/**
 * 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;
}
