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, 275 insertions, 0 deletions

diff --git a/channels/tsmf/client/alsa/CMakeLists.txt b/channels/tsmf/client/alsa/CMakeLists.txt
new file mode 100644
index 0000000..9910542
--- /dev/null
+++ b/channels/tsmf/client/alsa/CMakeLists.txt
@@ -0,0 +1,35 @@
+# 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)
+
+set(${MODULE_PREFIX}_SRCS
+	tsmf_alsa.c
+)
+
+set(${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/tsmf/client/alsa/tsmf_alsa.c b/channels/tsmf/client/alsa/tsmf_alsa.c
new file mode 100644
index 0000000..fcf30aa
--- /dev/null
+++ b/channels/tsmf/client/alsa/tsmf_alsa.c
@@ -0,0 +1,240 @@
+/**
+ * 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 = NULL;
+	snd_pcm_sw_params_t* sw_params = NULL;
+	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, NULL);
+	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)
+{
+	int len = 0;
+	int error = 0;
+	int frames = 0;
+	const BYTE* end = NULL;
+	const BYTE* pindex = NULL;
+	int rbytes_per_frame = 0;
+	int sbytes_per_frame = 0;
+	TSMFAlsaAudioDevice* alsa = (TSMFAlsaAudioDevice*)audio;
+	DEBUG_TSMF("data_size %" PRIu32 "", data_size);
+
+	if (alsa->out_handle)
+	{
+		sbytes_per_frame = alsa->source_channels * alsa->bytes_per_sample;
+		rbytes_per_frame = alsa->actual_channels * alsa->bytes_per_sample;
+		pindex = src;
+		end = pindex + data_size;
+
+		while (pindex < end)
+		{
+			len = end - pindex;
+			frames = len / rbytes_per_frame;
+			error = snd_pcm_writei(alsa->out_handle, pindex, frames);
+
+			if (error == -EPIPE)
+			{
+				snd_pcm_recover(alsa->out_handle, error, 0);
+				error = 0;
+			}
+			else if (error < 0)
+			{
+				DEBUG_TSMF("error len %d", 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 += 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(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(ITSMFAudioDevice* alsa_freerdp_tsmf_client_audio_subsystem_entry(void))
+{
+	TSMFAlsaAudioDevice* alsa = calloc(1, sizeof(TSMFAlsaAudioDevice));
+	if (!alsa)
+		return NULL;
+
+	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;
+	return &alsa->iface;
+}