/**

 * FreeRDP: A Remote Desktop Protocol Implementation

 * Digital Sound Processing - FFMPEG backend

 *

 * Copyright 2018 Armin Novak <armin.novak@thincast.com>

 * Copyright 2018 Thincast Technologies GmbH

 *

 * 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 <freerdp/log.h>

#include <libavcodec/avcodec.h>

#include <libavutil/avutil.h>

#include <libavutil/opt.h>

#if defined(SWRESAMPLE_FOUND)

#include <libswresample/swresample.h>

#elif defined(AVRESAMPLE_FOUND)

#include <libavresample/avresample.h>

#else

#error "libswresample or libavresample required"

#endif

#include "dsp.h"

#include "dsp_ffmpeg.h"

#define TAG FREERDP_TAG("dsp.ffmpeg")

struct _FREERDP_DSP_CONTEXT

{

	AUDIO_FORMAT format;

	BOOL isOpen;

	BOOL encoder;

	UINT32 bufferedSamples;

	enum AVCodecID id;

	AVCodec* codec;

	AVCodecContext* context;

	AVFrame* frame;

	AVFrame* resampled;

	AVFrame* buffered;

	AVPacket* packet;

#if defined(SWRESAMPLE_FOUND)

	SwrContext* rcontext;

#else

	AVAudioResampleContext* rcontext;

#endif

};

static BOOL ffmpeg_codec_is_filtered(enum AVCodecID id, BOOL encoder)

{

	switch (id)

	{

#if !defined(WITH_DSP_EXPERIMENTAL)

		case AV_CODEC_ID_ADPCM_IMA_OKI:

		case AV_CODEC_ID_MP3:

		case AV_CODEC_ID_ADPCM_MS:

		case AV_CODEC_ID_G723_1:

			return TRUE;

#endif

		case AV_CODEC_ID_NONE:

			return TRUE;

		default:

			return FALSE;

	}

}

static enum AVCodecID ffmpeg_get_avcodec(const AUDIO_FORMAT* format)

{

	const char* id;

	if (!format)

		return AV_CODEC_ID_NONE;

	id = audio_format_get_tag_string(format->wFormatTag);

	switch (format->wFormatTag)

	{

		case WAVE_FORMAT_UNKNOWN:

			return AV_CODEC_ID_NONE;

		case WAVE_FORMAT_PCM:

			switch (format->wBitsPerSample)

			{

				case 16:

					return AV_CODEC_ID_PCM_U16LE;

				case 8:

					return AV_CODEC_ID_PCM_U8;

				default:

					return AV_CODEC_ID_NONE;

			}

		case WAVE_FORMAT_DVI_ADPCM:

			return AV_CODEC_ID_ADPCM_IMA_OKI;

		case WAVE_FORMAT_ADPCM:

			return AV_CODEC_ID_ADPCM_MS;

		case WAVE_FORMAT_ALAW:

			return AV_CODEC_ID_PCM_ALAW;

		case WAVE_FORMAT_MULAW:

			return AV_CODEC_ID_PCM_MULAW;

		case WAVE_FORMAT_GSM610:

			return AV_CODEC_ID_GSM_MS;

		case WAVE_FORMAT_MSG723:

			return AV_CODEC_ID_G723_1;

		case WAVE_FORMAT_AAC_MS:

			return AV_CODEC_ID_AAC;

		default:

			return AV_CODEC_ID_NONE;

	}

}

static int ffmpeg_sample_format(const AUDIO_FORMAT* format)

{

	switch (format->wFormatTag)

	{

		case WAVE_FORMAT_PCM:

			switch (format->wBitsPerSample)

			{

				case 8:

					return AV_SAMPLE_FMT_U8;

				case 16:

					return AV_SAMPLE_FMT_S16;

				default:

					return FALSE;

			}

		case WAVE_FORMAT_DVI_ADPCM:

		case WAVE_FORMAT_ADPCM:

			return AV_SAMPLE_FMT_S16P;

		case WAVE_FORMAT_MPEGLAYER3:

		case WAVE_FORMAT_AAC_MS:

			return AV_SAMPLE_FMT_FLTP;

		case WAVE_FORMAT_MSG723:

		case WAVE_FORMAT_GSM610:

			return AV_SAMPLE_FMT_S16P;

		case WAVE_FORMAT_ALAW:

			return AV_SAMPLE_FMT_S16;

		default:

			return FALSE;

	}

}

static void ffmpeg_close_context(FREERDP_DSP_CONTEXT* context)

{

	if (context)

	{

		if (context->context)

			avcodec_free_context(&context->context);

		if (context->frame)

			av_frame_free(&context->frame);

		if (context->resampled)

			av_frame_free(&context->resampled);

		if (context->buffered)

			av_frame_free(&context->buffered);

		if (context->packet)

			av_packet_free(&context->packet);

		if (context->rcontext)

		{

#if defined(SWRESAMPLE_FOUND)

			swr_free(&context->rcontext);

#else

			avresample_free(&context->rcontext);

#endif

		}

		context->id = AV_CODEC_ID_NONE;

		context->codec = NULL;

		context->isOpen = FALSE;

		context->context = NULL;

		context->frame = NULL;

		context->resampled = NULL;

		context->packet = NULL;

		context->rcontext = NULL;

	}

}

static BOOL ffmpeg_open_context(FREERDP_DSP_CONTEXT* context)

{

	int ret;

	int layout;

	const AUDIO_FORMAT* format;

	if (!context || context->isOpen)

		return FALSE;

	format = &context->format;

	if (!format)

		return FALSE;

	layout = av_get_default_channel_layout(format->nChannels);

	context->id = ffmpeg_get_avcodec(format);

	if (ffmpeg_codec_is_filtered(context->id, context->encoder))

		goto fail;

	if (context->encoder)

		context->codec = avcodec_find_encoder(context->id);

	else

		context->codec = avcodec_find_decoder(context->id);

	if (!context->codec)

		goto fail;

	context->context = avcodec_alloc_context3(context->codec);

	if (!context->context)

		goto fail;

	switch (context->id)

	{

		/* We need support for multichannel and sample rates != 8000 */

		case AV_CODEC_ID_GSM_MS:

			context->context->strict_std_compliance = FF_COMPLIANCE_UNOFFICIAL;

			break;

		case AV_CODEC_ID_AAC:

			context->context->profile = FF_PROFILE_AAC_MAIN;

			break;

		default:

			break;

	}

	context->context->channels = format->nChannels;

	context->context->channel_layout = layout;

	context->context->sample_rate = format->nSamplesPerSec;

	context->context->block_align = format->nBlockAlign;

	context->context->bit_rate = format->nAvgBytesPerSec * 8;

	context->context->sample_fmt = ffmpeg_sample_format(format);

	context->context->time_base = av_make_q(1, context->context->sample_rate);

	if ((ret = avcodec_open2(context->context, context->codec, NULL)) < 0)

	{

		const char* err = av_err2str(ret);

		WLog_ERR(TAG, "Error avcodec_open2 %s [%d]", err, ret);

		goto fail;

	}

	context->packet = av_packet_alloc();

	if (!context->packet)

		goto fail;

	context->frame = av_frame_alloc();

	if (!context->frame)

		goto fail;

	context->resampled = av_frame_alloc();

	if (!context->resampled)

		goto fail;

	context->buffered = av_frame_alloc();

	if (!context->buffered)

		goto fail;

#if defined(SWRESAMPLE_FOUND)

	context->rcontext = swr_alloc();

#else

	context->rcontext = avresample_alloc_context();

#endif

	if (!context->rcontext)

		goto fail;

	context->frame->channel_layout = layout;

	context->frame->channels = format->nChannels;

	context->frame->sample_rate = format->nSamplesPerSec;

	context->frame->format = AV_SAMPLE_FMT_S16;

	if (context->encoder)

	{

		context->resampled->format = context->context->sample_fmt;

		context->resampled->sample_rate = context->context->sample_rate;

	}

	else

	{

		context->resampled->format = AV_SAMPLE_FMT_S16;

		context->resampled->sample_rate = format->nSamplesPerSec;

	}

	context->resampled->channel_layout = layout;

	context->resampled->channels = format->nChannels;

	if (context->context->frame_size > 0)

	{

		context->buffered->channel_layout = context->resampled->channel_layout;

		context->buffered->channels = context->resampled->channels;

		context->buffered->format = context->resampled->format;

		context->buffered->nb_samples = context->context->frame_size;

		if ((ret = av_frame_get_buffer(context->buffered, 1)) < 0)

			goto fail;

	}

	context->isOpen = TRUE;

	return TRUE;

fail:

	ffmpeg_close_context(context);

	return FALSE;

}

#if defined(SWRESAMPLE_FOUND)

static BOOL ffmpeg_resample_frame(SwrContext* context,

                                  AVFrame* in, AVFrame* out)

{

	int ret;

	if (!swr_is_initialized(context))

	{

		if ((ret = swr_config_frame(context, out, in)) < 0)

		{

			const char* err = av_err2str(ret);

			WLog_ERR(TAG, "Error during resampling %s [%d]", err, ret);

			return FALSE;

		}

		if ((ret = (swr_init(context))) < 0)

		{

			const char* err = av_err2str(ret);

			WLog_ERR(TAG, "Error during resampling %s [%d]", err, ret);

			return FALSE;

		}

	}

	if ((ret = swr_convert_frame(context, out, in)) < 0)

	{

		const char* err = av_err2str(ret);

		WLog_ERR(TAG, "Error during resampling %s [%d]", err, ret);

		return FALSE;

	}

	return TRUE;

}

#else

static BOOL ffmpeg_resample_frame(AVAudioResampleContext* context,

                                  AVFrame* in, AVFrame* out)

{

	int ret;

	if (!avresample_is_open(context))

	{

		if ((ret = avresample_config(context, out, in)) < 0)

		{

			const char* err = av_err2str(ret);

			WLog_ERR(TAG, "Error during resampling %s [%d]", err, ret);

			return FALSE;

		}

		if ((ret = (avresample_open(context))) < 0)

		{

			const char* err = av_err2str(ret);

			WLog_ERR(TAG, "Error during resampling %s [%d]", err, ret);

			return FALSE;

		}

	}

	if ((ret = avresample_convert_frame(context, out, in)) < 0)

	{

		const char* err = av_err2str(ret);

		WLog_ERR(TAG, "Error during resampling %s [%d]", err, ret);

		return FALSE;

	}

	return TRUE;

}

#endif

static BOOL ffmpeg_encode_frame(AVCodecContext* context, AVFrame* in,

                                AVPacket* packet, wStream* out)

{

	int ret;

	/* send the packet with the compressed data to the encoder */

	ret = avcodec_send_frame(context, in);

	if (ret < 0)

	{

		const char* err = av_err2str(ret);

		WLog_ERR(TAG, "Error submitting the packet to the encoder %s [%d]",

		         err, ret);

		return FALSE;

	}

	/* read all the output frames (in general there may be any number of them */

	while (ret >= 0)

	{

		ret = avcodec_receive_packet(context, packet);

		if ((ret == AVERROR(EAGAIN)) || (ret == AVERROR_EOF))

			return TRUE;

		else if (ret < 0)

		{

			const char* err = av_err2str(ret);

			WLog_ERR(TAG, "Error during encoding %s [%d]", err, ret);

			return FALSE;

		}

		if (!Stream_EnsureRemainingCapacity(out, packet->size))

			return FALSE;

		Stream_Write(out, packet->data, packet->size);

		av_packet_unref(packet);

	}

	return TRUE;

}

static BOOL ffmpeg_fill_frame(AVFrame* frame, const AUDIO_FORMAT* inputFormat,

                              const BYTE* data, size_t size)

{

	int ret, bpp;

	frame->channels = inputFormat->nChannels;

	frame->sample_rate = inputFormat->nSamplesPerSec;

	frame->format = ffmpeg_sample_format(inputFormat);

	frame->channel_layout = av_get_default_channel_layout(frame->channels);

	bpp = av_get_bytes_per_sample(frame->format);

	frame->nb_samples = size / inputFormat->nChannels / bpp;

	if ((ret = avcodec_fill_audio_frame(frame, frame->channels,

	                                    frame->format,

	                                    data, size, 1)) < 0)

	{

		const char* err = av_err2str(ret);

		WLog_ERR(TAG, "Error during audio frame fill %s [%d]", err, ret);

		return FALSE;

	}

	return TRUE;

}

#if defined(SWRESAMPLE_FOUND)

static BOOL ffmpeg_decode(AVCodecContext* dec_ctx, AVPacket* pkt,

                          AVFrame* frame,

                          SwrContext* resampleContext,

                          AVFrame* resampled, wStream* out)

#else

static BOOL ffmpeg_decode(AVCodecContext* dec_ctx, AVPacket* pkt,

                          AVFrame* frame,

                          AVAudioResampleContext* resampleContext,

                          AVFrame* resampled, wStream* out)

#endif

{

	int ret;

	/* send the packet with the compressed data to the decoder */

	ret = avcodec_send_packet(dec_ctx, pkt);

	if (ret < 0)

	{

		const char* err = av_err2str(ret);

		WLog_ERR(TAG, "Error submitting the packet to the decoder %s [%d]",

		         err, ret);

		return FALSE;

	}

	/* read all the output frames (in general there may be any number of them */

	while (ret >= 0)

	{

		ret = avcodec_receive_frame(dec_ctx, frame);

		if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)

			return TRUE;

		else if (ret < 0)

		{

			const char* err = av_err2str(ret);

			WLog_ERR(TAG, "Error during decoding %s [%d]", err, ret);

			return FALSE;

		}

#if defined(SWRESAMPLE_FOUND)

		if (!swr_is_initialized(resampleContext))

		{

			if ((ret = swr_config_frame(resampleContext, resampled, frame)) < 0)

			{

#else

		if (!avresample_is_open(resampleContext))

		{

			if ((ret = avresample_config(resampleContext, resampled, frame)) < 0)

			{

#endif

				const char* err = av_err2str(ret);

				WLog_ERR(TAG, "Error during resampling %s [%d]", err, ret);

				return FALSE;

			}

#if defined(SWRESAMPLE_FOUND)

			if ((ret = (swr_init(resampleContext))) < 0)

#else

			if ((ret = (avresample_open(resampleContext))) < 0)

#endif

			{

				const char* err = av_err2str(ret);

				WLog_ERR(TAG, "Error during resampling %s [%d]", err, ret);

				return FALSE;

			}

		}

#if defined(SWRESAMPLE_FOUND)

		if ((ret = swr_convert_frame(resampleContext, resampled, frame)) < 0)

#else

		if ((ret = avresample_convert_frame(resampleContext, resampled, frame)) < 0)

#endif

		{

			const char* err = av_err2str(ret);

			WLog_ERR(TAG, "Error during resampling %s [%d]", err, ret);

			return FALSE;

		}

		{

			const size_t data_size = resampled->channels * resampled->nb_samples * 2;

			Stream_EnsureRemainingCapacity(out, data_size);

			Stream_Write(out, resampled->data[0], data_size);

		}

	}

	return TRUE;

}

BOOL freerdp_dsp_ffmpeg_supports_format(const AUDIO_FORMAT* format, BOOL encode)

{

	enum AVCodecID id = ffmpeg_get_avcodec(format);

	if (ffmpeg_codec_is_filtered(id, encode))

		return FALSE;

	if (encode)

		return avcodec_find_encoder(id) != NULL;

	else

		return avcodec_find_decoder(id) != NULL;

}

FREERDP_DSP_CONTEXT* freerdp_dsp_ffmpeg_context_new(BOOL encode)

{

	FREERDP_DSP_CONTEXT* context;

	avcodec_register_all();

	context = calloc(1, sizeof(FREERDP_DSP_CONTEXT));

	if (!context)

		return NULL;

	context->encoder = encode;

	return context;

}

void freerdp_dsp_ffmpeg_context_free(FREERDP_DSP_CONTEXT* context)

{

	if (context)

	{

		ffmpeg_close_context(context);

		free(context);

	}

}

BOOL freerdp_dsp_ffmpeg_context_reset(FREERDP_DSP_CONTEXT* context,

                                      const AUDIO_FORMAT* targetFormat)

{

	if (!context || !targetFormat)

		return FALSE;

	ffmpeg_close_context(context);

	context->format = *targetFormat;

	return ffmpeg_open_context(context);

}

BOOL freerdp_dsp_ffmpeg_encode(FREERDP_DSP_CONTEXT* context, const AUDIO_FORMAT* format,

                               const BYTE* data, size_t length, wStream* out)

{

	int rc;

	if (!context || !format || !data || !out || !context->encoder)

		return FALSE;

	if (!context || !data || !out)

		return FALSE;

	/* Create input frame */

	if (!ffmpeg_fill_frame(context->frame, format, data, length))

		return FALSE;

	/* Resample to desired format. */

	if (!ffmpeg_resample_frame(context->rcontext,

	                           context->frame,

	                           context->resampled))

		return FALSE;

	if (context->context->frame_size <= 0)

	{

		return ffmpeg_encode_frame(context->context, context->resampled,

		                           context->packet, out);

	}

	else

	{

		int copied = 0;

		int rest = context->resampled->nb_samples;

		do

		{

			int inSamples = rest;

			if (inSamples + context->bufferedSamples > context->context->frame_size)

				inSamples = context->context->frame_size - context->bufferedSamples;

			rc = av_samples_copy(context->buffered->extended_data, context->resampled->extended_data,

			                     context->bufferedSamples, copied, inSamples,

			                     context->context->channels, context->context->sample_fmt);

			rest -= inSamples;

			copied += inSamples;

			context->bufferedSamples += inSamples;

			if (context->context->frame_size <= context->bufferedSamples)

			{

				/* Encode in desired format. */

				if (!ffmpeg_encode_frame(context->context, context->buffered,

				                         context->packet, out))

					return FALSE;

				context->bufferedSamples = 0;

			}

		}

		while (rest > 0);

		return TRUE;

	}

}

BOOL freerdp_dsp_ffmpeg_decode(FREERDP_DSP_CONTEXT* context, const AUDIO_FORMAT* srcFormat,

                               const BYTE* data, size_t length, wStream* out)

{

	if (!context || !srcFormat || !data || !out || context->encoder)

		return FALSE;

	av_init_packet(context->packet);

	context->packet->data = (uint8_t*)data;

	context->packet->size = length;

	return ffmpeg_decode(context->context, context->packet, context->frame,

	                     context->rcontext, context->resampled, out);

}