FreeRDP/channels/tsmf/client/alsa/tsmf_alsa.c

/**
 * 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.
 */

#ifdef HAVE_CONFIG_H
#include "config.h"
#endif

#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <pthread.h>
#include <unistd.h>
#include <alsa/asoundlib.h>
#include <freerdp/types.h>
#include <freerdp/utils/memory.h>
#include <freerdp/utils/dsp.h>

#include "tsmf_audio.h"

typedef struct _TSMFALSAAudioDevice
{
	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;

	FREERDP_DSP_CONTEXT* dsp_context;
} TSMFALSAAudioDevice;

static boolean tsmf_alsa_open_device(TSMFALSAAudioDevice* alsa)
{
	int error;

	error = snd_pcm_open(&alsa->out_handle, alsa->device, SND_PCM_STREAM_PLAYBACK, 0);
	if (error < 0)
	{
		DEBUG_WARN("failed to open device %s", alsa->device);
		return false;
	}

	DEBUG_DVC("open device %s", alsa->device);
	return true;
}

static boolean tsmf_alsa_open(ITSMFAudioDevice* audio, const char* device)
{
	TSMFALSAAudioDevice* alsa = (TSMFALSAAudioDevice*) audio;

	if (!device)
	{
		if (!alsa->device[0])
			strcpy(alsa->device, "default");
	}
	else
	{
		strcpy(alsa->device, device);
	}

	return tsmf_alsa_open_device(alsa);
}

static boolean tsmf_alsa_set_format(ITSMFAudioDevice* audio,
	uint32 sample_rate, uint32 channels, uint32 bits_per_sample)
{
	int error;
	snd_pcm_uframes_t frames;
	snd_pcm_hw_params_t* hw_params;
	snd_pcm_sw_params_t* sw_params;
	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)
	{
		DEBUG_WARN("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)
	{
		DEBUG_WARN("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_DVC("sample_rate %d channels %d bits_per_sample %d",
		sample_rate, channels, bits_per_sample);
	DEBUG_DVC("hardware buffer %d frames", (int)frames);
	if ((alsa->actual_rate != alsa->source_rate) ||
		(alsa->actual_channels != alsa->source_channels))
	{
		DEBUG_DVC("actual rate %d / channel %d is different "
			"from source rate %d / channel %d, resampling required.",
			alsa->actual_rate, alsa->actual_channels,
			alsa->source_rate, alsa->source_channels);
	}
	return true;
}

static boolean tsmf_alsa_play(ITSMFAudioDevice* audio, uint8* data, uint32 data_size)
{
	int len;
	int error;
	int frames;
	uint8* end;
	uint8* src;
	uint8* pindex;
	int rbytes_per_frame;
	int sbytes_per_frame;
	TSMFALSAAudioDevice* alsa = (TSMFALSAAudioDevice*) audio;

	DEBUG_DVC("data_size %d", 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;

		if ((alsa->source_rate == alsa->actual_rate) &&
			(alsa->source_channels == alsa->actual_channels))
		{
			src = data;
		}
		else
		{
			alsa->dsp_context->resample(alsa->dsp_context, data, alsa->bytes_per_sample,
				alsa->source_channels, alsa->source_rate, data_size / sbytes_per_frame,
				alsa->actual_channels, alsa->actual_rate);
			frames = alsa->dsp_context->resampled_frames;
			DEBUG_DVC("resampled %d frames at %d to %d frames at %d",
				data_size / sbytes_per_frame, alsa->source_rate, frames, alsa->actual_rate);
			data_size = frames * rbytes_per_frame;
			src = alsa->dsp_context->resampled_buffer;
		}

		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_DVC("error len %d", error);
				snd_pcm_close(alsa->out_handle);
				alsa->out_handle = 0;
				tsmf_alsa_open_device(alsa);
				break;
			}
			DEBUG_DVC("%d frames played.", error);
			if (error == 0)
				break;
			pindex += error * rbytes_per_frame;
		}
	}
	xfree(data);

	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 void tsmf_alsa_flush(ITSMFAudioDevice* audio)
{
}

static void tsmf_alsa_free(ITSMFAudioDevice* audio)
{
	TSMFALSAAudioDevice* alsa = (TSMFALSAAudioDevice*) audio;

	DEBUG_DVC("");

	if (alsa->out_handle)
	{
		snd_pcm_drain(alsa->out_handle);
		snd_pcm_close(alsa->out_handle);
	}
	freerdp_dsp_context_free(alsa->dsp_context);
	xfree(alsa);
}

ITSMFAudioDevice* TSMFAudioDeviceEntry(void)
{
	TSMFALSAAudioDevice* alsa;

	alsa = xnew(TSMFALSAAudioDevice);

	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;

	alsa->dsp_context = freerdp_dsp_context_new();

	return (ITSMFAudioDevice*) alsa;
}