googleapis · lukesneeringer · Feb 9, 2018 · Feb 5, 2018 · Feb 8, 2018
diff --git a/speech/google/cloud/speech_v1p1beta1/__init__.py b/speech/google/cloud/speech_v1p1beta1/__init__.py
@@ -0,0 +1,34 @@
+# Copyright 2018 Google LLC
+#
+# Licensed under the Apache License, Version 2.0 (the "License");
+# you may not use this file except in compliance with the License.
+# You may obtain a copy of the License at
+#
+#     https://www.apache.org/licenses/LICENSE-2.0
+#
+# Unless required by applicable law or agreed to in writing, software
+# distributed under the License is distributed on an "AS IS" BASIS,
+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+# See the License for the specific language governing permissions and
+# limitations under the License.
+
+from __future__ import absolute_import
+
+from google.cloud.speech_v1p1beta1 import types
+from google.cloud.speech_v1p1beta1.gapic import enums
+from google.cloud.speech_v1p1beta1.gapic import speech_client
+
+from google.cloud.speech_v1.helpers import SpeechHelpers
+
+
+class SpeechClient(SpeechHelpers, speech_client.SpeechClient):
+    __doc__ = speech_client.SpeechClient.__doc__
+    enums = enums
+    types = types
+
+
+__all__ = (
+    'enums',
+    'types',
+    'SpeechClient',
+)
diff --git a/speech/google/cloud/speech_v1p1beta1/gapic/__init__.py b/speech/google/cloud/speech_v1p1beta1/gapic/__init__.py
diff --git a/speech/google/cloud/speech_v1p1beta1/gapic/enums.py b/speech/google/cloud/speech_v1p1beta1/gapic/enums.py
@@ -0,0 +1,95 @@
+# Copyright 2018 Google LLC
+#
+# Licensed under the Apache License, Version 2.0 (the "License");
+# you may not use this file except in compliance with the License.
+# You may obtain a copy of the License at
+#
+#     https://www.apache.org/licenses/LICENSE-2.0
+#
+# Unless required by applicable law or agreed to in writing, software
+# distributed under the License is distributed on an "AS IS" BASIS,
+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+# See the License for the specific language governing permissions and
+# limitations under the License.
+"""Wrappers for protocol buffer enum types."""
+
+
+class RecognitionConfig(object):
+    class AudioEncoding(object):
+        """
+        The encoding of the audio data sent in the request.
+
+        All encodings support only 1 channel (mono) audio.
+
+        If you send a ``FLAC`` or ``WAV`` audio file format in the request,
+        then if you specify an encoding in ``AudioEncoding``, it must match the
+        encoding described in the audio header. If it does not match, then the
+        request returns an
+        ``google.rpc.Code.INVALID_ARGUMENT`` error code. You can request
+        recognition for ``WAV`` files that contain either ``LINEAR16`` or ``MULAW``
+        encoded audio.
+        For audio file formats other than ``FLAC`` or ``WAV``, you must
+        specify the audio encoding in your ``RecognitionConfig``.
+
+        For best results, the audio source should be captured and transmitted using
+        a lossless encoding (``FLAC`` or ``LINEAR16``). The accuracy of the speech
+        recognition can be reduced if lossy codecs, which include the other codecs
+        listed in this section, are used to capture or transmit the audio,
+        particularly if background noise is present.
+
+        Attributes:
+          ENCODING_UNSPECIFIED (int): Not specified.
+          LINEAR16 (int): Uncompressed 16-bit signed little-endian samples (Linear PCM).
+          FLAC (int): ```FLAC`` <https://xiph.org/flac/documentation.html>`_ (Free Lossless Audio
+          Codec) is the recommended encoding because it is
+          lossless--therefore recognition is not compromised--and
+          requires only about half the bandwidth of ``LINEAR16``. ``FLAC`` stream
+          encoding supports 16-bit and 24-bit samples, however, not all fields in
+          ``STREAMINFO`` are supported.
+          MULAW (int): 8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.
+          AMR (int): Adaptive Multi-Rate Narrowband codec. ``sample_rate_hertz`` must be 8000.
+          AMR_WB (int): Adaptive Multi-Rate Wideband codec. ``sample_rate_hertz`` must be 16000.
+          OGG_OPUS (int): Opus encoded audio frames in Ogg container
+          (`OggOpus <https://wiki.xiph.org/OggOpus>`_).
+          ``sample_rate_hertz`` must be one of 8000, 12000, 16000, 24000, or 48000.
+          SPEEX_WITH_HEADER_BYTE (int): Although the use of lossy encodings is not recommended, if a very low
+          bitrate encoding is required, ``OGG_OPUS`` is highly preferred over
+          Speex encoding. The `Speex <https://speex.org/>`_  encoding supported by
+          Cloud Speech API has a header byte in each block, as in MIME type
+          ``audio/x-speex-with-header-byte``.
+          It is a variant of the RTP Speex encoding defined in
+          `RFC 5574 <https://tools.ietf.org/html/rfc5574>`_.
+          The stream is a sequence of blocks, one block per RTP packet. Each block
+          starts with a byte containing the length of the block, in bytes, followed
+          by one or more frames of Speex data, padded to an integral number of
+          bytes (octets) as specified in RFC 5574. In other words, each RTP header
+          is replaced with a single byte containing the block length. Only Speex
+          wideband is supported. ``sample_rate_hertz`` must be 16000.
+        """
+        ENCODING_UNSPECIFIED = 0
+        LINEAR16 = 1
+        FLAC = 2
+        MULAW = 3
+        AMR = 4
+        AMR_WB = 5
+        OGG_OPUS = 6
+        SPEEX_WITH_HEADER_BYTE = 7
+
+
+class StreamingRecognizeResponse(object):
+    class SpeechEventType(object):
+        """
+        Indicates the type of speech event.
+
+        Attributes:
+          SPEECH_EVENT_UNSPECIFIED (int): No speech event specified.
+          END_OF_SINGLE_UTTERANCE (int): This event indicates that the server has detected the end of the user's
+          speech utterance and expects no additional speech. Therefore, the server
+          will not process additional audio (although it may subsequently return
+          additional results). The client should stop sending additional audio
+          data, half-close the gRPC connection, and wait for any additional results
+          until the server closes the gRPC connection. This event is only sent if
+          ``single_utterance`` was set to ``true``, and is not used otherwise.
+        """
+        SPEECH_EVENT_UNSPECIFIED = 0
+        END_OF_SINGLE_UTTERANCE = 1