mirror of
https://github.com/tuxbox-fork-migrations/recycled-ni-libstb-hal.git
synced 2025-08-26 23:12:44 +02:00
Origin commit data
------------------
Branch: master
Commit: 61903c41fd
Author: max_10 <max_10@gmx.de>
Date: 2017-12-15 (Fri, 15 Dec 2017)
------------------
No further description and justification available within origin commit message!
------------------
This commit was generated by Migit
377 lines
9.6 KiB
C++
377 lines
9.6 KiB
C++
/*
|
|
* 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;
|
|
|
|
if (swr) {
|
|
swr_free(&swr);
|
|
swr = NULL;
|
|
}
|
|
if (decoded_frame) {
|
|
av_frame_free(&decoded_frame);
|
|
decoded_frame = NULL;
|
|
}
|
|
|
|
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 in_rate = c->sample_rate;
|
|
// rates in descending order
|
|
int rates[] = {192000, 176400, 96000, 88200, 48000, 44100, 0};
|
|
int i = 0;
|
|
// find the next equal or smallest rate
|
|
while (rates[i] && in_rate < rates[i])
|
|
i++;
|
|
out_sample_rate = rates[i] ? rates[i] : 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)) {
|
|
int got_frame = 0;
|
|
|
|
if (!decoded_frame) {
|
|
if (!(decoded_frame = av_frame_alloc())) {
|
|
fprintf(stderr, "out of memory\n");
|
|
exit(1);
|
|
}
|
|
} else
|
|
av_frame_unref(decoded_frame);
|
|
|
|
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)));
|