT* format,
+                                   UINT32 latency)
+{
+	rdpsndAlsaPlugin* alsa = (rdpsndAlsaPlugin*)device;
+
+	if (format)
+	{
+		alsa->aformat = *format;
+		alsa->actual_rate = format->nSamplesPerSec;
+		alsa->actual_channels = format->nChannels;
+
+		switch (format->wFormatTag)
+		{
+			case WAVE_FORMAT_PCM:
+				switch (format->wBitsPerSample)
+				{
+					case 8:
+						alsa->format = SND_PCM_FORMAT_S8;
+						break;
+
+					case 16:
+						alsa->format = SND_PCM_FORMAT_S16_LE;
+						break;
+
+					default:
+						return FALSE;
+				}
+
+				break;
+
+			default:
+				return FALSE;
+		}
+	}
+
+	alsa->latency = latency;
+	return (rdpsnd_alsa_set_params(alsa) == 0);
+}
+
+static void rdpsnd_alsa_close_mixer(rdpsndAlsaPlugin* alsa)
+{
+	if (alsa && alsa->mixer_handle)
+	{
+		snd_mixer_close(alsa->mixer_handle);
+		alsa->mixer_handle = NULL;
+	}
+}
+
+static BOOL rdpsnd_alsa_open_mixer(rdpsndAlsaPlugin* alsa)
+{
+	int status = 0;
+
+	if (alsa->mixer_handle)
+		return TRUE;
+
+	status = snd_mixer_open(&alsa->mixer_handle, 0);
+
+	if (status < 0)
+	{
+		WLog_ERR(TAG, "snd_mixer_open failed");
+		goto fail;
+	}
+
+	status = snd_mixer_attach(alsa->mixer_handle, alsa->device_name);
+
+	if (status < 0)
+	{
+		WLog_ERR(TAG, "snd_mixer_attach failed");
+		goto fail;
+	}
+
+	status = snd_mixer_selem_register(alsa->mixer_handle, NULL, NULL);
+
+	if (status < 0)
+	{
+		WLog_ERR(TAG, "snd_mixer_selem_register failed");
+		goto fail;
+	}
+
+	status = snd_mixer_load(alsa->mixer_handle);
+
+	if (status < 0)
+	{
+		WLog_ERR(TAG, "snd_mixer_load failed");
+		goto fail;
+	}
+
+	return TRUE;
+fail:
+	rdpsnd_alsa_close_mixer(alsa);
+	return FALSE;
+}
+
+static void rdpsnd_alsa_pcm_close(rdpsndAlsaPlugin* alsa)
+{
+	if (alsa && alsa->pcm_handle)
+	{
+		snd_pcm_drain(alsa->pcm_handle);
+		snd_pcm_close(alsa->pcm_handle);
+		alsa->pcm_handle = 0;
+	}
+}
+
+static BOOL rdpsnd_alsa_open(rdpsndDevicePlugin* device, const AUDIO_FORMAT* format, UINT32 latency)
+{
+	int mode = 0;
+	int status = 0;
+	rdpsndAlsaPlugin* alsa = (rdpsndAlsaPlugin*)device;
+
+	if (alsa->pcm_handle)
+		return TRUE;
+
+	mode = 0;
+	/*mode |= SND_PCM_NONBLOCK;*/
+	status = snd_pcm_open(&alsa->pcm_handle, alsa->device_name, SND_PCM_STREAM_PLAYBACK, mode);
+
+	if (status < 0)
+	{
+		WLog_ERR(TAG, "snd_pcm_open failed");
+		return FALSE;
+	}
+
+	return rdpsnd_alsa_set_format(device, format, latency) && rdpsnd_alsa_open_mixer(alsa);
+}
+
+static void rdpsnd_alsa_close(rdpsndDevicePlugin* device)
+{
+	rdpsndAlsaPlugin* alsa = (rdpsndAlsaPlugin*)device;
+
+	if (!alsa)
+		return;
+
+	rdpsnd_alsa_close_mixer(alsa);
+}
+
+static void rdpsnd_alsa_free(rdpsndDevicePlugin* device)
+{
+	rdpsndAlsaPlugin* alsa = (rdpsndAlsaPlugin*)device;
+	rdpsnd_alsa_pcm_close(alsa);
+	rdpsnd_alsa_close_mixer(alsa);
+	free(alsa->device_name);
+	free(alsa);
+}
+
+static BOOL rdpsnd_alsa_format_supported(rdpsndDevicePlugin* device, const AUDIO_FORMAT* format)
+{
+	switch (format->wFormatTag)
+	{
+		case WAVE_FORMAT_PCM:
+			if (format->cbSize == 0 && format->nSamplesPerSec <= 48000 &&
+			    (format->wBitsPerSample == 8 || format->wBitsPerSample == 16) &&
+			    (format->nChannels == 1 || format->nChannels == 2))
+			{
+				return TRUE;
+			}
+
+			break;
+	}
+
+	return FALSE;
+}
+
+static UINT32 rdpsnd_alsa_get_volume(rdpsndDevicePlugin* device)
+{
+	long volume_min = 0;
+	long volume_max = 0;
+	long volume_left = 0;
+	long volume_right = 0;
+	UINT32 dwVolume = 0;
+	UINT16 dwVolumeLeft = 0;
+	UINT16 dwVolumeRight = 0;
+	rdpsndAlsaPlugin* alsa = (rdpsndAlsaPlugin*)device;
+	dwVolumeLeft = ((50 * 0xFFFF) / 100);  /* 50% */
+	dwVolumeRight = ((50 * 0xFFFF) / 100); /* 50% */
+
+	if (!rdpsnd_alsa_open_mixer(alsa))
+		return 0;
+
+	for (snd_mixer_elem_t* elem = snd_mixer_first_elem(alsa->mixer_handle); elem;
+	     elem = snd_mixer_elem_next(elem))
+	{
+		if (snd_mixer_selem_has_playback_volume(elem))
+		{
+			snd_mixer_selem_get_playback_volume_range(elem, &volume_min, &volume_max);
+			snd_mixer_selem_get_playback_volume(elem, SND_MIXER_SCHN_FRONT_LEFT, &volume_left);
+			snd_mixer_selem_get_playback_volume(elem, SND_MIXER_SCHN_FRONT_RIGHT, &volume_right);
+			dwVolumeLeft =
+			    (UINT16)(((volume_left * 0xFFFF) - volume_min) / (volume_max - volume_min));
+			dwVolumeRight =
+			    (UINT16)(((volume_right * 0xFFFF) - volume_min) / (volume_max - volume_min));
+			break;
+		}
+	}
+
+	dwVolume = (dwVolumeLeft << 16) | dwVolumeRight;
+	return dwVolume;
+}
+
+static BOOL rdpsnd_alsa_set_volume(rdpsndDevicePlugin* device, UINT32 value)
+{
+	long left = 0;
+	long right = 0;
+	long volume_min = 0;
+	long volume_max = 0;
+	long volume_left = 0;
+	long volume_right = 0;
+	rdpsndAlsaPlugin* alsa = (rdpsndAlsaPlugin*)device;
+
+	if (!rdpsnd_alsa_open_mixer(alsa))
+		return FALSE;
+
+	left = (value & 0xFFFF);
+	right = ((value >> 16) & 0xFFFF);
+
+	for (snd_mixer_elem_t* elem = snd_mixer_first_elem(alsa->mixer_handle); elem;
+	     elem = snd_mixer_elem_next(elem))
+	{
+		if (snd_mixer_selem_has_playback_volume(elem))
+		{
+			snd_mixer_selem_get_playback_volume_range(elem, &volume_min, &volume_max);
+			volume_left = volume_min + (left * (volume_max - volume_min)) / 0xFFFF;
+			volume_right = volume_min + (right * (volume_max - volume_min)) / 0xFFFF;
+
+			if ((snd_mixer_selem_set_playback_volume(elem, SND_MIXER_SCHN_FRONT_LEFT, volume_left) <
+			     0) ||
+			    (snd_mixer_selem_set_playback_volume(elem, SND_MIXER_SCHN_FRONT_RIGHT,
+			                                         volume_right) < 0))
+			{
+				WLog_ERR(TAG, "error setting the volume\n");
+				return FALSE;
+			}
+		}
+	}
+
+	return TRUE;
+}
+
+static UINT rdpsnd_alsa_play(rdpsndDevicePlugin* device, const BYTE* data, size_t size)
+{
+	UINT latency = 0;
+	size_t offset = 0;
+	int frame_size = 0;
+	rdpsndAlsaPlugin* alsa = (rdpsndAlsaPlugin*)device;
+	WINPR_ASSERT(alsa);
+	WINPR_ASSERT(data || (size == 0));
+	frame_size = alsa->actual_channels * alsa->aformat.wBitsPerSample / 8;
+	if (frame_size <= 0)
+		return 0;
+
+	while (offset < size)
+	{
+		snd_pcm_sframes_t status =
+		    snd_pcm_writei(alsa->pcm_handle, &data[offset], (size - offset) / frame_size);
+
+		if (status < 0)
+			status = snd_pcm_recover(alsa->pcm_handle, status, 0);
+
+		if (status < 0)
+		{
+			WLog_ERR(TAG, "status: %d\n", status);
+			rdpsnd_alsa_close(device);
+			rdpsnd_alsa_open(device, NULL, alsa->latency);
+			break;
+		}
+
+		offset += status * frame_size;
+	}
+
+	{
+		snd_pcm_sframes_t available = 0;
+		snd_pcm_sframes_t delay = 0;
+		int rc = snd_pcm_avail_delay(alsa->pcm_handle, &available, &delay);
+
+		if (rc != 0)
+			latency = 0;
+		else if (available == 0) /* Get [ms] from number of samples */
+			latency = delay * 1000 / alsa->actual_rate;
+		else
+			latency = 0;
+	}
+
+	return latency + alsa->latency;
+}
+
+/**
+ * Function description
+ *
+ * @return 0 on success, otherwise a Win32 error code
+ */
+static UINT rdpsnd_alsa_parse_addin_args(rdpsndDevicePlugin* device, const ADDIN_ARGV* args)
+{
+	int status = 0;
+	DWORD flags = 0;
+	const COMMAND_LINE_ARGUMENT_A* arg = NULL;
+	rdpsndAlsaPlugin* alsa = (rdpsndAlsaPlugin*)device;
+	COMMAND_LINE_ARGUMENT_A rdpsnd_alsa_args[] = { { "dev", COMMAND_LINE_VALUE_REQUIRED, "<device>",
+		                                             NULL, NULL, -1, NULL, "device" },
+		                                           { NULL, 0, NULL, NULL, NULL, -1, NULL, NULL } };
+	flags =
+	    COMMAND_LINE_SIGIL_NONE | COMMAND_LINE_SEPARATOR_COLON | COMMAND_LINE_IGN_UNKNOWN_KEYWORD;
+	status = CommandLineParseArgumentsA(args->argc, args->argv, rdpsnd_alsa_args, flags, alsa, NULL,
+	                                    NULL);
+
+	if (status < 0)
+	{
+		WLog_ERR(TAG, "CommandLineParseArgumentsA failed!");
+		return CHANNEL_RC_INITIALIZATION_ERROR;
+	}
+
+	arg = rdpsnd_alsa_args;
+
+	do
+	{
+		if (!(arg->Flags & COMMAND_LINE_VALUE_PRESENT))
+			continue;
+
+		CommandLineSwitchStart(arg) CommandLineSwitchCase(arg, "dev")
+		{
+			alsa->device_name = _strdup(arg->Value);
+
+			if (!alsa->device_name)
+				return CHANNEL_RC_NO_MEMORY;
+		}
+		CommandLineSwitchEnd(arg)
+	} while ((arg = CommandLineFindNextArgumentA(arg)) != NULL);
+
+	return CHANNEL_RC_OK;
+}
+
+/**
+ * Function description
+ *
+ * @return 0 on success, otherwise a Win32 error code
+ */
+FREERDP_ENTRY_POINT(UINT alsa_freerdp_rdpsnd_client_subsystem_entry(
+    PFREERDP_RDPSND_DEVICE_ENTRY_POINTS pEntryPoints))
+{
+	const ADDIN_ARGV* args = NULL;
+	rdpsndAlsaPlugin* alsa = NULL;
+	UINT error = 0;
+	alsa = (rdpsndAlsaPlugin*)calloc(1, sizeof(rdpsndAlsaPlugin));
+
+	if (!alsa)
+	{
+		WLog_ERR(TAG, "calloc failed!");
+		return CHANNEL_RC_NO_MEMORY;
+	}
+
+	alsa->device.Open = rdpsnd_alsa_open;
+	alsa->device.FormatSupported = rdpsnd_alsa_format_supported;
+	alsa->device.GetVolume = rdpsnd_alsa_get_volume;
+	alsa->device.SetVolume = rdpsnd_alsa_set_volume;
+	alsa->device.Play = rdpsnd_alsa_play;
+	alsa->device.Close = rdpsnd_alsa_close;
+	alsa->device.Free = rdpsnd_alsa_free;
+	args = pEntryPoints->args;
+
+	if (args->argc > 1)
+	{
+		if ((error = rdpsnd_alsa_parse_addin_args(&alsa->device, args)))
+		{
+			WLog_ERR(TAG, "rdpsnd_alsa_parse_addin_args failed with error %" PRIu32 "", error);
+			goto error_parse_args;
+		}
+	}
+
+	if (!alsa->device_name)
+	{
+		alsa->device_name = _strdup("default");
+
+		if (!alsa->device_name)
+		{
+			WLog_ERR(TAG, "_strdup failed!");
+			error = CHANNEL_RC_NO_MEMORY;
+			goto error_strdup;
+		}
+	}
+
+	alsa->pcm_handle = 0;
+	alsa->actual_rate = 22050;
+	alsa->format = SND_PCM_FORMAT_S16_LE;
+	alsa->actual_channels = 2;
+	pEntryPoints->pRegisterRdpsndDevice(pEntryPoints->rdpsnd, (rdpsndDevicePlugin*)alsa);
+	return CHANNEL_RC_OK;
+error_strdup:
+	free(alsa->device_name);
+error_parse_args:
+	free(alsa);
+	return error;
+}