(This is not an official Google product!)
Live Transcribe is an Android application that provides real-time captioning for people who are deaf or hard of hearing. This repository contains the Android client libraries for communicating with Google's Cloud Speech API that are used in Live Transcribe.
The automatic speech recognition (ASR) module has the following features:
- Infinite streaming
- Support for 70+ languages
- Robust to brief network loss (which occurs often when traveling and switching between network/wifi). Text is not lost, only delayed.
- Robust to extended network loss. Will reconnect again even if network has been out for hours. Of course, no speech recognition can be delivered without a connection.
- Robust to server errors
- Opus, AMR-WB, FLAC encoding can be easily enabled and configured.
- Contains a text formatting library for visualizing ASR confidence, speaker ID, and more
- Extensible to offline models
- Built-in support for speech detectors, which can be used to stop ASR during extended silences to save money and data (Note that speech detector implementation is not provided)
- Built-in support for speaker identification, which can be used to label or color text according to speaker number (Note that speaker identification implementation is not provided)
The libraries provided are nearly identical to those running in the production application Live Transcribe. They have been extensively field tested and unit tested. However, the tests themselves are not open sourced at this time.
Contact lt-speech-engine-maintainers@google.com with questions/issues.
To try this library out using our sample Android application, follow the instructions below. These instructions assume that the host operating system is Linux.
We have also provided APKs so that you can try out the library without building any code.
Whether you're using our code or our sample APKs, an API key is required. See the documentation on API keys to learn more. In the sample APK, you can copy/paste your API key into the pop-up dialog.
Requirements: CMake, Gradle, Android SDK/NDK
(1) Export the path of your Android SDK. The NDK is assumed to be located at
$ANDROID_SDK_PATH/ndk-bundle [Android NDK](https://developer.android.com/ndk/guides)
export ANDROID_SDK_PATH="/wherever/your/sdk/is"
(2) Build the APKs (location is app/build/outputs/apk/)
./build_all.sh
The purpose of this library is to simulate an infinite connection to Google's Cloud Speech API. The API itself currently does not support infinite streaming and forces a timeout after 5 minutes of streaming. It turns out that there are a lot of subtleties involved in covering up this timeout, and we believe that not everyone should have to figure this out on their own. The main logic for managing streaming requests is the RepeatingRecognitionSession class. This class maintains a single bidirectional streaming request, called a session, and takes measures to avoid hitting the timeout. Namely, we use the following strategies:
- Try to close sessions during pauses once the server-side timeout time is approaching (this is done using the is_final field from the returned speech results as they are an estimate of when a pause has happened).
- Close sessions that have been silent for a very long time prior to hitting the timeout. If someone were to start talking, it is better that that be towards the beginning of the session.
- Store a buffer of audio in between sessions and send it once the new session starts. The device may jump from one network to another, or from network to WiFi. Even on a steady connection, it can take more than the typical hundred or so milliseconds to open a new session after the previous one has closed. This buffer is crucial for making sure all audio gets to the server.
In most cases, the best model to use for general purpose recognition is the default model. For languages where it is available, the "video" model has much better performance.
// As seen in CloudSpeechSession.java
RecognitionConfig.Builder.newBuilder()
.setModel("video")
... // Other options.
.build()
At the time of writing this, the video model exists only for the "en-US" locale and is offered at a different price point.
Note that server performance can have a serious impact on result quality.
There are other concerns that arise when trying to stream infinitely. In some countries, data is expensive and sending uncompressed PCM formatted audio is simply not practical. At 16kHz, streaming uncompressed 16-bit PCM data requires 256 kilobits per second of data, ignoring the comparatively small overhead of header/auxiliary data. On low-bandwidth connections, this is too high of a data rate to reliably use the cloud for recognition. It becomes necessary to use a codec. Of the codecs supported by the speech APIs, we experimented with FLAC, AMR-WB, and Opus (in an Ogg container). For the former two, we leverage the Android framework's encoder. FLAC is a lossless codec (unlike most audio codecs) and will get you roughly a factor of 2 in data compression. It introduces a few hundred milliseconds of latency, but is quite acceptable in most cases. AMR-WB offers a much more appealing compression ratio, but in relatively noisy conditions performs very badly for speech recognition. We do not recommend using AMR-WB for speech recognition under any circumstances.
Finally, the Opus codec delivers quite impressive results for speech recognition. Unfortunately, the Android framework does not ship with an Opus encoder, so we included a native implementation in our library. At rates at least as low as 24 kilobits per second (a compression ratio of nearly 11), recognition quality does not seem to be impacted at all. We know that with captions, accuracy is critical, so in Live Transcribe, we configured our Opus codec to use a more conservative 32 kbps with variable bitrate (VBR) enabled. This is sufficient for minimizing data streaming costs (note that music streaming services use bitrates many times higher than this). However, there is still a bit of latency associated with Opus compression. There is one more fairly technical detail that we use to minimize latency in our Opus stream. For every block of audio that is pushed into the Ogg/Opus stream, we flush the Ogg stream rather than let the ogg library decide on its own when to push out the next block of data. This causes a slight increase in bitrate, but a significant reduction in latency. For the curious, deep in our encoder is a "low_latency_mode" flag. As a user of this library nothing need be done to enable that. Just request the following settings for your CloudSpeechSessionParams:
CloudSpeechSessionParams.newBuilder()
.setEncoderParams(CloudSpeechSessionParams.EncoderParams.newBuilder()
.setEnableEncoder(true)
.setAllowVbr(true)
.setCodec(CodecAndBitrate.OGG_OPUS_BITRATE_32KBPS))
.build()