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Initial commit of FLTP format binkaudio support. Converts to FLT format by manually interleaving decoded samples. swresample library is included with a view to use swr_convert() in future versions, but not used in this commit.

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deque
cc9cii 10 years ago
parent 03c0cbc65a
commit 944f99b23a
3 changed files with 150 additions and 30 deletions
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2
CMakeLists.txt
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@ -139,7 +139,7 @@ set(OPENMW_LIBS ${OENGINE_ALL})
set(OPENMW_LIBS_HEADER)
# Sound setup
set(FFmpeg_FIND_COMPONENTS AVCODEC AVFORMAT AVUTIL SWSCALE)
set(FFmpeg_FIND_COMPONENTS AVCODEC AVFORMAT AVUTIL SWSCALE SWRESAMPLE)
find_package(FFmpeg REQUIRED)
set(SOUND_INPUT_INCLUDES ${FFMPEG_INCLUDE_DIRS})
set(SOUND_INPUT_LIBRARY ${FFMPEG_LIBRARIES} ${SWSCALE_LIBRARIES})

158
apps/openmw/mwrender/videoplayer.cpp
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@ -32,10 +32,11 @@ extern "C"
#include <libavformat/avformat.h>
#include <libswscale/swscale.h>
// From libavformat version 55.0.100 and onward the declaration of av_gettime() is removed from libavformat/avformat.h and moved
// to libavutil/time.h
// From libavformat version 55.0.100 and onward the declaration of av_gettime() is
// removed from libavformat/avformat.h and moved to libavutil/time.h
// https://github.com/FFmpeg/FFmpeg/commit/06a83505992d5f49846c18507a6c3eb8a47c650e
#if AV_VERSION_INT(55, 0, 100) <= AV_VERSION_INT(LIBAVFORMAT_VERSION_MAJOR, LIBAVFORMAT_VERSION_MINOR, LIBAVFORMAT_VERSION_MICRO)
#if AV_VERSION_INT(55, 0, 100) <= AV_VERSION_INT(LIBAVFORMAT_VERSION_MAJOR, \
LIBAVFORMAT_VERSION_MINOR, LIBAVFORMAT_VERSION_MICRO)
#include <libavutil/time.h>
#endif
@ -46,30 +47,27 @@ extern "C"
LIBAVUTIL_VERSION_MINOR, LIBAVUTIL_VERSION_MICRO)
#include <libavutil/channel_layout.h>
#endif
}
#ifdef _WIN32
// Decide whether to play binkaudio.
#include <libavcodec/version.h>
// libavcodec versions 54.10.100 (or maybe earlier) to 54.54.100 potentially crashes Windows 64bit.
// From version 54.56 or higher, there's no sound due to the encoding format changing from S16 to FLTP
// (see https://gitorious.org/ffmpeg/ffmpeg/commit/7bfd1766d1c18f07b0a2dd042418a874d49ea60d and
// http://git.videolan.org/?p=ffmpeg.git;a=commitdiff;h=3049d5b9b32845c86aa5588bb3352bdeb2edfdb2;hp=43c6b45a53a186a187f7266e4d6bd3c2620519f1),
// but does not crash (or at least no known crash).
#if (LIBAVCODEC_VERSION_MAJOR > 54)
// From version 54.56 binkaudio encoding format changed from S16 to FLTP. See:
// https://gitorious.org/ffmpeg/ffmpeg/commit/7bfd1766d1c18f07b0a2dd042418a874d49ea60d
// http://ffmpeg.zeranoe.com/forum/viewtopic.php?f=15&t=872
#if AV_VERSION_INT(54, 56, 0) <= AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
LIBAVCODEC_VERSION_MINOR, LIBAVCODEC_VERSION_MICRO)
#include <libswresample/swresample.h> /* swr_init, swr_alloc, swr_convert, swr_free */
#include <libavutil/opt.h> /* av_opt_set_int, av_opt_set_sample_fmt */
#define FFMPEG_PLAY_BINKAUDIO
#else
#ifdef _WIN64
#elif defined(_WIN32) && defined(_WIN64)
// Versions up to 54.54.100 potentially crashes on Windows 64bit.
#if ((LIBAVCODEC_VERSION_MAJOR == 54) && (LIBAVCODEC_VERSION_MINOR >= 55))
#define FFMPEG_PLAY_BINKAUDIO
#endif
#else
#elif defined(_WIN32)
// 54.10.100 is a known working version on 32bit, but earlier ones may also work.
#if ((LIBAVCODEC_VERSION_MAJOR == 54) && (LIBAVCODEC_VERSION_MINOR >= 10))
#define FFMPEG_PLAY_BINKAUDIO
#endif
#endif
#endif
#endif
}
#define MAX_AUDIOQ_SIZE (5 * 16 * 1024)
#define MAX_VIDEOQ_SIZE (5 * 256 * 1024)
@ -317,8 +315,12 @@ class MovieAudioDecoder : public MWSound::Sound_Decoder
VideoState *mVideoState;
AVStream *mAVStream;
SwrContext *mSwr; /* non-zero indicates FLTP format */
int mSamplesAllChannels;
int mSampleSize;
AutoAVPacket mPacket;
AVFrame *mFrame;
AVFrame *mFrame; /* AVFrame is now defined in libavutil/frame.h (used to be libavcodec/avcodec.h) */
ssize_t mFramePos;
ssize_t mFrameSize;
@ -420,7 +422,11 @@ public:
MovieAudioDecoder(VideoState *is)
: mVideoState(is)
, mAVStream(*is->audio_st)
#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(55, 28, 1)
, mFrame(avcodec_alloc_frame())
#else
, mFrame(av_frame_alloc())
#endif
, mFramePos(0)
, mFrameSize(0)
, mAudioClock(0.0)
@ -429,10 +435,15 @@ public:
/* Correct audio only if larger error than this */
, mAudioDiffThreshold(2.0 * 0.050/* 50 ms */)
, mAudioDiffAvgCount(0)
, mSwr(0)
, mSamplesAllChannels(0)
, mSampleSize(0)
{ }
virtual ~MovieAudioDecoder()
{
av_freep(&mFrame);
if(mSwr)
swr_free(&mSwr);
}
void getInfo(int *samplerate, MWSound::ChannelConfig *chans, MWSound::SampleType * type)
@ -443,6 +454,12 @@ public:
*type = MWSound::SampleType_Int16;
else if(mAVStream->codec->sample_fmt == AV_SAMPLE_FMT_FLT)
*type = MWSound::SampleType_Float32;
else if(mAVStream->codec->sample_fmt == AV_SAMPLE_FMT_FLTP)
{
// resample to a known format for OpenAL_SoundStream
// TODO: allow different size sample format, e.g. Int16
*type = MWSound::SampleType_Float32;
}
else
fail(std::string("Unsupported sample format: ")+
av_get_sample_fmt_name(mAVStream->codec->sample_fmt));
@ -480,17 +497,65 @@ public:
}
*samplerate = mAVStream->codec->sample_rate;
// FIXME: error handling
if(mAVStream->codec->sample_fmt == AV_SAMPLE_FMT_FLTP)
{
mSwr = swr_alloc();
av_opt_set_int(mSwr, "in_channel_layout", mAVStream->codec->channel_layout, 0);
av_opt_set_int(mSwr, "out_channel_layout", mAVStream->codec->channel_layout, 0);
av_opt_set_int(mSwr, "in_sample_rate", mAVStream->codec->sample_rate, 0);
av_opt_set_int(mSwr, "out_sample_rate", mAVStream->codec->sample_rate, 0);
av_opt_set_sample_fmt(mSwr, "in_sample_fmt", AV_SAMPLE_FMT_FLTP, 0);
av_opt_set_sample_fmt(mSwr, "out_sample_fmt", AV_SAMPLE_FMT_FLT, 0); // TODO: Try S16
swr_init(mSwr);
}
mSamplesAllChannels = av_get_bytes_per_sample(mAVStream->codec->sample_fmt) *
mAVStream->codec->channels;
mSampleSize = av_get_bytes_per_sample(mAVStream->codec->sample_fmt);
}
/*
* stream is a ptr to vector<char> on the stack, see OpenAL_SoundStream::process in
* mwsound/openal_output.cpp (around line 481)
*
* len is the size of the output buffer (i.e. stream) based on the number of
* channels, rate and sample type (as reported by getInfo).
*
* sample_skip is the number of bytes to skip (from all channels) or repeat (i.e. negative)
*
* mFrameSize is the number of bytes decoded audio frame (from all channels), or -1 if finished
*
*
* +---------------------------------------------------------------------------------+
* | |
* |<------------------------------------------ len -------------------------------->|
* | |
* |<------ mFrameSize ------>| |
* | |
* +---------------------------------------------------------------------------------+
* ^
* |
* mFramePos >= 0
*
* |<----- len1 -------->|
*
*/
size_t read(char *stream, size_t len)
{
int sample_skip = synchronize_audio();
size_t total = 0;
float *outputStream = (float *)&stream[0];
//uint16_t *intStream = (uint16_t *)&stream[0];
while(total < len)
{
if(mFramePos >= mFrameSize)
{
// for FLT sample format mFrameSize returned by audio_decode_frame is:
// 1920 samples x 4 bytes/sample x 2 channels = 15360 bytes
/* We have already sent all our data; get more */
mFrameSize = audio_decode_frame(mFrame);
if(mFrameSize < 0)
@ -508,9 +573,56 @@ public:
if(mFramePos >= 0)
{
len1 = std::min<size_t>(len1, mFrameSize-mFramePos);
memcpy(stream, mFrame->data[0]+mFramePos, len1);
if(mSwr)
{
// Convert from AV_SAMPLE_FMT_FLTP to AV_SAMPLE_FMT_FLT
// FIXME: support channel formats other than stereo
// FIXME: support sample formats other than Float32
float* inputChannel0 = (float*)mFrame->extended_data[0];
float* inputChannel1 = (float*)mFrame->extended_data[1];
#if 0
uint16_t* inputChannel0 = (uint16_t*)mFrame->extended_data[0];
uint16_t* inputChannel1 = (uint16_t*)mFrame->extended_data[1];
#endif
inputChannel0 += mFramePos/mSamplesAllChannels;
inputChannel1 += mFramePos/mSamplesAllChannels;
//if(mFramePos > 0)
//std::cout << "mFramePos (bytes): " + std::to_string(mFramePos)
//<< " samples: " + std::to_string(mFramePos/mSamplesAllChannels) << std::endl;
// samples per channel = len1 bytes / bytes per sample / number of channels
unsigned int len1Samples = len1 / mSamplesAllChannels;
// stream offset = total bytes / bytes per sample
unsigned int totalOffset = total / mSampleSize;
float sample0 = 0;
float sample1 = 0;
for (unsigned int i = 0 ; i < len1Samples ; ++i)
{
sample0 = *inputChannel0++;
sample1 = *inputChannel1++;
#if 0
if(sample0<-1.0f)
sample0=-1.0f;
else if(sample0>1.0f)
sample0=1.0f;
if(sample1<-1.0f)
sample1=-1.0f;
else if(sample1>1.0f)
sample1=1.0f;
intStream[totalOffset+i*2] = (uint16_t) (sample0 * 32767.0f);
intStream[totalOffset+i*2+1] = (uint16_t) (sample1 * 32767.0f);
#endif
outputStream[totalOffset+i*2] = sample0;
outputStream[totalOffset+i*2+1] = sample1;
}
}
else
{
memcpy(stream, mFrame->data[0]+mFramePos, len1);
}
}
else // repeat some samples FIXME: support ftlp
{
len1 = std::min<size_t>(len1, -mFramePos);
@ -768,9 +880,15 @@ void VideoState::video_thread_loop(VideoState *self)
AVFrame *pFrame;
double pts;
#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(55, 28, 1)
pFrame = avcodec_alloc_frame();
self->rgbaFrame = avcodec_alloc_frame();
#else
pFrame = av_frame_alloc();
self->rgbaFrame = av_frame_alloc();
#endif
avpicture_alloc((AVPicture*)self->rgbaFrame, PIX_FMT_RGBA, (*self->video_st)->codec->width, (*self->video_st)->codec->height);
while(self->videoq.get(packet, self) >= 0)

6
cmake/FindFFmpeg.cmake
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@ -14,6 +14,7 @@
# - AVUTIL
# - POSTPROCESS
# - SWSCALE
# - SWRESAMPLE
# the following variables will be defined
# <component>_FOUND - System has <component>
# <component>_INCLUDE_DIRS - Include directory necessary for using the <component> headers
@ -32,7 +33,7 @@ include(FindPackageHandleStandardArgs)
# The default components were taken from a survey over other FindFFMPEG.cmake files
if (NOT FFmpeg_FIND_COMPONENTS)
set(FFmpeg_FIND_COMPONENTS AVCODEC AVFORMAT AVUTIL SWSCALE)
set(FFmpeg_FIND_COMPONENTS AVCODEC AVFORMAT AVUTIL SWSCALE SWRESAMPLE)
endif ()
#
@ -112,6 +113,7 @@ if (NOT FFMPEG_LIBRARIES)
find_component(AVUTIL libavutil avutil libavutil/avutil.h)
find_component(SWSCALE libswscale swscale libswscale/swscale.h)
find_component(POSTPROC libpostproc postproc libpostproc/postprocess.h)
find_component(SWRESAMPLE libswresample swresample libswresample/swresample.h)
# Check if the required components were found and add their stuff to the FFMPEG_* vars.
foreach (_component ${FFmpeg_FIND_COMPONENTS})
@ -142,7 +144,7 @@ if (NOT FFMPEG_LIBRARIES)
endif ()
# Now set the noncached _FOUND vars for the components.
foreach (_component AVCODEC AVDEVICE AVFORMAT AVUTIL POSTPROCESS SWSCALE)
foreach (_component AVCODEC AVDEVICE AVFORMAT AVUTIL POSTPROCESS SWSCALE SWRESAMPLE)
set_component_found(${_component})
endforeach ()

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