Merge pull request #595 from visualboyadvance-m/qol

Cleanup and WAV audio recording.

src/common/Patch.cpp

@@ -398,21 +398,14 @@ static bool patchApplyBPS(const char* patchname, uint8_t** rom, int* size)
     crc = crc32(0L, Z_NULL, 0);
     crc = crc32(crc, new_rom, dataSize);
 
-    // TODO
+    if(crc == dstCRC)
-    if(crc == dstCRC) {
+    {
-#if 1
         if (dataSize > *size) {
-            // SIGSEGV in /src/gba/GBA.cpp:3313 [*((uint16_t*)&rom[0x1fe209c]) = 0xdffa; // SWI 0xFA]
             *rom = (uint8_t*)realloc(*rom, dataSize);
         }
         memcpy(*rom, new_rom, dataSize);
         *size = dataSize;
         free(new_rom);
-#else
-        free(*rom);
-        *rom = new_rom;
-        *size = dataSize;
-#endif
     }
 
     fclose(f);
@@ -602,10 +595,6 @@ static bool patchApplyPPF(const char* patchname, uint8_t** rom, int* size)
 
 #endif
 
-// HINT: new format(.ext) => aditional changes in 
-//              sdl/SDL.cpp>>>main (~1600)  
-//              wx/panel.cpp>>>GameArea::LoadGame (~130)
-
 bool applyPatch(const char* patchname, uint8_t** rom, int* size)
 {
 #ifndef __LIBRETRO__

src/common/ffmpeg.cpp

@@ -12,7 +12,8 @@ struct supportedCodecs {
 
 const supportedCodecs audioSupported[] = {
     { AV_CODEC_ID_MP3, "MP3 (MPEG audio layer 3)", "mp3" },
-    { AV_CODEC_ID_AAC, "ADTS AAC (Advanced Audio Coding)", "aac,adts" }
+    { AV_CODEC_ID_AAC, "ADTS AAC (Advanced Audio Coding)", "aac,adts" },
+    { AV_CODEC_ID_PCM_S16LE, "WAV / WAVE (Waveform Audio)", "wav" }
 };
 
 const supportedCodecs videoSupported[] = {
@@ -120,7 +121,7 @@ recording::MediaRet recording::MediaRecorder::setup_audio_stream()
             }
         }
     }
-    if (!isSupported) return MRET_ERR_NOCODEC;
+    if (!isSupported && acodec->supported_samplerates) return MRET_ERR_NOCODEC;
     aenc->channels = av_get_channel_layout_nb_channels(aenc->channel_layout);
     aenc->channel_layout = AV_CH_LAYOUT_STEREO;
     if (acodec->channel_layouts)
@@ -143,7 +144,10 @@ recording::MediaRet recording::MediaRecorder::setup_audio_stream()
         return MRET_ERR_BUFSIZE;
     // number of samples per frame
     if (aenc->codec->capabilities & AV_CODEC_CAP_VARIABLE_FRAME_SIZE)
-        nb_samples = 10000;
+    {
+        //nb_samples = 10000; // can be any value, but we use our aud buffer size
+        nb_samples = 1470;
+    }
     else
         nb_samples = aenc->frame_size;
     // audio frame for input
@@ -552,6 +556,16 @@ recording::MediaRet recording::MediaRecorder::AddFrame(const uint16_t *aud, int
     int realLength = length / sizeof *aud;
     bool isMissing = false;
     int cp = -1;
+
+    if (c->frame_size == 0) // no compression/limit for audio frames
+    {
+        int maxCopy = realLength;
+        memcpy(audioBuffer + posInAudioBuffer, aud, maxCopy * 2);
+        posInAudioBuffer += maxCopy;
+        samplesInAudioBuffer += (maxCopy / 2);
+        aud += maxCopy;
+    }
+
     if (samplesInAudioBuffer < c->frame_size)
     {
         int missingSamples = (c->frame_size - samplesInAudioBuffer);
@@ -568,7 +582,7 @@ recording::MediaRet recording::MediaRecorder::AddFrame(const uint16_t *aud, int
             cp = realLength - maxCopy;
         }
     }
-    if (samplesInAudioBuffer != c->frame_size) // not enough samples
+    if (samplesInAudioBuffer != c->frame_size && (c->frame_size > 0 || samplesInAudioBuffer != realLength)) // not enough samples
     {
         return MRET_OK;
     }

src/wx/cmdevents.cpp

@@ -1197,7 +1197,7 @@ EVT_HANDLER_MASK(RecordSoundStartRecording, "Start sound recording...", CMDEN_NS
             ext.Replace(wxT(","), wxT(";*."));
             ext.insert(0, wxT("*."));
 
-            if (sound_extno < 0 && ext.find(wxT("*.mp3")) != wxString::npos)
+            if (sound_extno < 0 && ext.find(wxT("*.wav")) != wxString::npos)
                 sound_extno = extno;
 
             sound_exts.append(ext);