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libeplayer3: rename files
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martii
371
libeplayer3/writer/pcm.cpp
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371
libeplayer3/writer/pcm.cpp
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/*
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* linuxdvb output/writer handling.
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*
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* konfetti 2010 based on linuxdvb.c code from libeplayer2
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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 <sys/uio.h>
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#include <errno.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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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 unsigned char 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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unsigned char lpcm_prv[14];
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unsigned char breakBuffer[8192];
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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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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(int fd, AVFormatContext *avfc, AVStream *stream, AVPacket *packet, int64_t &pts);
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bool prepareClipPlay();
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int writePCM(int fd, int64_t Pts, uint8_t *data, unsigned int size);
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void Init();
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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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// FIXME: PES header size was hardcoded to 18 in earlier code. Actual size returned by InsertPesHeader is 14.
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SubFramesPerPES = ((2048 - 18) - 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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int WriterPCM::writePCM(int fd, int64_t Pts, uint8_t *data, unsigned int size)
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{
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unsigned char 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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}
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unsigned int n;
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unsigned char *injectBuffer = (unsigned char *) malloc(SubFrameLen);
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unsigned int pos;
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for (pos = 0; pos < size;) {
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//printf("PCM %s - Position=%d\n", __FUNCTION__, pos);
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if ((size - pos) < SubFrameLen) {
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breakBufferFillSize = size - pos;
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memcpy(breakBuffer, &data[pos],
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sizeof(unsigned char) * breakBufferFillSize);
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//printf("PCM %s - Unplayed=%d\n", __FUNCTION__, breakBufferFillSize);
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break;
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}
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//get first PES's worth
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if (breakBufferFillSize > 0) {
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memcpy(injectBuffer, breakBuffer, sizeof(unsigned char) * breakBufferFillSize);
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memcpy(&injectBuffer[breakBufferFillSize], &data[pos], sizeof(unsigned char) * (SubFrameLen - breakBufferFillSize));
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pos += (SubFrameLen - breakBufferFillSize);
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breakBufferFillSize = 0;
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} else {
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memcpy(injectBuffer, &data[pos], sizeof(unsigned char) * SubFrameLen);
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pos += SubFrameLen;
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}
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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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//write the PCM data
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if (uBitsPerSample == 16) {
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for (n = 0; n < SubFrameLen; n += 2) {
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unsigned char tmp;
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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 (n = 0; n < SubFrameLen; n += 12) {
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unsigned char 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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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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break;
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}
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free(injectBuffer);
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return size;
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}
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void WriterPCM::Init()
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{
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initialHeader = true;
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restart_audio_resampling = true;
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}
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extern int64_t calcPts(AVFormatContext *, AVStream *, int64_t);
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bool WriterPCM::Write(int fd, AVFormatContext *avfc, AVStream *stream, AVPacket *packet, int64_t &pts)
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{
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if (fd < 0)
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return false;
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if (!packet) {
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fprintf(stderr, "%s %s %d\n", __FILE__, __func__, __LINE__);
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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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unsigned int packet_size = packet->size;
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if (restart_audio_resampling) {
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fprintf(stderr, "%s %s %d\n", __FILE__, __func__, __LINE__);
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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; //FIXME: Needed?
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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; //FIXME: Needed?
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}
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AVCodec *codec = avcodec_find_decoder(c->codec_id);
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if (!codec || avcodec_open2(c, codec, NULL))
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fprintf(stderr, "%s %d: avcodec_open2 failed\n", __func__, __LINE__);
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}
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while (packet_size > 0) {
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int got_frame = 0;
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if (decoded_frame)
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av_frame_unref(decoded_frame);
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else 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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int len = avcodec_decode_audio4(c, decoded_frame, &got_frame, packet);
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if (len < 0) {
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fprintf(stderr, "%s %s %d\n", __FILE__, __func__, __LINE__);
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restart_audio_resampling = true;
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break;
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}
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packet_size -= len;
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if (!got_frame)
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continue;
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int e;
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if (!swr) {
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int rates[] = { 48000, 96000, 192000, 44100, 88200, 176400, 0 };
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int *rate = rates;
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int in_rate = c->sample_rate;
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while (*rate && ((*rate / in_rate) * in_rate != *rate) && (in_rate / *rate) * *rate != in_rate)
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rate++;
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out_sample_rate = *rate ? *rate : 44100;
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swr = swr_alloc();
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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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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_int(swr, "in_sample_fmt", c->sample_fmt, 0);
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av_opt_set_int(swr, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0);
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e = swr_init(swr);
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if (e < 0) {
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fprintf(stderr,
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"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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swr_free(&swr);
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swr = NULL;
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}
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}
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uint8_t *output = NULL;
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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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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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continue;
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}
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// FIXME. PTS calculation is probably broken.
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int64_t pts;
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int64_t next_in_pts = av_rescale(av_frame_get_best_effort_timestamp(decoded_frame),
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stream->time_base.num * (int64_t) out_sample_rate * c->sample_rate,
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stream->time_base.den);
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int64_t next_out_pts = av_rescale(swr_next_pts(swr, next_in_pts),
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stream->time_base.den,
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stream->time_base.num * (int64_t) out_sample_rate * c->sample_rate);
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pts = calcPts(avfc, stream, next_out_pts);
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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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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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writePCM(fd, pts, output, out_samples * sizeof(short) * out_channels);
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av_freep(&output);
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}
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return true;
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}
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WriterPCM::WriterPCM()
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{
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swr = NULL;
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decoded_frame = NULL;
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out_sample_rate = 44100;
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out_channels = 2;
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out_channel_layout = AV_CH_LAYOUT_STEREO;
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restart_audio_resampling = true;
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Register(this, AV_CODEC_ID_PCM_S16LE/*FIXME*/, AUDIO_ENCODING_LPCMA);
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}
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static WriterPCM writer_pcm __attribute__ ((init_priority (300)));
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