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https://github.com/tuxbox-neutrino/libstb-hal.git
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- move libeplayer3-sh4 code to libeplayer3-sh4 directory; ...
rename library to libeplayer3_sh4.la Signed-off-by: Thilo Graf <dbt@novatux.de>
This commit is contained in:
svenhoefer
376
libeplayer3-sh4/writer/pcm.cpp
Normal file
376
libeplayer3-sh4/writer/pcm.cpp
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/*
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* linuxdvb output/writer handling.
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*
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* Copyright (C) 2010 konfetti (based on code from libeplayer2)
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* Copyright (C) 2014 martii (based on code from libeplayer3)
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with this program; if not, write to the Free Software
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* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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*
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*/
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#include <stdio.h>
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#include <stdlib.h>
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#include <stdint.h>
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#include <string.h>
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#include <errno.h>
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#include <sys/ioctl.h>
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#include <sys/uio.h>
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#include <linux/dvb/audio.h>
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#include "misc.h"
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#include "pes.h"
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#include "writer.h"
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#include "player.h"
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extern "C" {
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#include <libavutil/avutil.h>
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#include <libavutil/time.h>
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#include <libavformat/avformat.h>
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#include <libswresample/swresample.h>
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#include <libavutil/opt.h>
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}
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// reference: search for TypeLpcmDVDAudio in player/frame_parser/frame_parser_audio_lpcm.cpp
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static const uint8_t clpcm_prv[14] = {
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0xA0, //sub_stream_id
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0, 0, //resvd and UPC_EAN_ISRC stuff, unused
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0x0A, //private header length
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0, 9, //first_access_unit_pointer
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0x00, //emph,rsvd,stereo,downmix
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0x0F, //quantisation word length 1,2
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0x0F, //audio sampling freqency 1,2
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0, //resvd, multi channel type
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0, //bit shift on channel GR2, assignment
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0x80, //dynamic range control
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0, 0 //resvd for copyright management
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};
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class WriterPCM : public Writer
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{
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private:
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unsigned int SubFrameLen;
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unsigned int SubFramesPerPES;
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uint8_t lpcm_prv[14];
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uint8_t injectBuffer[2048];
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uint8_t breakBuffer[sizeof(injectBuffer)];
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uint8_t *output;
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uint8_t out_samples_max;
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unsigned int breakBufferFillSize;
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int uNoOfChannels;
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int uSampleRate;
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int uBitsPerSample;
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AVStream *stream;
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SwrContext *swr;
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AVFrame *decoded_frame;
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int out_sample_rate;
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int out_channels;
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uint64_t out_channel_layout;
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bool initialHeader;
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bool restart_audio_resampling;
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public:
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bool Write(AVPacket *packet, int64_t pts);
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bool prepareClipPlay();
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bool writePCM(int64_t Pts, uint8_t *data, unsigned int size);
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void Init(int _fd, AVStream *_stream, Player *_player);
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WriterPCM();
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};
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bool WriterPCM::prepareClipPlay()
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{
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SubFrameLen = 0;
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SubFramesPerPES = 0;
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breakBufferFillSize = 0;
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memcpy(lpcm_prv, clpcm_prv, sizeof(lpcm_prv));
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// figure out size of subframe and set up sample rate
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switch (uSampleRate) {
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case 48000:
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SubFrameLen = 40;
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break;
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case 96000:
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lpcm_prv[8] |= 0x10;
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SubFrameLen = 80;
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break;
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case 192000:
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lpcm_prv[8] |= 0x20;
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SubFrameLen = 160;
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break;
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case 44100:
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lpcm_prv[8] |= 0x80;
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SubFrameLen = 40;
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break;
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case 88200:
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lpcm_prv[8] |= 0x90;
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SubFrameLen = 80;
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break;
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case 176400:
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lpcm_prv[8] |= 0xA0;
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SubFrameLen = 160;
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break;
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default:
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break;
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}
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SubFrameLen *= uNoOfChannels;
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SubFrameLen *= uBitsPerSample / 8;
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//rewrite PES size to have as many complete subframes per PES as we can
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SubFramesPerPES = ((sizeof(injectBuffer) - 14) - sizeof(lpcm_prv)) / SubFrameLen;
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SubFrameLen *= SubFramesPerPES;
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//set number of channels
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lpcm_prv[10] = uNoOfChannels - 1;
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switch (uBitsPerSample) {
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case 24:
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lpcm_prv[7] |= 0x20;
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case 16:
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break;
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default:
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printf("inappropriate bits per sample (%d) - must be 16 or 24\n", uBitsPerSample);
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return false;
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}
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return true;
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}
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bool WriterPCM::writePCM(int64_t Pts, uint8_t *data, unsigned int size)
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{
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bool res = true;
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uint8_t PesHeader[PES_MAX_HEADER_SIZE];
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if (initialHeader) {
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initialHeader = false;
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prepareClipPlay();
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ioctl(fd, AUDIO_CLEAR_BUFFER, NULL);
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}
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size += breakBufferFillSize;
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while (size >= SubFrameLen) {
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if (breakBufferFillSize)
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memcpy(injectBuffer, breakBuffer, breakBufferFillSize);
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memcpy(injectBuffer + breakBufferFillSize, data, SubFrameLen - breakBufferFillSize);
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size -= SubFrameLen;
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data += SubFrameLen - breakBufferFillSize;
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breakBufferFillSize = 0;
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//write the PCM data
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if (uBitsPerSample == 16) {
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for (unsigned int n = 0; n < SubFrameLen; n += 2) {
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uint8_t tmp = injectBuffer[n];
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injectBuffer[n] = injectBuffer[n + 1];
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injectBuffer[n + 1] = tmp;
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}
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} else {
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// 0 1 2 3 4 5 6 7 8 9 10 11
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// A1c A1b A1a B1c B1b B1a A2c A2b A2a B2c B2b B2a
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// to A1a A1b B1a B1b A2a A2b B2a B2b A1c B1c A2c B2c
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for (unsigned int n = 0; n < SubFrameLen; n += 12) {
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uint8_t t, *p = injectBuffer + n;
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t = p[0];
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p[0] = p[2];
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p[2] = p[5];
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p[5] = p[7];
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p[7] = p[11];
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p[11] = p[9];
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p[9] = p[3];
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p[3] = p[4];
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p[4] = p[8];
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p[8] = t;
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}
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}
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//increment err... subframe count?
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lpcm_prv[1] = ((lpcm_prv[1] + SubFramesPerPES) & 0x1F);
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struct iovec iov[3];
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iov[0].iov_base = PesHeader;
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iov[1].iov_base = lpcm_prv;
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iov[1].iov_len = sizeof(lpcm_prv);
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iov[2].iov_base = injectBuffer;
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iov[2].iov_len = SubFrameLen;
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iov[0].iov_len = InsertPesHeader(PesHeader, iov[1].iov_len + iov[2].iov_len, PCM_PES_START_CODE, Pts, 0);
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int len = writev(fd, iov, 3);
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if (len < 0) {
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res = false;
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break;
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}
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}
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if (size && res) {
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breakBufferFillSize = size;
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memcpy(breakBuffer, data, size);
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}
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return res;
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}
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void WriterPCM::Init(int _fd, AVStream *_stream, Player *_player)
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{
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fd = _fd;
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stream = _stream;
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player = _player;
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initialHeader = true;
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restart_audio_resampling = true;
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}
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bool WriterPCM::Write(AVPacket *packet, int64_t pts)
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{
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if (!packet) {
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restart_audio_resampling = true;
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return true;
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}
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AVCodecContext *c = stream->codec;
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if (restart_audio_resampling) {
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restart_audio_resampling = false;
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initialHeader = true;
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if (swr) {
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swr_free(&swr);
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swr = NULL;
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}
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if (decoded_frame) {
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av_frame_free(&decoded_frame);
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decoded_frame = NULL;
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}
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AVCodec *codec = avcodec_find_decoder(c->codec_id);
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if (!codec) {
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fprintf(stderr, "%s %d: avcodec_find_decoder(%llx)\n", __func__, __LINE__, (unsigned long long) c->codec_id);
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return false;
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}
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avcodec_close(c);
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if (avcodec_open2(c, codec, NULL)) {
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fprintf(stderr, "%s %d: avcodec_open2 failed\n", __func__, __LINE__);
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return false;
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}
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}
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if (!swr) {
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int in_rate = c->sample_rate;
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// rates in descending order
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int rates[] = {192000, 176400, 96000, 88200, 48000, 44100, 0};
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int i = 0;
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// find the next equal or smallest rate
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while (rates[i] && in_rate < rates[i])
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i++;
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out_sample_rate = rates[i] ? rates[i] : 44100;
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out_channels = c->channels;
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if (c->channel_layout == 0) {
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// FIXME -- need to guess, looks pretty much like a bug in the FFMPEG WMA decoder
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c->channel_layout = AV_CH_LAYOUT_STEREO;
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}
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out_channel_layout = c->channel_layout;
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// player2 won't play mono
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if (out_channel_layout == AV_CH_LAYOUT_MONO) {
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out_channel_layout = AV_CH_LAYOUT_STEREO;
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out_channels = 2;
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}
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uSampleRate = out_sample_rate;
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uNoOfChannels = av_get_channel_layout_nb_channels(out_channel_layout);
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uBitsPerSample = 16;
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swr = swr_alloc();
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if (!swr) {
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fprintf(stderr, "%s %d: swr_alloc failed\n", __func__, __LINE__);
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return false;
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}
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av_opt_set_int(swr, "in_channel_layout", c->channel_layout, 0);
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av_opt_set_int(swr, "out_channel_layout", out_channel_layout, 0);
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av_opt_set_int(swr, "in_sample_rate", c->sample_rate, 0);
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av_opt_set_int(swr, "out_sample_rate", out_sample_rate, 0);
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av_opt_set_sample_fmt(swr, "in_sample_fmt", c->sample_fmt, 0);
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av_opt_set_sample_fmt(swr, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0);
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int e = swr_init(swr);
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if (e < 0) {
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fprintf(stderr, "swr_init: %d (icl=%d ocl=%d isr=%d osr=%d isf=%d osf=%d)\n",
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-e, (int) c->channel_layout,
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(int) out_channel_layout, c->sample_rate, out_sample_rate, c->sample_fmt, AV_SAMPLE_FMT_S16);
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restart_audio_resampling = true;
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return false;
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}
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}
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unsigned int packet_size = packet->size;
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while (packet_size > 0 || (!packet_size && !packet->data)) {
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int got_frame = 0;
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if (!decoded_frame) {
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if (!(decoded_frame = av_frame_alloc())) {
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fprintf(stderr, "out of memory\n");
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exit(1);
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}
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} else
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av_frame_unref(decoded_frame);
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int len = avcodec_decode_audio4(c, decoded_frame, &got_frame, packet);
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if (len < 0) {
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restart_audio_resampling = true;
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break;
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}
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if (packet->data)
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packet_size -= len;
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if (!got_frame) {
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if (!packet->data || !packet_size)
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break;
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continue;
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}
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pts = player->input.calcPts(stream, av_frame_get_best_effort_timestamp(decoded_frame));
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int in_samples = decoded_frame->nb_samples;
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int out_samples = av_rescale_rnd(swr_get_delay(swr, c->sample_rate) + in_samples, out_sample_rate, c->sample_rate, AV_ROUND_UP);
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if (out_samples > out_samples_max) {
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if (output)
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av_freep(&output);
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int e = av_samples_alloc(&output, NULL, out_channels, out_samples, AV_SAMPLE_FMT_S16, 1);
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if (e < 0) {
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fprintf(stderr, "av_samples_alloc: %d\n", -e);
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break;
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}
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out_samples_max = out_samples;
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}
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out_samples = swr_convert(swr, &output, out_samples, (const uint8_t **) &decoded_frame->data[0], in_samples);
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if (!writePCM(pts, output, out_samples * sizeof(short) * out_channels)) {
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restart_audio_resampling = true;
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break;
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}
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}
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return !packet_size;
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}
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WriterPCM::WriterPCM()
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{
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swr = NULL;
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output = NULL;
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out_samples_max = 0;
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decoded_frame = av_frame_alloc();
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Register(this, AV_CODEC_ID_INJECTPCM, AUDIO_ENCODING_LPCMA);
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}
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static WriterPCM writer_pcm __attribute__ ((init_priority (300)));
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Block a user