Commit 6b2582da authored by Alex Loiko's avatar Alex Loiko Committed by Commit Bot

Multistream opus in Chrome.

This change is required to let Chrome encode and decode multistream
Opus audio. It adds WebRTC multistream Opus decoders and encoders as
non-advertised codecs to the audio codec factory.

The WebRTC default audio codec factory has contained the multistream
opus encoder/decoder since April 25 and April 8 respectively. The
multistream encoders have been tested in WebRTC.

Bug: webrtc:8649
Change-Id: Ie350ab2e052edfd7f18dfa4c67bb397dfa541d1d
Reviewed-on: https://chromium-review.googlesource.com/c/chromium/src/+/1609782Reviewed-by: default avatarHenrik Boström <hbos@chromium.org>
Commit-Queue: Alex Loiko <aleloi@chromium.org>
Cr-Commit-Position: refs/heads/master@{#659048}
parent 5994cc70
......@@ -653,7 +653,9 @@ target(link_target_type, "renderer") {
"//third_party/webrtc/api/audio_codecs/g722:audio_encoder_g722",
"//third_party/webrtc/api/audio_codecs/isac:audio_decoder_isac",
"//third_party/webrtc/api/audio_codecs/isac:audio_encoder_isac",
"//third_party/webrtc/api/audio_codecs/opus:audio_decoder_multiopus",
"//third_party/webrtc/api/audio_codecs/opus:audio_decoder_opus",
"//third_party/webrtc/api/audio_codecs/opus:audio_encoder_multiopus",
"//third_party/webrtc/api/audio_codecs/opus:audio_encoder_opus",
"//third_party/webrtc/api/video:video_bitrate_allocation",
"//third_party/webrtc/api/video:video_frame",
......
......@@ -17,7 +17,9 @@
#include "third_party/webrtc/api/audio_codecs/g722/audio_encoder_g722.h"
#include "third_party/webrtc/api/audio_codecs/isac/audio_decoder_isac.h"
#include "third_party/webrtc/api/audio_codecs/isac/audio_encoder_isac.h"
#include "third_party/webrtc/api/audio_codecs/opus/audio_decoder_multi_channel_opus.h"
#include "third_party/webrtc/api/audio_codecs/opus/audio_decoder_opus.h"
#include "third_party/webrtc/api/audio_codecs/opus/audio_encoder_multi_channel_opus.h"
#include "third_party/webrtc/api/audio_codecs/opus/audio_encoder_opus.h"
namespace content {
......@@ -73,7 +75,8 @@ CreateWebrtcAudioEncoderFactory() {
return webrtc::CreateAudioEncoderFactory<
webrtc::AudioEncoderOpus, webrtc::AudioEncoderIsac,
webrtc::AudioEncoderG722, webrtc::AudioEncoderG711,
NotAdvertisedEncoder<webrtc::AudioEncoderL16>>();
NotAdvertisedEncoder<webrtc::AudioEncoderL16>,
NotAdvertisedEncoder<webrtc::AudioEncoderMultiChannelOpus>>();
}
rtc::scoped_refptr<webrtc::AudioDecoderFactory>
......@@ -81,7 +84,8 @@ CreateWebrtcAudioDecoderFactory() {
return webrtc::CreateAudioDecoderFactory<
webrtc::AudioDecoderOpus, webrtc::AudioDecoderIsac,
webrtc::AudioDecoderG722, webrtc::AudioDecoderG711,
NotAdvertisedDecoder<webrtc::AudioDecoderL16>>();
NotAdvertisedDecoder<webrtc::AudioDecoderL16>,
NotAdvertisedDecoder<webrtc::AudioDecoderMultiChannelOpus>>();
}
} // namespace content
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