forked from mia/Aegisub
ab8f6e6239
Rename lots of things, THIS BREAKS THE AEGISUB BUILD because of changed exported type and function names. Fixed an uninitialized memory bug that would make it crash on unindexed audio tracks in mastroska. Made ffms.h C-friendlier. Exports the start time of an audio track in the audio properties. Less signedness and type conversion warnings. Originally committed to SVN as r2940.
476 lines
14 KiB
C++
476 lines
14 KiB
C++
// Copyright (c) 2007-2009 Fredrik Mellbin
|
|
//
|
|
// Permission is hereby granted, free of charge, to any person obtaining a copy
|
|
// of this software and associated documentation files (the "Software"), to deal
|
|
// in the Software without restriction, including without limitation the rights
|
|
// to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
|
|
// copies of the Software, and to permit persons to whom the Software is
|
|
// furnished to do so, subject to the following conditions:
|
|
//
|
|
// The above copyright notice and this permission notice shall be included in
|
|
// all copies or substantial portions of the Software.
|
|
//
|
|
// THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
|
|
// IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
|
|
// FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
|
|
// AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
|
|
// LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
|
|
// OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
|
|
// THE SOFTWARE.
|
|
|
|
#include "ffaudiosource.h"
|
|
#include <errno.h>
|
|
|
|
#ifdef __UNIX__
|
|
#define _snprintf snprintf
|
|
#endif
|
|
|
|
TAudioBlock::TAudioBlock(int64_t Start, int64_t Samples, uint8_t *SrcData, size_t SrcBytes) {
|
|
this->Start = Start;
|
|
this->Samples = Samples;
|
|
Data = new uint8_t[SrcBytes];
|
|
memcpy(Data, SrcData, SrcBytes);
|
|
}
|
|
|
|
TAudioBlock::~TAudioBlock() {
|
|
delete[] Data;
|
|
}
|
|
|
|
TAudioCache::TAudioCache() {
|
|
MaxCacheBlocks = 0;
|
|
BytesPerSample = 0;
|
|
}
|
|
|
|
TAudioCache::~TAudioCache() {
|
|
for (TAudioCache::iterator it=begin(); it != end(); it++)
|
|
delete *it;
|
|
}
|
|
|
|
void TAudioCache::Initialize(int BytesPerSample, int MaxCacheBlocks) {
|
|
this->BytesPerSample = BytesPerSample;
|
|
this->MaxCacheBlocks = MaxCacheBlocks;
|
|
}
|
|
|
|
void TAudioCache::CacheBlock(int64_t Start, int64_t Samples, uint8_t *SrcData) {
|
|
if (BytesPerSample > 0) {
|
|
for (TAudioCache::iterator it=begin(); it != end(); it++) {
|
|
if ((*it)->Start == Start) {
|
|
delete *it;
|
|
erase(it);
|
|
break;
|
|
}
|
|
}
|
|
|
|
push_front(new TAudioBlock(Start, Samples, SrcData, Samples * BytesPerSample));
|
|
if (size() >= MaxCacheBlocks) {
|
|
delete back();
|
|
pop_back();
|
|
}
|
|
}
|
|
}
|
|
|
|
bool TAudioCache::AudioBlockComp(TAudioBlock *A, TAudioBlock *B) {
|
|
return A->Start < B->Start;
|
|
}
|
|
|
|
int64_t TAudioCache::FillRequest(int64_t Start, int64_t Samples, uint8_t *Dst) {
|
|
// May be better to move used blocks to the front
|
|
std::list<TAudioBlock *> UsedBlocks;
|
|
for (TAudioCache::iterator it=begin(); it != end(); it++) {
|
|
int64_t SrcOffset = FFMAX(0, Start - (*it)->Start);
|
|
int64_t DstOffset = FFMAX(0, (*it)->Start - Start);
|
|
int64_t CopySamples = FFMIN((*it)->Samples - SrcOffset, Samples - DstOffset);
|
|
if (CopySamples > 0) {
|
|
memcpy(Dst + DstOffset * BytesPerSample, (*it)->Data + SrcOffset * BytesPerSample, CopySamples * BytesPerSample);
|
|
UsedBlocks.push_back(*it);
|
|
}
|
|
}
|
|
UsedBlocks.sort(AudioBlockComp);
|
|
int64_t Ret = Start;
|
|
for (std::list<TAudioBlock *>::iterator it = UsedBlocks.begin(); it != UsedBlocks.end(); it++) {
|
|
if (it == UsedBlocks.begin() || Ret == (*it)->Start)
|
|
Ret = (*it)->Start + (*it)->Samples;
|
|
else
|
|
break;
|
|
}
|
|
return FFMIN(Ret, Start + Samples);
|
|
}
|
|
|
|
FFAudio::FFAudio() {
|
|
CurrentSample = 0;
|
|
DecodingBuffer = new uint8_t[AVCODEC_MAX_AUDIO_FRAME_SIZE * 10];
|
|
};
|
|
|
|
FFAudio::~FFAudio() {
|
|
delete[] DecodingBuffer;
|
|
};
|
|
|
|
size_t FFAudio::FindClosestAudioKeyFrame(int64_t Sample) {
|
|
for (size_t i = 0; i < Frames.size(); i++) {
|
|
if (Frames[i].SampleStart == Sample && Frames[i].KeyFrame)
|
|
return i;
|
|
else if (Frames[i].SampleStart > Sample && Frames[i].KeyFrame)
|
|
return i - 1;
|
|
}
|
|
return Frames.size() - 1;
|
|
}
|
|
|
|
void FFAudioSource::Free(bool CloseCodec) {
|
|
if (CloseCodec)
|
|
avcodec_close(CodecContext);
|
|
av_close_input_file(FormatContext);
|
|
}
|
|
|
|
FFAudioSource::FFAudioSource(const char *SourceFile, int Track, FFIndex *Index, char *ErrorMsg, unsigned MsgSize) {
|
|
FormatContext = NULL;
|
|
AVCodec *Codec = NULL;
|
|
AudioTrack = Track;
|
|
Frames = (*Index)[AudioTrack];
|
|
|
|
if (Frames.size() == 0) {
|
|
Free(false);
|
|
_snprintf(ErrorMsg, MsgSize, "Audio track contains no frames, was it indexed properly?");
|
|
throw ErrorMsg;
|
|
}
|
|
|
|
if (av_open_input_file(&FormatContext, SourceFile, NULL, 0, NULL) != 0) {
|
|
_snprintf(ErrorMsg, MsgSize, "Couldn't open '%s'", SourceFile);
|
|
throw ErrorMsg;
|
|
}
|
|
|
|
if (av_find_stream_info(FormatContext) < 0) {
|
|
Free(false);
|
|
_snprintf(ErrorMsg, MsgSize, "Couldn't find stream information");
|
|
throw ErrorMsg;
|
|
}
|
|
|
|
CodecContext = FormatContext->streams[AudioTrack]->codec;
|
|
|
|
Codec = avcodec_find_decoder(CodecContext->codec_id);
|
|
if (Codec == NULL) {
|
|
Free(false);
|
|
_snprintf(ErrorMsg, MsgSize, "Audio codec not found");
|
|
throw ErrorMsg;
|
|
}
|
|
|
|
if (avcodec_open(CodecContext, Codec) < 0) {
|
|
Free(false);
|
|
_snprintf(ErrorMsg, MsgSize, "Could not open audio codec");
|
|
throw ErrorMsg;
|
|
}
|
|
|
|
// Always try to decode a frame to make sure all required parameters are known
|
|
int64_t Dummy;
|
|
if (DecodeNextAudioBlock(DecodingBuffer, &Dummy, ErrorMsg, MsgSize) < 0) {
|
|
Free(true);
|
|
throw ErrorMsg;
|
|
}
|
|
av_seek_frame(FormatContext, AudioTrack, Frames[0].DTS, AVSEEK_FLAG_BACKWARD);
|
|
avcodec_flush_buffers(CodecContext);
|
|
|
|
FillAP(AP, CodecContext, Frames);
|
|
|
|
if (AP.SampleRate <= 0 || AP.BitsPerSample <= 0) {
|
|
Free(true);
|
|
_snprintf(ErrorMsg, MsgSize, "Codec returned zero size audio");
|
|
throw ErrorMsg;
|
|
}
|
|
|
|
AudioCache.Initialize((AP.Channels *AP.BitsPerSample) / 8, 50);
|
|
}
|
|
|
|
int FFAudioSource::DecodeNextAudioBlock(uint8_t *Buf, int64_t *Count, char *ErrorMsg, unsigned MsgSize) {
|
|
const size_t SizeConst = (av_get_bits_per_sample_format(CodecContext->sample_fmt) * CodecContext->channels) / 8;
|
|
int Ret = -1;
|
|
*Count = 0;
|
|
AVPacket Packet, TempPacket;
|
|
InitNullPacket(&Packet);
|
|
InitNullPacket(&TempPacket);
|
|
|
|
while (av_read_frame(FormatContext, &Packet) >= 0) {
|
|
if (Packet.stream_index == AudioTrack) {
|
|
TempPacket.data = Packet.data;
|
|
TempPacket.size = Packet.size;
|
|
|
|
while (TempPacket.size > 0) {
|
|
int TempOutputBufSize = AVCODEC_MAX_AUDIO_FRAME_SIZE * 10;
|
|
Ret = avcodec_decode_audio3(CodecContext, (int16_t *)Buf, &TempOutputBufSize, &TempPacket);
|
|
|
|
if (Ret < 0) {// throw error or something?
|
|
av_free_packet(&Packet);
|
|
goto Done;
|
|
}
|
|
|
|
if (Ret > 0) {
|
|
TempPacket.size -= Ret;
|
|
TempPacket.data += Ret;
|
|
Buf += TempOutputBufSize;
|
|
if (SizeConst)
|
|
*Count += TempOutputBufSize / SizeConst;
|
|
}
|
|
}
|
|
|
|
av_free_packet(&Packet);
|
|
goto Done;
|
|
}
|
|
|
|
av_free_packet(&Packet);
|
|
}
|
|
|
|
Done:
|
|
return Ret;
|
|
}
|
|
|
|
int FFAudioSource::GetAudio(void *Buf, int64_t Start, int64_t Count, char *ErrorMsg, unsigned MsgSize) {
|
|
const int64_t SizeConst = (av_get_bits_per_sample_format(CodecContext->sample_fmt) * CodecContext->channels) / 8;
|
|
memset(Buf, 0, SizeConst * Count);
|
|
|
|
int PreDecBlocks = 0;
|
|
uint8_t *DstBuf = static_cast<uint8_t *>(Buf);
|
|
|
|
// Fill with everything in the cache
|
|
int64_t CacheEnd = AudioCache.FillRequest(Start, Count, DstBuf);
|
|
// Was everything in the cache?
|
|
if (CacheEnd == Start + Count)
|
|
return 0;
|
|
|
|
size_t CurrentAudioBlock;
|
|
// Is seeking required to decode the requested samples?
|
|
// if (!(CurrentSample >= Start && CurrentSample <= CacheEnd)) {
|
|
if (CurrentSample != CacheEnd) {
|
|
PreDecBlocks = 15;
|
|
CurrentAudioBlock = FFMAX((int64_t)FindClosestAudioKeyFrame(CacheEnd) - PreDecBlocks - 20, (int64_t)0);
|
|
av_seek_frame(FormatContext, AudioTrack, Frames[CurrentAudioBlock].DTS, AVSEEK_FLAG_BACKWARD);
|
|
avcodec_flush_buffers(CodecContext);
|
|
|
|
AVPacket Packet;
|
|
InitNullPacket(&Packet);
|
|
|
|
// Establish where we actually are
|
|
// Trigger on packet dts difference since groups can otherwise be indistinguishable
|
|
int64_t LastDTS = - 1;
|
|
while (av_read_frame(FormatContext, &Packet) >= 0) {
|
|
if (Packet.stream_index == AudioTrack) {
|
|
if (LastDTS < 0) {
|
|
LastDTS = Packet.dts;
|
|
} else if (LastDTS != Packet.dts) {
|
|
for (size_t i = 0; i < Frames.size(); i++)
|
|
if (Frames[i].DTS == Packet.dts) {
|
|
// The current match was consumed
|
|
CurrentAudioBlock = i + 1;
|
|
break;
|
|
}
|
|
|
|
av_free_packet(&Packet);
|
|
break;
|
|
}
|
|
}
|
|
|
|
av_free_packet(&Packet);
|
|
}
|
|
} else {
|
|
CurrentAudioBlock = FindClosestAudioKeyFrame(CurrentSample);
|
|
}
|
|
|
|
int64_t DecodeCount;
|
|
|
|
do {
|
|
int Ret = DecodeNextAudioBlock(DecodingBuffer, &DecodeCount, ErrorMsg, MsgSize);
|
|
if (Ret < 0) {
|
|
// FIXME
|
|
//Env->ThrowError("Bleh, bad audio decoding");
|
|
}
|
|
|
|
// Cache the block if enough blocks before it have been decoded to avoid garbage
|
|
if (PreDecBlocks == 0) {
|
|
AudioCache.CacheBlock(Frames[CurrentAudioBlock].SampleStart, DecodeCount, DecodingBuffer);
|
|
CacheEnd = AudioCache.FillRequest(CacheEnd, Start + Count - CacheEnd, DstBuf + (CacheEnd - Start) * SizeConst);
|
|
} else {
|
|
PreDecBlocks--;
|
|
}
|
|
|
|
CurrentAudioBlock++;
|
|
if (CurrentAudioBlock < Frames.size())
|
|
CurrentSample = Frames[CurrentAudioBlock].SampleStart;
|
|
} while (Start + Count - CacheEnd > 0 && CurrentAudioBlock < Frames.size());
|
|
|
|
return 0;
|
|
}
|
|
|
|
FFAudioSource::~FFAudioSource() {
|
|
Free(true);
|
|
}
|
|
|
|
void MatroskaAudioSource::Free(bool CloseCodec) {
|
|
if (CS)
|
|
cs_Destroy(CS);
|
|
if (MC.ST.fp) {
|
|
mkv_Close(MF);
|
|
fclose(MC.ST.fp);
|
|
}
|
|
if (CloseCodec)
|
|
avcodec_close(CodecContext);
|
|
av_free(CodecContext);
|
|
}
|
|
|
|
MatroskaAudioSource::MatroskaAudioSource(const char *SourceFile, int Track, FFIndex *Index, char *ErrorMsg, unsigned MsgSize) {
|
|
CodecContext = NULL;
|
|
AVCodec *Codec = NULL;
|
|
TrackInfo *TI = NULL;
|
|
CS = NULL;
|
|
Frames = (*Index)[Track];
|
|
|
|
if (Frames.size() == 0) {
|
|
Free(false);
|
|
_snprintf(ErrorMsg, MsgSize, "Audio track contains no frames, was it indexed properly?");
|
|
throw ErrorMsg;
|
|
}
|
|
|
|
MC.ST.fp = fopen(SourceFile, "rb");
|
|
if (MC.ST.fp == NULL) {
|
|
_snprintf(ErrorMsg, MsgSize, "Can't open '%s': %s", SourceFile, strerror(errno));
|
|
throw ErrorMsg;
|
|
}
|
|
|
|
setvbuf(MC.ST.fp, NULL, _IOFBF, CACHESIZE);
|
|
|
|
MF = mkv_OpenEx(&MC.ST.base, 0, 0, ErrorMessage, sizeof(ErrorMessage));
|
|
if (MF == NULL) {
|
|
fclose(MC.ST.fp);
|
|
_snprintf(ErrorMsg, MsgSize, "Can't parse Matroska file: %s", ErrorMessage);
|
|
throw ErrorMsg;
|
|
}
|
|
|
|
mkv_SetTrackMask(MF, ~(1 << Track));
|
|
TI = mkv_GetTrackInfo(MF, Track);
|
|
|
|
if (TI->CompEnabled) {
|
|
CS = cs_Create(MF, Track, ErrorMessage, sizeof(ErrorMessage));
|
|
if (CS == NULL) {
|
|
Free(false);
|
|
_snprintf(ErrorMsg, MsgSize, "Can't create decompressor: %s", ErrorMessage);
|
|
throw ErrorMsg;
|
|
}
|
|
}
|
|
|
|
CodecContext = avcodec_alloc_context();
|
|
CodecContext->extradata = (uint8_t *)TI->CodecPrivate;
|
|
CodecContext->extradata_size = TI->CodecPrivateSize;
|
|
|
|
Codec = avcodec_find_decoder(MatroskaToFFCodecID(TI->CodecID, TI->CodecPrivate));
|
|
if (Codec == NULL) {
|
|
Free(false);
|
|
_snprintf(ErrorMsg, MsgSize, "Video codec not found");
|
|
throw ErrorMsg;
|
|
}
|
|
|
|
if (avcodec_open(CodecContext, Codec) < 0) {
|
|
Free(false);
|
|
_snprintf(ErrorMsg, MsgSize, "Could not open video codec");
|
|
throw ErrorMsg;
|
|
}
|
|
|
|
// Always try to decode a frame to make sure all required parameters are known
|
|
int64_t Dummy;
|
|
if (DecodeNextAudioBlock(DecodingBuffer, &Dummy, Frames[0].FilePos, Frames[0].FrameSize, ErrorMsg, MsgSize) < 0) {
|
|
Free(true);
|
|
throw ErrorMsg;
|
|
}
|
|
avcodec_flush_buffers(CodecContext);
|
|
|
|
FillAP(AP, CodecContext, Frames);
|
|
|
|
if (AP.SampleRate <= 0 || AP.BitsPerSample <= 0) {
|
|
Free(true);
|
|
_snprintf(ErrorMsg, MsgSize, "Codec returned zero size audio");
|
|
throw ErrorMsg;
|
|
}
|
|
|
|
AudioCache.Initialize((AP.Channels *AP.BitsPerSample) / 8, 50);
|
|
}
|
|
|
|
MatroskaAudioSource::~MatroskaAudioSource() {
|
|
Free(true);
|
|
}
|
|
|
|
int MatroskaAudioSource::GetAudio(void *Buf, int64_t Start, int64_t Count, char *ErrorMsg, unsigned MsgSize) {
|
|
const int64_t SizeConst = (av_get_bits_per_sample_format(CodecContext->sample_fmt) * CodecContext->channels) / 8;
|
|
memset(Buf, 0, SizeConst * Count);
|
|
|
|
int PreDecBlocks = 0;
|
|
uint8_t *DstBuf = static_cast<uint8_t *>(Buf);
|
|
|
|
// Fill with everything in the cache
|
|
int64_t CacheEnd = AudioCache.FillRequest(Start, Count, DstBuf);
|
|
// Was everything in the cache?
|
|
if (CacheEnd == Start + Count)
|
|
return 0;
|
|
|
|
size_t CurrentAudioBlock;
|
|
// Is seeking required to decode the requested samples?
|
|
// if (!(CurrentSample >= Start && CurrentSample <= CacheEnd)) {
|
|
if (CurrentSample != CacheEnd) {
|
|
PreDecBlocks = 15;
|
|
CurrentAudioBlock = FFMAX((int64_t)FindClosestAudioKeyFrame(CacheEnd) - PreDecBlocks, (int64_t)0);
|
|
avcodec_flush_buffers(CodecContext);
|
|
} else {
|
|
CurrentAudioBlock = FindClosestAudioKeyFrame(CurrentSample);
|
|
}
|
|
|
|
int64_t DecodeCount;
|
|
|
|
do {
|
|
int Ret = DecodeNextAudioBlock(DecodingBuffer, &DecodeCount, Frames[CurrentAudioBlock].FilePos, Frames[CurrentAudioBlock].FrameSize, ErrorMsg, MsgSize);
|
|
if (Ret < 0) {
|
|
// FIXME
|
|
//Env->ThrowError("Bleh, bad audio decoding");
|
|
}
|
|
|
|
// Cache the block if enough blocks before it have been decoded to avoid garbage
|
|
if (PreDecBlocks == 0) {
|
|
AudioCache.CacheBlock(Frames[CurrentAudioBlock].SampleStart, DecodeCount, DecodingBuffer);
|
|
CacheEnd = AudioCache.FillRequest(CacheEnd, Start + Count - CacheEnd, DstBuf + (CacheEnd - Start) * SizeConst);
|
|
} else {
|
|
PreDecBlocks--;
|
|
}
|
|
|
|
CurrentAudioBlock++;
|
|
if (CurrentAudioBlock < Frames.size())
|
|
CurrentSample = Frames[CurrentAudioBlock].SampleStart;
|
|
} while (Start + Count - CacheEnd > 0 && CurrentAudioBlock < Frames.size());
|
|
|
|
return 0;
|
|
}
|
|
|
|
int MatroskaAudioSource::DecodeNextAudioBlock(uint8_t *Buf, int64_t *Count, uint64_t FilePos, unsigned int FrameSize, char *ErrorMsg, unsigned MsgSize) {
|
|
const size_t SizeConst = (av_get_bits_per_sample_format(CodecContext->sample_fmt) * CodecContext->channels) / 8;
|
|
int Ret = -1;
|
|
*Count = 0;
|
|
AVPacket TempPacket;
|
|
InitNullPacket(&TempPacket);
|
|
|
|
// FIXME check return
|
|
ReadFrame(FilePos, FrameSize, CS, MC, ErrorMsg, MsgSize);
|
|
TempPacket.size = FrameSize;
|
|
TempPacket.data = MC.Buffer;
|
|
|
|
while (TempPacket.size > 0) {
|
|
int TempOutputBufSize = AVCODEC_MAX_AUDIO_FRAME_SIZE;
|
|
Ret = avcodec_decode_audio3(CodecContext, (int16_t *)Buf, &TempOutputBufSize, &TempPacket);
|
|
|
|
if (Ret < 0) // throw error or something?
|
|
goto Done;
|
|
|
|
if (Ret > 0) {
|
|
TempPacket.size -= Ret;
|
|
TempPacket.data += Ret;
|
|
Buf += TempOutputBufSize;
|
|
if (SizeConst)
|
|
*Count += TempOutputBufSize / SizeConst;
|
|
}
|
|
}
|
|
|
|
Done:
|
|
return Ret;
|
|
}
|