def audio_encoding(self, codec_name):
try:
codec = Codec(codec_name, 'w')
except UnknownCodecError:
raise SkipTest()
ctx = codec.create()
if ctx.codec.experimental:
raise SkipTest()
sample_fmt = ctx.codec.audio_formats[-1].name
sample_rate = 48000
channel_layout = "stereo"
channels = 2
ctx.time_base = Fraction(1) / sample_rate
ctx.sample_rate = sample_rate
ctx.format = sample_fmt
ctx.layout = channel_layout
ctx.channels = channels
ctx.open()
resampler = AudioResampler(sample_fmt, channel_layout, sample_rate)
container = av.open(
fate_suite('audio-reference/chorusnoise_2ch_44kHz_s16.wav'))
audio_stream = container.streams.audio[0]
path = self.sandboxed('encoder.%s' % codec_name)
samples = 0
packet_sizes = []
with open(path, 'wb') as f:
for frame in iter_frames(container, audio_stream):
# We need to let the encoder retime.
frame.pts = None
"""
bad_resampler = AudioResampler(sample_fmt, "mono", sample_rate)
bad_frame = bad_resampler.resample(frame)
with self.assertRaises(ValueError):
next(encoder.encode(bad_frame))
bad_resampler = AudioResampler(sample_fmt, channel_layout, 3000)
bad_frame = bad_resampler.resample(frame)
with self.assertRaises(ValueError):
next(encoder.encode(bad_frame))
bad_resampler = AudioResampler('u8', channel_layout, 3000)
bad_frame = bad_resampler.resample(frame)
with self.assertRaises(ValueError):
next(encoder.encode(bad_frame))
"""
resampled_frame = resampler.resample(frame)
samples += resampled_frame.samples
for packet in ctx.encode(resampled_frame):
# bytearray because python can
# freaks out if the first byte is NULL
f.write(bytearray(packet))
packet_sizes.append(packet.size)
for packet in ctx.encode(None):
packet_sizes.append(packet.size)
f.write(bytearray(packet))
ctx = Codec(codec_name, 'r').create()
ctx.time_base = Fraction(1) / sample_rate
ctx.sample_rate = sample_rate
ctx.format = sample_fmt
ctx.layout = channel_layout
ctx.channels = channels
ctx.open()
result_samples = 0
# should have more asserts but not sure what to check
# libav and ffmpeg give different results
# so can really use checksums
for frame in iter_raw_frames(path, packet_sizes, ctx):
result_samples += frame.samples
self.assertEqual(frame.rate, sample_rate)
self.assertEqual(len(frame.layout.channels), channels)
def audio_encoding(self, codec_name):
try:
codec = Codec(codec_name, "w")
except UnknownCodecError:
raise SkipTest()
ctx = codec.create()
if ctx.codec.experimental:
raise SkipTest()
sample_fmt = ctx.codec.audio_formats[-1].name
sample_rate = 48000
channel_layout = "stereo"
channels = 2
ctx.time_base = Fraction(1) / sample_rate
ctx.sample_rate = sample_rate
ctx.format = sample_fmt
ctx.layout = channel_layout
ctx.channels = channels
ctx.open()
resampler = AudioResampler(sample_fmt, channel_layout, sample_rate)
container = av.open(
fate_suite("audio-reference/chorusnoise_2ch_44kHz_s16.wav"))
audio_stream = container.streams.audio[0]
path = self.sandboxed("encoder.%s" % codec_name)
samples = 0
packet_sizes = []
with open(path, "wb") as f:
for frame in iter_frames(container, audio_stream):
resampled_frames = resampler.resample(frame)
for resampled_frame in resampled_frames:
samples += resampled_frame.samples
for packet in ctx.encode(resampled_frame):
packet_sizes.append(packet.size)
f.write(packet)
for packet in ctx.encode(None):
packet_sizes.append(packet.size)
f.write(packet)
ctx = Codec(codec_name, "r").create()
ctx.sample_rate = sample_rate
ctx.format = sample_fmt
ctx.layout = channel_layout
ctx.channels = channels
ctx.open()
result_samples = 0
# should have more asserts but not sure what to check
# libav and ffmpeg give different results
# so can really use checksums
for frame in iter_raw_frames(path, packet_sizes, ctx):
result_samples += frame.samples
self.assertEqual(frame.rate, sample_rate)
self.assertEqual(len(frame.layout.channels), channels)
def audio_encoding(self, codec_name):
try:
codec = Codec(codec_name, 'w')
except UnknownCodecError:
raise SkipTest()
ctx = codec.create()
if ctx.codec.experimental:
raise SkipTest()
sample_fmt = ctx.codec.audio_formats[-1].name
sample_rate = 48000
channel_layout = "stereo"
channels = 2
ctx.time_base = Fraction(1) / sample_rate
ctx.sample_rate = sample_rate
ctx.format = sample_fmt
ctx.layout = channel_layout
ctx.channels = channels
ctx.open()
resampler = AudioResampler(sample_fmt, channel_layout, sample_rate)
container = av.open(fate_suite('audio-reference/chorusnoise_2ch_44kHz_s16.wav'))
audio_stream = container.streams.audio[0]
path = self.sandboxed('encoder.%s' % codec_name)
samples = 0
packet_sizes = []
with open(path, 'wb') as f:
for frame in iter_frames(container, audio_stream):
# We need to let the encoder retime.
frame.pts = None
"""
bad_resampler = AudioResampler(sample_fmt, "mono", sample_rate)
bad_frame = bad_resampler.resample(frame)
with self.assertRaises(ValueError):
next(encoder.encode(bad_frame))
bad_resampler = AudioResampler(sample_fmt, channel_layout, 3000)
bad_frame = bad_resampler.resample(frame)
with self.assertRaises(ValueError):
next(encoder.encode(bad_frame))
bad_resampler = AudioResampler('u8', channel_layout, 3000)
bad_frame = bad_resampler.resample(frame)
with self.assertRaises(ValueError):
next(encoder.encode(bad_frame))
"""
resampled_frame = resampler.resample(frame)
samples += resampled_frame.samples
for packet in ctx.encode(resampled_frame):
# bytearray because python can
# freaks out if the first byte is NULL
f.write(bytearray(packet))
packet_sizes.append(packet.size)
for packet in ctx.encode(None):
packet_sizes.append(packet.size)
f.write(bytearray(packet))
ctx = Codec(codec_name, 'r').create()
ctx.time_base = Fraction(1) / sample_rate
ctx.sample_rate = sample_rate
ctx.format = sample_fmt
ctx.layout = channel_layout
ctx.channels = channels
ctx.open()
result_samples = 0
# should have more asserts but not sure what to check
# libav and ffmpeg give different results
# so can really use checksums
for frame in iter_raw_frames(path, packet_sizes, ctx):
result_samples += frame.samples
self.assertEqual(frame.rate, sample_rate)
self.assertEqual(len(frame.layout.channels), channels)