def _al_play(self): context.lock() state = al.ALint() al.alGetSourcei(self._al_source, al.AL_SOURCE_STATE, state) if state.value != al.AL_PLAYING: al.alSourcePlay(self._al_source) context.unlock()
def _al_play(self): if _debug: print 'OpenALAudioPlayer._al_play()' with context.lock: state = al.ALint() al.alGetSourcei(self._al_source, al.AL_SOURCE_STATE, state) if state.value != al.AL_PLAYING: al.alSourcePlay(self._al_source)
def refill(self, write_size): if _debug: print 'refill', write_size self._lock.acquire() while write_size > self._min_buffer_size: audio_data = self.source_group.get_audio_data(write_size) if not audio_data: self._eos = True self._events.append( (self._write_cursor, MediaEvent(0, 'on_eos'))) self._events.append( (self._write_cursor, MediaEvent(0, 'on_source_group_eos'))) break for event in audio_data.events: cursor = self._write_cursor + event.timestamp * \ self.source_group.audio_format.bytes_per_second self._events.append((cursor, event)) context.lock() buffer = bufferPool.getBuffer(self._al_source) al.alBufferData(buffer, self._al_format, audio_data.data, audio_data.length, self.source_group.audio_format.sample_rate) if _debug_buffers: error = al.alGetError() if error != 0: print("BUFFER DATA ERROR: " + str(error)) al.alSourceQueueBuffers(self._al_source, 1, ctypes.byref(buffer)) if _debug_buffers: error = al.alGetError() if error != 0: print("QUEUE BUFFER ERROR: " + str(error)) context.unlock() self._write_cursor += audio_data.length self._buffer_sizes.append(audio_data.length) self._buffer_timestamps.append(audio_data.timestamp) write_size -= audio_data.length # Check for underrun stopping playback if self._playing: state = al.ALint() context.lock() al.alGetSourcei(self._al_source, al.AL_SOURCE_STATE, state) if state.value != al.AL_PLAYING: if _debug: print 'underrun' al.alSourcePlay(self._al_source) context.unlock() self._lock.release()
def refill(self, write_size): if _debug: print "refill", write_size self._lock.acquire() while write_size > self._min_buffer_size: audio_data = self.source_group.get_audio_data(write_size) if not audio_data: self._eos = True self._events.append((self._write_cursor, MediaEvent(0, "on_eos"))) self._events.append((self._write_cursor, MediaEvent(0, "on_source_group_eos"))) break for event in audio_data.events: cursor = self._write_cursor + event.timestamp * self.source_group.audio_format.bytes_per_second self._events.append((cursor, event)) buffer = al.ALuint() context.lock() al.alGenBuffers(1, buffer) al.alBufferData( buffer, self._al_format, audio_data.data, audio_data.length, self.source_group.audio_format.sample_rate ) al.alSourceQueueBuffers(self._al_source, 1, ctypes.byref(buffer)) context.unlock() self._write_cursor += audio_data.length self._buffer_sizes.append(audio_data.length) self._buffer_timestamps.append(audio_data.timestamp) write_size -= audio_data.length # Check for underrun stopping playback if self._playing: state = al.ALint() context.lock() al.alGetSourcei(self._al_source, al.AL_SOURCE_STATE, state) if state.value != al.AL_PLAYING: if _debug: print "underrun" al.alSourcePlay(self._al_source) context.unlock() self._lock.release()
alcontext = alc.alcCreateContext(device, None) alc.alcMakeContextCurrent(alcontext) source = al.ALuint() al.alGenSources(1, source) pool = BufferPool() # avcodec init() container = open(sys.argv[1]) stream = get_audio_stream(container) decoder = AVCodecDecoder(container, stream) while True: buffer = pool.get_buffer() if not decoder.fill_buffer(buffer): break al.alSourceQueueBuffers(source, 1, buffer) al.alSourcePlay(source) while True: value = al.ALint() al.alGetSourcei(source, al.AL_SOURCE_STATE, value) if value.value == al.AL_STOPPED: break time.sleep(1)
class AVCodecDecoder: def __init__(self, container, stream): self.codec = get_codec(container, stream) if self.codec.contents.channels == 1: self.buffer_format = al.AL_FORMAT_MONO16 elif self.codec.contents.channels == 2: self.buffer_format = al.AL_FORMAT_STEREO16 else: raise Exception('Invalid number of channels') self.sample_rate = self.codec.contents.sample_rate self.container = container self.stream = stream self.packet = avformat.AVPacket() self.sample_buffer = \ (ctypes.c_int16 * (avcodec.AVCODEC_MAX_AUDIO_FRAME_SIZE / 2))() self.read_packet() def read_packet(self): while True: if self.packet.data: self.packet.data = None self.packet.size = 0 if avformat.av_read_packet(self.container, self.packet) < 0: break if self.packet.stream_index == self.stream: break self.packet_data = self.packet.data self.packet_size = self.packet.size def fill_buffer(self, buffer): if self.packet_size <= 0: self.read_packet() if self.packet_size == 0: # EOS return False sample_buffer_size = ctypes.c_int() len = avcodec.avcodec_decode_audio(self.codec, self.sample_buffer, sample_buffer_size, self.packet_data, self.packet_size) if len < 0: # Error, skip frame raise Exception('frame error TODO') if sample_buffer_size.value > 0: al.alBufferData(buffer, self.buffer_format, self.sample_buffer, sample_buffer_size.value, self.sample_rate) # Advance buffer pointer self.packet_data = ctypes.c_uint8.from_address( ctypes.addressof(self.packet_data) + len) self.packet_size -= len return True import sys # openal device = alc.alcOpenDevice(None) if not device: raise Exception('No OpenAL device.') alcontext = alc.alcCreateContext(device, None) alc.alcMakeContextCurrent(alcontext) source = al.ALuint() al.alGenSources(1, source) pool = BufferPool() # avcodec init() container = open(sys.argv[1]) stream = get_audio_stream(container) decoder = AVCodecDecoder(container, stream) while True: buffer = pool.get_buffer() if not decoder.fill_buffer(buffer): break al.alSourceQueueBuffers(source, 1, buffer) al.alSourcePlay(source) while True: value = al.ALint() al.alGetSourcei(source, al.AL_SOURCE_STATE, value) if value.value == al.AL_STOPPED: break time.sleep(1)