diff options
| author | Sebastian Jansson <srte@webrtc.org> | 2020-01-29 17:42:52 +0100 |
|---|---|---|
| committer | Commit Bot <commit-bot@chromium.org> | 2020-01-29 18:42:34 +0000 |
| commit | c3eb9fd49f7343ab7ea2ea49ae1fa576aae5231d (patch) | |
| tree | 4aa1a4077b92f4e4441d6b27cf5b46a243f0ae20 /modules/audio_coding | |
| parent | 0cda7b832a9e86b7fb5d48d00b94a8d321602cdb (diff) | |
| download | webrtc-c3eb9fd49f7343ab7ea2ea49ae1fa576aae5231d.tar.gz | |
Reland "Reland "Only include overhead if using send side bandwidth estimation.""
This is a reland of 086055d0fd9b9b9efe8bcf85884324a019e9bd33
ANA was accitendly disabled even when transport sequence numbers were
negotiated due to a bug in how the audio send stream is configured. To
solve this we simply continue to always allow enabling ANA and leave it
up to the application to ensure that it's not used together with receive
side estimation.
Original change's description:
> Reland "Only include overhead if using send side bandwidth estimation."
>
> This is a reland of 8c79c6e1af354c526497082c79ccbe12af03a33e
>
> Original change's description:
> > Only include overhead if using send side bandwidth estimation.
> >
> > Bug: webrtc:11298
> > Change-Id: Ia2daf690461b55d394c1b964d6a7977a98be8be2
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166820
> > Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> > Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> > Reviewed-by: Ali Tofigh <alito@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30382}
>
> Bug: webrtc:11298
> Change-Id: I33205e869a8ae27c15ffe991f6d985973ed6d15a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167524
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30390}
Bug: webrtc:11298
Change-Id: If2ad91e17ebfc85dc51edcd9607996e18c5d1f13
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167883
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30413}
Diffstat (limited to 'modules/audio_coding')
| -rw-r--r-- | modules/audio_coding/codecs/opus/audio_encoder_opus.cc | 5 | ||||
| -rw-r--r-- | modules/audio_coding/codecs/opus/audio_encoder_opus.h | 1 |
2 files changed, 6 insertions, 0 deletions
diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/modules/audio_coding/codecs/opus/audio_encoder_opus.cc index 44cfe9e5a2..168bcec241 100644 --- a/modules/audio_coding/codecs/opus/audio_encoder_opus.cc +++ b/modules/audio_coding/codecs/opus/audio_encoder_opus.cc @@ -593,6 +593,11 @@ void AudioEncoderOpusImpl::OnReceivedUplinkPacketLossFraction( ApplyAudioNetworkAdaptor(); } +void AudioEncoderOpusImpl::OnReceivedTargetAudioBitrate( + int target_audio_bitrate_bps) { + SetTargetBitrate(target_audio_bitrate_bps); +} + void AudioEncoderOpusImpl::OnReceivedUplinkBandwidth( int target_audio_bitrate_bps, absl::optional<int64_t> bwe_period_ms, diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus.h b/modules/audio_coding/codecs/opus/audio_encoder_opus.h index 66c489f79b..40fd167c10 100644 --- a/modules/audio_coding/codecs/opus/audio_encoder_opus.h +++ b/modules/audio_coding/codecs/opus/audio_encoder_opus.h @@ -104,6 +104,7 @@ class AudioEncoderOpusImpl final : public AudioEncoder { void DisableAudioNetworkAdaptor() override; void OnReceivedUplinkPacketLossFraction( float uplink_packet_loss_fraction) override; + void OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps) override; void OnReceivedUplinkBandwidth( int target_audio_bitrate_bps, absl::optional<int64_t> bwe_period_ms) override; |