Revert "AVCodecDecoder: use swresample to interleave audio channels."

Commit 856cc59e58 didn't really "fix"
anything; it just broke audio pretty much everywhere but YouTube,
and there videos play at 2x speed so it wasn't really worth it.

Stopgap solution for #12509.

src/add-ons/media/plugins/ffmpeg/AVCodecDecoder.cpp

@@ -5,14 +5,12 @@
  * Copyright (C) 2004 Marcus Overhagen
  * Copyright (C) 2009 Stephan Amßus <[email protected]>
  * Copyright (C) 2014 Colin Günther <[email protected]>
- * Copyright (C) 2015 Adrien Destugues <[email protected]>
  *
  * All rights reserved. Distributed under the terms of the MIT License.
  */
 
 //! libavcodec based decoder for Haiku
 
-
 #include "AVCodecDecoder.h"
 
 #include <new>
@@ -89,7 +87,6 @@ AVCodecDecoder::AVCodecDecoder()
 	fIsAudio(false),
 	fCodec(NULL),
 	fContext(avcodec_alloc_context3(NULL)),
-	fResampleContext(NULL),
 	fDecodedData(NULL),
 	fDecodedDataSizeInBytes(0),
 	fPostProcessedDecodedPicture(avcodec_alloc_frame()),
@@ -118,6 +115,7 @@ AVCodecDecoder::AVCodecDecoder()
 	fAudioDecodeError(false),
 
 	fDecodedDataBuffer(avcodec_alloc_frame()),
+	fDecodedDataBufferOffset(0),
 	fDecodedDataBufferSize(0)
 {
 	TRACE("AVCodecDecoder::AVCodecDecoder()\n");
@@ -146,7 +144,6 @@ AVCodecDecoder::~AVCodecDecoder()
 	if (fCodecInitDone)
 		avcodec_close(fContext);
 
-	swr_free(&fResampleContext);
 	free(fChunkBuffer);
 	free(fDecodedData);
 
@@ -232,7 +229,8 @@ AVCodecDecoder::Setup(media_format* ioEncodedFormat, const void* infoBuffer,
 		} else {
 			if (fIsAudio) {
 				fBlockAlign
-					= ioEncodedFormat->u.encoded_audio.output.buffer_size;
+					= ioEncodedFormat->u.encoded_audio.output
+						.buffer_size;
 				TRACE("  using buffer_size as block align: %d\n",
 					fBlockAlign);
 			}
@@ -272,6 +270,7 @@ AVCodecDecoder::SeekedTo(int64 frame, bigtime_t time)
 	free(fChunkBuffer);
 	fChunkBuffer = NULL;
 	fChunkBufferSize = 0;
+	fDecodedDataBufferOffset = 0;
 	fDecodedDataBufferSize = 0;
 	fDecodedDataSizeInBytes = 0;
 
@@ -359,6 +358,7 @@ AVCodecDecoder::_NegotiateAudioOutputFormat(media_format* inOutFormat)
 	fChunkBuffer = NULL;
 	fChunkBufferSize = 0;
 	fAudioDecodeError = false;
+	fDecodedDataBufferOffset = 0;
 	fDecodedDataBufferSize = 0;
 
 	_ResetTempPacket();
@@ -405,13 +405,6 @@ AVCodecDecoder::_NegotiateAudioOutputFormat(media_format* inOutFormat)
 	if (fRawDecodedAudio->opaque == NULL)
 		return B_NO_MEMORY;
 
-	fResampleContext = swr_alloc_set_opts(NULL,
-		fContext->channel_layout, fContext->request_sample_fmt,
-		fContext->sample_rate,
-		fContext->channel_layout, fContext->sample_fmt, fContext->sample_rate,
-		0, NULL);
-	swr_init(fResampleContext);
-
 	TRACE("  bit_rate = %d, sample_rate = %d, channels = %d, "
 		"output frame size: %d, count: %ld, rate: %.2f\n",
 		fContext->bit_rate, fContext->sample_rate, fContext->channels,
@@ -901,20 +894,24 @@ AVCodecDecoder::_MoveAudioFramesToRawDecodedAudioAndUpdateStartTimes()
 	assert(fRawDecodedAudio->nb_samples < fOutputFrameCount);
 	assert(fOutputFrameRate > 0);
 
-	// Some decoders do not support format conversion on themselves, or use
+	int32 frames = min_c(fOutputFrameCount - fRawDecodedAudio->nb_samples,
-	// "planar" audio (each channel separated instead of interleaved samples).
+		fDecodedDataBufferSize / fOutputFrameSize);
-	// In that case, we use swresample to convert the data (and it is
+	if (frames == 0)
-	// smart enough to do just a copy, when possible)
+		debugger("fDecodedDataBufferSize not multiple of frame size!");
-	int32 frames = swr_convert(fResampleContext, fRawDecodedAudio->data,
+
-		fOutputFrameCount - fRawDecodedAudio->nb_samples,
-		(const uint8_t**)fDecodedDataBuffer->data,
-		fDecodedDataBuffer->nb_samples);
-	if (frames < 0)
-		debugger("resampling failed");
 	size_t remainingSize = frames * fOutputFrameSize;
 
-	// libswresample handles all the buffering for us, how nice of them!
+#if 0
-	fDecodedDataBufferSize = 0;
+	// Some decoders do not support format conversion on themselves, or use
+	// "planar" audio (each channel separated instead of interleaved samples).
+	// If this is a problem we will need to use swresample to convert the data
+	// here, instead of directly copying it.
+	swr_convert(fResampleContext, fRawDecodedAudio->data,
+		fDecodedDataBuffer->data + fDecodedDataBufferOffset, frames);
+#else
+	memcpy(fRawDecodedAudio->data[0], fDecodedDataBuffer->data[0]
+		+ fDecodedDataBufferOffset, remainingSize);
+#endif
 
 	bool firstAudioFramesCopiedToRawDecodedAudio
 		= fRawDecodedAudio->data[0] != fDecodedData;
@@ -931,6 +928,19 @@ AVCodecDecoder::_MoveAudioFramesToRawDecodedAudioAndUpdateStartTimes()
 	fRawDecodedAudio->data[0] += remainingSize;
 	fRawDecodedAudio->linesize[0] += remainingSize;
 	fRawDecodedAudio->nb_samples += frames;
+
+	fDecodedDataBufferOffset += remainingSize;
+	fDecodedDataBufferSize -= remainingSize;
+
+	// Update start times accordingly
+	bigtime_t framesTimeInterval = static_cast<bigtime_t>(
+		(1000000LL * frames) / fOutputFrameRate);
+	fDecodedDataBuffer->pkt_dts += framesTimeInterval;
+		// Start time of buffer is updated in case that it contains
+		// more audio frames to move.
+	fTempPacket.dts += framesTimeInterval;
+		// Start time of fTempPacket is updated in case the fTempPacket
+		// contains more audio frames to decode.
 }
 
 
@@ -952,7 +962,8 @@ AVCodecDecoder::_MoveAudioFramesToRawDecodedAudioAndUpdateStartTimes()
 	After this function returns successfully the caller can safely make the
 	following assumptions:
 		1. fDecodedDataBufferSize is greater than zero.
-		2. fDecodedDataBuffer contains audio frames.
+		2. fDecodedDataBufferOffset is set to zero.
+		3. fDecodedDataBuffer contains audio frames.
 
 	\returns B_OK on successfully decoding one audio frame chunk.
 	\returns B_LAST_BUFFER_ERROR No more audio frame chunks available. From
@@ -964,7 +975,7 @@ AVCodecDecoder::_DecodeNextAudioFrameChunk()
 {
 	assert(fDecodedDataBufferSize == 0);
 
-	while (fDecodedDataBufferSize == 0) {
+	while(fDecodedDataBufferSize == 0) {
 		status_t loadingChunkStatus
 			= _LoadNextChunkIfNeededAndAssignStartTime();
 		if (loadingChunkStatus != B_OK)
@@ -1011,6 +1022,7 @@ AVCodecDecoder::_DecodeNextAudioFrameChunk()
 		   Also see "Note" below.
 		2. fTempPacket was updated to exclude the data chunk that was consumed
 		   by avcodec_decode_audio4().
+		3. fDecodedDataBufferOffset is set to zero.
 
 	When this function failed to decode at least one audio frame due to a
 	decoding error the caller can safely make the following assumptions:
@@ -1037,6 +1049,7 @@ AVCodecDecoder::_DecodeSomeAudioFramesIntoEmptyDecodedDataBuffer()
 	assert(fTempPacket.size > 0);
 
 	avcodec_get_frame_defaults(fDecodedDataBuffer);
+	fDecodedDataBufferOffset = 0;
 	int gotAudioFrame = 0;
 
 	int encodedDataSizeInBytes = avcodec_decode_audio4(fContext,

src/add-ons/media/plugins/ffmpeg/AVCodecDecoder.h

@@ -4,7 +4,6 @@
  * Copyright (C) 2001 Axel Dörfler.
  * Copyright (C) 2004 Marcus Overhagen.
  * Copyright (C) 2009 Stephan Aßmus <[email protected]>.
- * Copyright (C) 2015 Adrien Destugues <[email protected]>.
  *
  * All rights reserved. Distributed under the terms of the MIT License.
  */
@@ -13,17 +12,13 @@
 
 //! libavcodec based decoder for Haiku
 
-
 #include <MediaFormats.h>
 
-
 extern "C" {
 	#include "avcodec.h"
-	#include "swresample.h"
 	#include "swscale.h"
 }
 
-
 #include "DecoderPlugin.h"
 #include "ReaderPlugin.h"
 
@@ -108,7 +103,6 @@ private:
 			// FFmpeg related members
 			AVCodec*			fCodec;
 			AVCodecContext*		fContext;
-			SwrContext*			fResampleContext;
 			uint8_t*			fDecodedData;
 			size_t				fDecodedDataSizeInBytes;
 			AVFrame*			fPostProcessedDecodedPicture;
@@ -118,9 +112,7 @@ private:
 			bool 				fCodecInitDone;
 
 			gfx_convert_func	fFormatConversionFunc;
-#if USE_SWS_FOR_COLOR_SPACE_CONVERSION
 			SwsContext*			fSwsContext;
-#endif
 
 			char*				fExtraData;
 			int					fExtraDataSize;
@@ -137,6 +129,7 @@ private:
 			bool				fAudioDecodeError;
 
 			AVFrame*			fDecodedDataBuffer;
+			int32				fDecodedDataBufferOffset;
 			int32				fDecodedDataBufferSize;
 
 			AVPacket			fTempPacket;

src/add-ons/media/plugins/ffmpeg/Jamfile

@@ -47,7 +47,6 @@ for architectureObject in [ MultiArchSubDirSetup ] {
 		UseHeaders [ FDirName $(ffmpegHeaders) libavformat ] ;
 		UseHeaders [ FDirName $(ffmpegHeaders) libavutil ] ;
 		UseHeaders [ FDirName $(ffmpegHeaders) libswscale ] ;
-		UseHeaders [ FDirName $(ffmpegHeaders) libswresample ] ;
 
 		Addon [ MultiArchDefaultGristFiles ffmpeg ] :
 			$(sources)