Use MediaTransportInterface, for audio streams.
Bug: webrtc:9719
Change-Id: I6d3db66b781173b207de51d84193fbd34a7f3239
Reviewed-on: https://webrtc-review.googlesource.com/c/104642
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25385}diff --git a/media/engine/webrtcvoiceengine.cc b/media/engine/webrtcvoiceengine.cc
index 03cc543..889c7ec 100644
--- a/media/engine/webrtcvoiceengine.cc
+++ b/media/engine/webrtcvoiceengine.cc
@@ -22,6 +22,7 @@
#include "absl/strings/match.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/call/audio_sink.h"
+#include "api/media_transport_interface.h"
#include "media/base/audiosource.h"
#include "media/base/mediaconstants.h"
#include "media/base/streamparams.h"
@@ -709,12 +710,13 @@
const absl::optional<std::string>& audio_network_adaptor_config,
webrtc::Call* call,
webrtc::Transport* send_transport,
+ webrtc::MediaTransportInterface* media_transport,
const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& encoder_factory,
const absl::optional<webrtc::AudioCodecPairId> codec_pair_id,
rtc::scoped_refptr<webrtc::FrameEncryptorInterface> frame_encryptor,
const webrtc::CryptoOptions& crypto_options)
: call_(call),
- config_(send_transport),
+ config_(send_transport, media_transport),
send_side_bwe_with_overhead_(
webrtc::field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")),
max_send_bitrate_bps_(max_send_bitrate_bps),
@@ -1076,6 +1078,7 @@
const std::vector<webrtc::RtpExtension>& extensions,
webrtc::Call* call,
webrtc::Transport* rtcp_send_transport,
+ webrtc::MediaTransportInterface* media_transport,
const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
const std::map<int, webrtc::SdpAudioFormat>& decoder_map,
absl::optional<webrtc::AudioCodecPairId> codec_pair_id,
@@ -1091,6 +1094,7 @@
config_.rtp.nack.rtp_history_ms = use_nack ? kNackRtpHistoryMs : 0;
config_.rtp.extensions = extensions;
config_.rtcp_send_transport = rtcp_send_transport;
+ config_.media_transport = media_transport;
config_.jitter_buffer_max_packets = jitter_buffer_max_packets;
config_.jitter_buffer_fast_accelerate = jitter_buffer_fast_accelerate;
if (!stream_ids.empty()) {
@@ -1792,7 +1796,8 @@
WebRtcAudioSendStream* stream = new WebRtcAudioSendStream(
ssrc, mid_, sp.cname, sp.id, send_codec_spec_, send_rtp_extensions_,
max_send_bitrate_bps_, audio_network_adaptor_config, call_, this,
- engine()->encoder_factory_, codec_pair_id_, nullptr, crypto_options_);
+ media_transport(), engine()->encoder_factory_, codec_pair_id_, nullptr,
+ crypto_options_);
send_streams_.insert(std::make_pair(ssrc, stream));
// At this point the stream's local SSRC has been updated. If it is the first
@@ -1873,13 +1878,14 @@
// Create a new channel for receiving audio data.
recv_streams_.insert(std::make_pair(
- ssrc, new WebRtcAudioReceiveStream(
- ssrc, receiver_reports_ssrc_, recv_transport_cc_enabled_,
- recv_nack_enabled_, sp.stream_ids(), recv_rtp_extensions_,
- call_, this, engine()->decoder_factory_, decoder_map_,
- codec_pair_id_, engine()->audio_jitter_buffer_max_packets_,
- engine()->audio_jitter_buffer_fast_accelerate_,
- unsignaled_frame_decryptor_, crypto_options_)));
+ ssrc,
+ new WebRtcAudioReceiveStream(
+ ssrc, receiver_reports_ssrc_, recv_transport_cc_enabled_,
+ recv_nack_enabled_, sp.stream_ids(), recv_rtp_extensions_, call_,
+ this, media_transport(), engine()->decoder_factory_, decoder_map_,
+ codec_pair_id_, engine()->audio_jitter_buffer_max_packets_,
+ engine()->audio_jitter_buffer_fast_accelerate_,
+ unsignaled_frame_decryptor_, crypto_options_)));
recv_streams_[ssrc]->SetPlayout(playout_);
return true;