def __init__(self, codec):
"""Initialisation. Create a decoder for the specified codec.
Codec is specified by file extension. Available codecs are
listed in pymedia.audio.acodec.extensions.
"""
super(AudioDecoder, self).__init__()
if float(acodec.version) > 2:
self.codecid = acodec.getCodecId(codec)
self.decoder = acodec.Decoder({"id": self.codecid})
else:
self.decoder = acodec.Decoder(codec)
def dumpWAV(name):
import pymedia.audio.acodec as acodec
import pymedia.muxer as muxer
import time, wave, string, os
name1 = str.split(name, '.')
name2 = string.join(name1[:len(name1) - 1])
# Open demuxer first
dm = muxer.Demuxer(name1[-1].lower())
dec = None
f = open(name, 'rb')
snd = None
s = " "
while len(s):
s = f.read(20000)
if len(s):
frames = dm.parse(s)
for fr in frames:
if dec == None:
# Open decoder
dec = acodec.Decoder(dm.streams[0])
r = dec.decode(fr[1])
if r and r.data:
if snd == None:
snd = wave.open(name2 + '.wav', 'wb')
snd.setparams(
(r.channels, 2, r.sample_rate, 0, 'NONE', ''))
snd.writeframes(r.data)
def __init__(self, file_name):
super(audio_decoder, self).__init__()
self.file_name = file_name
ext = file_name.strip().split('.')[-1].lower()
dm = muxer.Demuxer(ext)
fin = open(file_name, 'rb')
if not fin:
raise "cannot find file %s" % file_name
s = fin.read(3000000)
r = dm.parse(s)
print dm.streams
self.decoder = None
for aindex in xrange( len( dm.streams )):
if dm.streams[ aindex ] and dm.streams[ aindex ][ 'type' ]== muxer.CODEC_TYPE_AUDIO:
self.decoder = acodec.Decoder( dm.streams[ aindex ] )
self.aindex = aindex
break
if not self.decoder:
raise "no audio track found in given media file!"
self.resampler = sound.Resampler( (dm.streams[ aindex ][ 'sample_rate' ], dm.streams[ aindex ][ 'channels' ]),
(constants.AUDIO_SAMPLE_RATE , 1) )
self.ostream = data_stream(constants.AUDIO_SAMPLE_RATE, data_format = {'dtype':np.int16})
self.odtype = np.int16
self.demuxer = dm
self.frames = r
self.fin = fin
def play(file_path):
#file_path = "test.mp3"
root, ext = path.splitext(file_path)
demuxer = muxer.Demuxer(ext[1:].lower())
decoder = None
output = None
file = open(file_path, 'rb')
data = ' '
while data:
data = file.read(90000)
if len(data):
frames = demuxer.parse(data)
for frame in frames:
if decoder == None:
decoder = acodec.Decoder(demuxer.streams[0])
audio_frame = decoder.decode(frame[1])
if audio_frame and audio_frame.data:
if output == None:
output = sound.Output(audio_frame.sample_rate,
audio_frame.channels,
sound.AFMT_S16_LE)
#while self.stop:
#time.sleep(1)
output.play(audio_frame.data)
while output.isPlaying():
time.sleep(0.05)
def aplayer( name, card, rate, tt ):
import pymedia.muxer as muxer
import pymedia.audio.acodec as acodec
#import pymedia.audio.sound as sound
import time
dm= muxer.Demuxer( str.split( name, '.' )[ -1 ].lower() )
f= open( name, 'rb' )
#snd= resampler= dec= None
dec=None
s= f.read( 32000 )
t= 0
while len( s ):
frames= dm.parse( s )
if frames:
for fr in frames:
# Assume for now only audio streams
if dec== None:
print dm.getInfo(), dm.streams
dec= acodec.Decoder( dm.streams[ fr[ 0 ] ] )
r= dec.decode( fr[ 1 ] )
if r:
pass
#print repr(r)
s= f.read( 512 )
def dumpPCM(name):
import pymedia.audio.acodec as acodec
import pymedia.muxer as muxer
import time, wave, string, os
name1 = str.split(name, '.')
name2 = string.join(name1[:len(name1) - 1])
# Open demuxer first
dm = muxer.Demuxer(name1[-1].lower())
dec = None
f = open(name, 'rb')
snd = None
s = " "
while len(s):
s = f.read(20000)
if len(s):
frames = dm.parse(s)
for fr in frames:
if dec == None:
# Open decoder
dec = acodec.Decoder(dm.streams[0])
r = dec.decode(fr[1])
if r and r.data:
if snd == None:
snd = open(name2 + '.pcm', 'wb')
snd.write(r.data)
def run(self):
dm = muxer.Demuxer(str.split(self.mfile, '.')[-1].lower())
self.snd = dec = None
s = self.getbuff()
while len(s) and not self.stop:
frames = dm.parse(s)
if frames:
for fr in frames:
if dec == None:
dec = acodec.Decoder(dm.streams[fr[0]])
r = dec.decode(fr[1])
if r and r.data:
if self.snd == None:
self.snd = sound.Output(int(r.sample_rate),
r.channels,
sound.AFMT_S16_LE)
data = r.data
self.snd.play(data)
s = self.getbuff()
# 淡出
fade_out(self.snd)
# 停止
self.snd.stop()
# 设置音量正常
self.snd.setVolume(65535)
def aplayer(name):
import pymedia.muxer as muxer, pymedia.audio.acodec as acodec, pymedia.audio.sound as sound
import time
dm = muxer.Demuxer(str.split(name, '.')[-1].lower())
f = open(name, 'rb')
f.seek(0, 2)
size = f.tell()
f.seek(0, 0)
snd = resampler = dec = None
s = f.read(32000)
t = time.time()
while len(s):
frames = dm.parse(s)
if frames:
for fr in frames:
# Assume for now only audio streams
if dec == None:
print dm.getInfo(), dm.streams
dec = acodec.Decoder(dm.streams[fr[0]])
r = dec.decode(fr[1])
if time.time() - t > SEEK_INTERVAL:
newPos = random.randint(0, size / 2)
print 'Seek to', newPos
dec.reset()
dm.reset()
f.seek(newPos, 0)
t = time.time()
s = f.read(512)
def get_audio_analysis(audio):
dm = muxer.Demuxer('ogg')
frames = dm.parse(audio)
print len(frames)
#return dm
#get_audio_decoder
dec = acodec.Decoder(dm.streams[0])
def playAudio(filename):
import pymedia.audio.acodec as acodec
import pymedia.muxer as muxer
import pymedia.audio.sound as sound
import time
name1 = str.split(filename, '.')
# Open demuxer first
dm = muxer.Demuxer(name1[-1].lower())
dec = None
snd = None
s = " "
f = open(filename, 'rb')
while len(s):
s = f.read(20000)
if len(s):
# 解析出最初的几帧音频数据
frames = dm.parse(s)
for fr in frames:
if dec == None:
# Open decoder
dec = acodec.Decoder(dm.streams[0])
#音频数据在 frame 数组的第二个元素中
r = dec.decode(fr[1])
if r and r.data:
if snd == None:
snd = sound.Output(r.sample_rate, r.channels,
sound.AFMT_S16_LE)
snd.play(r.data)
#8.延时，直到播放完毕
while snd.isPlaying():
time.sleep(0.5)
def _getStreamParams( self, vindex, aindex, vparams, aparams, r ):
# Decode one video frame and 1 audio frame to get stream data
vDec= aDec= None
for d in r:
try:
# Demux file into streams
if d[ 0 ]== vindex:
if not vDec:
vDec= vcodec.Decoder( vparams )
vfr= vDec.decode( d[ 1 ] )
if vfr and vfr.data:
if vfr.aspect_ratio> .0:
self.pictureSize= ( int( vfr.size[ 1 ]* vfr.aspect_ratio ), vfr.size[ 1 ] )
else:
self.pictureSize= vfr.size
self.vBitRate= vfr.bitrate
if aindex== -1 or aDec:
break
elif d[ 0 ]== aindex:
if not aDec:
aDec= acodec.Decoder( aparams )
afr= aDec.decode( d[ 1 ] )
if afr and afr.data:
self.aBitRate= afr.bitrate
self.aSampleRate= afr.sample_rate
self.aChannels= afr.channels
if vindex== -1 or vDec:
break
except:
self.err.append( sys.exc_info() )
break
def decompressWAV(fname):
snd = None
fname1 = str.split(fname, '.')
fname2 = string.join(fname1[:len(fname1) - 1])
# Open demuxer first
dm = muxer.Demuxer(fname1[-1].lower())
dec = None
f = open(fname, 'rb')
s = " "
while len(s):
s = f.read(20000)
if len(s):
frames = dm.parse(s)
for fr in frames:
if dec == None:
# Open decoder
dec = acodec.Decoder(dm.streams[0])
r = dec.decode(fr[1])
if r and r.data:
if snd == None:
snd = wave.open(fname2 + '.wav', 'wb')
snd.setparams(
(r.channels, 2, r.sample_rate, 0, 'NONE', ''))
snd.writeframes(r.data)
return snd
def play(self):
self.stopFlag = True
self.mp3 = []
self.i = 0
#dm = muxer.Demuxer(str.split(self.file_name, '.')[-1].lower())
dm = muxer.Demuxer('mp3')
#f = open(self.file_name, 'rb')
snd = dec = None
header = {
#'Referer': 'http://www.xiami.com/',
'User-Agent':
'User-Agent:Mozilla/5.0 (Windows NT 6.1) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/28.0.1500.72 Safari/537.36',
}
request = urllib2.Request(self.file_name, headers=header)
self.f = urllib2.urlopen(request)
print 'Player Init Time:%s' % (time.time() - TIME)
#s = self.f.read(32000)
s = self.f.read(10000)
#thread.start_new_thread(self.download,())
self.tcpFlag = True
#self.TcpThread = tcpThread(self)
#self.TcpThread.start()
#self.downloadthread = downloadThread(self.f, self.mp3)
#self.downloadthread.start()
while self.stopFlag:
pass
print 'first read long:%s' % (len(s))
while len(s) and self.tcpFlag:
frames = dm.parse(s)
if frames:
for fr in frames:
if dec == None:
dec = acodec.Decoder(dm.streams[fr[0]])
r = dec.decode(fr[1])
if r and r.data:
if snd == None:
snd = sound.Output(int(r.sample_rate), r.channels,
sound.AFMT_S16_LE)
data = r.data
try:
snd.play(data)
except Exception, e:
print e
#sys.exit(0)
'''
if self.i<len(self.mp3):
s = self.mp3[self.i]
else:
s = []
self.i = self.i + 1
'''
s = self.f.read(2000)
def recodeAudio(fName, fOutput, type, bitrate=None):
# ------------------------------------
import pymedia.audio.acodec as acodec
import pymedia.muxer as muxer
# Open demuxer
dm = muxer.Demuxer(fName.split('.')[-1].lower())
f = open(fName, 'rb')
s = f.read(90000)
dec = enc = mx = None
print 'Recoding %s into %s' % (fName, fOutput)
while len(s):
frames = dm.parse(s)
if frames:
for fr in frames:
# Assume for now only audio streams
if dec == None:
# Open decoder
dec = acodec.Decoder(dm.streams[fr[0]])
print 'Decoder params:', dm.streams[fr[0]]
# Decode audio frame
r = dec.decode(fr[1])
if r:
if bitrate == None:
bitrate = r.bitrate
# Open muxer and encoder
if enc == None:
params = {
'id': acodec.getCodecID(type),
'bitrate': bitrate,
'sample_rate': r.sample_rate,
'channels': r.channels
}
print 'Encoder params:', params
mx = muxer.Muxer(type)
stId = mx.addStream(muxer.CODEC_TYPE_AUDIO, params)
enc = acodec.Encoder(params)
fw = open(fOutput, 'wb')
ss = mx.start()
fw.write(ss)
enc_frames = enc.encode(r.data)
if enc_frames:
for efr in enc_frames:
ss = mx.write(stId, efr)
if ss:
fw.write(ss)
s = f.read(100000)
f.close()
if fw:
if mx:
ss = mx.end()
if ss:
fw.write(ss)
fw.close()
def _initAudio( self, params ):
try:
# Reset audio stream
self._resetAudio()
# Initializing audio codec
self.ac= acodec.Decoder( params )
self.aindex= -1
except:
self.err.append( sys.exc_info() )
def aplayer(name, card, rate, tt):
import pymedia.muxer as muxer, pymedia.audio.acodec as acodec, pymedia.audio.sound as sound
import time
dm = muxer.Demuxer(str.split(name, '.')[-1].lower())
snds = sound.getODevices()
if card not in range(len(snds)):
raise 'Cannot play sound to non existent device %d out of %d' % (
card + 1, len(snds))
f = open(name, 'rb')
snd = resampler = dec = None
s = f.read(32000)
t = 0
while len(s):
frames = dm.parse(s)
if frames:
for fr in frames:
# Assume for now only audio streams
if dec == None:
print dm.getInfo(), dm.streams
dec = acodec.Decoder(dm.streams[fr[0]])
r = dec.decode(fr[1])
if r and r.data:
if snd == None:
print 'Opening sound with %d channels -> %s' % (
r.channels, snds[card]['name'])
snd = sound.Output(int(r.sample_rate * rate),
r.channels, sound.AFMT_S16_LE, card)
if rate < 1 or rate > 1:
resampler = sound.Resampler(
(r.sample_rate, r.channels),
(int(r.sample_rate / rate), r.channels))
print 'Sound resampling %d->%d' % (
r.sample_rate, r.sample_rate / rate)
data = r.data
if resampler:
data = resampler.resample(data)
if EMULATE:
# Calc delay we should wait to emulate snd.play()
d = len(data) / float(r.sample_rate * r.channels * 2)
time.sleep(d)
if int(t + d) != int(t):
print 'playing: %d sec\r' % (t + d),
t += d
else:
snd.play(data)
if tt > 0:
if snd and snd.getPosition() > tt:
break
s = f.read(512)
while snd.isPlaying():
time.sleep(.05)
def initAudio(self, params):
try:
# Reset audio stream
self.resetAudio()
# Initializing audio codec
self.ac = acodec.Decoder(params)
except:
traceback.print_exc()
self.err.append(sys.exc_info()[1])
def videoDecodeBenchmark( inFile, opt ):
dm= muxer.Demuxer( inFile.split( '.' )[ -1 ] )
f= open( inFile, 'rb' )
s= f.read( 400000 )
r= dm.parse( s )
v= filter( lambda x: x[ 'type' ]== muxer.CODEC_TYPE_VIDEO, dm.streams )
if len( v )== 0:
raise 'There is no video stream in a file %s' % inFile
v_id= v[ 0 ][ 'index' ]
print 'Assume video stream at %d index: ' % v_id
a= filter( lambda x: x[ 'type' ]== muxer.CODEC_TYPE_AUDIO, dm.streams )
if len( a )== 0:
print 'There is no audio stream in a file %s. Ignoring audio.' % inFile
opt= 'noaudio'
else:
a_id= a[ 0 ][ 'index' ]
t= time.time()
vc= vcodec.Decoder( dm.streams[ v_id ] )
print dm.streams[ v_id ]
if opt!= 'noaudio':
import pymedia.audio.acodec as acodec
import pymedia.audio.sound as sound
ac= acodec.Decoder( dm.streams[ a_id ] )
resampler= None
frames= 0
while len( s )> 0:
for fr in r:
if fr[ 0 ]== v_id:
d= vc.decode( fr[ 1 ] )
if fr[ 3 ]> 0:
print 'Video PTS', fr[ 3 ]
if d and d.data:
frames+= 1
#ff= open( 'c:\\test', 'wb' )
#ff.write( d.data[ 0 ] )
#ff.close()
elif opt!= 'noaudio' and fr[ 0 ]== a_id:
d= ac.decode( fr[ 1 ] )
if resampler== None:
if d and d.channels> 2:
resampler= sound.Resampler( (d.sample_rate,d.channels), (d.sample_rate,2) )
else:
data= resampler.resample( d.data )
s= f.read( 400000 )
r= dm.parse( s )
tt= time.time()- t
print '%d frames in %d secs( %.02f fps )' % ( frames, tt, float(frames)/tt )
def run(self):
logging.info('play started on ' + self.filename)
dm = muxer.Demuxer(str.split(self.filename, '.')[-1].lower())
snds = sound.getODevices()
if 0 not in range(len(snds)):
print 'Cannot play sound to non existent device %d out of %d' % (
1, len(snds))
self.endfunc()
return
f = open(self.filename, 'rb')
snd = resampler = dec = None
s = f.read(32000)
t = 0
while len(s):
frames = dm.parse(s)
if frames:
for fr in frames:
# Assume for now only audio streams
if dec == None:
print dm.getInfo(), dm.streams
dec = acodec.Decoder(dm.streams[fr[0]])
r = dec.decode(fr[1])
if r and r.data:
if snd == None:
print 'Opening sound with %d channels -> %s' % (
r.channels, snds[0]['name'])
snd = sound.Output(int(r.sample_rate), r.channels,
sound.AFMT_S16_LE, 0)
data = r.data
while (self.state == PAUSE):
time.sleep(.05)
if (self.state == STOP):
self.endfunc()
return
snd.play(data)
s = f.read(BUFFER_SIZE)
count = 0
while snd.isPlaying():
count = count + 1
print "playing: ", str(count)
time.sleep(.05)
while (self.state == PAUSE):
time.sleep(.05)
if (self.state == STOP):
self.endfunc()
return
self.endfunc()
def play_by_pymedia(sound_file):
file_type = str.split( sound_file, '.' )[ -1 ].lower()
#1.二进制方法读取前 10000 个字节，保证能读到第一帧音频数据
f = open( sound_file, 'rb' )
data= f.read(4096)
#data= f.read()
#2.创建合成器对象，解析出最初的几帧音频数据
import pymedia.muxer as muxer
dm = muxer.Demuxer(file_type)
frames = dm.parse( data )
#print len(frames)
#3.根据解析出来的 Mp3 编码信息，创建解码器对象
import pymedia.audio.acodec as acodec
#dec = acodec.Decoder( dm.streams[ 0 ] )
#像下面这样也行
params = {'id': acodec.getCodecID(file_type), 'bitrate': 128000, 'sample_rate': 44100, 'ext': 'mp3', 'channels': 2}
dec= acodec.Decoder(params)
if len(frames) > 1:
#4.解码第一帧音频数据
frame = frames[0]
#音频数据在 frame 数组的第二个元素中
r= dec.decode( frame[ 1 ] )
print "sample_rate:%s , channels:%s " % (r.sample_rate,r.channels)
#注意：这一步可以直接解码 r=dec.decode( data)，而不用读出第一帧音频数据
#但是开始会有一下噪音，如果是网络流纯音频数据，不包含标签信息，则不会出现杂音
else:
r=dec.decode(data)
#5.创建音频输出对象
import pymedia.audio.sound as sound
snd = sound.Output( r.sample_rate, r.channels, sound.AFMT_S16_LE )
#6.播放
if r: snd.play_all(r.data)
#7.继续读取、解码、播放
while True:
data = f.read(512)
if len(data)>0:
r = dec.decode( data )
if r: snd.play_all(r.data)
else:
break
#8.延时，直到播放完毕
import time
while snd.isPlaying(): time.sleep( .5 )
def main(self):
dm = muxer.Demuxer(self.extension)
shutdown = False
decoder = None
while self.anyReady() or not shutdown:
while self.dataReady("inbox"):
data = self.recv("inbox")
frames = dm.parse(data)
for frame in frames:
if not decoder:
# first time we get data from the demuxer
# we create the decoder
stream_index = frame[0]
decoder = acodec.Decoder(dm.streams[stream_index])
# decode our first frame
decoded = decoder.decode(frame[1])
# output the format of the audio
format = {
'channels': decoded.channels,
'sample_rate': decoded.sample_rate,
'format': pyMediaFormat2format[sound.AFMT_S16_LE],
}
self.send(format, "format")
else:
# otherwise we just decode a frame
decoded = decoder.decode(frame[1])
self.send(str(decoded.data), "outbox")
self.send(4096, "needData")
while self.dataReady("control"):
msg = self.recv("control")
if isinstance(msg, (producerFinished, shutdownMicroprocess)):
shutdown = True
self.send(msg, "signal")
if not shutdown:
self.pause()
yield 1
def aplayer( name ):
import pymedia.audio.acodec as acodec
import pymedia.audio.sound as sound
import pymedia.muxer as muxer
import time
dm= muxer.Demuxer( name.split( '.' )[ -1 ].lower() )
f= open( name, 'rb' )
s= f.read( 90000 )
frames= dm.parse( s )
dec= acodec.Decoder( dm.streams[ frames[ 0 ] [ 0 ] ] )
# dec= acodec.Decoder( str.split( name, '.' )[ -1 ].lower() )
br= None
snd= None
f.seek(0)
s= f.read( 50000 )
while len( s ):
r= dec.decode( s )
if snd== None:
print 'Opening sound with %d channels( %d bitrate )' % ( r.channels, r.bitrate )
snd= sound.Output( r.sample_rate, r.channels, sound.AFMT_S16_LE )
br= r.bitrate
if r:
snd.play( r.data )
ev= pygame.event.get()
s= f.read( 512 )
for e in ev:
if e.type== pygame.KEYDOWN:
if e.key== pygame.K_RIGHT:
# Seek forward
f.seek( SEEK_SEC* br/ 8, 1 )
dec.reset()
if e.key== pygame.K_LEFT:
# Seek forward
if f.tell()> SEEK_SEC* br/ 8:
f.seek( -SEEK_SEC* br/ 8, 1 )
dec.reset()
if e.key== pygame.K_ESCAPE:
s= ''
break
while snd.isPlaying():
time.sleep( .05 )
def playCash(self, temp):
try:
d = None
dns = 'Provider=Microsoft.Jet.OLEDB.4.0;User ID=ikmdb;Data Source=%s;Persist Security Info=False;Jet OLEDB:System database=%s' % (self._insiderdb, self._sysdb)
sql = 'SELECT TOP 1 iif ( IsNull(SCardType), "", SCardType ) AS SCardType, iif ( IsNull(InsiderMoney), "", InsiderMoney ) AS InsiderMoney FROM [InsiderInfo] WHERE InsiderNumber = "%s"' % temp['InsiderNumber']
conn = win32com.client.Dispatch(r'ADODB.Connection')
conn.Open(dns)
rs = win32com.client.Dispatch('ADODB.Recordset')
rs.Open(sql, conn, 1, 3)
while not rs.EOF:
d = { }
d['InsiderMoney'] = string.atof(str(rs('InsiderMoney')).strip())
d['SCardType'] = string.atoi(str(rs('SCardType')).strip())
rs.MoveNext()
rs.Close()
conn.Close()
if d:
d['type'] = 'cash'
d['level'] = config.memtype[d['SCardType']] + 'CASH'
temp = d
else:
return None
tt = []
if self.queue.qsize() == 0:
tt.append(temp['level'])
if int(temp['InsiderMoney']) > config.switch_money:
tt.append(str(int(temp['InsiderMoney'])))
suffix = '.mp3'
prefix = config.src
for tem in tt:
f = open((prefix + tem + suffix).encode('gbk'), 'rb')
data = f.read()
dm = muxer.Demuxer('mp3')
frames = dm.parse(data)
dec = acodec.Decoder(dm.streams[0])
r = dec.decode(data)
snd = sound.Output(r.sample_rate, r.channels, 32, config.cashSpeaker)
snd.play(r.data)
while snd.isPlaying():
time.sleep(0.1)
except BaseException:
config.logger.error(traceback.format_exc())
def run(self):
f = open(self.addr, 'rb')
data = f.read(10000)
dm = muxer.Demuxer(self.codec)
frames = dm.parse(data)
print len(frames)
dec = acodec.Decoder(dm.streams[0])
frame = frames[0]
r = dec.decode(frame[1]) #音频数据在 frame 数组的第二个元素中
print "sample_rate:%s , channels:%s " % (r.sample_rate, r.channels)
snd = sound.Output(r.sample_rate, r.channels, sound.AFMT_S16_LE)
if r: snd.play(r.data)
while True:
if not self.playing: break
data = f.read(512)
if len(data) > 0:
r = dec.decode(data)
if r: snd.play(r.data)
else:
while snd.isPlaying():
time.sleep(.5)
break
def main(self):
dm = muxer.Demuxer(self.extension)
shutdown = False
decoder = None
while self.anyReady() or not shutdown:
while self.dataReady("inbox"):
data = self.recv("inbox")
frames = dm.parse(data)
for frame in frames:
if not decoder:
stream_index = frame[0]
decoder = acodec.Decoder(dm.streams[stream_index])
raw = decoder.decode(frame[1])
data = {}
data['type'] = 'audio'
data['audio'] = str(raw.data)
data['channels'] = raw.channels
data['sample_rate'] = raw.sample_rate
data['format'] = mapping_format_from_pymedia[
sound.AFMT_S16_LE]
self.send(data, "outbox")
while self.dataReady("control"):
msg = self.recv("control")
if isinstance(msg, (producerFinished, shutdownMicroprocess)):
shutdown = True
self.send(msg, "signal")
if not shutdown:
self.pause()
yield 1
def aPlayer( name ):
global sampleFreqs, exit, snd
# Open analyzer for a mono signal
analyzer= sound.SpectrAnalyzer( 1, SAMPLES, NUM_FREQS )
resampler= None
dec= acodec.Decoder( str.split( name, '.' )[ -1 ].lower() )
f= open( name, 'rb' )
snd= None
s= f.read( 20000 )
while len( s ) and exit== 0:
r= dec.decode( s )
if snd== None:
print 'Opening sound with sample rate %d and channels %d' % ( r.sample_rate, r.channels )
snd= sound.Output( r.sample_rate, r.channels, sound.AFMT_S16_LE )
resampler= sound.Resampler( (r.sample_rate,r.channels), (r.sample_rate,1) )
if r and len( r.data ):
s1= resampler.resample( r.data )
fTmp= analyzer.asBands( BANDS, s1 )
sampleFreqs.append( ( snd.getPosition()+ snd.getLeft(), fTmp ) )
snd.play( r.data )
pp= snd.getPosition()
x= 0
while x< len( sampleFreqs ) and pp> sampleFreqs[ x ][ 0 ]:
x+= 1
for i in range( 0, x- 1 ):
del( sampleFreqs[ i ] )
s= f.read( 512 )
# Wait until complete all sounds
while snd.isPlaying():
time.sleep( .05 )
exit= 1
def mp3Thread(self, s_name):
import pymedia.audio.acodec as acodec
import pymedia.muxer as muxer
import pymedia.audio.sound as sound
dm = muxer.Demuxer( str.split( str(sName), '.' )[ -1 ].lower() )
f = open( sName, 'rb' )
s = f.read(8192)
fr = dm.parse(s)
dec = acodec.Decoder(dm.streams[0])
r = dec.decode(s)
snd= sound.Output(r.sample_rate, r.channels, sound.AFMT_S16_LE)
while len(s) > 0:
if self.stop_mp3:
break
if r: snd.play(r.data)
s = f.read(512)
try:
r = dec.decode(s)
except:
break
while snd.is_playing(): time.sleep(.05)
self.is_playing = False
def voiceRecorderPlayer(secs, delay, interval, name):
# Turn on the microphone input
open(name, 'wb').close()
mixer = sound.Mixer()
controls = mixer.getControls()
micControls= filter( lambda x: \
x[ 'connection' ]== 'Microphone' and \
x[ 'destination' ]== 'Recording Control' and \
x[ 'name' ].count( 'Volume' )> 0, controls )
if len(micControls):
micControls[0]['control'].setActive()
print 'Setting microphone input as active'
# Minimum set of parameters we need to create Encoder
cparams = {
'id': acodec.getCodecID('mp3'),
'bitrate': 128000,
'sample_rate': 44100,
'channels': 2
}
ac = acodec.Encoder(cparams)
dm = muxer.Demuxer('mp3')
dc = acodec.Decoder(cparams)
sndIn = sound.Input(44100, 2, sound.AFMT_S16_LE)
sndOut = sound.Output(44100, 2, sound.AFMT_S16_LE)
sndIn.start()
bytesWritten = 0
bytesRead = 0
t = time.time()
paused = False
# Loop until recorded position greater than the limit specified
while sndIn.getPosition() <= secs:
s = sndIn.getData()
if s and len(s):
f = open(name, 'ab')
for fr in ac.encode(s):
# We definitely should use mux first, but for
# simplicity reasons this way it'll work also
f.write(fr)
bytesWritten += len(fr)
f.close()
else:
# 1/10 of the second buffer for playback
if sndOut.getLeft() < 0.1 and not paused:
if bytesRead < bytesWritten + READ_CHUNK:
f = open(name, 'rb')
f.seek(bytesRead)
s = f.read(READ_CHUNK)
f.close()
bytesRead += len(s)
frames = dm.parse(s)
if frames:
for fr in frames:
r = dc.decode(fr[1])
if r and r.data:
sndOut.play(r.data)
else:
time.sleep(.01)
# See if pause should be working
i = int((time.time() - t) % (interval + delay))
paused = i > interval
# Stop listening the incoming sound from the microphone or line in
sndIn.stop()
# 建立Demuxer
demuxer = muxer.Demuxer(str.split(name, '.')[-1].lower())
# 讀取MP3檔案
f = open(name, 'rb')
stream = f.read(32000)
# 持續的播放stream
while len(stream):
frames = demuxer.parse(stream)
if frames:
for fr in frames:
if dec == None:
# 輸出MP3資料
print demuxer.getHeaderInfo()
print demuxer.streams
dec = acodec.Decoder(demuxer.streams[fr[0]])
r = dec.decode(fr[1])
if r and r.data:
if mysound == None:
mysound = sound.Output(r.sample_rate, r.channels, sound.AFMT_S16_LE)
data = r.data
# 播放
mysound.play(data)
stream = f.read(512)
def videoDecodeBenchmark(inFile, opt):
pygame.init()
pygame.display.set_mode((800, 600), 0)
ovl = None
dm = muxer.Demuxer(inFile.split('.')[-1])
f = open(inFile, 'rb')
s = f.read(400000)
r = dm.parse(s)
v = filter(lambda x: x['type'] == muxer.CODEC_TYPE_VIDEO, dm.streams)
if len(v) == 0:
raise 'There is no video stream in a file %s' % inFile
v_id = v[0]['index']
print 'Assume video stream at %d index: ' % v_id
a = filter(lambda x: x['type'] == muxer.CODEC_TYPE_AUDIO, dm.streams)
if len(a) == 0:
print 'There is no audio stream in a file %s. Ignoring audio.' % inFile
opt = 'noaudio'
else:
a_id = a[0]['index']
t = time.time()
vc = vcodec.Decoder(dm.streams[v_id])
print dm.streams[v_id]
if opt != 'noaudio':
ac = acodec.Decoder(dm.streams[a_id])
resampler = None
frames = 0
q = []
while len(s) > 0:
for fr in r:
if fr[0] == v_id:
d = vc.decode(fr[1])
if d and d.data:
frames += 1
#ff= open( 'c:\\test', 'wb' )
#ff.write( d.data[ 0 ] )
#ff.close()
if not ovl:
ovl = pygame.Overlay(YV12, d.size)
q.append(d)
if len(q) > 4:
try:
ovl.set_data(q[0].data)
ovl.display()
except:
ovl.display(q[0].data)
del (q[0])
elif opt != 'noaudio' and fr[0] == a_id:
d = ac.decode(fr[1])
if resampler == None:
if d and d.channels > 2:
resampler = sound.Resampler(
(d.sample_rate, d.channels), (d.sample_rate, 2))
else:
data = resampler.resample(d.data)
s = f.read(400000)
r = dm.parse(s)
tt = time.time() - t
print '%d frames in %d secs( %.02f fps )' % (frames, tt,
float(frames) / tt)
ev = pygame.event.get()
for e in ev:
if e.type == pygame.KEYDOWN and e.key == pygame.K_ESCAPE:
s = ''
break
