254 lines
7.2 KiB
C
254 lines
7.2 KiB
C
/**
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* FreeRDP: A Remote Desktop Protocol Implementation
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* Video Redirection Virtual Channel - ALSA Audio Device
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*
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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 <unistd.h>
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#include <winpr/crt.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 "tsmf_audio.h"
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typedef struct _TSMFALSAAudioDevice
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{
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ITSMFAudioDevice iface;
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char device[32];
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snd_pcm_t *out_handle;
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UINT32 source_rate;
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UINT32 actual_rate;
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UINT32 source_channels;
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UINT32 actual_channels;
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UINT32 bytes_per_sample;
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FREERDP_DSP_CONTEXT *dsp_context;
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} TSMFAlsaAudioDevice;
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static BOOL tsmf_alsa_open_device(TSMFAlsaAudioDevice *alsa)
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{
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int error;
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error = snd_pcm_open(&alsa->out_handle, alsa->device, SND_PCM_STREAM_PLAYBACK, 0);
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if(error < 0)
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{
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WLog_ERR(TAG, "failed to open device %s", alsa->device);
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return FALSE;
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}
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DEBUG_TSMF("open device %s", alsa->device);
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return TRUE;
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}
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static BOOL tsmf_alsa_open(ITSMFAudioDevice *audio, const char *device)
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{
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TSMFAlsaAudioDevice *alsa = (TSMFAlsaAudioDevice *) audio;
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if(!device)
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{
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strncpy(alsa->device, "default", sizeof(alsa->device));
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}
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else
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{
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strncpy(alsa->device, device, sizeof(alsa->device));
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}
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return tsmf_alsa_open_device(alsa);
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}
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static BOOL tsmf_alsa_set_format(ITSMFAudioDevice *audio,
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UINT32 sample_rate, UINT32 channels, UINT32 bits_per_sample)
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{
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int error;
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snd_pcm_uframes_t frames;
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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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TSMFAlsaAudioDevice *alsa = (TSMFAlsaAudioDevice *) audio;
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if(!alsa->out_handle)
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return FALSE;
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snd_pcm_drop(alsa->out_handle);
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alsa->actual_rate = alsa->source_rate = sample_rate;
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alsa->actual_channels = alsa->source_channels = channels;
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alsa->bytes_per_sample = bits_per_sample / 8;
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error = snd_pcm_hw_params_malloc(&hw_params);
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if(error < 0)
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{
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WLog_ERR(TAG, "snd_pcm_hw_params_malloc failed");
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return FALSE;
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}
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snd_pcm_hw_params_any(alsa->out_handle, hw_params);
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snd_pcm_hw_params_set_access(alsa->out_handle, hw_params,
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SND_PCM_ACCESS_RW_INTERLEAVED);
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snd_pcm_hw_params_set_format(alsa->out_handle, hw_params,
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SND_PCM_FORMAT_S16_LE);
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snd_pcm_hw_params_set_rate_near(alsa->out_handle, hw_params,
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&alsa->actual_rate, NULL);
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snd_pcm_hw_params_set_channels_near(alsa->out_handle, hw_params,
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&alsa->actual_channels);
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frames = sample_rate;
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snd_pcm_hw_params_set_buffer_size_near(alsa->out_handle, hw_params,
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&frames);
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snd_pcm_hw_params(alsa->out_handle, hw_params);
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snd_pcm_hw_params_free(hw_params);
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error = snd_pcm_sw_params_malloc(&sw_params);
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if(error < 0)
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{
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WLog_ERR(TAG, "snd_pcm_sw_params_malloc");
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return FALSE;
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}
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snd_pcm_sw_params_current(alsa->out_handle, sw_params);
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snd_pcm_sw_params_set_start_threshold(alsa->out_handle, sw_params,
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frames / 2);
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snd_pcm_sw_params(alsa->out_handle, sw_params);
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snd_pcm_sw_params_free(sw_params);
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snd_pcm_prepare(alsa->out_handle);
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DEBUG_TSMF("sample_rate %d channels %d bits_per_sample %d",
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sample_rate, channels, bits_per_sample);
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DEBUG_TSMF("hardware buffer %d frames", (int)frames);
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if((alsa->actual_rate != alsa->source_rate) ||
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(alsa->actual_channels != alsa->source_channels))
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{
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DEBUG_TSMF("actual rate %d / channel %d is different "
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"from source rate %d / channel %d, resampling required.",
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alsa->actual_rate, alsa->actual_channels,
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alsa->source_rate, alsa->source_channels);
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}
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return TRUE;
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}
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static BOOL tsmf_alsa_play(ITSMFAudioDevice *audio, BYTE *data, UINT32 data_size)
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{
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int len;
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int error;
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int frames;
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BYTE *end;
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BYTE *src;
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BYTE *pindex;
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int rbytes_per_frame;
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int sbytes_per_frame;
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TSMFAlsaAudioDevice *alsa = (TSMFAlsaAudioDevice *) audio;
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DEBUG_TSMF("data_size %d", data_size);
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if(alsa->out_handle)
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{
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sbytes_per_frame = alsa->source_channels * alsa->bytes_per_sample;
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rbytes_per_frame = alsa->actual_channels * alsa->bytes_per_sample;
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if((alsa->source_rate == alsa->actual_rate) &&
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(alsa->source_channels == alsa->actual_channels))
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{
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src = data;
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}
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else
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{
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alsa->dsp_context->resample(alsa->dsp_context, data, alsa->bytes_per_sample,
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alsa->source_channels, alsa->source_rate, data_size / sbytes_per_frame,
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alsa->actual_channels, alsa->actual_rate);
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frames = alsa->dsp_context->resampled_frames;
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DEBUG_TSMF("resampled %d frames at %d to %d frames at %d",
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data_size / sbytes_per_frame, alsa->source_rate, frames, alsa->actual_rate);
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data_size = frames * rbytes_per_frame;
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src = alsa->dsp_context->resampled_buffer;
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}
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pindex = src;
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end = pindex + data_size;
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while(pindex < end)
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{
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len = end - pindex;
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frames = len / rbytes_per_frame;
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error = snd_pcm_writei(alsa->out_handle, pindex, frames);
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if(error == -EPIPE)
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{
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snd_pcm_recover(alsa->out_handle, error, 0);
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error = 0;
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}
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else if(error < 0)
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{
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DEBUG_TSMF("error len %d", error);
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snd_pcm_close(alsa->out_handle);
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alsa->out_handle = 0;
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tsmf_alsa_open_device(alsa);
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break;
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}
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DEBUG_TSMF("%d frames played.", error);
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if(error == 0)
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break;
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pindex += error * rbytes_per_frame;
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}
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}
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free(data);
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return TRUE;
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}
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static UINT64 tsmf_alsa_get_latency(ITSMFAudioDevice *audio)
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{
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UINT64 latency = 0;
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snd_pcm_sframes_t frames = 0;
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TSMFAlsaAudioDevice *alsa = (TSMFAlsaAudioDevice *) audio;
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if(alsa->out_handle && alsa->actual_rate > 0 &&
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snd_pcm_delay(alsa->out_handle, &frames) == 0 &&
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frames > 0)
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{
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latency = ((UINT64)frames) * 10000000LL / (UINT64) alsa->actual_rate;
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}
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return latency;
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}
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static BOOL tsmf_alsa_flush(ITSMFAudioDevice *audio)
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{
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return TRUE;
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}
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static void tsmf_alsa_free(ITSMFAudioDevice *audio)
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{
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TSMFAlsaAudioDevice *alsa = (TSMFAlsaAudioDevice *) audio;
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DEBUG_TSMF("");
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if(alsa->out_handle)
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{
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snd_pcm_drain(alsa->out_handle);
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snd_pcm_close(alsa->out_handle);
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}
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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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#ifdef STATIC_CHANNELS
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#define freerdp_tsmf_client_audio_subsystem_entry alsa_freerdp_tsmf_client_audio_subsystem_entry
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#else
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#define freerdp_tsmf_client_audio_subsystem_entry FREERDP_API freerdp_tsmf_client_audio_subsystem_entry
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#endif
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ITSMFAudioDevice *freerdp_tsmf_client_audio_subsystem_entry(void)
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{
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TSMFAlsaAudioDevice *alsa;
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alsa = (TSMFAlsaAudioDevice *) malloc(sizeof(TSMFAlsaAudioDevice));
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ZeroMemory(alsa, sizeof(TSMFAlsaAudioDevice));
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alsa->iface.Open = tsmf_alsa_open;
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alsa->iface.SetFormat = tsmf_alsa_set_format;
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alsa->iface.Play = tsmf_alsa_play;
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alsa->iface.GetLatency = tsmf_alsa_get_latency;
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alsa->iface.Flush = tsmf_alsa_flush;
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alsa->iface.Free = tsmf_alsa_free;
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alsa->dsp_context = freerdp_dsp_context_new();
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return (ITSMFAudioDevice *) alsa;
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}
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