Systems and methods for providing real-time audio and data
A method of delivering audio to one or more client computing devices includes receiving, by an audio server computing device, a live audio signal starting at a first time; processing, by the audio server computing device, the live audio signal, thereby creating a data representation of the live audio signal; transmitting, by the audio server computing device, via a wireless network in electronic communication with the audio server computing device, the data representation of the live audio signal to the one or more client computing devices; interpreting, by the one or more client computing devices, the data representation of the live audio signal, thereby producing an interpreted audio signal; and providing, by the one or more client computing devices, the interpreted audio signal to a user listening device starting at a second time. A latency between the first time and the second time is less than 100 milliseconds.
1. A computerized method of delivering audio to one or more client computing devices, wherein the audio is any real-time audio of a live event and the one or more client computing devices comprise smartphones of attendees physically present at the live event, the method comprising:
receiving, by an audio server computing device, a live audio signal starting at a first time;
processing, by the audio server computing device, the live audio signal, thereby creating a data representation of the live audio signal using a compression codec, the data representation including one or more audio data packets;
transmitting, by the audio server computing device, via a UDP-configured wireless network access point in electronic communication with the audio server computing device, the data representation of the live audio signal to the one or more client computing devices;
interpreting, by the one or more client computing devices, the data representation of the live audio signal, thereby producing a continuous, stable interpreted audio signal; and
providing, by the one or more client computing devices, the continuous, stable interpreted audio signal to a user listening device starting at a second time, wherein a latency between the first time and the second time is less than 100 milliseconds to minimize asynchronization between a live performance aspect of the live event and a corresponding portion of the continuous, stable interpreted audio signal.
2. The computerized method of claim 1 , wherein the wireless network is configured to transmit multiple duplicates of each audio data packet within a same transmission to mitigate data packet loss.
3. The computerized method of claim 1 , wherein the audio server computing device uses location information provided by the one or more client computing devices to optimize an audio mix delivered to the one or more client computing devices, wherein the audio mix includes the continuous, stable interpreted audio signal.
4. The computerized method of claim 1 , wherein the interpreted audio signal is optimized to account for surrounding or ambient audio to avoid an echo effect.
5. The computerized method of claim 1 , wherein the audio delivered to the one or more client computing devices includes two or more simultaneous live streaming audio signals.
6. The computerized method of claim 5 , wherein the one or more client computing devices are configured to mix the two or more simultaneous live streaming audio signals according to user-specified criteria.
7. The computerized method of claim 1 , wherein the one or more client computing devices are configured to mix the interpreted audio signal with audio provided by microphone equipment of the one or more client computing devices.
8. The computerized method of claim 1 , wherein the one or more client computing devices are configured to synchronize the interpreted audio signal with video provided by camera equipment of the one or more client computing devices.
9. The computerized method of claim 1 , wherein the latency between the first time and the second time is buffered by manually setting a buffered delay on an audio application installed on the one or more client computing devices.
10. The computerized method of claim 1 , wherein the latency between the first time and the second time is automatically buffered by an audio application installed on the one or more client computing devices.
11. A system for delivering audio to a listener, wherein the audio is any real-time audio of a live event, the system comprising:
an audio server computing device configured to receive a live audio signal starting at a first time and to convert the live audio signal into a data representation of the live audio signal using a compression codec, the data representation including one or more audio data packets, the audio server computing device including:
an audio hardware interface;
a server computer in electronic communication with the audio hardware interface; and
specialized software installed on the server computer, the specialized software including a hardware driver and media processing software;
a UDP-configured wireless network access point configured to communicate electronically with network hardware of the audio server computing device and to transmit the data representation of the live audio signal; and
one or more client computing devices configured to receive the data representation of the live audio signal via the wireless network access point, interpret the data representation of the live audio signal, thereby producing a continuous, stable interpreted audio signal, and
provide the continuous, stable interpreted audio signal to one or more user listening devices starting at a second time, wherein the one or more client computing devices comprise smartphones of attendees physically present at the live event, the one or more client computing devices each including:
a hardware wireless network interface; and
an audio application installed on the client computing device configured to interpret the data representation of the live audio signal, thereby producing the continuous, stable interpreted audio signal, wherein a latency between the first time and the second time is less than 100 milliseconds to minimize asynchronization between a live performance aspect of the live event and a corresponding portion of the continuous, stable interpreted audio signal.
12. The system of claim 11 , wherein the wireless network is configured to transmit multiple duplicates of each audio data packet within a same transmission to mitigate data packet loss.
13. The system of claim 11 , wherein the audio server computing device uses location information provided by the client computing device to optimize an audio mix delivered to the one or more client computing devices, wherein the audio mix includes the continuous, stable interpreted audio signal.
14. The system of claim 11 , wherein the interpreted audio signal is optimized to account for surrounding or ambient audio to avoid an echo effect.
15. The system of claim 11 , wherein the audio delivered to the one or more client computing devices includes two or more simultaneous live streaming audio signals.
16. The system of claim 15 , wherein the one or more client computing devices are configured to mix the two or more simultaneous live streaming audio signals according to user-specified criteria.
17. The system of claim 11 , wherein the one or more client computing devices are configured to mix the interpreted audio signal with audio provided by microphone equipment of the one or more client computing devices.
18. The system of claim 11 , wherein the one or more client computing devices are configured to synchronize the interpreted audio signal with video provided by camera equipment of the one or more client computing devices.
19. The system of claim 11 , wherein the audio application installed on the client computing device is further configured to buffer the latency between the first time and the second time based on a manually set buffered delay.
20. The system of claim 11 , wherein the audio application installed on the client computing device is further configured to automatically buffer the latency between the first time and the second time.