diff options
author | XhmikosR <xhmikosr@users.sourceforge.net> | 2012-09-24 14:29:35 +0400 |
---|---|---|
committer | XhmikosR <xhmikosr@users.sourceforge.net> | 2012-09-24 14:30:07 +0400 |
commit | 1ff29856b25c0fb1ab1d194977b52482d09b292f (patch) | |
tree | 78f40764d48b0d510d48c999982886b011c940b0 /src/DSUtil/AudioParser.cpp | |
parent | f1b7b0c68fcb3cf155b3e2d7a70f3f320430bf62 (diff) |
AudioParser.cpp: cosmetics
Diffstat (limited to 'src/DSUtil/AudioParser.cpp')
-rw-r--r-- | src/DSUtil/AudioParser.cpp | 22 |
1 files changed, 11 insertions, 11 deletions
diff --git a/src/DSUtil/AudioParser.cpp b/src/DSUtil/AudioParser.cpp index 44499a5db..9fa91e138 100644 --- a/src/DSUtil/AudioParser.cpp +++ b/src/DSUtil/AudioParser.cpp @@ -53,7 +53,7 @@ int GetAC3FrameSize(const BYTE* buf) }
int frame_size;
- static const int rates[] = {32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384, 448, 512, 576, 640};
+ static const int rates[] = { 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384, 448, 512, 576, 640 };
int rate = rates[frmsizecod >> 1];
switch (buf[4] & 0xc0) {
@@ -112,9 +112,9 @@ int ParseAC3Header(const BYTE* buf, int* samplerate, int* channels, int* framele return 0;
}
- static const int rates[] = {32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384, 448, 512, 576, 640};
- static const unsigned char lfeon[8] = {0x10, 0x10, 0x04, 0x04, 0x04, 0x01, 0x04, 0x01};
- static const unsigned char halfrate[12] = {0, 0, 0, 0, 0, 0, 0, 0, 0, 1, 2, 3};
+ static const int rates[] = { 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384, 448, 512, 576, 640 };
+ static const unsigned char lfeon[8] = { 0x10, 0x10, 0x04, 0x04, 0x04, 0x01, 0x04, 0x01 };
+ static const unsigned char halfrate[12] = { 0, 0, 0, 0, 0, 0, 0, 0, 0, 1, 2, 3 };
int frmsizecod = buf[4] & 0x3F;
if (frmsizecod >= 38) {
@@ -322,8 +322,8 @@ void dts14le_to_dts16be(const BYTE* source, BYTE* destination, int size) int ParseDTSHeader(const BYTE* buf, int* samplerate, int* channels, int* framelength, int* tr_bitrate)
{
- static const int dts_channels[16] = {1, 2, 2, 2, 2, 3, 3, 4, 4, 5, 6, 6, 6, 7, 8, 8};
- static const int core_sample_rates[] = {0, 8000, 16000, 32000, 0, 0, 11025, 22050, 44100, 0, 0, 12000, 24000, 48000, 96000, 192000};
+ static const int dts_channels[16] = { 1, 2, 2, 2, 2, 3, 3, 4, 4, 5, 6, 6, 6, 7, 8, 8} ;
+ static const int core_sample_rates[] = { 0, 8000, 16000, 32000, 0, 0, 11025, 22050, 44100, 0, 0, 12000, 24000, 48000, 96000, 192000 };
static const int transmission_bitrates[32] = {
32000, 56000, 64000, 96000,
112000, 128000, 192000, 224000,
@@ -410,7 +410,7 @@ int ParseHdmvLPCMHeader(const BYTE* buf, int* samplerate, int* channels) int frame_size = buf[0] << 8 | buf[1];
frame_size += 4; // add header size;
- static int channels_layout[] = {0, 1, 0, 2, 3, 3, 4, 4, 5, 6, 7, 8, 0, 0, 0, 0};
+ static int channels_layout[] = { 0, 1, 0, 2, 3, 3, 4, 4, 5, 6, 7, 8, 0, 0, 0, 0 };
BYTE channel_layout = buf[2] >> 4;
*channels = channels_layout[channel_layout];
if (!*channels) {
@@ -448,7 +448,7 @@ static UINT32 LATMGetValue(CGolombBuffer& gb) static bool ParseLATMHeaderAudioSpecificConfig(CGolombBuffer& gb, int* samplerate, int* channels)
{
static int sampling_frequency[] = { 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050, 16000, 12000, 11025, 8000, 7350 };
- static int channel_configuration[] = {0, 1, 2, 3, 4, 5, 6, 8};
+ static int channel_configuration[] = { 0, 1, 2, 3, 4, 5, 6, 8 };
short sAudioObjectType = (short)gb.BitRead(5);
if (sAudioObjectType == 31) {
@@ -506,7 +506,7 @@ static bool ParseLATMHeaderStreamMuxConfig(CGolombBuffer& gb, int* samplerate, i {
BYTE bAudioMuxVersion = (BYTE)gb.BitRead(1);
BYTE bAudioMuxVersionA = 0; // default value is 0
- if (bAudioMuxVersion) { //
+ if (bAudioMuxVersion) {
bAudioMuxVersionA = (BYTE)gb.BitRead(1);
}
@@ -531,7 +531,7 @@ static bool ParseLATMHeaderStreamMuxConfig(CGolombBuffer& gb, int* samplerate, i if (!bUseSameConfig) {
if (bAudioMuxVersion == 0) {
success = ParseLATMHeaderAudioSpecificConfig(gb, samplerate, channels);
- } else{
+ } else {
UINT32 uAscLen = LATMGetValue(gb);
success = ParseLATMHeaderAudioSpecificConfig(gb, samplerate, channels); // TODO uAscLen -= AudioSpecificConfig();
// gb.BitRead((int)uAscLen); // fillBits
@@ -554,7 +554,7 @@ bool ParseAACLATMHeader(BYTE* buf, int len, int* samplerate, int* channels) return false;
}
gb.BitRead(13); // audioMuxLengthBytes;
-
+
bool success = false;
// Parsing AudioMuxElement
if (!gb.BitRead(1)) { // useSameStreamMux
|