Aegisub/FFmpegSource2/ffaudiosource.cpp
Fredrik Mellbin ff8d019d58 FFMS2 beta 4
This commit breaks the shit out of linux

Originally committed to SVN as r2571.
2008-12-30 23:57:22 +00:00

366 lines
11 KiB
C++

// Copyright (c) 2007-2008 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>
AudioBase::AudioBase() {
DecodingBuffer = new uint8_t[AVCODEC_MAX_AUDIO_FRAME_SIZE * 10];
};
AudioBase::~AudioBase() {
delete[] DecodingBuffer;
};
size_t AudioBase::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, FrameIndex *TrackIndices, char *ErrorMsg, unsigned MsgSize) {
FormatContext = NULL;
AVCodec *Codec = NULL;
AudioTrack = Track;
Frames = (*TrackIndices)[AudioTrack];
if (Frames.size() == 0) {
Free(false);
_snprintf(ErrorMsg, MsgSize, "Audio track contains no frames");
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
uint8_t DummyBuf[512];
if (GetAudio(DummyBuf, 0, 1, ErrorMsg, MsgSize)) {
Free(true);
throw ErrorMsg;
}
AP.BitsPerSample = av_get_bits_per_sample_format(CodecContext->sample_fmt);
AP.Channels = CodecContext->channels;;
AP.Float = AudioFMTIsFloat(CodecContext->sample_fmt);
AP.SampleRate = CodecContext->sample_rate;
AP.NumSamples = (Frames.back()).SampleStart;
if (AP.SampleRate <= 0 || AP.BitsPerSample <= 0) {
Free(true);
_snprintf(ErrorMsg, MsgSize, "Codec returned zero size audio");
throw ErrorMsg;
}
}
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;
while (av_read_frame(FormatContext, &Packet) >= 0) {
if (Packet.stream_index == AudioTrack) {
uint8_t *Data = Packet.data;
int Size = Packet.size;
while (Size > 0) {
int TempOutputBufSize = AVCODEC_MAX_AUDIO_FRAME_SIZE * 10;
Ret = avcodec_decode_audio2(CodecContext, (int16_t *)Buf, &TempOutputBufSize, Data, Size);
if (Ret < 0) {// throw error or something?
av_free_packet(&Packet);
goto Done;
}
if (Ret > 0) {
Size -= Ret;
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 size_t SizeConst = (av_get_bits_per_sample_format(CodecContext->sample_fmt) * CodecContext->channels) / 8;
size_t CurrentAudioBlock = FFMAX((int64_t)FindClosestAudioKeyFrame(Start) - 50, (int64_t)0);
memset(Buf, 0, SizeConst * Count);
AVPacket Packet;
avcodec_flush_buffers(CodecContext);
av_seek_frame(FormatContext, AudioTrack, Frames[CurrentAudioBlock].DTS, AVSEEK_FLAG_BACKWARD);
// 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);
}
uint8_t *DstBuf = (uint8_t *)Buf;
int64_t RemainingSamples = Count;
int64_t DecodeCount;
do {
int64_t DecodeStart = Frames[CurrentAudioBlock].SampleStart;
int Ret = DecodeNextAudioBlock(DecodingBuffer, &DecodeCount, ErrorMsg, MsgSize);
if (Ret < 0) {
// FIXME
//Env->ThrowError("Bleh, bad audio decoding");
}
CurrentAudioBlock++;
int64_t OffsetBytes = SizeConst * FFMAX(0, Start - DecodeStart);
int64_t CopyBytes = FFMAX(0, SizeConst * FFMIN(RemainingSamples, DecodeCount - FFMAX(0, Start - DecodeStart)));
memcpy(DstBuf, DecodingBuffer + OffsetBytes, CopyBytes);
DstBuf += CopyBytes;
if (SizeConst)
RemainingSamples -= CopyBytes / SizeConst;
} while (RemainingSamples > 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, FrameIndex *TrackIndices, char *ErrorMsg, unsigned MsgSize) {
CodecContext = NULL;
AVCodec *Codec = NULL;
TrackInfo *TI = NULL;
CS = NULL;
Frames = (*TrackIndices)[Track];
if (Frames.size() == 0) {
Free(false);
_snprintf(ErrorMsg, MsgSize, "Audio track contains no frames");
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
uint8_t DummyBuf[512];
if (GetAudio(DummyBuf, 0, 1, ErrorMsg, MsgSize)) {
Free(true);
throw ErrorMsg;
}
AP.BitsPerSample = av_get_bits_per_sample_format(CodecContext->sample_fmt);
AP.Channels = CodecContext->channels;;
AP.Float = AudioFMTIsFloat(CodecContext->sample_fmt);
AP.SampleRate = CodecContext->sample_rate;
AP.NumSamples = (Frames.back()).SampleStart;
if (AP.SampleRate <= 0 || AP.BitsPerSample <= 0) {
Free(true);
_snprintf(ErrorMsg, MsgSize, "Codec returned zero size audio");
throw ErrorMsg;
}
}
MatroskaAudioSource::~MatroskaAudioSource() {
Free(true);
}
int MatroskaAudioSource::GetAudio(void *Buf, int64_t Start, int64_t Count, char *ErrorMsg, unsigned MsgSize) {
const size_t SizeConst = (av_get_bits_per_sample_format(CodecContext->sample_fmt) * CodecContext->channels) / 8;
size_t CurrentAudioBlock = FFMAX((int64_t)FindClosestAudioKeyFrame(Start) - 10, (int64_t)0);
avcodec_flush_buffers(CodecContext);
memset(Buf, 0, SizeConst * Count);
uint8_t *DstBuf = (uint8_t *)Buf;
int64_t RemainingSamples = Count;
int64_t DecodeCount;
do {
int64_t DecodeStart = Frames[CurrentAudioBlock].SampleStart;
int Ret = DecodeNextAudioBlock(DecodingBuffer, &DecodeCount, Frames[CurrentAudioBlock].FilePos, Frames[CurrentAudioBlock].FrameSize, ErrorMsg, MsgSize);
if (Ret < 0) {
// FIXME
//Env->ThrowError("Bleh, bad audio decoding");
}
CurrentAudioBlock++;
int64_t OffsetBytes = SizeConst * FFMAX(0, Start - DecodeStart);
int64_t CopyBytes = FFMAX(0, SizeConst * FFMIN(RemainingSamples, DecodeCount - FFMAX(0, Start - DecodeStart)));
memcpy(DstBuf, DecodingBuffer + OffsetBytes, CopyBytes);
DstBuf += CopyBytes;
if (SizeConst)
RemainingSamples -= CopyBytes / SizeConst;
} while (RemainingSamples > 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;
// FIXME check return
ReadFrame(FilePos, FrameSize, CS, MC, ErrorMsg, MsgSize);
int Size = FrameSize;
uint8_t *Data = MC.Buffer;
while (Size > 0) {
int TempOutputBufSize = AVCODEC_MAX_AUDIO_FRAME_SIZE;
Ret = avcodec_decode_audio2(CodecContext, (int16_t *)Buf, &TempOutputBufSize, Data, Size);
if (Ret < 0) // throw error or something?
goto Done;
if (Ret > 0) {
Size -= Ret;
Data += Ret;
Buf += TempOutputBufSize;
if (SizeConst)
*Count += TempOutputBufSize / SizeConst;
}
}
Done:
return Ret;
}