mirror of
https://github.com/Swiftgram/Telegram-iOS.git
synced 2025-07-31 15:37:01 +00:00
Fixed opus encoding completion
This commit is contained in:
Ilya Laktyushin
parent
41c468c139
commit
b4541f7495
@ -81,13 +81,13 @@ typedef enum {
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NSDictionary *outputSettings = @
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{
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AVFormatIDKey: @(kAudioFormatLinearPCM),
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AVSampleRateKey: @(TGBridgeAudioEncoderSampleRate),
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AVNumberOfChannelsKey: @1,
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AVLinearPCMBitDepthKey: @16,
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AVLinearPCMIsFloatKey: @false,
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AVLinearPCMIsBigEndianKey: @false,
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AVLinearPCMIsNonInterleaved: @false
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AVFormatIDKey: @(kAudioFormatLinearPCM),
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AVSampleRateKey: @(TGBridgeAudioEncoderSampleRate),
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AVNumberOfChannelsKey: @1,
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AVLinearPCMBitDepthKey: @16,
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AVLinearPCMIsFloatKey: @false,
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AVLinearPCMIsBigEndianKey: @false,
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AVLinearPCMIsNonInterleaved: @false
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};
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_readerOutput = [AVAssetReaderAudioMixOutput assetReaderAudioMixOutputWithAudioTracks:asset.tracks audioSettings:outputSettings];
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@ -114,9 +114,9 @@ typedef enum {
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static ATQueue *queue = nil;
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static dispatch_once_t onceToken;
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dispatch_once(&onceToken, ^
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{
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queue = [[ATQueue alloc] initWithName:@"org.telegram.opusAudioEncoderQueue"];
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});
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{
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queue = [[ATQueue alloc] initWithName:@"org.telegram.opusAudioEncoderQueue"];
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});
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return queue;
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}
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@ -126,74 +126,74 @@ static const int encoderPacketSizeInBytes = TGBridgeAudioEncoderSampleRate / 100
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- (void)startWithCompletion:(void (^)(NSString *, int32_t))completion
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{
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[[TGBridgeAudioEncoder processingQueue] dispatch:^
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{
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_oggWriter = [[TGOggOpusWriter alloc] init];
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if (![_oggWriter beginWithDataItem:_tempFileItem])
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{
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[self cleanup];
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return;
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}
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[_assetReader startReading];
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while (_assetReader.status != AVAssetReaderStatusCompleted)
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{
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if (_assetReader.status == AVAssetReaderStatusReading)
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{
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CMSampleBufferRef nextBuffer = [_readerOutput copyNextSampleBuffer];
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if (nextBuffer)
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{
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AudioBufferList abl;
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CMBlockBufferRef blockBuffer;
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CMSampleBufferGetAudioBufferListWithRetainedBlockBuffer(nextBuffer, NULL, &abl, sizeof(abl), NULL, NULL, kCMSampleBufferFlag_AudioBufferList_Assure16ByteAlignment, &blockBuffer);
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[[TGBridgeAudioEncoder processingQueue] dispatch:^
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{
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[self _processBuffer:&abl.mBuffers[0]];
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CFRelease(nextBuffer);
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CFRelease(blockBuffer);
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}];
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}
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else
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{
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[[TGBridgeAudioEncoder processingQueue] dispatch:^
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{
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if (_tailLength > 0) {
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[_oggWriter writeFrame:(uint8_t *)_audioBuffer.bytes frameByteCount:(NSUInteger)_tailLength];
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}
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}];
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break;
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}
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}
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}
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[[TGBridgeAudioEncoder processingQueue] dispatch:^
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{
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TGFileDataItem *dataItemResult = nil;
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NSTimeInterval durationResult = 0.0;
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NSUInteger totalBytes = 0;
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if (_assetReader.status == AVAssetReaderStatusCompleted)
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{
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NSLog(@"finished");
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if (_oggWriter != nil)
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{
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dataItemResult = _tempFileItem;
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durationResult = [_oggWriter encodedDuration];
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totalBytes = [_oggWriter encodedBytes];
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}
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[self cleanup];
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}
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//TGLog(@"[TGBridgeAudioEncoder#%x convert time: %f ms]", self, (CFAbsoluteTimeGetCurrent() - startTime) * 1000.0);
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if (completion != nil)
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completion(dataItemResult.path, (int32_t)durationResult);
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}];
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}];
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{
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_oggWriter = [[TGOggOpusWriter alloc] init];
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if (![_oggWriter beginWithDataItem:_tempFileItem])
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{
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[self cleanup];
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return;
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}
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[_assetReader startReading];
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while (_assetReader.status != AVAssetReaderStatusCompleted)
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{
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if (_assetReader.status == AVAssetReaderStatusReading)
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{
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CMSampleBufferRef nextBuffer = [_readerOutput copyNextSampleBuffer];
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if (nextBuffer)
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{
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AudioBufferList abl;
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CMBlockBufferRef blockBuffer;
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CMSampleBufferGetAudioBufferListWithRetainedBlockBuffer(nextBuffer, NULL, &abl, sizeof(abl), NULL, NULL, kCMSampleBufferFlag_AudioBufferList_Assure16ByteAlignment, &blockBuffer);
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[[TGBridgeAudioEncoder processingQueue] dispatch:^
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{
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[self _processBuffer:&abl.mBuffers[0]];
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CFRelease(nextBuffer);
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CFRelease(blockBuffer);
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}];
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}
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else
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{
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[[TGBridgeAudioEncoder processingQueue] dispatch:^
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{
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if (_tailLength > 0) {
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[_oggWriter writeFrame:(uint8_t *)_audioBuffer.bytes frameByteCount:(NSUInteger)_tailLength];
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}
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}];
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break;
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}
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}
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}
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[[TGBridgeAudioEncoder processingQueue] dispatch:^
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{
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TGFileDataItem *dataItemResult = nil;
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NSTimeInterval durationResult = 0.0;
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NSUInteger totalBytes = 0;
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if (_assetReader.status == AVAssetReaderStatusCompleted)
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{
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NSLog(@"finished");
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if (_oggWriter != nil && [_oggWriter writeFrame:NULL frameByteCount:0])
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{
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dataItemResult = _tempFileItem;
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durationResult = [_oggWriter encodedDuration];
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totalBytes = [_oggWriter encodedBytes];
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}
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[self cleanup];
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}
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//TGLog(@"[TGBridgeAudioEncoder#%x convert time: %f ms]", self, (CFAbsoluteTimeGetCurrent() - startTime) * 1000.0);
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if (completion != nil)
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completion(dataItemResult.path, (int32_t)durationResult);
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}];
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}];
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}
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- (void)_processBuffer:(AudioBuffer const *)buffer
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@ -306,11 +306,11 @@ static const int encoderPacketSizeInBytes = TGBridgeAudioEncoderSampleRate / 100
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[_queue dispatch:^
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{
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_fileName = filePath;
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_length = [[[NSFileManager defaultManager] attributesOfItemAtPath:_fileName error:nil][NSFileSize] unsignedIntegerValue];
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_fileExists = [[NSFileManager defaultManager] fileExistsAtPath:_fileName];
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}];
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{
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_fileName = filePath;
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_length = [[[NSFileManager defaultManager] attributesOfItemAtPath:_fileName error:nil][NSFileSize] unsignedIntegerValue];
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_fileExists = [[NSFileManager defaultManager] fileExistsAtPath:_fileName];
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}];
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}
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return self;
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}
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@ -322,38 +322,38 @@ static const int encoderPacketSizeInBytes = TGBridgeAudioEncoderSampleRate / 100
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- (void)moveToPath:(NSString *)path
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{
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[_queue dispatch:^
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{
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[[NSFileManager defaultManager] moveItemAtPath:_fileName toPath:path error:nil];
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_fileName = path;
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}];
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{
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[[NSFileManager defaultManager] moveItemAtPath:_fileName toPath:path error:nil];
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_fileName = path;
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}];
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}
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- (void)remove
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{
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[_queue dispatch:^
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{
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[[NSFileManager defaultManager] removeItemAtPath:_fileName error:nil];
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}];
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{
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[[NSFileManager defaultManager] removeItemAtPath:_fileName error:nil];
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}];
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}
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- (void)appendData:(NSData *)data
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{
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[_queue dispatch:^
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{
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if (!_fileExists)
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{
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[[NSFileManager defaultManager] createFileAtPath:_fileName contents:nil attributes:nil];
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_fileExists = true;
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}
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NSFileHandle *file = [NSFileHandle fileHandleForUpdatingAtPath:_fileName];
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[file seekToEndOfFile];
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[file writeData:data];
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[file synchronizeFile];
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[file closeFile];
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_length += data.length;
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[_data appendData:data];
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}];
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{
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if (!_fileExists)
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{
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[[NSFileManager defaultManager] createFileAtPath:_fileName contents:nil attributes:nil];
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_fileExists = true;
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}
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NSFileHandle *file = [NSFileHandle fileHandleForUpdatingAtPath:_fileName];
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[file seekToEndOfFile];
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[file writeData:data];
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[file synchronizeFile];
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[file closeFile];
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_length += data.length;
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[_data appendData:data];
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}];
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}
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- (NSData *)readDataAtOffset:(NSUInteger)offset length:(NSUInteger)length
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@ -361,14 +361,14 @@ static const int encoderPacketSizeInBytes = TGBridgeAudioEncoderSampleRate / 100
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__block NSData *data = nil;
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[_queue dispatch:^
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{
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NSFileHandle *file = [NSFileHandle fileHandleForUpdatingAtPath:_fileName];
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[file seekToFileOffset:(unsigned long long)offset];
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data = [file readDataOfLength:length];
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if (data.length != length)
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//TGLog(@"Read data length mismatch");
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[file closeFile];
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} synchronous:true];
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{
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NSFileHandle *file = [NSFileHandle fileHandleForUpdatingAtPath:_fileName];
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[file seekToFileOffset:(unsigned long long)offset];
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data = [file readDataOfLength:length];
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if (data.length != length)
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//TGLog(@"Read data length mismatch");
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[file closeFile];
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} synchronous:true];
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return data;
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}
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@ -377,9 +377,9 @@ static const int encoderPacketSizeInBytes = TGBridgeAudioEncoderSampleRate / 100
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{
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__block NSUInteger result = 0;
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[_queue dispatch:^
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{
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result = _length;
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} synchronous:true];
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{
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result = _length;
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} synchronous:true];
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return result;
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}
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@ -420,11 +420,11 @@ static const int encoderPacketSizeInBytes = TGBridgeAudioEncoderSampleRate / 100
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static ATQueue *queue = nil;
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static dispatch_once_t onceToken;
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dispatch_once(&onceToken, ^
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{
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queue = [[ATQueue alloc] init];
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queue->_nativeQueue = dispatch_get_main_queue();
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queue->_isMainQueue = true;
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});
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{
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queue = [[ATQueue alloc] init];
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queue->_nativeQueue = dispatch_get_main_queue();
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queue->_isMainQueue = true;
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});
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return queue;
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}
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@ -434,9 +434,9 @@ static const int encoderPacketSizeInBytes = TGBridgeAudioEncoderSampleRate / 100
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static ATQueue *queue = nil;
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static dispatch_once_t onceToken;
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dispatch_once(&onceToken, ^
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{
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queue = [[ATQueue alloc] initWithNativeQueue:dispatch_get_global_queue(DISPATCH_QUEUE_PRIORITY_DEFAULT, 0)];
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});
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{
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queue = [[ATQueue alloc] initWithNativeQueue:dispatch_get_global_queue(DISPATCH_QUEUE_PRIORITY_DEFAULT, 0)];
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});
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return queue;
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}
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@ -446,9 +446,9 @@ static const int encoderPacketSizeInBytes = TGBridgeAudioEncoderSampleRate / 100
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static ATQueue *queue = nil;
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static dispatch_once_t onceToken;
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dispatch_once(&onceToken, ^
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{
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queue = [[ATQueue alloc] initWithNativeQueue:dispatch_get_global_queue(DISPATCH_QUEUE_PRIORITY_BACKGROUND, 0)];
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});
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{
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queue = [[ATQueue alloc] initWithNativeQueue:dispatch_get_global_queue(DISPATCH_QUEUE_PRIORITY_BACKGROUND, 0)];
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});
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return queue;
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}
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@ -319,7 +319,7 @@ final class ManagedAudioRecorderContext {
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self.idleTimerExtensionDisposable = (Signal<Void, NoError> { subscriber in
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return pushIdleTimerExtension()
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} |> delay(5.0, queue: queue)).start()
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} |> delay(5.0, queue: queue)).start()
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}
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deinit {
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@ -406,19 +406,19 @@ final class ManagedAudioRecorderContext {
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strongSelf.audioSessionAcquired(headset: state.isHeadsetConnected)
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}
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}
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}, deactivate: { [weak self] in
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return Signal { subscriber in
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queue.async {
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if let strongSelf = self {
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strongSelf.hasAudioSession = false
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strongSelf.stop()
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strongSelf.recordingState.set(.stopped)
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subscriber.putCompletion()
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}
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}, deactivate: { [weak self] in
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return Signal { subscriber in
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queue.async {
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if let strongSelf = self {
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strongSelf.hasAudioSession = false
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strongSelf.stop()
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strongSelf.recordingState.set(.stopped)
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subscriber.putCompletion()
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}
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return EmptyDisposable
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}
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return EmptyDisposable
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}
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})
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}
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}
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@ -592,53 +592,57 @@ final class ManagedAudioRecorderContext {
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}
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func takeData() -> RecordedAudioData? {
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var scaledSamplesMemory = malloc(100 * 2)!
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var scaledSamples: UnsafeMutablePointer<Int16> = scaledSamplesMemory.assumingMemoryBound(to: Int16.self)
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defer {
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free(scaledSamplesMemory)
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}
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memset(scaledSamples, 0, 100 * 2);
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var waveform: Data?
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let count = self.compressedWaveformSamples.count / 2
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self.compressedWaveformSamples.withUnsafeMutableBytes { (samples: UnsafeMutablePointer<Int16>) -> Void in
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for i in 0 ..< count {
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let sample = samples[i]
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let index = i * 100 / count
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if (scaledSamples[index] < sample) {
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scaledSamples[index] = sample;
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if self.oggWriter.writeFrame(nil, frameByteCount: 0) {
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var scaledSamplesMemory = malloc(100 * 2)!
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var scaledSamples: UnsafeMutablePointer<Int16> = scaledSamplesMemory.assumingMemoryBound(to: Int16.self)
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defer {
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free(scaledSamplesMemory)
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}
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memset(scaledSamples, 0, 100 * 2);
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var waveform: Data?
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let count = self.compressedWaveformSamples.count / 2
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self.compressedWaveformSamples.withUnsafeMutableBytes { (samples: UnsafeMutablePointer<Int16>) -> Void in
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for i in 0 ..< count {
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let sample = samples[i]
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let index = i * 100 / count
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if (scaledSamples[index] < sample) {
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scaledSamples[index] = sample;
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}
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}
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}
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var peak: Int16 = 0
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var sumSamples: Int64 = 0
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for i in 0 ..< 100 {
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let sample = scaledSamples[i]
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if peak < sample {
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peak = sample
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var peak: Int16 = 0
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var sumSamples: Int64 = 0
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for i in 0 ..< 100 {
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let sample = scaledSamples[i]
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if peak < sample {
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peak = sample
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}
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sumSamples += Int64(sample)
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}
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sumSamples += Int64(sample)
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}
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var calculatedPeak: UInt16 = 0
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calculatedPeak = UInt16((Double(sumSamples) * 1.8 / 100.0))
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if calculatedPeak < 2500 {
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calculatedPeak = 2500
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var calculatedPeak: UInt16 = 0
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calculatedPeak = UInt16((Double(sumSamples) * 1.8 / 100.0))
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if calculatedPeak < 2500 {
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calculatedPeak = 2500
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}
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for i in 0 ..< 100 {
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let sample: UInt16 = UInt16(Int64(scaledSamples[i]))
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let minPeak = min(Int64(sample), Int64(calculatedPeak))
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let resultPeak = minPeak * 31 / Int64(calculatedPeak)
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scaledSamples[i] = Int16(clamping: min(31, resultPeak))
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}
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let resultWaveform = AudioWaveform(samples: Data(bytes: scaledSamplesMemory, count: 100 * 2), peak: 31)
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let bitstream = resultWaveform.makeBitstream()
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waveform = AudioWaveform(bitstream: bitstream, bitsPerSample: 5).makeBitstream()
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}
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for i in 0 ..< 100 {
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let sample: UInt16 = UInt16(Int64(scaledSamples[i]))
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let minPeak = min(Int64(sample), Int64(calculatedPeak))
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let resultPeak = minPeak * 31 / Int64(calculatedPeak)
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scaledSamples[i] = Int16(clamping: min(31, resultPeak))
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}
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let resultWaveform = AudioWaveform(samples: Data(bytes: scaledSamplesMemory, count: 100 * 2), peak: 31)
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let bitstream = resultWaveform.makeBitstream()
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waveform = AudioWaveform(bitstream: bitstream, bitsPerSample: 5).makeBitstream()
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return RecordedAudioData(compressedData: self.dataItem.data(), duration: self.oggWriter.encodedDuration(), waveform: waveform)
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} else {
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return nil
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}
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return RecordedAudioData(compressedData: self.dataItem.data(), duration: self.oggWriter.encodedDuration(), waveform: waveform)
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}
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}
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@ -426,39 +426,24 @@ static inline int writeOggPage(ogg_page *page, TGDataItem *fileItem)
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pages_out++;
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}
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op.packet = (unsigned char *)_packet;
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op.bytes = nbBytes;
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op.b_o_s = 0;
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op.granulepos = enc_granulepos;
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if (op.e_o_s)
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{
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/* We compute the final GP as ceil(len*48k/input_rate). When a resampling
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decoder does the matching floor(len*input/48k) conversion the length will
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be exactly the same as the input.
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*/
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op.granulepos = ((total_samples * 48000 + rate - 1) / rate) + header.preskip;
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}
|
||||
op.packetno = 2 + _packetId;
|
||||
ogg_stream_packetin(&os, &op);
|
||||
last_segments += size_segments;
|
||||
if (framePcmBytes != NULL) {
|
||||
op.packet = (unsigned char *)_packet;
|
||||
op.bytes = nbBytes;
|
||||
op.b_o_s = 0;
|
||||
op.granulepos = enc_granulepos;
|
||||
|
||||
/* The downside of early reading is if the input is an exact
|
||||
multiple of the frame_size you'll get an extra frame that needs
|
||||
to get cropped off. The downside of late reading is added delay.
|
||||
If your ogg_delay is 120ms or less we'll assume you want the
|
||||
low delay behavior.
|
||||
*/
|
||||
/*if ((!op.e_o_s) && max_ogg_delay > 5760)
|
||||
{
|
||||
nb_samples = inopt.read_samples(inopt.readdata, input, frame_size);
|
||||
total_samples += nb_samples;
|
||||
if (nb_samples < frame_size)
|
||||
eos = 1;
|
||||
if (nb_samples == 0)
|
||||
op.e_o_s = 1;
|
||||
if (op.e_o_s)
|
||||
{
|
||||
/* We compute the final GP as ceil(len*48k/input_rate). When a resampling
|
||||
decoder does the matching floor(len*input/48k) conversion the length will
|
||||
be exactly the same as the input.
|
||||
*/
|
||||
op.granulepos = ((total_samples * 48000 + rate - 1) / rate) + header.preskip;
|
||||
}
|
||||
op.packetno = 2 + _packetId;
|
||||
ogg_stream_packetin(&os, &op);
|
||||
last_segments += size_segments;
|
||||
}
|
||||
else
|
||||
nb_samples = -1;*/
|
||||
|
||||
// If the stream is over or we're sure that the delayed flush will fire, go ahead and flush now to avoid adding delay
|
||||
while ((op.e_o_s || (enc_granulepos + (frame_size * 48000 / coding_rate) - last_granulepos > max_ogg_delay) ||
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user