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-rw-r--r--channels/rdpsnd/client/alsa/rdpsnd_alsa.c574
1 files changed, 574 insertions, 0 deletions
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;
+}