WebRtcAudioTrack is hardcoded to configure AudioAttributes with
1. usage=USAGE_VOICE_COMMUNICATIOON
2. contentType=CONTENT_TYPE_SPEECH
This change allows AudioAttributes to be configured via the
JavaAudioDeviceModule.
Bug: webrtc:12153
Change-Id: I67c7f6e572c5a9f3a8fde674b6600d2adaf17895
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191941
Commit-Queue: Gaurav Vaish <gvaish@chromium.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32583}
This is a reland of 1b8ef63876ebfa55a51c8ca9b1d8206bf8233e01. It was
previously reverted (https://webrtc-review.googlesource.com/c/src/+/175008)
but the revert was found to be unnecessary.
Original change's description:
> Add an optional override for AudioRecord device
>
> This is important when we have multiple named devices connected over
> USB (eg. "Webcam", "Microphone", "Headset") and there is some way to
> choose a specific input device to route from.
>
> Bug: b/154440591
> Change-Id: I8dc1801a5e4db7f7bb439e855d43897c1f7d8bc4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173748
> Commit-Queue: Robin Lee <rgl@google.com>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31130}
TBR=henrika@webrtc.org,sakal@webrtc.org,rgl@google.com
Bug: b/154440591, b/155256727
Change-Id: Ic9bf8305c85552a0dc0d2cde6190988423e7fc70
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175084
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31255}
This reverts commit 1b8ef63876ebfa55a51c8ca9b1d8206bf8233e01.
Reason for revert: Breaks downstream projects. b/155256727
Original change's description:
> Add an optional override for AudioRecord device
>
> This is important when we have multiple named devices connected over
> USB (eg. "Webcam", "Microphone", "Headset") and there is some way to
> choose a specific input device to route from.
>
> Bug: b/154440591
> Change-Id: I8dc1801a5e4db7f7bb439e855d43897c1f7d8bc4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173748
> Commit-Queue: Robin Lee <rgl@google.com>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31130}
TBR=henrika@webrtc.org,sakal@webrtc.org,rgl@google.com
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: b/154440591, b/155256727
Change-Id: I6836676096d47d9da5702a40b9d127569ad50dda
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175008
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31238}
This is important when we have multiple named devices connected over
USB (eg. "Webcam", "Microphone", "Headset") and there is some way to
choose a specific input device to route from.
Bug: b/154440591
Change-Id: I8dc1801a5e4db7f7bb439e855d43897c1f7d8bc4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173748
Commit-Queue: Robin Lee <rgl@google.com>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31130}
We may sometimes want to override only input or only output, or
override them with different values. This change allows setting the
overrides separately.
Change-Id: Ib0c44cb7a3cfa834f997fb6cd54f7cad68705f41
Bug: webrtc:10441
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128763
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27236}
Added audio format field and set method to Builder. - WebRTCAudioRecord. Added audio format field, added to constructor. Default audio format value AudioFormat.ENCODING_PCM_16BIT. initRecord(), added how to calculate bytesPerFrame, depends on audioFormat.
First commit and contribution, updated AUTHORS file
Bug: None
Change-Id: I16f660d42350ec9ce2e329b239bd7f6324e76dfe
Reviewed-on: https://webrtc-review.googlesource.com/c/122302
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26775}
This CL removes the use of the @JNINamespace annotation and instead
sets the correct JNI namespace in the build file.
Bug: webrtc:8278
Change-Id: Ia4490399e45a97d56b02c260fd80df4edfa092bf
Reviewed-on: https://webrtc-review.googlesource.com/76440
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23299}
The new ADM code removed some redundancies, which led to a decrease in
log output. This especially affected NS and AEC logs. This change
reintroduces these log messages, making debugging easier. "Acoustic
Echo Canceler" has been changed to AEC for easier grepping.
Some new logging is also added.
Bug: webrtc:7452
Change-Id: I9bfb91895931d73d92f3187c8c7c5b7524ac05ba
Reviewed-on: https://webrtc-review.googlesource.com/71401
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23003}
Any native call before PeerConnectionFactory.initialize() will fail.
This means creation of JavaAudioDeviceModule will fail if it's created
before PeerConnectionFactory.initialize(). Clients should technically
always call PeerConnectionFactory.initialize() first, but we can help
the situation by deferring creation of the native ADM until it's
actually needed.
Bug: webrtc:7452
Change-Id: I53df2bdb980a8bdc413975f1cea6bcf297b453d5
Reviewed-on: https://webrtc-review.googlesource.com/70763
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22936}
This CL introduces sdk/android/api/org/webrtc/audio/AudioDeviceModule.java,
which is the new interface for audio device modules on Android.
This CL also refactors the main AudioDeviceModule implementation, which
is sdk/android/api/org/webrtc/audio/JavaAudioDeviceModule.java and makes
it conform to the new interface. The old code used global static methods
to configure the audio device code. This CL gets rid of all that and uses
a builder pattern in JavaAudioDeviceModule instead. The only two dynamic
methods left in the interface are setSpeakerMute() and setMicrophoneMute().
Removing the global static methods allowed a significant cleanup, and e.g.
the file sdk/android/src/jni/audio_device/audio_manager.cc has been
completely removed.
The PeerConnectionFactory interface is also updated to allow passing in
an external AudioDeviceModule. The current built-in ADM is encapsulated
under LegacyAudioDeviceModule.java, which is the default for now to
ensure backwards compatibility.
Bug: webrtc:7452
Change-Id: I64d5f4dba9a004da001f1acb2bd0c1b1f2b64f21
Reviewed-on: https://webrtc-review.googlesource.com/65360
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22765}
The class called AudioDeviceModule today is an implementation of a
future interface. We want to reserve the name AudioDeviceModule for
the actual interface. The implementation class has been renamed to
JavaAudioDeviceModule. 'Java' here refers to the fact that the
implementation is using android.media.AudioRecord as input and
android.media.AudioTrack as output, and this is opposed to native
AudioDeviceModule implementations such as OpenSLES and AAudio.
Bug: webrtc:7452
Change-Id: Ifc243c2e169b12a50128ee3252f06d574aa7b358
Reviewed-on: https://webrtc-review.googlesource.com/65400
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22673}
