frodovdr
/
vdr-plugin-softhdcuvid
mirror of https://github.com/jojo61/vdr-plugin-softhdcuvid.git
1
0
Fork 0
Code Issues Releases Wiki Activity
vdr-plugin-softhdcuvid/codec.c

1854 lines
51 KiB
C
Raw Permalink Blame History

///
/// @file codec.c @brief Codec functions
///
/// Copyright (c) 2009 - 2015 by Johns. All Rights Reserved.
///
/// Contributor(s):
///
/// License: AGPLv3
///
/// This program is free software: you can redistribute it and/or modify
/// it under the terms of the GNU Affero General Public License as
/// published by the Free Software Foundation, either version 3 of the
/// License.
///
/// 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 Affero General Public License for more details.
///
/// $Id: d285eb28485bea02cd205fc8be47320dfe0376cf $
//////////////////////////////////////////////////////////////////////////////
///
/// @defgroup Codec The codec module.
///
/// This module contains all decoder and codec functions.
/// It is uses ffmpeg (http://ffmpeg.org) as backend.
///
/// It may work with libav (http://libav.org), but the tests show
/// many bugs and incompatiblity in it. Don't use this shit.
///
/// compile with pass-through support (stable, AC-3, E-AC-3 only)
#define USE_PASSTHROUGH
/// compile audio drift correction support (very experimental)
#define USE_AUDIO_DRIFT_CORRECTION
/// compile AC-3 audio drift correction support (very experimental)
#define USE_AC3_DRIFT_CORRECTION
/// use ffmpeg libswresample API (autodected, Makefile)
#define noUSE_SWRESAMPLE
/// use libav libavresample API (autodected, Makefile)
#define noUSE_AVRESAMPLE
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <unistd.h>
#ifdef __FreeBSD__
#include <sys/endian.h>
#else
#include <endian.h>
#endif
#include <sys/types.h>
#include <sys/stat.h>
#include <fcntl.h>
#include <libintl.h>
#define _(str) gettext(str) ///< gettext shortcut
#define _N(str) str ///< gettext_noop shortcut
#include <libavcodec/avcodec.h>
#include <libavutil/mem.h>
// support old ffmpeg versions <1.0
#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(55,18,102)
#define AVCodecID CodecID
#define AV_CODEC_ID_AC3 CODEC_ID_AC3
#define AV_CODEC_ID_EAC3 CODEC_ID_EAC3
#define AV_CODEC_ID_MPEG2VIDEO CODEC_ID_MPEG2VIDEO
#define AV_CODEC_ID_H264 CODEC_ID_H264
#endif
#ifdef USE_SWRESAMPLE
#include <libswresample/swresample.h>
#endif
#ifdef USE_AVRESAMPLE
#include <libavresample/avresample.h>
#include <libavutil/opt.h>
#endif
#ifndef __USE_GNU
#define __USE_GNU
#endif
#include <pthread.h>
#ifdef MAIN_H
#include MAIN_H
#endif
#include "iatomic.h"
#include "misc.h"
#include "video.h"
#include "audio.h"
#include "codec.h"
//----------------------------------------------------------------------------
// correct is AV_VERSION_INT(56,35,101) but some gentoo i* think
// they must change it.
#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(56,26,100)
/// ffmpeg 2.6 started to show artifacts after channel switch
/// to SDTV channels
#define FFMPEG_WORKAROUND_ARTIFACTS 1
#endif
//----------------------------------------------------------------------------
// Global
//----------------------------------------------------------------------------
///
/// ffmpeg lock mutex
///
/// new ffmpeg dislikes simultanous open/close
/// this breaks our code, until this is fixed use lock.
///
static pthread_mutex_t CodecLockMutex;
/// Flag prefer fast channel switch
char CodecUsePossibleDefectFrames;
AVBufferRef *hw_device_ctx;
//----------------------------------------------------------------------------
// Video
//----------------------------------------------------------------------------
#if 0
///
/// Video decoder typedef.
///
//typedef struct _video_decoder_ Decoder;
#endif
#if 0
///
/// Video decoder structure.
///
struct _video_decoder_
{
VideoHwDecoder *HwDecoder; ///< video hardware decoder
int GetFormatDone; ///< flag get format called!
AVCodec *VideoCodec; ///< video codec
AVCodecContext *VideoCtx; ///< video codec context
#ifdef FFMPEG_WORKAROUND_ARTIFACTS
int FirstKeyFrame; ///< flag first frame
#endif
AVFrame *Frame; ///< decoded video frame
};
#endif
//----------------------------------------------------------------------------
// Call-backs
//----------------------------------------------------------------------------
/**
** Callback to negotiate the PixelFormat.
**
** @param video_ctx codec context
** @param fmt is the list of formats which are supported by
** the codec, it is terminated by -1 as 0 is a
** valid format, the formats are ordered by
** quality.
*/
static enum AVPixelFormat Codec_get_format(AVCodecContext * video_ctx,
const enum AVPixelFormat *fmt)
{
VideoDecoder *decoder;
decoder = video_ctx->opaque;
#if LIBAVCODEC_VERSION_INT == AV_VERSION_INT(54,86,100)
// this begins to stink, 1.1.2 calls get_format for each frame
// 1.1.3 has the same version, but works again
if (decoder->GetFormatDone) {
if (decoder->GetFormatDone < 10) {
++decoder->GetFormatDone;
Error
("codec/video: ffmpeg/libav buggy: get_format called again\n");
}
return *fmt; // FIXME: this is hack
}
#endif
// bug in ffmpeg 1.1.1, called with zero width or height
if (!video_ctx->width || !video_ctx->height) {
Error("codec/video: ffmpeg/libav buggy: width or height zero\n");
}
decoder->GetFormatDone = 1;
return Video_get_format(decoder->HwDecoder, video_ctx, fmt);
}
//static void Codec_free_buffer(void *opaque, uint8_t *data);
/**
** Video buffer management, get buffer for frame.
**
** Called at the beginning of each frame to get a buffer for it.
**
** @param video_ctx Codec context
** @param frame Get buffer for this frame
*/
static int Codec_get_buffer2(AVCodecContext * video_ctx, AVFrame * frame, int flags)
{
VideoDecoder *decoder;
decoder = video_ctx->opaque;
#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(54,86,100)
// ffmpeg has this already fixed
// libav 0.8.5 53.35.0 still needs this
#endif
if (!decoder->GetFormatDone) { // get_format missing
enum AVPixelFormat fmts[2];
fprintf(stderr, "codec: buggy libav, use ffmpeg\n");
Warning(_("codec: buggy libav, use ffmpeg\n"));
fmts[0] = video_ctx->pix_fmt;
fmts[1] = AV_PIX_FMT_NONE;
Codec_get_format(video_ctx, fmts);
}
if (decoder->hwaccel_get_buffer && (AV_PIX_FMT_VDPAU == decoder->hwaccel_pix_fmt || AV_PIX_FMT_CUDA == decoder->hwaccel_pix_fmt)) {
//Debug(3,"hwaccel get_buffer\n");
return decoder->hwaccel_get_buffer(video_ctx, frame, flags);
}
//Debug(3, "codec: fallback to default get_buffer\n");
return avcodec_default_get_buffer2(video_ctx, frame, flags);
}
//----------------------------------------------------------------------------
// Test
//----------------------------------------------------------------------------
/**
** Allocate a new video decoder context.
**
** @param hw_decoder video hardware decoder
**
** @returns private decoder pointer for video decoder.
*/
VideoDecoder *CodecVideoNewDecoder(VideoHwDecoder * hw_decoder)
{
VideoDecoder *decoder;
if (!(decoder = calloc(1, sizeof(*decoder)))) {
Fatal(_("codec: can't allocate vodeo decoder\n"));
}
decoder->HwDecoder = hw_decoder;
return decoder;
}
/**
** Deallocate a video decoder context.
**
** @param decoder private video decoder
*/
void CodecVideoDelDecoder(VideoDecoder * decoder)
{
free(decoder);
}
/**
** Open video decoder.
**
** @param decoder private video decoder
** @param codec_id video codec id
*/
void CodecVideoOpen(VideoDecoder * decoder, int codec_id)
{
AVCodec *video_codec;
const char *name;
int ret;
Debug(3, "***************codec: Video Open using video codec ID %#06x (%s)\n", codec_id,
avcodec_get_name(codec_id));
if (decoder->VideoCtx) {
Error(_("codec: missing close\n"));
}
name = "NULL";
if (!strcasecmp(VideoGetDriverName(), "cuvid")) {
switch (codec_id) {
case AV_CODEC_ID_MPEG2VIDEO:
name = "mpeg2_cuvid";
break;
case AV_CODEC_ID_H264:
name = "h264_cuvid";
break;
case AV_CODEC_ID_HEVC:
name = "hevc_cuvid";
break;
}
}
if (name && (video_codec = avcodec_find_decoder_by_name(name))) {
Debug(3, "codec: decoder found\n");
} else {
Debug(3,"Decoder %s not supported\n",name);
Fatal(_(" No decoder found"));
}
decoder->VideoCodec = video_codec;
Debug(3, "codec: video '%s'\n", decoder->VideoCodec->long_name);
if (!(decoder->VideoCtx = avcodec_alloc_context3(video_codec))) {
Fatal(_("codec: can't allocate video codec context\n"));
}
if (!HwDeviceContext) {
Fatal("codec: no hw device context to be used");
}
decoder->VideoCtx->hw_device_ctx = av_buffer_ref(HwDeviceContext);
// FIXME: for software decoder use all cpus, otherwise 1
decoder->VideoCtx->thread_count = 1;
decoder->VideoCtx->pkt_timebase.num = 1;
decoder->VideoCtx->pkt_timebase.den = 90000;
decoder->VideoCtx->framerate.num = 50;
decoder->VideoCtx->framerate.den = 1;
pthread_mutex_lock(&CodecLockMutex);
// open codec
if (strcmp(decoder->VideoCodec->long_name,"Nvidia CUVID MPEG2VIDEO decoder") == 0) { // deinterlace for mpeg2 is somehow broken
if (av_opt_set_int(decoder->VideoCtx->priv_data, "deint", 2 ,0) < 0) { // adaptive
pthread_mutex_unlock(&CodecLockMutex);
Fatal(_("codec: can't set option deint to video codec!\n"));
}
#if 1
if (av_opt_set_int(decoder->VideoCtx->priv_data, "surfaces", 9 ,0) < 0) {
pthread_mutex_unlock(&CodecLockMutex);
Fatal(_("codec: can't set option surfces to video codec!\n"));
}
#endif
if (av_opt_set(decoder->VideoCtx->priv_data, "drop_second_field", "false" ,0) < 0) {
pthread_mutex_unlock(&CodecLockMutex);
Fatal(_("codec: can't set option drop 2.field to video codec!\n"));
}
}
else if (strstr(decoder->VideoCodec->long_name,"Nvidia CUVID") != NULL) {
if (av_opt_set_int(decoder->VideoCtx->priv_data, "deint", 2 ,0) < 0) { // adaptive
pthread_mutex_unlock(&CodecLockMutex);
Fatal(_("codec: can't set option deint to video codec!\n"));
}
if (av_opt_set(decoder->VideoCtx->priv_data, "drop_second_field", "false" ,0) < 0) {
pthread_mutex_unlock(&CodecLockMutex);
Fatal(_("codec: can't set option drop 2.field to video codec!\n"));
}
}
if ((ret = avcodec_open2(decoder->VideoCtx, video_codec, NULL)) < 0) {
pthread_mutex_unlock(&CodecLockMutex);
Fatal(_("codec: can't open video codec!\n"));
}
Debug(3," Codec open %d\n",ret);
pthread_mutex_unlock(&CodecLockMutex);
decoder->VideoCtx->opaque = decoder; // our structure
//decoder->VideoCtx->debug = FF_DEBUG_STARTCODE;
//decoder->VideoCtx->err_recognition |= AV_EF_EXPLODE;
decoder->VideoCtx->get_format = Codec_get_format;
decoder->VideoCtx->get_buffer2 = Codec_get_buffer2;
decoder->VideoCtx->thread_count = 1;
decoder->VideoCtx->active_thread_type = 0;
decoder->VideoCtx->draw_horiz_band = NULL;
if (strstr(decoder->VideoCodec->long_name,"Nvidia CUVID") != NULL)
decoder->VideoCtx->hwaccel_context = VideoGetHwAccelContext(decoder->HwDecoder);
//
// Prepare frame buffer for decoder
//
if (!(decoder->Frame = av_frame_alloc())) {
Fatal(_("codec: can't allocate video decoder frame buffer\n"));
}
// reset buggy ffmpeg/libav flag
decoder->GetFormatDone = 0;
#ifdef FFMPEG_WORKAROUND_ARTIFACTS
decoder->FirstKeyFrame = 1;
#endif
}
/**
** Close video decoder.
**
** @param video_decoder private video decoder
*/
void CodecVideoClose(VideoDecoder *video_decoder)
{
AVCodecContext *video_ctx = video_decoder->VideoCtx;;
int ret;
AVFrame *frame;
// FIXME: play buffered data
av_frame_free(&video_decoder->Frame); // callee does checks
Debug(3,"CodecVideoClose\n");
if (video_decoder->VideoCtx) {
pthread_mutex_lock(&CodecLockMutex);
// avcodec_send_packet(video_ctx, NULL);
// while (avcodec_receive_frame(video_ctx,video_decoder->Frame) >= 0);
avcodec_close(video_decoder->VideoCtx);
av_freep(&video_decoder->VideoCtx);
pthread_mutex_unlock(&CodecLockMutex);
}
}
#if 0
/**
** Display pts...
**
** ffmpeg-0.9 pts always AV_NOPTS_VALUE
** ffmpeg-0.9 pkt_pts nice monotonic (only with HD)
** ffmpeg-0.9 pkt_dts wild jumping -160 - 340 ms
**
** libav 0.8_pre20111116 pts always AV_NOPTS_VALUE
** libav 0.8_pre20111116 pkt_pts always 0 (could be fixed?)
** libav 0.8_pre20111116 pkt_dts wild jumping -160 - 340 ms
*/
void DisplayPts(AVCodecContext * video_ctx, AVFrame * frame)
{
int ms_delay;
int64_t pts;
static int64_t last_pts;
pts = frame->pkt_pts;
if (pts == (int64_t) AV_NOPTS_VALUE) {
printf("*");
}
ms_delay = (1000 * video_ctx->time_base.num) / video_ctx->time_base.den;
ms_delay += frame->repeat_pict * ms_delay / 2;
printf("codec: PTS %s%s %" PRId64 " %d %d/%d %d/%d %dms\n",
frame->repeat_pict ? "r" : " ", frame->interlaced_frame ? "I" : " ",
pts, (int)(pts - last_pts) / 90, video_ctx->time_base.num,
video_ctx->time_base.den, video_ctx->framerate.num, video_ctx->framerate.den, ms_delay);
if (pts != (int64_t) AV_NOPTS_VALUE) {
last_pts = pts;
}
}
#endif
/**
** Decode a video packet.
**
** @param decoder video decoder data
** @param avpkt video packet
*/
extern int CuvidTestSurfaces();
void CodecVideoDecode(VideoDecoder * decoder, const AVPacket * avpkt)
{
AVCodecContext *video_ctx;
AVFrame *frame;
int used,ret,ret1;
int got_frame;
int consumed = 0;
AVPacket *pkt;
static uint64_t last_time = 0;
static uint64_t lastpts = 0;
next_part:
video_ctx = decoder->VideoCtx;
frame = decoder->Frame;
pkt = avpkt; // use copy
got_frame = 0;
ret1 = avcodec_send_packet(video_ctx, pkt);
if (ret1 >= 0) {
consumed = 1;
}
if ((ret1 == AVERROR(EAGAIN) || ret1 == AVERROR_EOF || ret1 >= 0) && CuvidTestSurfaces()) {
ret = 0;
while ((ret >= 0) && CuvidTestSurfaces()) { // get frames until empty snd Surfaces avail.
ret = avcodec_receive_frame(video_ctx, frame); // get new frame
if (ret >= 0) { // one is avail.
got_frame = 1;
}
else {
got_frame = 0;
}
if (got_frame) { // frame completed
#ifdef FFMPEG_WORKAROUND_ARTIFACTS
if (!CodecUsePossibleDefectFrames && decoder->FirstKeyFrame) {
decoder->FirstKeyFrame++;
if (frame->key_frame || (decoder->FirstKeyFrame > 3)) { // key frame is not reliable
Debug(3, "codec: key frame after %d frames\n",decoder->FirstKeyFrame);
decoder->FirstKeyFrame = 0;
VideoRenderFrame(decoder->HwDecoder, video_ctx, frame);
}
av_frame_unref(frame);
} else {
//DisplayPts(video_ctx, frame);
VideoRenderFrame(decoder->HwDecoder, video_ctx, frame);
av_frame_unref(frame);
}
#else
//DisplayPts(video_ctx, frame);
VideoRenderFrame(decoder->HwDecoder, video_ctx, frame);
av_frame_unref(frame);
#endif
// printf("got frame\n");
} else {
// printf("codec: got no frame %d send %d\n",ret,ret1);
}
}
} else {
// consumed = 1;
}
if (!consumed) {
goto next_part; // try again to stuff decoder
}
}
/**
** Flush the video decoder.
**
** @param decoder video decoder data
*/
void CodecVideoFlushBuffers(VideoDecoder * decoder)
{
AVFrame *frame;
frame = decoder->Frame;
if (decoder->VideoCtx) {
avcodec_flush_buffers(decoder->VideoCtx);
}
}
//----------------------------------------------------------------------------
// Audio
//----------------------------------------------------------------------------
#if 0
///
/// Audio decoder typedef.
///
typedef struct _audio_decoder_ AudioDecoder;
#endif
///
/// Audio decoder structure.
///
struct _audio_decoder_
{
AVCodec *AudioCodec; ///< audio codec
AVCodecContext *AudioCtx; ///< audio codec context
char Passthrough; ///< current pass-through flags
int SampleRate; ///< current stream sample rate
int Channels; ///< current stream channels
int HwSampleRate; ///< hw sample rate
int HwChannels; ///< hw channels
#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(56,28,1)
AVFrame *Frame; ///< decoded audio frame buffer
#endif
#if !defined(USE_SWRESAMPLE) && !defined(USE_AVRESAMPLE)
ReSampleContext *ReSample; ///< old resampling context
#endif
#ifdef USE_SWRESAMPLE
#if LIBSWRESAMPLE_VERSION_INT < AV_VERSION_INT(0, 15, 100)
struct SwrContext *Resample; ///< ffmpeg software resample context
#else
SwrContext *Resample; ///< ffmpeg software resample context
#endif
#endif
#ifdef USE_AVRESAMPLE
AVAudioResampleContext *Resample; ///< libav software resample context
#endif
uint16_t Spdif[24576 / 2]; ///< SPDIF output buffer
int SpdifIndex; ///< index into SPDIF output buffer
int SpdifCount; ///< SPDIF repeat counter
int64_t LastDelay; ///< last delay
struct timespec LastTime; ///< last time
int64_t LastPTS; ///< last PTS
int Drift; ///< accumulated audio drift
int DriftCorr; ///< audio drift correction value
int DriftFrac; ///< audio drift fraction for ac3
#if !defined(USE_SWRESAMPLE) && !defined(USE_AVRESAMPLE)
struct AVResampleContext *AvResample; ///< second audio resample context
#define MAX_CHANNELS 8 ///< max number of channels supported
int16_t *Buffer[MAX_CHANNELS]; ///< deinterleave sample buffers
int BufferSize; ///< size of sample buffer
int16_t *Remain[MAX_CHANNELS]; ///< filter remaining samples
int RemainSize; ///< size of remain buffer
int RemainCount; ///< number of remaining samples
#endif
};
///
/// IEC Data type enumeration.
///
enum IEC61937
{
IEC61937_AC3 = 0x01, ///< AC-3 data
// FIXME: more data types
IEC61937_EAC3 = 0x15, ///< E-AC-3 data
};
#ifdef USE_AUDIO_DRIFT_CORRECTION
#define CORRECT_PCM 1 ///< do PCM audio-drift correction
#define CORRECT_AC3 2 ///< do AC-3 audio-drift correction
static char CodecAudioDrift; ///< flag: enable audio-drift correction
#else
static const int CodecAudioDrift = 0;
#endif
#ifdef USE_PASSTHROUGH
///
/// Pass-through flags: CodecPCM, CodecAC3, CodecEAC3, ...
///
static char CodecPassthrough;
#else
static const int CodecPassthrough = 0;
#endif
static char CodecDownmix; ///< enable AC-3 decoder downmix
/**
** Allocate a new audio decoder context.
**
** @returns private decoder pointer for audio decoder.
*/
AudioDecoder *CodecAudioNewDecoder(void)
{
AudioDecoder *audio_decoder;
if (!(audio_decoder = calloc(1, sizeof(*audio_decoder)))) {
Fatal(_("codec: can't allocate audio decoder\n"));
}
#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(56,28,1)
if (!(audio_decoder->Frame = av_frame_alloc())) {
Fatal(_("codec: can't allocate audio decoder frame buffer\n"));
}
#endif
return audio_decoder;
}
/**
** Deallocate an audio decoder context.
**
** @param decoder private audio decoder
*/
void CodecAudioDelDecoder(AudioDecoder * decoder)
{
#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(56,28,1)
av_frame_free(&decoder->Frame); // callee does checks
#endif
free(decoder);
}
/**
** Open audio decoder.
**
** @param audio_decoder private audio decoder
** @param codec_id audio codec id
*/
void CodecAudioOpen(AudioDecoder * audio_decoder, int codec_id)
{
AVCodec *audio_codec;
Debug(3, "codec: using audio codec ID %#06x (%s)\n", codec_id,
avcodec_get_name(codec_id));
if (!(audio_codec = avcodec_find_decoder_by_name(avcodec_get_name(codec_id)))) {
// if (!(audio_codec = avcodec_find_decoder(codec_id))) {
Fatal(_("codec: codec ID %#06x not found\n"), codec_id);
// FIXME: errors aren't fatal
}
audio_decoder->AudioCodec = audio_codec;
if (!(audio_decoder->AudioCtx = avcodec_alloc_context3(audio_codec))) {
Fatal(_("codec: can't allocate audio codec context\n"));
}
if (CodecDownmix) {
#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(53,61,100)
audio_decoder->AudioCtx->request_channels = 2;
#endif
audio_decoder->AudioCtx->request_channel_layout =
AV_CH_LAYOUT_STEREO_DOWNMIX;
}
#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(53,61,100)
// this has no effect (with ffmpeg and libav)
// audio_decoder->AudioCtx->request_sample_fmt = AV_SAMPLE_FMT_S16;
#endif
pthread_mutex_lock(&CodecLockMutex);
// open codec
#if LIBAVCODEC_VERSION_INT <= AV_VERSION_INT(53,5,0)
if (avcodec_open(audio_decoder->AudioCtx, audio_codec) < 0) {
pthread_mutex_unlock(&CodecLockMutex);
Fatal(_("codec: can't open audio codec\n"));
}
#else
if (1) {
AVDictionary *av_dict;
av_dict = NULL;
// FIXME: import settings
//av_dict_set(&av_dict, "dmix_mode", "0", 0);
//av_dict_set(&av_dict, "ltrt_cmixlev", "1.414", 0);
//av_dict_set(&av_dict, "loro_cmixlev", "1.414", 0);
if (avcodec_open2(audio_decoder->AudioCtx, audio_codec, &av_dict) < 0) {
pthread_mutex_unlock(&CodecLockMutex);
Fatal(_("codec: can't open audio codec\n"));
}
av_dict_free(&av_dict);
}
#endif
pthread_mutex_unlock(&CodecLockMutex);
Debug(3, "codec: audio '%s'\n", audio_decoder->AudioCodec->long_name);
audio_decoder->SampleRate = 0;
audio_decoder->Channels = 0;
audio_decoder->HwSampleRate = 0;
audio_decoder->HwChannels = 0;
audio_decoder->LastDelay = 0;
}
/**
** Close audio decoder.
**
** @param audio_decoder private audio decoder
*/
void CodecAudioClose(AudioDecoder * audio_decoder)
{
// FIXME: output any buffered data
#if !defined(USE_SWRESAMPLE) && !defined(USE_AVRESAMPLE)
if (audio_decoder->AvResample) {
int ch;
av_resample_close(audio_decoder->AvResample);
audio_decoder->AvResample = NULL;
audio_decoder->RemainCount = 0;
audio_decoder->BufferSize = 0;
audio_decoder->RemainSize = 0;
for (ch = 0; ch < MAX_CHANNELS; ++ch) {
free(audio_decoder->Buffer[ch]);
audio_decoder->Buffer[ch] = NULL;
free(audio_decoder->Remain[ch]);
audio_decoder->Remain[ch] = NULL;
}
}
if (audio_decoder->ReSample) {
audio_resample_close(audio_decoder->ReSample);
audio_decoder->ReSample = NULL;
}
#endif
#ifdef USE_SWRESAMPLE
if (audio_decoder->Resample) {
swr_free(&audio_decoder->Resample);
}
#endif
#ifdef USE_AVRESAMPLE
if (audio_decoder->Resample) {
avresample_free(&audio_decoder->Resample);
}
#endif
if (audio_decoder->AudioCtx) {
pthread_mutex_lock(&CodecLockMutex);
avcodec_close(audio_decoder->AudioCtx);
av_freep(&audio_decoder->AudioCtx);
pthread_mutex_unlock(&CodecLockMutex);
}
}
/**
** Set audio drift correction.
**
** @param mask enable mask (PCM, AC-3)
*/
void CodecSetAudioDrift(int mask)
{
#ifdef USE_AUDIO_DRIFT_CORRECTION
CodecAudioDrift = mask & (CORRECT_PCM | CORRECT_AC3);
#endif
(void)mask;
}
/**
** Set audio pass-through.
**
** @param mask enable mask (PCM, AC-3, E-AC-3)
*/
void CodecSetAudioPassthrough(int mask)
{
#ifdef USE_PASSTHROUGH
CodecPassthrough = mask & (CodecPCM | CodecAC3 | CodecEAC3);
#endif
(void)mask;
}
/**
** Set audio downmix.
**
** @param onoff enable/disable downmix.
*/
void CodecSetAudioDownmix(int onoff)
{
if (onoff == -1) {
CodecDownmix ^= 1;
return;
}
CodecDownmix = onoff;
}
/**
** Reorder audio frame.
**
** ffmpeg L R C Ls Rs -> alsa L R Ls Rs C
** ffmpeg L R C LFE Ls Rs -> alsa L R Ls Rs C LFE
** ffmpeg L R C LFE Ls Rs Rl Rr -> alsa L R Ls Rs C LFE Rl Rr
**
** @param buf[IN,OUT] sample buffer
** @param size size of sample buffer in bytes
** @param channels number of channels interleaved in sample buffer
*/
static void CodecReorderAudioFrame(int16_t * buf, int size, int channels)
{
int i;
int c;
int ls;
int rs;
int lfe;
switch (channels) {
case 5:
size /= 2;
for (i = 0; i < size; i += 5) {
c = buf[i + 2];
ls = buf[i + 3];
rs = buf[i + 4];
buf[i + 2] = ls;
buf[i + 3] = rs;
buf[i + 4] = c;
}
break;
case 6:
size /= 2;
for (i = 0; i < size; i += 6) {
c = buf[i + 2];
lfe = buf[i + 3];
ls = buf[i + 4];
rs = buf[i + 5];
buf[i + 2] = ls;
buf[i + 3] = rs;
buf[i + 4] = c;
buf[i + 5] = lfe;
}
break;
case 8:
size /= 2;
for (i = 0; i < size; i += 8) {
c = buf[i + 2];
lfe = buf[i + 3];
ls = buf[i + 4];
rs = buf[i + 5];
buf[i + 2] = ls;
buf[i + 3] = rs;
buf[i + 4] = c;
buf[i + 5] = lfe;
}
break;
}
}
/**
** Handle audio format changes helper.
**
** @param audio_decoder audio decoder data
** @param[out] passthrough pass-through output
*/
static int CodecAudioUpdateHelper(AudioDecoder * audio_decoder,
int *passthrough)
{
const AVCodecContext *audio_ctx;
int err;
audio_ctx = audio_decoder->AudioCtx;
Debug(3, "codec/audio: format change %s %dHz *%d channels%s%s%s%s%s\n",
av_get_sample_fmt_name(audio_ctx->sample_fmt), audio_ctx->sample_rate,
audio_ctx->channels, CodecPassthrough & CodecPCM ? " PCM" : "",
CodecPassthrough & CodecMPA ? " MPA" : "",
CodecPassthrough & CodecAC3 ? " AC-3" : "",
CodecPassthrough & CodecEAC3 ? " E-AC-3" : "",
CodecPassthrough ? " pass-through" : "");
*passthrough = 0;
audio_decoder->SampleRate = audio_ctx->sample_rate;
audio_decoder->HwSampleRate = audio_ctx->sample_rate;
audio_decoder->Channels = audio_ctx->channels;
audio_decoder->HwChannels = audio_ctx->channels;
audio_decoder->Passthrough = CodecPassthrough;
// SPDIF/HDMI pass-through
if ((CodecPassthrough & CodecAC3 && audio_ctx->codec_id == AV_CODEC_ID_AC3)
|| (CodecPassthrough & CodecEAC3
&& audio_ctx->codec_id == AV_CODEC_ID_EAC3)) {
if (audio_ctx->codec_id == AV_CODEC_ID_EAC3) {
// E-AC-3 over HDMI some receivers need HBR
audio_decoder->HwSampleRate *= 4;
}
audio_decoder->HwChannels = 2;
audio_decoder->SpdifIndex = 0; // reset buffer
audio_decoder->SpdifCount = 0;
*passthrough = 1;
}
// channels/sample-rate not support?
if ((err =
AudioSetup(&audio_decoder->HwSampleRate,
&audio_decoder->HwChannels, *passthrough))) {
// try E-AC-3 none HBR
audio_decoder->HwSampleRate /= 4;
if (audio_ctx->codec_id != AV_CODEC_ID_EAC3
|| (err =
AudioSetup(&audio_decoder->HwSampleRate,
&audio_decoder->HwChannels, *passthrough))) {
Debug(3, "codec/audio: audio setup error\n");
// FIXME: handle errors
audio_decoder->HwChannels = 0;
audio_decoder->HwSampleRate = 0;
return err;
}
}
Debug(3, "codec/audio: resample %s %dHz *%d -> %s %dHz *%d\n",
av_get_sample_fmt_name(audio_ctx->sample_fmt), audio_ctx->sample_rate,
audio_ctx->channels, av_get_sample_fmt_name(AV_SAMPLE_FMT_S16),
audio_decoder->HwSampleRate, audio_decoder->HwChannels);
return 0;
}
/**
** Audio pass-through decoder helper.
**
** @param audio_decoder audio decoder data
** @param avpkt undecoded audio packet
*/
static int CodecAudioPassthroughHelper(AudioDecoder * audio_decoder,
const AVPacket * avpkt)
{
#ifdef USE_PASSTHROUGH
const AVCodecContext *audio_ctx;
audio_ctx = audio_decoder->AudioCtx;
// SPDIF/HDMI passthrough
if (CodecPassthrough & CodecAC3 && audio_ctx->codec_id == AV_CODEC_ID_AC3) {
uint16_t *spdif;
int spdif_sz;
spdif = audio_decoder->Spdif;
spdif_sz = 6144;
#ifdef USE_AC3_DRIFT_CORRECTION
// FIXME: this works with some TVs/AVReceivers
// FIXME: write burst size drift correction, which should work with all
if (CodecAudioDrift & CORRECT_AC3) {
int x;
x = (audio_decoder->DriftFrac +
(audio_decoder->DriftCorr * spdif_sz)) / (10 *
audio_decoder->HwSampleRate * 100);
audio_decoder->DriftFrac =
(audio_decoder->DriftFrac +
(audio_decoder->DriftCorr * spdif_sz)) % (10 *
audio_decoder->HwSampleRate * 100);
// round to word border
x *= audio_decoder->HwChannels * 4;
if (x < -64) { // limit correction
x = -64;
} else if (x > 64) {
x = 64;
}
spdif_sz += x;
}
#endif
// build SPDIF header and append A52 audio to it
// avpkt is the original data
if (spdif_sz < avpkt->size + 8) {
Error(_("codec/audio: decoded data smaller than encoded\n"));
return -1;
}
spdif[0] = htole16(0xF872); // iec 61937 sync word
spdif[1] = htole16(0x4E1F);
spdif[2] = htole16(IEC61937_AC3 | (avpkt->data[5] & 0x07) << 8);
spdif[3] = htole16(avpkt->size * 8);
// copy original data for output
// FIXME: not 100% sure, if endian is correct on not intel hardware
swab(avpkt->data, spdif + 4, avpkt->size);
// FIXME: don't need to clear always
memset(spdif + 4 + avpkt->size / 2, 0, spdif_sz - 8 - avpkt->size);
// don't play with the ac-3 samples
AudioEnqueue(spdif, spdif_sz);
return 1;
}
if (CodecPassthrough & CodecEAC3
&& audio_ctx->codec_id == AV_CODEC_ID_EAC3) {
uint16_t *spdif;
int spdif_sz;
int repeat;
// build SPDIF header and append A52 audio to it
// avpkt is the original data
spdif = audio_decoder->Spdif;
spdif_sz = 24576; // 4 * 6144
if (audio_decoder->HwSampleRate == 48000) {
spdif_sz = 6144;
}
if (spdif_sz < audio_decoder->SpdifIndex + avpkt->size + 8) {
Error(_("codec/audio: decoded data smaller than encoded\n"));
return -1;
}
// check if we must pack multiple packets
repeat = 1;
if ((avpkt->data[4] & 0xc0) != 0xc0) { // fscod
static const uint8_t eac3_repeat[4] = { 6, 3, 2, 1 };
// fscod2
repeat = eac3_repeat[(avpkt->data[4] & 0x30) >> 4];
}
// fprintf(stderr, "repeat %d %d\n", repeat, avpkt->size);
// copy original data for output
// pack upto repeat EAC-3 pakets into one IEC 61937 burst
// FIXME: not 100% sure, if endian is correct on not intel hardware
swab(avpkt->data, spdif + 4 + audio_decoder->SpdifIndex, avpkt->size);
audio_decoder->SpdifIndex += avpkt->size;
if (++audio_decoder->SpdifCount < repeat) {
return 1;
}
spdif[0] = htole16(0xF872); // iec 61937 sync word
spdif[1] = htole16(0x4E1F);
spdif[2] = htole16(IEC61937_EAC3);
spdif[3] = htole16(audio_decoder->SpdifIndex * 8);
memset(spdif + 4 + audio_decoder->SpdifIndex / 2, 0,
spdif_sz - 8 - audio_decoder->SpdifIndex);
// don't play with the eac-3 samples
AudioEnqueue(spdif, spdif_sz);
audio_decoder->SpdifIndex = 0;
audio_decoder->SpdifCount = 0;
return 1;
}
#endif
return 0;
}
#if !defined(USE_SWRESAMPLE) && !defined(USE_AVRESAMPLE)
/**
** Set/update audio pts clock.
**
** @param audio_decoder audio decoder data
** @param pts presentation timestamp
*/
static void CodecAudioSetClock(AudioDecoder * audio_decoder, int64_t pts)
{
struct timespec nowtime;
int64_t delay;
int64_t tim_diff;
int64_t pts_diff;
int drift;
int corr;
AudioSetClock(pts);
delay = AudioGetDelay();
if (!delay) {
return;
}
clock_gettime(CLOCK_MONOTONIC, &nowtime);
if (!audio_decoder->LastDelay) {
audio_decoder->LastTime = nowtime;
audio_decoder->LastPTS = pts;
audio_decoder->LastDelay = delay;
audio_decoder->Drift = 0;
audio_decoder->DriftFrac = 0;
Debug(3, "codec/audio: inital drift delay %" PRId64 "ms\n",
delay / 90);
return;
}
// collect over some time
pts_diff = pts - audio_decoder->LastPTS;
if (pts_diff < 10 * 1000 * 90) {
return;
}
tim_diff = (nowtime.tv_sec - audio_decoder->LastTime.tv_sec)
* 1000 * 1000 * 1000 + (nowtime.tv_nsec -
audio_decoder->LastTime.tv_nsec);
drift =
(tim_diff * 90) / (1000 * 1000) - pts_diff + delay -
audio_decoder->LastDelay;
// adjust rounding error
nowtime.tv_nsec -= nowtime.tv_nsec % (1000 * 1000 / 90);
audio_decoder->LastTime = nowtime;
audio_decoder->LastPTS = pts;
audio_decoder->LastDelay = delay;
if (0) {
Debug(3,
"codec/audio: interval P:%5" PRId64 "ms T:%5" PRId64 "ms D:%4"
PRId64 "ms %f %d\n", pts_diff / 90, tim_diff / (1000 * 1000),
delay / 90, drift / 90.0, audio_decoder->DriftCorr);
}
// underruns and av_resample have the same time :(((
if (abs(drift) > 10 * 90) {
// drift too big, pts changed?
Debug(3, "codec/audio: drift(%6d) %3dms reset\n",
audio_decoder->DriftCorr, drift / 90);
audio_decoder->LastDelay = 0;
#ifdef DEBUG
corr = 0; // keep gcc happy
#endif
} else {
drift += audio_decoder->Drift;
audio_decoder->Drift = drift;
corr = (10 * audio_decoder->HwSampleRate * drift) / (90 * 1000);
// SPDIF/HDMI passthrough
if ((CodecAudioDrift & CORRECT_AC3) && (!(CodecPassthrough & CodecAC3)
|| audio_decoder->AudioCtx->codec_id != AV_CODEC_ID_AC3)
&& (!(CodecPassthrough & CodecEAC3)
|| audio_decoder->AudioCtx->codec_id != AV_CODEC_ID_EAC3)) {
audio_decoder->DriftCorr = -corr;
}
if (audio_decoder->DriftCorr < -20000) { // limit correction
audio_decoder->DriftCorr = -20000;
} else if (audio_decoder->DriftCorr > 20000) {
audio_decoder->DriftCorr = 20000;
}
}
// FIXME: this works with libav 0.8, and only with >10ms with ffmpeg 0.10
if (audio_decoder->AvResample && audio_decoder->DriftCorr) {
int distance;
// try workaround for buggy ffmpeg 0.10
if (abs(audio_decoder->DriftCorr) < 2000) {
distance = (pts_diff * audio_decoder->HwSampleRate) / (900 * 1000);
} else {
distance = (pts_diff * audio_decoder->HwSampleRate) / (90 * 1000);
}
av_resample_compensate(audio_decoder->AvResample,
audio_decoder->DriftCorr / 10, distance);
}
if (1) {
static int c;
if (!(c++ % 10)) {
Debug(3, "codec/audio: drift(%6d) %8dus %5d\n",
audio_decoder->DriftCorr, drift * 1000 / 90, corr);
}
}
}
/**
** Handle audio format changes.
**
** @param audio_decoder audio decoder data
**
** @note this is the old not good supported version
*/
static void CodecAudioUpdateFormat(AudioDecoder * audio_decoder)
{
int passthrough;
const AVCodecContext *audio_ctx;
int err;
if (audio_decoder->ReSample) {
audio_resample_close(audio_decoder->ReSample);
audio_decoder->ReSample = NULL;
}
if (audio_decoder->AvResample) {
av_resample_close(audio_decoder->AvResample);
audio_decoder->AvResample = NULL;
audio_decoder->RemainCount = 0;
}
audio_ctx = audio_decoder->AudioCtx;
if ((err = CodecAudioUpdateHelper(audio_decoder, &passthrough))) {
Debug(3, "codec/audio: resample %dHz *%d -> %dHz *%d err %d\n",
audio_ctx->sample_rate, audio_ctx->channels,
audio_decoder->HwSampleRate, audio_decoder->HwChannels,err);
if (err == 1) {
audio_decoder->ReSample =
av_audio_resample_init(audio_decoder->HwChannels,
audio_ctx->channels, audio_decoder->HwSampleRate,
audio_ctx->sample_rate, audio_ctx->sample_fmt,
audio_ctx->sample_fmt, 16, 10, 0, 0.8);
// libav-0.8_pre didn't support 6 -> 2 channels
if (!audio_decoder->ReSample) {
Error(_("codec/audio: resample setup error\n"));
audio_decoder->HwChannels = 0;
audio_decoder->HwSampleRate = 0;
}
return;
}
Debug(3, "codec/audio: audio setup error\n");
// FIXME: handle errors
audio_decoder->HwChannels = 0;
audio_decoder->HwSampleRate = 0;
return;
}
if (passthrough) { // pass-through no conversion allowed
return;
}
// prepare audio drift resample
#ifdef USE_AUDIO_DRIFT_CORRECTION
if (CodecAudioDrift & CORRECT_PCM) {
if (audio_decoder->AvResample) {
Error(_("codec/audio: overwrite resample\n"));
}
audio_decoder->AvResample =
av_resample_init(audio_decoder->HwSampleRate,
audio_decoder->HwSampleRate, 16, 10, 0, 0.8);
if (!audio_decoder->AvResample) {
Error(_("codec/audio: AvResample setup error\n"));
} else {
// reset drift to some default value
audio_decoder->DriftCorr /= 2;
audio_decoder->DriftFrac = 0;
av_resample_compensate(audio_decoder->AvResample,
audio_decoder->DriftCorr / 10,
10 * audio_decoder->HwSampleRate);
}
}
#endif
}
/**
** Codec enqueue audio samples.
**
** @param audio_decoder audio decoder data
** @param data samples data
** @param count number of bytes in sample data
*/
void CodecAudioEnqueue(AudioDecoder * audio_decoder, int16_t * data, int count)
{
#ifdef USE_AUDIO_DRIFT_CORRECTION
if ((CodecAudioDrift & CORRECT_PCM) && audio_decoder->AvResample) {
int16_t buf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 4 +
FF_INPUT_BUFFER_PADDING_SIZE] __attribute__ ((aligned(16)));
int16_t buftmp[MAX_CHANNELS][(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 4];
int consumed;
int i;
int n;
int ch;
int bytes_n;
bytes_n = count / audio_decoder->HwChannels;
// resize sample buffer, if needed
if (audio_decoder->RemainCount + bytes_n > audio_decoder->BufferSize) {
audio_decoder->BufferSize = audio_decoder->RemainCount + bytes_n;
for (ch = 0; ch < MAX_CHANNELS; ++ch) {
audio_decoder->Buffer[ch] =
realloc(audio_decoder->Buffer[ch],
audio_decoder->BufferSize);
}
}
// copy remaining bytes into sample buffer
for (ch = 0; ch < audio_decoder->HwChannels; ++ch) {
memcpy(audio_decoder->Buffer[ch], audio_decoder->Remain[ch],
audio_decoder->RemainCount);
}
// deinterleave samples into sample buffer
for (i = 0; i < bytes_n / 2; i++) {
for (ch = 0; ch < audio_decoder->HwChannels; ++ch) {
audio_decoder->Buffer[ch][audio_decoder->RemainCount / 2 + i]
= data[i * audio_decoder->HwChannels + ch];
}
}
bytes_n += audio_decoder->RemainSize;
n = 0; // keep gcc lucky
// resample the sample buffer into tmp buffer
for (ch = 0; ch < audio_decoder->HwChannels; ++ch) {
n = av_resample(audio_decoder->AvResample, buftmp[ch],
audio_decoder->Buffer[ch], &consumed, bytes_n / 2,
sizeof(buftmp[ch]) / 2, ch == audio_decoder->HwChannels - 1);
// fixme remaining channels
if (bytes_n - consumed * 2 > audio_decoder->RemainSize) {
audio_decoder->RemainSize = bytes_n - consumed * 2;
}
audio_decoder->Remain[ch] =
realloc(audio_decoder->Remain[ch], audio_decoder->RemainSize);
memcpy(audio_decoder->Remain[ch],
audio_decoder->Buffer[ch] + consumed,
audio_decoder->RemainSize);
audio_decoder->RemainCount = audio_decoder->RemainSize;
}
// interleave samples from sample buffer
for (i = 0; i < n; i++) {
for (ch = 0; ch < audio_decoder->HwChannels; ++ch) {
buf[i * audio_decoder->HwChannels + ch] = buftmp[ch][i];
}
}
n *= 2;
n *= audio_decoder->HwChannels;
if (!(audio_decoder->Passthrough & CodecPCM)) {
CodecReorderAudioFrame(buf, n, audio_decoder->HwChannels);
}
AudioEnqueue(buf, n);
return;
}
#endif
if (!(audio_decoder->Passthrough & CodecPCM)) {
CodecReorderAudioFrame(data, count, audio_decoder->HwChannels);
}
AudioEnqueue(data, count);
}
int myavcodec_decode_audio3(AVCodecContext *avctx, int16_t *samples,
int *frame_size_ptr,
AVPacket *avpkt)
{
AVFrame *frame = av_frame_alloc();
int ret, got_frame = 0;
if (!frame)
return AVERROR(ENOMEM);
ret = avcodec_decode_audio4(avctx, frame, &got_frame, avpkt);
if (ret >= 0 && got_frame) {
int i,ch;
int planar = av_sample_fmt_is_planar(avctx->sample_fmt);
int data_size = av_get_bytes_per_sample(avctx->sample_fmt);
if (data_size < 0) {
/* This should not occur, checking just for paranoia */
fprintf(stderr, "Failed to calculate data size\n");
exit(1);
}
for (i=0; i<frame->nb_samples; i++) {
for (ch=0; ch < avctx->channels; ch++) {
memcpy(samples,frame->extended_data[ch]+data_size*i,data_size);
samples = (char *) samples + data_size;
}
}
//Debug(3,"data_size %d nb_samples %d sample_fmt %d channels %d planar %d\n",data_size,frame->nb_samples,avctx->sample_fmt,avctx->channels,planar);
*frame_size_ptr = data_size * avctx->channels * frame->nb_samples;
} else {
*frame_size_ptr = 0;
}
av_frame_free(&frame);
return ret;
}
/**
** Decode an audio packet.
**
** PTS must be handled self.
**
** @param audio_decoder audio decoder data
** @param avpkt audio packet
*/
void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
{
int16_t buf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 4 +
FF_INPUT_BUFFER_PADDING_SIZE] __attribute__ ((aligned(16)));
int buf_sz;
int l;
AVCodecContext *audio_ctx;
audio_ctx = audio_decoder->AudioCtx;
// FIXME: don't need to decode pass-through codecs
buf_sz = sizeof(buf);
l = myavcodec_decode_audio3(audio_ctx, buf, &buf_sz, (AVPacket *) avpkt);
if (avpkt->size != l) {
if (l == AVERROR(EAGAIN)) {
Error(_("codec: latm\n"));
return;
}
if (l < 0) { // no audio frame could be decompressed
Error(_("codec: error audio data\n"));
return;
}
Error(_("codec: error more than one frame data\n"));
}
// update audio clock
if (avpkt->pts != (int64_t) AV_NOPTS_VALUE) {
CodecAudioSetClock(audio_decoder, avpkt->pts);
}
// FIXME: must first play remainings bytes, than change and play new.
if (audio_decoder->Passthrough != CodecPassthrough
|| audio_decoder->SampleRate != audio_ctx->sample_rate
|| audio_decoder->Channels != audio_ctx->channels) {
CodecAudioUpdateFormat(audio_decoder);
}
if (audio_decoder->HwSampleRate && audio_decoder->HwChannels) {
// need to resample audio
if (audio_decoder->ReSample) {
int16_t outbuf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 4 +
FF_INPUT_BUFFER_PADDING_SIZE]
__attribute__ ((aligned(16)));
int outlen;
// FIXME: libav-0.7.2 crash here
outlen = audio_resample(audio_decoder->ReSample, outbuf, buf, buf_sz);
#ifdef DEBUG
if (outlen != buf_sz) {
Debug(3, "codec/audio: possible fixed ffmpeg\n");
}
#endif
if (outlen) {
// outlen seems to be wrong in ffmpeg-0.9
outlen /= audio_decoder->Channels *
av_get_bytes_per_sample(audio_ctx->sample_fmt);
outlen *=
audio_decoder->HwChannels *
av_get_bytes_per_sample(audio_ctx->sample_fmt);
Debug(4, "codec/audio: %d -> %d\n", buf_sz, outlen);
CodecAudioEnqueue(audio_decoder, outbuf, outlen);
}
} else {
if (CodecAudioPassthroughHelper(audio_decoder, avpkt)) {
return;
}
CodecAudioEnqueue(audio_decoder, buf, buf_sz);
}
}
}
#endif
#if defined(USE_SWRESAMPLE) || defined(USE_AVRESAMPLE)
/**
** Set/update audio pts clock.
**
** @param audio_decoder audio decoder data
** @param pts presentation timestamp
*/
static void CodecAudioSetClock(AudioDecoder * audio_decoder, int64_t pts)
{
#ifdef USE_AUDIO_DRIFT_CORRECTION
struct timespec nowtime;
int64_t delay;
int64_t tim_diff;
int64_t pts_diff;
int drift;
int corr;
AudioSetClock(pts);
delay = AudioGetDelay();
if (!delay) {
return;
}
clock_gettime(CLOCK_MONOTONIC, &nowtime);
if (!audio_decoder->LastDelay) {
audio_decoder->LastTime = nowtime;
audio_decoder->LastPTS = pts;
audio_decoder->LastDelay = delay;
audio_decoder->Drift = 0;
audio_decoder->DriftFrac = 0;
Debug(3, "codec/audio: inital drift delay %" PRId64 "ms\n",
delay / 90);
return;
}
// collect over some time
pts_diff = pts - audio_decoder->LastPTS;
if (pts_diff < 10 * 1000 * 90) {
return;
}
tim_diff = (nowtime.tv_sec - audio_decoder->LastTime.tv_sec)
* 1000 * 1000 * 1000 + (nowtime.tv_nsec -
audio_decoder->LastTime.tv_nsec);
drift =
(tim_diff * 90) / (1000 * 1000) - pts_diff + delay -
audio_decoder->LastDelay;
// adjust rounding error
nowtime.tv_nsec -= nowtime.tv_nsec % (1000 * 1000 / 90);
audio_decoder->LastTime = nowtime;
audio_decoder->LastPTS = pts;
audio_decoder->LastDelay = delay;
if (0) {
Debug(3,
"codec/audio: interval P:%5" PRId64 "ms T:%5" PRId64 "ms D:%4"
PRId64 "ms %f %d\n", pts_diff / 90, tim_diff / (1000 * 1000),
delay / 90, drift / 90.0, audio_decoder->DriftCorr);
}
// underruns and av_resample have the same time :(((
if (abs(drift) > 10 * 90) {
// drift too big, pts changed?
Debug(3, "codec/audio: drift(%6d) %3dms reset\n",
audio_decoder->DriftCorr, drift / 90);
audio_decoder->LastDelay = 0;
#ifdef DEBUG
corr = 0; // keep gcc happy
#endif
} else {
drift += audio_decoder->Drift;
audio_decoder->Drift = drift;
corr = (10 * audio_decoder->HwSampleRate * drift) / (90 * 1000);
// SPDIF/HDMI passthrough
if ((CodecAudioDrift & CORRECT_AC3) && (!(CodecPassthrough & CodecAC3)
|| audio_decoder->AudioCtx->codec_id != AV_CODEC_ID_AC3)
&& (!(CodecPassthrough & CodecEAC3)
|| audio_decoder->AudioCtx->codec_id != AV_CODEC_ID_EAC3)) {
audio_decoder->DriftCorr = -corr;
}
if (audio_decoder->DriftCorr < -20000) { // limit correction
audio_decoder->DriftCorr = -20000;
} else if (audio_decoder->DriftCorr > 20000) {
audio_decoder->DriftCorr = 20000;
}
}
#ifdef USE_SWRESAMPLE
if (audio_decoder->Resample && audio_decoder->DriftCorr) {
int distance;
// try workaround for buggy ffmpeg 0.10
if (abs(audio_decoder->DriftCorr) < 2000) {
distance = (pts_diff * audio_decoder->HwSampleRate) / (900 * 1000);
} else {
distance = (pts_diff * audio_decoder->HwSampleRate) / (90 * 1000);
}
if (swr_set_compensation(audio_decoder->Resample,
audio_decoder->DriftCorr / 10, distance)) {
Debug(3, "codec/audio: swr_set_compensation failed\n");
}
}
#endif
#ifdef USE_AVRESAMPLE
if (audio_decoder->Resample && audio_decoder->DriftCorr) {
int distance;
distance = (pts_diff * audio_decoder->HwSampleRate) / (900 * 1000);
if (avresample_set_compensation(audio_decoder->Resample,
audio_decoder->DriftCorr / 10, distance)) {
Debug(3, "codec/audio: swr_set_compensation failed\n");
}
}
#endif
if (1) {
static int c;
if (!(c++ % 10)) {
Debug(3, "codec/audio: drift(%6d) %8dus %5d\n",
audio_decoder->DriftCorr, drift * 1000 / 90, corr);
}
}
#else
AudioSetClock(pts);
#endif
}
/**
** Handle audio format changes.
**
** @param audio_decoder audio decoder data
*/
static void CodecAudioUpdateFormat(AudioDecoder * audio_decoder)
{
int passthrough;
const AVCodecContext *audio_ctx;
if (CodecAudioUpdateHelper(audio_decoder, &passthrough)) {
// FIXME: handle swresample format conversions.
return;
}
if (passthrough) { // pass-through no conversion allowed
return;
}
audio_ctx = audio_decoder->AudioCtx;
#ifdef DEBUG
if (audio_ctx->sample_fmt == AV_SAMPLE_FMT_S16
&& audio_ctx->sample_rate == audio_decoder->HwSampleRate
&& !CodecAudioDrift) {
// FIXME: use Resample only, when it is needed!
fprintf(stderr, "no resample needed\n");
}
#endif
#ifdef USE_SWRESAMPLE
audio_decoder->Resample =
swr_alloc_set_opts(audio_decoder->Resample, audio_ctx->channel_layout,
AV_SAMPLE_FMT_S16, audio_decoder->HwSampleRate,
audio_ctx->channel_layout, audio_ctx->sample_fmt,
audio_ctx->sample_rate, 0, NULL);
if (audio_decoder->Resample) {
swr_init(audio_decoder->Resample);
} else {
Error(_("codec/audio: can't setup resample\n"));
}
#endif
#ifdef USE_AVRESAMPLE
if (!(audio_decoder->Resample = avresample_alloc_context())) {
Error(_("codec/audio: can't setup resample\n"));
return;
}
av_opt_set_int(audio_decoder->Resample, "in_channel_layout",
audio_ctx->channel_layout, 0);
av_opt_set_int(audio_decoder->Resample, "in_sample_fmt",
audio_ctx->sample_fmt, 0);
av_opt_set_int(audio_decoder->Resample, "in_sample_rate",
audio_ctx->sample_rate, 0);
av_opt_set_int(audio_decoder->Resample, "out_channel_layout",
audio_ctx->channel_layout, 0);
av_opt_set_int(audio_decoder->Resample, "out_sample_fmt",
AV_SAMPLE_FMT_S16, 0);
av_opt_set_int(audio_decoder->Resample, "out_sample_rate",
audio_decoder->HwSampleRate, 0);
if (avresample_open(audio_decoder->Resample)) {
avresample_free(&audio_decoder->Resample);
audio_decoder->Resample = NULL;
Error(_("codec/audio: can't open resample\n"));
return;
}
#endif
}
/**
** Decode an audio packet.
**
** PTS must be handled self.
**
** @note the caller has not aligned avpkt and not cleared the end.
**
** @param audio_decoder audio decoder data
** @param avpkt audio packet
*/
void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
{
AVCodecContext *audio_ctx;
#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(56,28,1)
AVFrame frame[1];
#else
AVFrame *frame;
#endif
int got_frame;
int n;
audio_ctx = audio_decoder->AudioCtx;
// FIXME: don't need to decode pass-through codecs
// new AVFrame API
#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(56,28,1)
avcodec_get_frame_defaults(frame);
#else
frame = audio_decoder->Frame;
av_frame_unref(frame);
#endif
got_frame = 0;
n = avcodec_decode_audio4(audio_ctx, frame, &got_frame,
(AVPacket *) avpkt);
if (n != avpkt->size) {
if (n == AVERROR(EAGAIN)) {
Error(_("codec/audio: latm\n"));
return;
}
if (n < 0) { // no audio frame could be decompressed
Error(_("codec/audio: bad audio frame\n"));
return;
}
Error(_("codec/audio: error more than one frame data\n"));
}
if (!got_frame) {
Error(_("codec/audio: no frame\n"));
return;
}
// update audio clock
if (avpkt->pts != (int64_t) AV_NOPTS_VALUE) {
CodecAudioSetClock(audio_decoder, avpkt->pts);
}
// format change
if (audio_decoder->Passthrough != CodecPassthrough
|| audio_decoder->SampleRate != audio_ctx->sample_rate
|| audio_decoder->Channels != audio_ctx->channels) {
CodecAudioUpdateFormat(audio_decoder);
}
if (!audio_decoder->HwSampleRate || !audio_decoder->HwChannels) {
return; // unsupported sample format
}
if (CodecAudioPassthroughHelper(audio_decoder, avpkt)) {
return;
}
if (0) {
char strbuf[32];
int data_sz;
int plane_sz;
data_sz =
av_samples_get_buffer_size(&plane_sz, audio_ctx->channels,
frame->nb_samples, audio_ctx->sample_fmt, 1);
fprintf(stderr, "codec/audio: sample_fmt %s\n",
av_get_sample_fmt_name(audio_ctx->sample_fmt));
av_get_channel_layout_string(strbuf, 32, audio_ctx->channels,
audio_ctx->channel_layout);
fprintf(stderr, "codec/audio: layout %s\n", strbuf);
fprintf(stderr,
"codec/audio: channels %d samples %d plane %d data %d\n",
audio_ctx->channels, frame->nb_samples, plane_sz, data_sz);
}
#ifdef USE_SWRESAMPLE
if (audio_decoder->Resample) {
uint8_t outbuf[8192 * 2 * 8];
uint8_t *out[1];
out[0] = outbuf;
n = swr_convert(audio_decoder->Resample, out,
sizeof(outbuf) / (2 * audio_decoder->HwChannels),
(const uint8_t **)frame->extended_data, frame->nb_samples);
if (n > 0) {
if (!(audio_decoder->Passthrough & CodecPCM)) {
CodecReorderAudioFrame((int16_t *) outbuf,
n * 2 * audio_decoder->HwChannels,
audio_decoder->HwChannels);
}
AudioEnqueue(outbuf, n * 2 * audio_decoder->HwChannels);
}
return;
}
#endif
#ifdef USE_AVRESAMPLE
if (audio_decoder->Resample) {
uint8_t outbuf[8192 * 2 * 8];
uint8_t *out[1];
out[0] = outbuf;
n = avresample_convert(audio_decoder->Resample, out, 0,
sizeof(outbuf) / (2 * audio_decoder->HwChannels),
(uint8_t **) frame->extended_data, 0, frame->nb_samples);
// FIXME: set out_linesize, in_linesize correct
if (n > 0) {
if (!(audio_decoder->Passthrough & CodecPCM)) {
CodecReorderAudioFrame((int16_t *) outbuf,
n * 2 * audio_decoder->HwChannels,
audio_decoder->HwChannels);
}
AudioEnqueue(outbuf, n * 2 * audio_decoder->HwChannels);
}
return;
}
#endif
#ifdef DEBUG
// should be never reached
fprintf(stderr, "oops\n");
#endif
}
#endif
/**
** Flush the audio decoder.
**
** @param decoder audio decoder data
*/
void CodecAudioFlushBuffers(AudioDecoder * decoder)
{
avcodec_flush_buffers(decoder->AudioCtx);
}
//----------------------------------------------------------------------------
// Codec
//----------------------------------------------------------------------------
/**
** Empty log callback
*/
static void CodecNoopCallback( __attribute__ ((unused))
void *ptr, __attribute__ ((unused))
int level, __attribute__ ((unused))
const char *fmt, __attribute__ ((unused)) va_list vl)
{
}
/**
** Codec init
*/
void CodecInit(void)
{
pthread_mutex_init(&CodecLockMutex, NULL);
#ifndef DEBUG
// disable display ffmpeg error messages
av_log_set_callback(CodecNoopCallback);
#else
(void)CodecNoopCallback;
#endif
avcodec_register_all(); // register all formats and codecs
}
/**
** Codec exit.
*/
void CodecExit(void)
{
pthread_mutex_destroy(&CodecLockMutex);
}
Reference in New Issue View Git Blame Copy Permalink