Aegisub/aegisub/libffms/src/core/audiosource.cpp

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// Copyright (c) 2010 Thomas Goyne <tgoyne@gmail.com>
//
// 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 "audiosource.h"
#include <algorithm>
#include <cassert>
FFMS_AudioSource::FFMS_AudioSource(const char *SourceFile, FFMS_Index &Index, int Track)
: Delay(0)
, MaxCacheBlocks(50)
, BytesPerSample(0)
, Decoded(0)
, CurrentSample(-1)
, PacketNumber(0)
, CurrentFrame(NULL)
, TrackNumber(Track)
, SeekOffset(0)
, DecodingBuffer(AVCODEC_MAX_AUDIO_FRAME_SIZE * 10)
{
if (Track < 0 || Track >= static_cast<int>(Index.size()))
throw FFMS_Exception(FFMS_ERROR_INDEX, FFMS_ERROR_INVALID_ARGUMENT,
"Out of bounds track index selected");
if (Index[Track].TT != FFMS_TYPE_AUDIO)
throw FFMS_Exception(FFMS_ERROR_INDEX, FFMS_ERROR_INVALID_ARGUMENT,
"Not an audio track");
if (Index[Track].empty())
throw FFMS_Exception(FFMS_ERROR_INDEX, FFMS_ERROR_INVALID_ARGUMENT,
"Audio track contains no audio frames");
Frames = Index[Track];
if (!Index.CompareFileSignature(SourceFile))
throw FFMS_Exception(FFMS_ERROR_INDEX, FFMS_ERROR_FILE_MISMATCH,
"The index does not match the source file");
}
void FFMS_AudioSource::Init(FFMS_Index &Index, int DelayMode) {
// The first packet after a seek is often decoded incorrectly, which
// makes it impossible to ever correctly seek back to the beginning, so
// store the first block now
// In addition, anything with the same PTS as the first packet can't be
// distinguished from the first packet and so can't be seeked to, so
// store those as well
// Some of LAVF's splitters don't like to seek to the beginning of the
// file (ts and?), so cache a few blocks even if PTSes are unique
// Packet 7 is the last packet I've had be unseekable to, so cache up to
// 10 for a bit of an extra buffer
CacheIterator end = Cache.end();
while (PacketNumber < Frames.size() &&
((Frames[0].PTS != ffms_av_nopts_value && Frames[PacketNumber].PTS == Frames[0].PTS) ||
Cache.size() < 10)) {
DecodeNextBlock();
if (Decoded)
CacheBlock(end, CurrentSample, Decoded, &DecodingBuffer[0]);
}
// Store the iterator to the last element of the cache which is used for
// correctness rather than speed, so that when looking for one to delete
// we know how much to skip
CacheNoDelete = Cache.end();
--CacheNoDelete;
// Read properties of the audio which may not be available until the first
// frame has been decoded
FillAP(AP, CodecContext, Frames);
if (AP.SampleRate <= 0 || AP.BitsPerSample <= 0)
throw FFMS_Exception(FFMS_ERROR_DECODING, FFMS_ERROR_CODEC,
"Codec returned zero size audio");
if (DelayMode < FFMS_DELAY_NO_SHIFT)
throw FFMS_Exception(FFMS_ERROR_INDEX, FFMS_ERROR_INVALID_ARGUMENT,
"Bad audio delay compensation mode");
if (DelayMode == FFMS_DELAY_NO_SHIFT) return;
if (DelayMode > (signed)Index.size())
throw FFMS_Exception(FFMS_ERROR_INDEX, FFMS_ERROR_INVALID_ARGUMENT,
"Out of bounds track index selected for audio delay compensation");
if (DelayMode >= 0 && Index[DelayMode].TT != FFMS_TYPE_VIDEO)
throw FFMS_Exception(FFMS_ERROR_INDEX, FFMS_ERROR_INVALID_ARGUMENT,
"Audio delay compensation must be relative to a video track");
double AdjustRelative = 0;
if (DelayMode != FFMS_DELAY_TIME_ZERO) {
if (DelayMode == FFMS_DELAY_FIRST_VIDEO_TRACK) {
for (size_t i = 0; i < Index.size(); ++i) {
if (Index[i].TT == FFMS_TYPE_VIDEO) {
DelayMode = i;
break;
}
}
}
if (DelayMode >= 0) {
const FFMS_Track &VTrack = Index[DelayMode];
AdjustRelative = VTrack[0].PTS * VTrack.TB.Num / (double)VTrack.TB.Den / 1000.;
}
}
Delay = static_cast<int64_t>((AdjustRelative - Frames[0].PTS) * AP.SampleRate + .5);
AP.NumSamples -= Delay;
}
void FFMS_AudioSource::CacheBlock(CacheIterator &pos, int64_t Start, size_t Samples, uint8_t *SrcData) {
Cache.insert(pos, AudioBlock(Start, Samples, SrcData, Samples * BytesPerSample));
if (Cache.size() >= MaxCacheBlocks) {
// Kill the oldest one
CacheIterator min = CacheNoDelete;
// Never drop the first one as the first packet decoded after a seek
// is often decoded incorrectly and we can't seek to before the first one
++min;
for (CacheIterator it = min; it != Cache.end(); ++it)
if (it->Age < min->Age) min = it;
if (min == pos) ++pos;
Cache.erase(min);
}
}
void FFMS_AudioSource::DecodeNextBlock() {
if (BytesPerSample == 0) BytesPerSample = (av_get_bits_per_sample_fmt(CodecContext->sample_fmt) * CodecContext->channels) / 8;
CurrentFrame = &Frames[PacketNumber];
AVPacket Packet;
if (!ReadPacket(&Packet))
throw FFMS_Exception(FFMS_ERROR_PARSER, FFMS_ERROR_UNKNOWN, "ReadPacket unexpectedly failed to read a packet");
// ReadPacket may have changed the packet number
CurrentFrame = &Frames[PacketNumber];
CurrentSample = CurrentFrame->SampleStart;
++PacketNumber;
uint8_t *Buf = &DecodingBuffer[0];
uint8_t *Data = Packet.data;
while (Packet.size > 0) {
int TempOutputBufSize = AVCODEC_MAX_AUDIO_FRAME_SIZE * 10 - (Buf - &DecodingBuffer[0]);
int Ret = avcodec_decode_audio3(CodecContext, (int16_t *)Buf, &TempOutputBufSize, &Packet);
// Should only ever happen if the user chose to ignore decoding errors
// during indexing, so continue to just ignore decoding errors
if (Ret < 0) break;
if (Ret > 0) {
Packet.size -= Ret;
Packet.data += Ret;
Buf += TempOutputBufSize;
}
}
Packet.data = Data;
FreePacket(&Packet);
Decoded = (Buf - &DecodingBuffer[0]) / BytesPerSample;
if (Decoded == 0) {
// zero sample packets aren't included in the index so we didn't
// actually move to the next packet
--PacketNumber;
}
}
static bool SampleStartComp(const TFrameInfo &a, const TFrameInfo &b) {
return a.SampleStart < b.SampleStart;
}
void FFMS_AudioSource::GetAudio(void *Buf, int64_t Start, int64_t Count) {
if (Start < 0 || Start + Count > AP.NumSamples || Count < 0)
throw FFMS_Exception(FFMS_ERROR_DECODING, FFMS_ERROR_INVALID_ARGUMENT,
"Out of bounds audio samples requested");
uint8_t *Dst = static_cast<uint8_t*>(Buf);
// Apply audio delay (if any) and fill any samples before the start time with zero
Start -= Delay;
if (Start < 0) {
size_t Bytes = static_cast<size_t>(BytesPerSample * FFMIN(-Start, Count));
memset(Dst, 0, Bytes);
Count += Start;
// Entire request was before the start of the audio
if (Count <= 0) return;
Start = 0;
Dst += Bytes;
}
CacheIterator it = Cache.begin();
while (Count > 0) {
// Find first useful cache block
while (it != Cache.end() && it->Start + it->Samples <= Start) ++it;
// Cache has the next block we want
if (it != Cache.end() && it->Start <= Start) {
int64_t SrcOffset = FFMAX(0, Start - it->Start);
int64_t DstOffset = FFMAX(0, it->Start - Start);
int64_t CopySamples = FFMIN(it->Samples - SrcOffset, Count - DstOffset);
size_t Bytes = static_cast<size_t>(CopySamples * BytesPerSample);
memcpy(Dst + DstOffset * BytesPerSample, &it->Data[SrcOffset * BytesPerSample], Bytes);
Start += CopySamples;
Count -= CopySamples;
Dst += Bytes;
++it;
}
// Decode another block
else {
if (Start < CurrentSample && SeekOffset == -1)
throw FFMS_Exception(FFMS_ERROR_SEEKING, FFMS_ERROR_CODEC, "Audio stream is not seekable");
if (SeekOffset >= 0 && (Start < CurrentSample || Start > CurrentSample + Decoded * 5)) {
TFrameInfo f;
f.SampleStart = Start;
int NewPacketNumber = std::distance(Frames.begin(), std::lower_bound(Frames.begin(), Frames.end(), f, SampleStartComp));
NewPacketNumber = FFMAX(0, NewPacketNumber - SeekOffset - 15);
while (NewPacketNumber > 0 && !Frames[NewPacketNumber].KeyFrame) --NewPacketNumber;
// Only seek forward if it'll actually result in moving forward
if (Start < CurrentSample || static_cast<size_t>(NewPacketNumber) > PacketNumber) {
PacketNumber = NewPacketNumber;
Decoded = 0;
CurrentSample = -1;
avcodec_flush_buffers(CodecContext);
Seek();
}
}
// Decode everything between the last keyframe and the block we want
while (CurrentSample + Decoded <= Start) DecodeNextBlock();
if (CurrentSample > Start)
throw FFMS_Exception(FFMS_ERROR_SEEKING, FFMS_ERROR_CODEC, "Seeking is severely broken");
CacheBlock(it, CurrentSample, Decoded, &DecodingBuffer[0]);
size_t FirstSample = static_cast<size_t>(Start - CurrentSample);
size_t Samples = static_cast<size_t>(Decoded - FirstSample);
size_t Bytes = FFMIN(Samples, static_cast<size_t>(Count)) * BytesPerSample;
memcpy(Dst, &DecodingBuffer[FirstSample * BytesPerSample], Bytes);
Start += Samples;
Count -= Samples;
Dst += Bytes;
}
}
}
size_t GetSeekablePacketNumber(FFMS_Track const& Frames, size_t PacketNumber) {
// Packets don't always have unique PTSes, so we may not be able to
// uniquely identify the packet we want. This function attempts to find
// a PTS we can seek to which will let us figure out which packet we're
// on before we get to the packet we actually wanted
// MatroskaAudioSource doesn't need this, as it seeks by byte offset
// rather than PTS. LAVF theoretically can seek by byte offset, but we
// don't use it as not all demuxers support it and it's broken in some of
// those that claim to support it
// However much we might wish to, we can't seek to before packet zero
if (PacketNumber == 0) return PacketNumber;
// Desired packet's PTS is unique, so don't do anything
if (Frames[PacketNumber].PTS != Frames[PacketNumber - 1].PTS &&
(PacketNumber + 1 == Frames.size() || Frames[PacketNumber].PTS != Frames[PacketNumber + 1].PTS))
return PacketNumber;
// When decoding, we only reliably know what packet we're at when the
// newly parsed packet has a different PTS from the previous one. As such,
// we walk backwards until we hit a different PTS and then seek to there,
// so that we can then decode until we hit the PTS group we actually wanted
// (and thereby know that we're at the first packet in the group rather
// than whatever the splitter happened to choose)
// This doesn't work if our desired packet has the same PTS as the first
// packet, but this scenario should never come up anyway; we permanently
// cache the decoded results from those packets, so there's no need to ever
// seek to them
int64_t PTS = Frames[PacketNumber].PTS;
while (PacketNumber > 0 && PTS == Frames[PacketNumber].PTS)
--PacketNumber;
return PacketNumber;
}