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aac: removed method file_decode()

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aac_stream_decode() was basically copy+pasted from aac_decode().
Since stream_decode() can also decode files, eliminate aac_decode().
This commit is contained in:
Max Kellermann 2008-11-12 08:17:50 +01:00
parent e19b53f814
commit 39fa60769c
1 changed files with 1 additions and 124 deletions
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125
src/decoder/aac_plugin.c
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@ -322,6 +322,7 @@ aac_stream_decode(struct decoder *mpd_decoder, struct input_stream *inStream)
bool initialized = false;
initAacBuffer(&b, mpd_decoder, inStream);
aac_parse_header(&b, &totalTime);
decoder = faacDecOpen();
@ -422,129 +423,6 @@ aac_stream_decode(struct decoder *mpd_decoder, struct input_stream *inStream)
free(b.buffer);
}
static void
aac_decode(struct decoder *mpd_decoder, const char *path)
{
float file_time;
float totalTime;
faacDecHandle decoder;
faacDecFrameInfo frameInfo;
faacDecConfigurationPtr config;
long bread;
struct audio_format audio_format;
uint32_t sample_rate;
unsigned char channels;
unsigned int sampleCount;
char *sampleBuffer;
size_t sampleBufferLen;
/*float * seekTable;
long seekTableEnd = -1;
int seekPositionFound = 0; */
uint16_t bitRate = 0;
AacBuffer b;
struct input_stream inStream;
bool initialized = false;
if ((totalTime = getAacFloatTotalTime(path)) < 0)
return;
if (!input_stream_open(&inStream, path))
return;
initAacBuffer(&b, mpd_decoder, &inStream);
aac_parse_header(&b, NULL);
decoder = faacDecOpen();
config = faacDecGetCurrentConfiguration(decoder);
config->outputFormat = FAAD_FMT_16BIT;
#ifdef HAVE_FAACDECCONFIGURATION_DOWNMATRIX
config->downMatrix = 1;
#endif
#ifdef HAVE_FAACDECCONFIGURATION_DONTUPSAMPLEIMPLICITSBR
config->dontUpSampleImplicitSBR = 0;
#endif
faacDecSetConfiguration(decoder, config);
fillAacBuffer(&b);
#ifdef HAVE_FAAD_BUFLEN_FUNCS
bread = faacDecInit(decoder, b.buffer, b.bytesIntoBuffer,
&sample_rate, &channels);
#else
bread = faacDecInit(decoder, b.buffer, &sample_rate, &channels);
#endif
if (bread < 0) {
ERROR("Error not a AAC stream.\n");
faacDecClose(decoder);
if (b.buffer)
free(b.buffer);
return;
}
audio_format.bits = 16;
file_time = 0.0;
advanceAacBuffer(&b, bread);
do {
fillAacBuffer(&b);
if (b.bytesIntoBuffer == 0)
break;
#ifdef HAVE_FAAD_BUFLEN_FUNCS
sampleBuffer = faacDecDecode(decoder, &frameInfo, b.buffer,
b.bytesIntoBuffer);
#else
sampleBuffer = faacDecDecode(decoder, &frameInfo, b.buffer);
#endif
if (frameInfo.error > 0) {
ERROR("error decoding AAC file: %s\n", path);
ERROR("faad2 error: %s\n",
faacDecGetErrorMessage(frameInfo.error));
break;
}
#ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE
sample_rate = frameInfo.samplerate;
#endif
if (!initialized) {
audio_format.channels = frameInfo.channels;
audio_format.sample_rate = sample_rate;
decoder_initialized(mpd_decoder, &audio_format,
false, totalTime);
initialized = true;
}
advanceAacBuffer(&b, frameInfo.bytesconsumed);
sampleCount = (unsigned long)(frameInfo.samples);
if (sampleCount > 0) {
bitRate = frameInfo.bytesconsumed * 8.0 *
frameInfo.channels * sample_rate /
frameInfo.samples / 1000 + 0.5;
file_time +=
(float)(frameInfo.samples) / frameInfo.channels /
sample_rate;
}
sampleBufferLen = sampleCount * 2;
decoder_data(mpd_decoder, NULL, sampleBuffer,
sampleBufferLen, file_time,
bitRate, NULL);
} while (decoder_get_command(mpd_decoder) == DECODE_COMMAND_NONE);
faacDecClose(decoder);
if (b.buffer)
free(b.buffer);
}
static struct tag *aacTagDup(const char *file)
{
struct tag *ret = NULL;
@ -568,7 +446,6 @@ static const char *const aac_mimeTypes[] = { "audio/aac", "audio/aacp", NULL };
const struct decoder_plugin aacPlugin = {
.name = "aac",
.stream_decode = aac_stream_decode,
.file_decode = aac_decode,
.tag_dup = aacTagDup,
.suffixes = aac_suffixes,
.mime_types = aac_mimeTypes