libstb-hal/libeplayer3/writer/pcm.cpp

/*
 * linuxdvb output/writer handling.
 *
 * Copyright (C) 2010  konfetti (based on code from libeplayer2)
 * Copyright (C) 2014  martii   (based on code from libeplayer3)
 *
 * This program is free software; you can redistribute it and/or modify
 * it under the terms of the GNU General Public License as published by
 * the Free Software Foundation; either version 2 of the License, or
 * (at your option) any later version.
 *
 * This program is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 * GNU General Public License for more details.
 *
 * You should have received a copy of the GNU General Public License
 * along with this program; if not, write to the Free Software
 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
 *
 */

#include <stdio.h>
#include <stdlib.h>
#include <stdint.h>
#include <string.h>
#include <errno.h>
#include <sys/ioctl.h>
#include <sys/uio.h>
#include <linux/dvb/audio.h>

#include "misc.h"
#include "pes.h"
#include "writer.h"
#include "player.h"

extern "C" {
#include <libavutil/avutil.h>
#include <libavutil/time.h>
#include <libavformat/avformat.h>
#include <libswresample/swresample.h>
#include <libavutil/opt.h>
}

// reference: search for TypeLpcmDVDAudio in player/frame_parser/frame_parser_audio_lpcm.cpp
static const uint8_t clpcm_prv[14] = {
	0xA0,	//sub_stream_id
	0, 0,	//resvd and UPC_EAN_ISRC stuff, unused
	0x0A,	//private header length
	0, 9,	//first_access_unit_pointer
	0x00,	//emph,rsvd,stereo,downmix
	0x0F,	//quantisation word length 1,2
	0x0F,	//audio sampling freqency 1,2
	0,	//resvd, multi channel type
	0,	//bit shift on channel GR2, assignment
	0x80,	//dynamic range control
	0, 0	//resvd for copyright management
};

class WriterPCM : public Writer
{
	private:
		unsigned int SubFrameLen;
		unsigned int SubFramesPerPES;
		uint8_t lpcm_prv[14];
		uint8_t injectBuffer[2048];
		uint8_t breakBuffer[sizeof(injectBuffer)];
		uint8_t *output;
		uint8_t out_samples_max;
		unsigned int breakBufferFillSize;
		int uNoOfChannels;
		int uSampleRate;
		int uBitsPerSample;

		AVStream *stream;
		SwrContext *swr;
		AVFrame *decoded_frame;
		int out_sample_rate;
		int out_channels;
		uint64_t out_channel_layout;
		bool initialHeader;
		bool restart_audio_resampling;

	public:
		bool Write(AVPacket *packet, int64_t pts);
		bool prepareClipPlay();
		bool writePCM(int64_t Pts, uint8_t *data, unsigned int size);
		void Init(int _fd, AVStream *_stream, Player *_player);
		WriterPCM();
};

bool WriterPCM::prepareClipPlay()
{
	SubFrameLen = 0;
	SubFramesPerPES = 0;
	breakBufferFillSize = 0;

	memcpy(lpcm_prv, clpcm_prv, sizeof(lpcm_prv));

	// figure out size of subframe and set up sample rate
	switch (uSampleRate) {
		case 48000:
			SubFrameLen = 40;
			break;
		case 96000:
			lpcm_prv[8] |= 0x10;
			SubFrameLen = 80;
			break;
		case 192000:
			lpcm_prv[8] |= 0x20;
			SubFrameLen = 160;
			break;
		case 44100:
			lpcm_prv[8] |= 0x80;
			SubFrameLen = 40;
			break;
		case 88200:
			lpcm_prv[8] |= 0x90;
			SubFrameLen = 80;
			break;
		case 176400:
			lpcm_prv[8] |= 0xA0;
			SubFrameLen = 160;
			break;
		default:
			break;
	}

	SubFrameLen *= uNoOfChannels;
	SubFrameLen *= uBitsPerSample / 8;

	//rewrite PES size to have as many complete subframes per PES as we can
	SubFramesPerPES = ((sizeof(injectBuffer) - 14) - sizeof(lpcm_prv)) / SubFrameLen;
	SubFrameLen *= SubFramesPerPES;

	//set number of channels
	lpcm_prv[10] = uNoOfChannels - 1;

	switch (uBitsPerSample) {
		case 24:
			lpcm_prv[7] |= 0x20;
		case 16:
			break;
		default:
			printf("inappropriate bits per sample (%d) - must be 16 or 24\n", uBitsPerSample);
			return false;
	}

	return true;
}

bool WriterPCM::writePCM(int64_t Pts, uint8_t *data, unsigned int size)
{
	bool res = true;
	uint8_t PesHeader[PES_MAX_HEADER_SIZE];

	if (initialHeader) {
		initialHeader = false;
		prepareClipPlay();
		ioctl(fd, AUDIO_CLEAR_BUFFER, NULL);
	}

	size += breakBufferFillSize;

	while (size >= SubFrameLen) {
		if (breakBufferFillSize)
			memcpy(injectBuffer, breakBuffer, breakBufferFillSize);
		memcpy(injectBuffer + breakBufferFillSize, data, SubFrameLen - breakBufferFillSize);
		size -= SubFrameLen;
		data += SubFrameLen - breakBufferFillSize;
		breakBufferFillSize = 0;

		//write the PCM data
		if (uBitsPerSample == 16) {
			for (unsigned int n = 0; n < SubFrameLen; n += 2) {
				uint8_t tmp = injectBuffer[n];
				injectBuffer[n] = injectBuffer[n + 1];
				injectBuffer[n + 1] = tmp;
			}
		} else {
			//      0   1   2   3   4   5   6   7   8   9  10  11
			//    A1c A1b A1a B1c B1b B1a A2c A2b A2a B2c B2b B2a
			// to A1a A1b B1a B1b A2a A2b B2a B2b A1c B1c A2c B2c
			for (unsigned int n = 0; n < SubFrameLen; n += 12) {
				uint8_t t, *p = injectBuffer + n;
				t = p[0];
				p[0] = p[2];
				p[2] = p[5];
				p[5] = p[7];
				p[7] = p[11];
				p[11] = p[9];
				p[9] = p[3];
				p[3] = p[4];
				p[4] = p[8];
				p[8] = t;
			}
		}

		//increment err... subframe count?
		lpcm_prv[1] = ((lpcm_prv[1] + SubFramesPerPES) & 0x1F);

		struct iovec iov[3];
		iov[0].iov_base = PesHeader;
		iov[1].iov_base = lpcm_prv;
		iov[1].iov_len = sizeof(lpcm_prv);
		iov[2].iov_base = injectBuffer;
		iov[2].iov_len = SubFrameLen;
		iov[0].iov_len = InsertPesHeader(PesHeader, iov[1].iov_len + iov[2].iov_len, PCM_PES_START_CODE, Pts, 0);
		int len = writev(fd, iov, 3);
		if (len < 0) {
			res = false;
			break;
		}
	}
	if (size && res) {
		breakBufferFillSize = size;
		memcpy(breakBuffer, data, size);
	}

	return res;
}

void WriterPCM::Init(int _fd, AVStream *_stream, Player *_player)
{
	fd = _fd;
	stream = _stream;
	player = _player;
	initialHeader = true;
	restart_audio_resampling = true;
}

bool WriterPCM::Write(AVPacket *packet, int64_t pts)
{
	if (!packet) {
		restart_audio_resampling = true;
		return true;
	}

	AVCodecContext *c = stream->codec;

	if (restart_audio_resampling) {
		restart_audio_resampling = false;
		initialHeader = true;

		swr_free(&swr);

		AVCodec *codec = avcodec_find_decoder(c->codec_id);
		if (!codec) {
			fprintf(stderr, "%s %d: avcodec_find_decoder(%llx)\n", __func__, __LINE__, (unsigned long long) c->codec_id);
			return false;
		}
		avcodec_close(c);
		if (avcodec_open2(c, codec, NULL)) {
			fprintf(stderr, "%s %d: avcodec_open2 failed\n", __func__, __LINE__);
			return false;
		}
	}

	if (!swr) {
		int rates[] = { 48000, 96000, 192000, 44100, 88200, 176400, 0 };
		int *rate = rates;
		int in_rate = c->sample_rate;
		while (*rate && ((*rate / in_rate) * in_rate != *rate) && (in_rate / *rate) * *rate != in_rate)
			rate++;
		out_sample_rate = *rate ? *rate : 44100;
		out_channels = c->channels;
		if (c->channel_layout == 0) {
			// FIXME -- need to guess, looks pretty much like a bug in the FFMPEG WMA decoder
			c->channel_layout = AV_CH_LAYOUT_STEREO;
		}

		out_channel_layout = c->channel_layout;
		// player2 won't play mono
		if (out_channel_layout == AV_CH_LAYOUT_MONO) {
			out_channel_layout = AV_CH_LAYOUT_STEREO;
			out_channels = 2;
		}

		uSampleRate = out_sample_rate;
		uNoOfChannels = av_get_channel_layout_nb_channels(out_channel_layout);
		uBitsPerSample = 16;

		swr = swr_alloc();
		if (!swr) {
			fprintf(stderr, "%s %d: swr_alloc failed\n", __func__, __LINE__);
			return false;
		}
		av_opt_set_int(swr, "in_channel_layout", c->channel_layout, 0);
		av_opt_set_int(swr, "out_channel_layout", out_channel_layout, 0);
		av_opt_set_int(swr, "in_sample_rate", c->sample_rate, 0);
		av_opt_set_int(swr, "out_sample_rate", out_sample_rate, 0);
		av_opt_set_sample_fmt(swr, "in_sample_fmt", c->sample_fmt, 0);
		av_opt_set_sample_fmt(swr, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0);

		int e = swr_init(swr);
		if (e < 0) {
			fprintf(stderr, "swr_init: %d (icl=%d ocl=%d isr=%d osr=%d isf=%d osf=%d)\n",
				-e, (int) c->channel_layout,
				(int) out_channel_layout, c->sample_rate, out_sample_rate, c->sample_fmt, AV_SAMPLE_FMT_S16);
			restart_audio_resampling = true;
			return false;
		}
	}

	unsigned int packet_size = packet->size;
	while (packet_size > 0 || (!packet_size && !packet->data)) {
		av_frame_unref(decoded_frame);

		int got_frame = 0;
		int len = avcodec_decode_audio4(c, decoded_frame, &got_frame, packet);
		if (len < 0) {
			restart_audio_resampling = true;
			break;
		}

		if (packet->data)
			packet_size -= len;

		if (!got_frame) {
			if (!packet->data || !packet_size)
				break;
			continue;
		}

		pts = player->input.calcPts(stream, av_frame_get_best_effort_timestamp(decoded_frame));

		int in_samples = decoded_frame->nb_samples;
		int out_samples = av_rescale_rnd(swr_get_delay(swr, c->sample_rate) + in_samples, out_sample_rate, c->sample_rate, AV_ROUND_UP);
		if (out_samples > out_samples_max) {
			if (output)
				av_freep(&output);
			int e = av_samples_alloc(&output, NULL, out_channels, out_samples, AV_SAMPLE_FMT_S16, 1);
			if (e < 0) {
				fprintf(stderr, "av_samples_alloc: %d\n", -e);
				break;
			}
			out_samples_max = out_samples;
		}

		out_samples = swr_convert(swr, &output, out_samples, (const uint8_t **) &decoded_frame->data[0], in_samples);

		if (!writePCM(pts, output, out_samples * sizeof(short) * out_channels)) {
			restart_audio_resampling = true;
			break;
		}
	}
	return !packet_size;
}

WriterPCM::WriterPCM()
{
	swr = NULL;
	output = NULL;
	out_samples_max = 0;
	decoded_frame = av_frame_alloc();

	Register(this, AV_CODEC_ID_INJECTPCM, AUDIO_ENCODING_LPCMA);
}

static WriterPCM writer_pcm __attribute__ ((init_priority (300)));