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authorMax Kellermann <max@duempel.org>2008-10-10 14:40:54 +0200
committerMax Kellermann <max@duempel.org>2008-10-10 14:40:54 +0200
commitde2cb3f37568e7680549057f8d7b6d748c388480 (patch)
tree46f9f43a1f83b49945c8a4fc77f933fad9230e01 /src/inputPlugins/aac_plugin.c
parent6101dc6c768b09dbcdc1840a84b619a5a6a20129 (diff)
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audio_format: renamed sampleRate to sample_rate
The last bit of CamelCase in audio_format.h. Additionally, rename a bunch of local variables.
Diffstat (limited to '')
-rw-r--r--src/inputPlugins/aac_plugin.c46
1 files changed, 23 insertions, 23 deletions
diff --git a/src/inputPlugins/aac_plugin.c b/src/inputPlugins/aac_plugin.c
index a96623e1b..e9b2d7476 100644
--- a/src/inputPlugins/aac_plugin.c
+++ b/src/inputPlugins/aac_plugin.c
@@ -148,7 +148,7 @@ static size_t adts_find_frame(AacBuffer * b)
static void adtsParse(AacBuffer * b, float *length)
{
unsigned int frames, frameLength;
- int sampleRate = 0;
+ int sample_rate = 0;
float framesPerSec;
/* Read all frames to ensure correct time and bitrate */
@@ -158,9 +158,9 @@ static void adtsParse(AacBuffer * b, float *length)
frameLength = adts_find_frame(b);
if (frameLength > 0) {
if (frames == 0) {
- sampleRate = adtsSampleRates[(b->
- buffer[2] & 0x3c)
- >> 2];
+ sample_rate = adtsSampleRates[(b->
+ buffer[2] & 0x3c)
+ >> 2];
}
if (frameLength > b->bytesIntoBuffer)
@@ -171,7 +171,7 @@ static void adtsParse(AacBuffer * b, float *length)
break;
}
- framesPerSec = (float)sampleRate / 1024.0;
+ framesPerSec = (float)sample_rate / 1024.0;
if (framesPerSec != 0)
*length = (float)frames / framesPerSec;
}
@@ -253,7 +253,7 @@ static float getAacFloatTotalTime(char *file)
float length;
faacDecHandle decoder;
faacDecConfigurationPtr config;
- uint32_t sampleRate;
+ uint32_t sample_rate;
unsigned char channels;
InputStream inStream;
long bread;
@@ -274,11 +274,11 @@ static float getAacFloatTotalTime(char *file)
fillAacBuffer(&b);
#ifdef HAVE_FAAD_BUFLEN_FUNCS
bread = faacDecInit(decoder, b.buffer, b.bytesIntoBuffer,
- &sampleRate, &channels);
+ &sample_rate, &channels);
#else
- bread = faacDecInit(decoder, b.buffer, &sampleRate, &channels);
+ bread = faacDecInit(decoder, b.buffer, &sample_rate, &channels);
#endif
- if (bread >= 0 && sampleRate > 0 && channels > 0)
+ if (bread >= 0 && sample_rate > 0 && channels > 0)
length = 0;
faacDecClose(decoder);
@@ -312,7 +312,7 @@ static int aac_stream_decode(struct decoder * mpd_decoder,
faacDecConfigurationPtr config;
long bread;
struct audio_format audio_format;
- uint32_t sampleRate;
+ uint32_t sample_rate;
unsigned char channels;
unsigned int sampleCount;
char *sampleBuffer;
@@ -346,9 +346,9 @@ static int aac_stream_decode(struct decoder * mpd_decoder,
#ifdef HAVE_FAAD_BUFLEN_FUNCS
bread = faacDecInit(decoder, b.buffer, b.bytesIntoBuffer,
- &sampleRate, &channels);
+ &sample_rate, &channels);
#else
- bread = faacDecInit(decoder, b.buffer, &sampleRate, &channels);
+ bread = faacDecInit(decoder, b.buffer, &sample_rate, &channels);
#endif
if (bread < 0) {
ERROR("Error not a AAC stream.\n");
@@ -386,12 +386,12 @@ static int aac_stream_decode(struct decoder * mpd_decoder,
break;
}
#ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE
- sampleRate = frameInfo.samplerate;
+ sample_rate = frameInfo.samplerate;
#endif
if (!initialized) {
audio_format.channels = frameInfo.channels;
- audio_format.sampleRate = sampleRate;
+ audio_format.sample_rate = sample_rate;
decoder_initialized(mpd_decoder, &audio_format, totalTime);
initialized = 1;
}
@@ -402,11 +402,11 @@ static int aac_stream_decode(struct decoder * mpd_decoder,
if (sampleCount > 0) {
bitRate = frameInfo.bytesconsumed * 8.0 *
- frameInfo.channels * sampleRate /
+ frameInfo.channels * sample_rate /
frameInfo.samples / 1000 + 0.5;
file_time +=
(float)(frameInfo.samples) / frameInfo.channels /
- sampleRate;
+ sample_rate;
}
sampleBufferLen = sampleCount * 2;
@@ -446,7 +446,7 @@ static int aac_decode(struct decoder * mpd_decoder, char *path)
faacDecConfigurationPtr config;
long bread;
struct audio_format audio_format;
- uint32_t sampleRate;
+ uint32_t sample_rate;
unsigned char channels;
unsigned int sampleCount;
char *sampleBuffer;
@@ -484,9 +484,9 @@ static int aac_decode(struct decoder * mpd_decoder, char *path)
#ifdef HAVE_FAAD_BUFLEN_FUNCS
bread = faacDecInit(decoder, b.buffer, b.bytesIntoBuffer,
- &sampleRate, &channels);
+ &sample_rate, &channels);
#else
- bread = faacDecInit(decoder, b.buffer, &sampleRate, &channels);
+ bread = faacDecInit(decoder, b.buffer, &sample_rate, &channels);
#endif
if (bread < 0) {
ERROR("Error not a AAC stream.\n");
@@ -522,12 +522,12 @@ static int aac_decode(struct decoder * mpd_decoder, char *path)
break;
}
#ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE
- sampleRate = frameInfo.samplerate;
+ sample_rate = frameInfo.samplerate;
#endif
if (!initialized) {
audio_format.channels = frameInfo.channels;
- audio_format.sampleRate = sampleRate;
+ audio_format.sample_rate = sample_rate;
decoder_initialized(mpd_decoder, &audio_format,
totalTime);
initialized = 1;
@@ -539,11 +539,11 @@ static int aac_decode(struct decoder * mpd_decoder, char *path)
if (sampleCount > 0) {
bitRate = frameInfo.bytesconsumed * 8.0 *
- frameInfo.channels * sampleRate /
+ frameInfo.channels * sample_rate /
frameInfo.samples / 1000 + 0.5;
file_time +=
(float)(frameInfo.samples) / frameInfo.channels /
- sampleRate;
+ sample_rate;
}
sampleBufferLen = sampleCount * 2;