703 lines
17 KiB
C
703 lines
17 KiB
C
/**
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* FreeRDP: A Remote Desktop Protocol Implementation
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* Audio Output Virtual Channel
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*
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* Copyright 2009-2011 Jay Sorg
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* Copyright 2010-2011 Vic Lee
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*
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* Licensed under the Apache License, Version 2.0 (the "License");
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* you may not use this file except in compliance with the License.
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* You may obtain a copy of the License at
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*
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* http://www.apache.org/licenses/LICENSE-2.0
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*
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* Unless required by applicable law or agreed to in writing, software
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* distributed under the License is distributed on an "AS IS" BASIS,
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* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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* See the License for the specific language governing permissions and
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* limitations under the License.
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <stdio.h>
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#include <stdlib.h>
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#include <string.h>
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#include <winpr/crt.h>
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#include <winpr/cmdline.h>
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#include <winpr/sysinfo.h>
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#include <winpr/collections.h>
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#include <alsa/asoundlib.h>
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#include <freerdp/types.h>
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#include <freerdp/codec/dsp.h>
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#include <freerdp/utils/svc_plugin.h>
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#include "rdpsnd_main.h"
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typedef struct rdpsnd_alsa_plugin rdpsndAlsaPlugin;
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struct rdpsnd_alsa_plugin
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{
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rdpsndDevicePlugin device;
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HANDLE thread;
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char* device_name;
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snd_pcm_t* pcm_handle;
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snd_mixer_t* mixer_handle;
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UINT32 source_rate;
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UINT32 actual_rate;
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snd_pcm_format_t format;
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UINT32 source_channels;
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UINT32 actual_channels;
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int bytes_per_channel;
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int wformat;
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int block_size;
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int latency;
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BYTE* audio_data;
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UINT32 audio_data_size;
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UINT32 audio_data_left;
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wMessageQueue* queue;
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snd_pcm_uframes_t period_size;
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snd_async_handler_t* pcm_callback;
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FREERDP_DSP_CONTEXT* dsp_context;
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};
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struct _RDPSND_WAVE_INFO
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{
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BYTE* data;
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int length;
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BYTE cBlockNo;
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UINT16 wTimeStamp;
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UINT16 wFormatNo;
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UINT32 wTimeA;
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UINT32 wTimeB;
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};
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typedef struct _RDPSND_WAVE_INFO RDPSND_WAVE_INFO;
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static void rdpsnd_alsa_set_params(rdpsndAlsaPlugin* alsa)
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{
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int status;
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snd_pcm_hw_params_t* hw_params;
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snd_pcm_sw_params_t* sw_params;
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snd_pcm_uframes_t start_threshold;
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snd_pcm_uframes_t buffer_size;
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snd_pcm_drop(alsa->pcm_handle);
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status = snd_pcm_hw_params_malloc(&hw_params);
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if (status < 0)
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{
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DEBUG_WARN("snd_pcm_hw_params_malloc failed");
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return;
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}
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snd_pcm_hw_params_any(alsa->pcm_handle, hw_params);
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snd_pcm_hw_params_set_access(alsa->pcm_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
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snd_pcm_hw_params_set_format(alsa->pcm_handle, hw_params, alsa->format);
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snd_pcm_hw_params_set_rate_near(alsa->pcm_handle, hw_params, &alsa->actual_rate, NULL);
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snd_pcm_hw_params_set_channels_near(alsa->pcm_handle, hw_params, &alsa->actual_channels);
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snd_pcm_hw_params_get_period_size(hw_params, &alsa->period_size, 0);
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alsa->audio_data_left = 0;
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if (alsa->latency < 0)
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buffer_size = alsa->actual_rate * 4 / 10; /* Default to 400ms buffer */
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else
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buffer_size = alsa->latency * alsa->actual_rate * 2 / 1000; /* Double of the latency */
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if (buffer_size < alsa->actual_rate / 2)
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buffer_size = alsa->actual_rate / 2; /* Minimum 0.5-second buffer */
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snd_pcm_hw_params_set_buffer_size_near(alsa->pcm_handle, hw_params, &buffer_size);
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//snd_pcm_hw_params_set_period_size_near(alsa->pcm_handle, hw_params, &alsa->period_size, NULL);
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snd_pcm_hw_params(alsa->pcm_handle, hw_params);
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snd_pcm_hw_params_free(hw_params);
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status = snd_pcm_sw_params_malloc(&sw_params);
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if (status < 0)
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{
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DEBUG_WARN("snd_pcm_sw_params_malloc failed");
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return;
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}
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snd_pcm_sw_params_current(alsa->pcm_handle, sw_params);
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if (alsa->latency == 0)
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start_threshold = 0;
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else
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start_threshold = buffer_size / 2;
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snd_pcm_sw_params_set_start_threshold(alsa->pcm_handle, sw_params, start_threshold);
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snd_pcm_sw_params(alsa->pcm_handle, sw_params);
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snd_pcm_sw_params_free(sw_params);
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snd_pcm_prepare(alsa->pcm_handle);
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DEBUG_SVC("hardware buffer %d frames, playback buffer %.2g seconds",
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(int) buffer_size, (double) buffer_size / 2.0 / (double) alsa->actual_rate);
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if ((alsa->actual_rate != alsa->source_rate) || (alsa->actual_channels != alsa->source_channels))
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{
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DEBUG_SVC("actual rate %d / channel %d is different from source rate %d / channel %d, resampling required.",
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alsa->actual_rate, alsa->actual_channels, alsa->source_rate, alsa->source_channels);
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}
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}
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static void rdpsnd_alsa_set_format(rdpsndDevicePlugin* device, rdpsndFormat* format, int latency)
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{
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rdpsndAlsaPlugin* alsa = (rdpsndAlsaPlugin*) device;
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if (format != NULL)
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{
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alsa->source_rate = format->nSamplesPerSec;
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alsa->actual_rate = format->nSamplesPerSec;
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alsa->source_channels = format->nChannels;
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alsa->actual_channels = format->nChannels;
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switch (format->wFormatTag)
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{
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case WAVE_FORMAT_PCM:
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switch (format->wBitsPerSample)
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{
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case 4:
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break;
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case 8:
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alsa->format = SND_PCM_FORMAT_S8;
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alsa->bytes_per_channel = 1;
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break;
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case 16:
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alsa->format = SND_PCM_FORMAT_S16_LE;
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alsa->bytes_per_channel = 2;
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break;
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}
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break;
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case WAVE_FORMAT_ADPCM:
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case WAVE_FORMAT_DVI_ADPCM:
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alsa->format = SND_PCM_FORMAT_S16_LE;
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alsa->bytes_per_channel = 2;
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break;
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}
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alsa->wformat = format->wFormatTag;
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alsa->block_size = format->nBlockAlign;
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}
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alsa->latency = latency;
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rdpsnd_alsa_set_params(alsa);
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}
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static void rdpsnd_alsa_open_mixer(rdpsndAlsaPlugin* alsa)
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{
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int status;
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snd_mixer_t* handle;
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status = snd_mixer_open(&handle, 0);
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if (status < 0)
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{
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DEBUG_WARN("snd_mixer_open failed");
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return;
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}
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status = snd_mixer_attach(handle, alsa->device_name);
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if (status < 0)
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{
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DEBUG_WARN("snd_mixer_attach failed");
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snd_mixer_close(handle);
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return;
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}
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status = snd_mixer_selem_register(handle, NULL, NULL);
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if (status < 0)
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{
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DEBUG_WARN("snd_mixer_selem_register failed");
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snd_mixer_close(handle);
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return;
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}
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status = snd_mixer_load(handle);
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if (status < 0)
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{
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DEBUG_WARN("snd_mixer_load failed");
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snd_mixer_close(handle);
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return;
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}
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alsa->mixer_handle = handle;
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}
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static void rdpsnd_alsa_open(rdpsndDevicePlugin* device, rdpsndFormat* format, int latency)
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{
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int mode;
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int status;
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rdpsndAlsaPlugin* alsa = (rdpsndAlsaPlugin*) device;
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if (alsa->pcm_handle)
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return;
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DEBUG_SVC("opening");
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mode = 0;
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//mode |= SND_PCM_NONBLOCK;
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status = snd_pcm_open(&alsa->pcm_handle, alsa->device_name, SND_PCM_STREAM_PLAYBACK, mode);
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if (status < 0)
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{
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DEBUG_WARN("snd_pcm_open failed");
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}
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else
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{
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freerdp_dsp_context_reset_adpcm(alsa->dsp_context);
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rdpsnd_alsa_set_format(device, format, latency);
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rdpsnd_alsa_open_mixer(alsa);
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}
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}
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static void rdpsnd_alsa_close(rdpsndDevicePlugin* device)
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{
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rdpsndAlsaPlugin* alsa = (rdpsndAlsaPlugin*)device;
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if (alsa->pcm_handle)
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{
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DEBUG_SVC("close");
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snd_pcm_drain(alsa->pcm_handle);
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snd_pcm_close(alsa->pcm_handle);
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alsa->pcm_handle = 0;
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}
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if (alsa->mixer_handle)
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{
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snd_mixer_close(alsa->mixer_handle);
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alsa->mixer_handle = NULL;
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}
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}
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static void rdpsnd_alsa_free(rdpsndDevicePlugin* device)
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{
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rdpsndAlsaPlugin* alsa = (rdpsndAlsaPlugin*) device;
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rdpsnd_alsa_close(device);
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free(alsa->device_name);
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if (alsa->audio_data)
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free(alsa->audio_data);
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freerdp_dsp_context_free(alsa->dsp_context);
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free(alsa);
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}
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static BOOL rdpsnd_alsa_format_supported(rdpsndDevicePlugin* device, rdpsndFormat* format)
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{
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switch (format->wFormatTag)
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{
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case WAVE_FORMAT_PCM:
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if (format->cbSize == 0 &&
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format->nSamplesPerSec <= 48000 &&
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(format->wBitsPerSample == 8 || format->wBitsPerSample == 16) &&
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(format->nChannels == 1 || format->nChannels == 2))
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{
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return TRUE;
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}
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break;
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case WAVE_FORMAT_ADPCM:
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case WAVE_FORMAT_DVI_ADPCM:
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if (format->nSamplesPerSec <= 48000 &&
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format->wBitsPerSample == 4 &&
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(format->nChannels == 1 || format->nChannels == 2))
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{
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return TRUE;
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}
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break;
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case WAVE_FORMAT_ALAW:
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break;
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case WAVE_FORMAT_MULAW:
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break;
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case WAVE_FORMAT_GSM610:
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break;
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}
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return FALSE;
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}
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static void rdpsnd_alsa_set_volume(rdpsndDevicePlugin* device, UINT32 value)
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{
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long left;
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long right;
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long volume_min;
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long volume_max;
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long volume_left;
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long volume_right;
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snd_mixer_elem_t* elem;
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rdpsndAlsaPlugin* alsa = (rdpsndAlsaPlugin*) device;
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if (!alsa->mixer_handle)
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return;
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left = (value & 0xFFFF);
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right = ((value >> 16) & 0xFFFF);
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for (elem = snd_mixer_first_elem(alsa->mixer_handle); elem; elem = snd_mixer_elem_next(elem))
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{
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if (snd_mixer_selem_has_playback_volume(elem))
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{
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snd_mixer_selem_get_playback_volume_range(elem, &volume_min, &volume_max);
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volume_left = volume_min + (left * (volume_max - volume_min)) / 0xFFFF;
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volume_right = volume_min + (right * (volume_max - volume_min)) / 0xFFFF;
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snd_mixer_selem_set_playback_volume(elem, SND_MIXER_SCHN_FRONT_LEFT, volume_left);
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snd_mixer_selem_set_playback_volume(elem, SND_MIXER_SCHN_FRONT_RIGHT, volume_right);
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}
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}
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}
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static void* rdpsnd_alsa_schedule_thread(void* arg)
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{
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BYTE* data;
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int length;
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int status;
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int offset;
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int frames;
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int frame_size;
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wMessage message;
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UINT32 wInitialTime;
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UINT32 wCurrentTime;
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UINT32 wSessionTime;
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UINT16 wFixedLatency;
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UINT16 wAverageLatency;
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UINT16 wAverageSleepLatency;
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RDPSND_WAVE_INFO* waveInfo;
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snd_pcm_sframes_t available_input;
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snd_pcm_sframes_t available_output;
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rdpsndAlsaPlugin* alsa = (rdpsndAlsaPlugin*) arg;
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rdpsndDevicePlugin* device = (rdpsndDevicePlugin*) arg;
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wInitialTime = GetTickCount();
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wFixedLatency = 250;
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wAverageLatency = wFixedLatency / 2;
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wAverageSleepLatency = wFixedLatency / 2;
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while (1)
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{
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if (!MessageQueue_Wait(alsa->queue))
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break;
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if (!MessageQueue_Peek(alsa->queue, &message, TRUE))
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break;
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if (message.id == WMQ_QUIT)
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break;
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frame_size = alsa->actual_channels * alsa->bytes_per_channel;
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if (message.id == 0)
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{
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data = (BYTE*) message.wParam;
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length = (int) (size_t) message.lParam;
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}
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else if (message.id == 1)
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{
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UINT16 wAudioLength;
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waveInfo = (RDPSND_WAVE_INFO*) message.wParam;
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data = waveInfo->data;
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length = waveInfo->length;
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frames = length / frame_size;
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wAudioLength = frames / (alsa->actual_rate / 1000);
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}
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offset = 0;
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available_output = snd_pcm_avail_update(alsa->pcm_handle);
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while (offset < length)
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{
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available_input = (length - offset) / frame_size;
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if (available_output < 1)
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{
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snd_pcm_wait(alsa->pcm_handle, -1);
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available_output = snd_pcm_avail_update(alsa->pcm_handle);
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}
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status = snd_pcm_writei(alsa->pcm_handle, &data[offset],
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(available_input < available_output) ? available_input : available_output);
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available_output = 0;
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if (status == -EPIPE)
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{
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snd_pcm_recover(alsa->pcm_handle, status, 0);
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status = 0;
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}
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else if (status == -EAGAIN)
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{
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status = 0;
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}
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else if (status < 0)
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{
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DEBUG_WARN("snd_pcm_writei status %d", status);
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snd_pcm_close(alsa->pcm_handle);
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alsa->pcm_handle = NULL;
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rdpsnd_alsa_open((rdpsndDevicePlugin*) alsa, NULL, alsa->latency);
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break;
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}
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available_output = snd_pcm_avail_update(alsa->pcm_handle);
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offset += status * frame_size;
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}
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free(data);
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if (message.id == 1)
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{
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UINT16 wLatency;
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UINT16 wTimeStamp;
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UINT16 wAudioLength;
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UINT16 wSleepLatency;
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waveInfo = (RDPSND_WAVE_INFO*) message.wParam;
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waveInfo->wTimeB = GetTickCount();
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wLatency = (UINT16) (waveInfo->wTimeB - waveInfo->wTimeA);
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wTimeStamp = waveInfo->wTimeStamp + wLatency;
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frames = length / frame_size;
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wAudioLength = frames / (alsa->actual_rate / 1000);
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wAverageLatency = (wAverageLatency + wLatency) / 2;
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if (wFixedLatency > wLatency)
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wSleepLatency = wFixedLatency - wLatency;
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else
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wSleepLatency = 0;
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wAverageSleepLatency = (wAverageSleepLatency + wSleepLatency) / 2;
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Sleep(wSleepLatency);
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wCurrentTime = GetTickCount();
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wSessionTime = wCurrentTime - wInitialTime;
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printf("[%06d.%03d] FixedLatency: %d ms AvLatency: %d ms CurrentLatency: %d ms "
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"SleepLatency: %d ms AvSleepLatency: %d ms Frames: %d Length: %d ms Channels: %d Rate: %d\n",
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wSessionTime / 1000, wSessionTime % 1000,
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wFixedLatency, wAverageLatency, wLatency,
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wSleepLatency, wAverageSleepLatency,
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frames, wAudioLength, alsa->actual_channels, alsa->actual_rate);
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wTimeStamp += wSleepLatency;
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device->WaveConfirm(device, wTimeStamp, waveInfo->cBlockNo);
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free(waveInfo);
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}
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}
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return NULL;
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}
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BYTE* rdpsnd_process_audio_sample(rdpsndDevicePlugin* device, BYTE* data, int* size)
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{
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int frames;
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BYTE* srcData;
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int srcFrameSize;
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int dstFrameSize;
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rdpsndAlsaPlugin* alsa = (rdpsndAlsaPlugin*) device;
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if (!alsa->pcm_handle)
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return NULL;
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if (alsa->wformat == WAVE_FORMAT_ADPCM)
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{
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alsa->dsp_context->decode_ms_adpcm(alsa->dsp_context,
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data, *size, alsa->source_channels, alsa->block_size);
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*size = alsa->dsp_context->adpcm_size;
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srcData = alsa->dsp_context->adpcm_buffer;
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}
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else if (alsa->wformat == WAVE_FORMAT_DVI_ADPCM)
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{
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alsa->dsp_context->decode_ima_adpcm(alsa->dsp_context,
|
|
data, *size, alsa->source_channels, alsa->block_size);
|
|
|
|
*size = alsa->dsp_context->adpcm_size;
|
|
srcData = alsa->dsp_context->adpcm_buffer;
|
|
}
|
|
else
|
|
{
|
|
srcData = data;
|
|
}
|
|
|
|
srcFrameSize = alsa->source_channels * alsa->bytes_per_channel;
|
|
dstFrameSize = alsa->actual_channels * alsa->bytes_per_channel;
|
|
|
|
if ((*size % srcFrameSize) != 0)
|
|
return NULL;
|
|
|
|
if (!((alsa->source_rate == alsa->actual_rate) && (alsa->source_channels == alsa->actual_channels)))
|
|
{
|
|
alsa->dsp_context->resample(alsa->dsp_context, srcData, alsa->bytes_per_channel,
|
|
alsa->source_channels, alsa->source_rate, *size / srcFrameSize,
|
|
alsa->actual_channels, alsa->actual_rate);
|
|
|
|
frames = alsa->dsp_context->resampled_frames;
|
|
|
|
DEBUG_SVC("resampled %d frames at %d to %d frames at %d",
|
|
length / srcFrameSize, alsa->source_rate, frames, alsa->actual_rate);
|
|
|
|
*size = frames * dstFrameSize;
|
|
srcData = alsa->dsp_context->resampled_buffer;
|
|
}
|
|
|
|
data = srcData;
|
|
|
|
return data;
|
|
}
|
|
|
|
static void rdpsnd_alsa_play(rdpsndDevicePlugin* device, BYTE* data, int size)
|
|
{
|
|
BYTE* sample;
|
|
rdpsndAlsaPlugin* alsa = (rdpsndAlsaPlugin*) device;
|
|
|
|
data = rdpsnd_process_audio_sample(device, data, &size);
|
|
|
|
sample = (BYTE*) malloc(size);
|
|
CopyMemory(sample, data, size);
|
|
|
|
MessageQueue_Post(alsa->queue, (void*) alsa, 0, (void*) sample, (void*) (size_t) size);
|
|
}
|
|
|
|
static void rdpsnd_alsa_wave_play(rdpsndDevicePlugin* device,
|
|
UINT16 wTimeStamp, UINT16 wFormatNo, BYTE cBlockNo, BYTE* data, int size)
|
|
{
|
|
RDPSND_WAVE_INFO* waveInfo;
|
|
rdpsndAlsaPlugin* alsa = (rdpsndAlsaPlugin*) device;
|
|
|
|
waveInfo = (RDPSND_WAVE_INFO*) malloc(sizeof(RDPSND_WAVE_INFO));
|
|
|
|
waveInfo->wTimeA = GetTickCount();
|
|
waveInfo->wTimeStamp = wTimeStamp;
|
|
waveInfo->wFormatNo = wFormatNo;
|
|
waveInfo->cBlockNo = cBlockNo;
|
|
|
|
data = rdpsnd_process_audio_sample(device, data, &size);
|
|
|
|
waveInfo->data = (BYTE*) malloc(size);
|
|
CopyMemory(waveInfo->data, data, size);
|
|
waveInfo->length = size;
|
|
|
|
MessageQueue_Post(alsa->queue, (void*) alsa, 1, (void*) waveInfo, (void*) (size_t) size);
|
|
}
|
|
|
|
static void rdpsnd_alsa_start(rdpsndDevicePlugin* device)
|
|
{
|
|
rdpsndAlsaPlugin* alsa = (rdpsndAlsaPlugin*) device;
|
|
|
|
if (!alsa->pcm_handle)
|
|
return;
|
|
|
|
snd_pcm_start(alsa->pcm_handle);
|
|
}
|
|
|
|
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 }
|
|
};
|
|
|
|
static void rdpsnd_alsa_parse_addin_args(rdpsndDevicePlugin* device, ADDIN_ARGV* args)
|
|
{
|
|
int status;
|
|
DWORD flags;
|
|
COMMAND_LINE_ARGUMENT_A* arg;
|
|
rdpsndAlsaPlugin* alsa = (rdpsndAlsaPlugin*) device;
|
|
|
|
flags = COMMAND_LINE_SIGIL_NONE | COMMAND_LINE_SEPARATOR_COLON;
|
|
|
|
status = CommandLineParseArgumentsA(args->argc, (const char**) args->argv, rdpsnd_alsa_args, flags, alsa, NULL, NULL);
|
|
|
|
arg = rdpsnd_alsa_args;
|
|
|
|
do
|
|
{
|
|
if (!(arg->Flags & COMMAND_LINE_VALUE_PRESENT))
|
|
continue;
|
|
|
|
CommandLineSwitchStart(arg)
|
|
|
|
CommandLineSwitchCase(arg, "dev")
|
|
{
|
|
alsa->device_name = _strdup(arg->Value);
|
|
}
|
|
|
|
CommandLineSwitchEnd(arg)
|
|
}
|
|
while ((arg = CommandLineFindNextArgumentA(arg)) != NULL);
|
|
}
|
|
|
|
#ifdef STATIC_CHANNELS
|
|
#define freerdp_rdpsnd_client_subsystem_entry alsa_freerdp_rdpsnd_client_subsystem_entry
|
|
#endif
|
|
|
|
int freerdp_rdpsnd_client_subsystem_entry(PFREERDP_RDPSND_DEVICE_ENTRY_POINTS pEntryPoints)
|
|
{
|
|
ADDIN_ARGV* args;
|
|
rdpsndAlsaPlugin* alsa;
|
|
|
|
alsa = (rdpsndAlsaPlugin*) malloc(sizeof(rdpsndAlsaPlugin));
|
|
ZeroMemory(alsa, sizeof(rdpsndAlsaPlugin));
|
|
|
|
alsa->device.Open = rdpsnd_alsa_open;
|
|
alsa->device.FormatSupported = rdpsnd_alsa_format_supported;
|
|
alsa->device.SetFormat = rdpsnd_alsa_set_format;
|
|
alsa->device.SetVolume = rdpsnd_alsa_set_volume;
|
|
alsa->device.Play = rdpsnd_alsa_play;
|
|
alsa->device.WavePlay = rdpsnd_alsa_wave_play;
|
|
alsa->device.Start = rdpsnd_alsa_start;
|
|
alsa->device.Close = rdpsnd_alsa_close;
|
|
alsa->device.Free = rdpsnd_alsa_free;
|
|
|
|
args = pEntryPoints->args;
|
|
rdpsnd_alsa_parse_addin_args((rdpsndDevicePlugin*) alsa, args);
|
|
|
|
if (!alsa->device_name)
|
|
alsa->device_name = _strdup("default");
|
|
|
|
alsa->pcm_handle = 0;
|
|
alsa->source_rate = 22050;
|
|
alsa->actual_rate = 22050;
|
|
alsa->format = SND_PCM_FORMAT_S16_LE;
|
|
alsa->source_channels = 2;
|
|
alsa->actual_channels = 2;
|
|
alsa->bytes_per_channel = 2;
|
|
|
|
alsa->dsp_context = freerdp_dsp_context_new();
|
|
|
|
pEntryPoints->pRegisterRdpsndDevice(pEntryPoints->rdpsnd, (rdpsndDevicePlugin*) alsa);
|
|
|
|
alsa->queue = MessageQueue_New();
|
|
alsa->thread = CreateThread(NULL, 0, (LPTHREAD_START_ROUTINE) rdpsnd_alsa_schedule_thread, alsa, 0, NULL);
|
|
|
|
return 0;
|
|
}
|