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audio_core: hle: mf: address another batch of reviews from @B3N30

This commit is contained in:
liushuyu 2019-01-05 22:28:56 -07:00 committed by B3N30
parent 7f5b54fda4
commit c03861c2d9
6 changed files with 39 additions and 43 deletions
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4
src/audio_core/CMakeLists.txt
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@ -4,6 +4,8 @@ add_library(audio_core STATIC
codec.h
dsp_interface.cpp
dsp_interface.h
hle/adts.h
hle/adts_reader.cpp
hle/common.h
hle/decoder.cpp
hle/decoder.h
@ -30,7 +32,7 @@ add_library(audio_core STATIC
$<$<BOOL:${SDL2_FOUND}>:sdl2_sink.cpp sdl2_sink.h>
$<$<BOOL:${ENABLE_CUBEB}>:cubeb_sink.cpp cubeb_sink.h>
$<$<BOOL:${FFMPEG_FOUND}>:hle/ffmpeg_decoder.cpp hle/ffmpeg_decoder.h hle/ffmpeg_dl.cpp hle/ffmpeg_dl.h>
$<$<BOOL:${ENABLE_MF}>:hle/wmf_decoder.cpp hle/wmf_decoder.h hle/wmf_decoder_utils.cpp hle/wmf_decoder_utils.h hle/adts_reader.cpp>
$<$<BOOL:${ENABLE_MF}>:hle/wmf_decoder.cpp hle/wmf_decoder.h hle/wmf_decoder_utils.cpp hle/wmf_decoder_utils.h>
)
create_target_directory_groups(audio_core)

8
src/audio_core/hle/adts.h
View File

@ -17,14 +17,6 @@ struct ADTSData {
u32 samplerate;
};
typedef struct ADTSData ADTSData;
#ifdef __cplusplus
extern "C" {
#endif // __cplusplus
u32 parse_adts(char* buffer, struct ADTSData* out);
// last two bytes of MF AAC decoder user data
u16 mf_get_aac_tag(struct ADTSData input);
#ifdef __cplusplus
}
#endif // __cplusplus

15
src/audio_core/hle/wmf_decoder.cpp
View File

@ -73,7 +73,7 @@ std::optional<BinaryResponse> WMFDecoder::Impl::Initalize(const BinaryRequest& r
std::memcpy(&response, &request, sizeof(response));
response.unknown1 = 0x0;
if (MFDecoderInit(&transform) != 0) {
if (!MFDecoderInit(&transform)) {
LOG_CRITICAL(Audio_DSP, "Can't init decoder");
return response;
}
@ -104,7 +104,7 @@ void WMFDecoder::Impl::Clear() {
int WMFDecoder::Impl::DecodingLoop(ADTSData adts_header,
std::array<std::vector<u8>, 2>& out_streams) {
int output_status = 0;
MFOutputState output_status = OK;
char* output_buffer = nullptr;
DWORD output_len = 0;
IMFSample* output = nullptr;
@ -113,7 +113,7 @@ int WMFDecoder::Impl::DecodingLoop(ADTSData adts_header,
output_status = ReceiveSample(transform, out_stream_id, &output);
// 0 -> okay; 3 -> okay but more data available (buffer too small)
if (output_status == 0 || output_status == 3) {
if (output_status == OK || output_status == HAVE_MORE_DATA) {
CopySampleToBuffer(output, (void**)&output_buffer, &output_len);
// the following was taken from ffmpeg version of the decoder
@ -133,20 +133,21 @@ int WMFDecoder::Impl::DecodingLoop(ADTSData adts_header,
}
// in case of "ok" only, just return quickly
if (output_status == 0)
if (output_status == OK)
return 0;
// for status = 2, reset MF
if (output_status == 2) {
if (output_status == NEED_RECONFIG) {
Clear();
return -1;
}
// for status = 3, try again with new buffer
if (output_status == 3)
if (output_status == HAVE_MORE_DATA)
continue;
return output_status; // return on other status
LOG_ERROR(Audio_DSP, "Errors occurred when receiving output: {}", output_status);
return -1; // return on other status
}
return -1;

35
src/audio_core/hle/wmf_decoder_utils.cpp
View File

@ -22,7 +22,7 @@ void ReportError(std::string msg, HRESULT hr) {
LOG_CRITICAL(Audio_DSP, "{}: {:08x}", msg, hr);
}
int MFCoInit() {
bool MFCoInit() {
HRESULT hr = S_OK;
// lite startup is faster and all what we need is included
@ -30,15 +30,15 @@ int MFCoInit() {
if (hr != S_OK) {
// Do you know you can't initialize MF in test mode or safe mode?
ReportError("Failed to initialize Media Foundation", hr);
return -1;
return false;
}
LOG_INFO(Audio_DSP, "Media Foundation activated");
return 0;
return true;
}
int MFDecoderInit(IMFTransform** transform, GUID audio_format) {
bool MFDecoderInit(IMFTransform** transform, GUID audio_format) {
HRESULT hr = S_OK;
MFT_REGISTER_TYPE_INFO reg = {0};
GUID category = MFT_CATEGORY_AUDIO_DECODER;
@ -54,7 +54,7 @@ int MFDecoderInit(IMFTransform** transform, GUID audio_format) {
if (FAILED(hr) || num_activate < 1) {
ReportError("Failed to enumerate decoders", hr);
CoTaskMemFree(activate);
return -1;
return false;
}
LOG_INFO(Audio_DSP, "Windows(R) Media Foundation found {} suitable decoder(s)", num_activate);
for (unsigned int n = 0; n < num_activate; n++) {
@ -66,10 +66,10 @@ int MFDecoderInit(IMFTransform** transform, GUID audio_format) {
if (*transform == nullptr) {
ReportError("Failed to initialize MFT", hr);
CoTaskMemFree(activate);
return -1;
return false;
}
CoTaskMemFree(activate);
return 0;
return true;
}
void MFDeInit(IMFTransform** transform) {
@ -117,7 +117,7 @@ IMFSample* CreateSample(void* data, DWORD len, DWORD alignment, LONGLONG duratio
return sample;
}
bool SelectInputMediaType(IMFTransform* transform, int in_stream_id, ADTSData adts,
bool SelectInputMediaType(IMFTransform* transform, int in_stream_id, const ADTSData& adts,
UINT8* user_data, UINT32 user_data_len, GUID audio_format) {
HRESULT hr = S_OK;
IMFMediaType* t;
@ -261,8 +261,7 @@ int SendSample(IMFTransform* transform, DWORD in_stream_id, IMFSample* in_sample
return 0;
}
// return: 0: okay; 1: needs more sample; 2: needs reconfiguring; 3: more data available
int ReceiveSample(IMFTransform* transform, DWORD out_stream_id, IMFSample** out_sample) {
MFOutputState ReceiveSample(IMFTransform* transform, DWORD out_stream_id, IMFSample** out_sample) {
HRESULT hr;
MFT_OUTPUT_DATA_BUFFER out_buffers;
IMFSample* sample = nullptr;
@ -272,14 +271,14 @@ int ReceiveSample(IMFTransform* transform, DWORD out_stream_id, IMFSample** out_
if (!out_sample) {
ReportError("nullptr pointer passed to receive_sample()", MF_E_SAMPLE_NOT_WRITABLE);
return -1;
return FATAL_ERROR;
}
hr = transform->GetOutputStreamInfo(out_stream_id, &out_info);
if (FAILED(hr)) {
ReportError("MFT: Failed to get stream info", hr);
return -1;
return FATAL_ERROR;
}
mft_create_sample = (out_info.dwFlags & MFT_OUTPUT_STREAM_PROVIDES_SAMPLES) ||
(out_info.dwFlags & MFT_OUTPUT_STREAM_CAN_PROVIDE_SAMPLES);
@ -293,7 +292,7 @@ int ReceiveSample(IMFTransform* transform, DWORD out_stream_id, IMFSample** out_
sample = CreateSample(nullptr, out_info.cbSize, out_info.cbAlignment);
if (!sample) {
ReportError("MFT: Unable to allocate memory for samples", hr);
return -1;
return FATAL_ERROR;
}
}
@ -309,12 +308,12 @@ int ReceiveSample(IMFTransform* transform, DWORD out_stream_id, IMFSample** out_
if (hr == MF_E_TRANSFORM_NEED_MORE_INPUT) {
// Most likely reasons: data corrupted; your actions not expected by MFT
return 1;
return NEED_MORE_INPUT;
}
if (hr == MF_E_TRANSFORM_STREAM_CHANGE) {
ReportError("MFT: stream format changed, re-configuration required", hr);
return 2;
return NEED_RECONFIG;
}
break;
@ -322,15 +321,15 @@ int ReceiveSample(IMFTransform* transform, DWORD out_stream_id, IMFSample** out_
if (out_buffers.dwStatus & MFT_OUTPUT_DATA_BUFFER_INCOMPLETE) {
// this status is also unreliable but whatever
return 3;
return HAVE_MORE_DATA;
}
if (*out_sample == nullptr) {
ReportError("MFT: decoding failure", hr);
return -1;
return FATAL_ERROR;
}
return 0;
return OK;
}
int CopySampleToBuffer(IMFSample* sample, void** output, DWORD* len) {

10
src/audio_core/hle/wmf_decoder_utils.h
View File

@ -19,6 +19,8 @@
#include "adts.h"
enum MFOutputState { FATAL_ERROR = -1, OK = 0, NEED_MORE_INPUT, NEED_RECONFIG, HAVE_MORE_DATA };
// utility functions
template <class T>
void SafeRelease(T** ppT) {
@ -31,11 +33,11 @@ void SafeRelease(T** ppT) {
void ReportError(std::string msg, HRESULT hr);
// exported functions
int MFCoInit();
int MFDecoderInit(IMFTransform** transform, GUID audio_format = MFAudioFormat_AAC);
bool MFCoInit();
bool MFDecoderInit(IMFTransform** transform, GUID audio_format = MFAudioFormat_AAC);
void MFDeInit(IMFTransform** transform);
IMFSample* CreateSample(void* data, DWORD len, DWORD alignment = 1, LONGLONG duration = 0);
bool SelectInputMediaType(IMFTransform* transform, int in_stream_id, ADTSData adts,
bool SelectInputMediaType(IMFTransform* transform, int in_stream_id, const ADTSData& adts,
UINT8* user_data, UINT32 user_data_len,
GUID audio_format = MFAudioFormat_AAC);
int DetectMediaType(char* buffer, size_t len, ADTSData* output, char** aac_tag);
@ -43,5 +45,5 @@ bool SelectOutputMediaType(IMFTransform* transform, int out_stream_id,
GUID audio_format = MFAudioFormat_PCM);
void MFFlush(IMFTransform** transform);
int SendSample(IMFTransform* transform, DWORD in_stream_id, IMFSample* in_sample);
int ReceiveSample(IMFTransform* transform, DWORD out_stream_id, IMFSample** out_sample);
MFOutputState ReceiveSample(IMFTransform* transform, DWORD out_stream_id, IMFSample** out_sample);
int CopySampleToBuffer(IMFSample* sample, void** output, DWORD* len);