commit 5fa6d363c605083f0beb6f6d21910849a06952db
parent 6c2e49c88221012b7c5a928d59c26b054a3c9e23
Author: Dan Baker <dbaker@mozilla.com>
Date: Mon, 27 Oct 2025 13:35:54 -0600
Bug 1995393 - Vendor libwebrtc from de4667eb4f
Upstream commit: https://webrtc.googlesource.com/src/+/de4667eb4f071c691cfe5304210480ee61a112b0
Propagate Environment into voice engine channels
This way Call no longer need to expose Environment or field trials
Bug: webrtc:440271885
Change-Id: Ib680e9943b1b35f789eac6228cd33dd55569d382
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/406380
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#45448}
Diffstat:
14 files changed, 76 insertions(+), 82 deletions(-)
diff --git a/third_party/libwebrtc/README.mozilla.last-vendor b/third_party/libwebrtc/README.mozilla.last-vendor
@@ -1,4 +1,4 @@
# ./mach python dom/media/webrtc/third_party_build/vendor-libwebrtc.py --from-local /Users/danielbaker/elm/.moz-fast-forward/moz-libwebrtc --commit mozpatches libwebrtc
-libwebrtc updated from /Users/danielbaker/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2025-10-27T19:33:32.697294+00:00.
+libwebrtc updated from /Users/danielbaker/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2025-10-27T19:35:39.571378+00:00.
# base of lastest vendoring
-9aa4278cf5
+de4667eb4f
diff --git a/third_party/libwebrtc/call/call.cc b/third_party/libwebrtc/call/call.cc
@@ -28,7 +28,6 @@
#include "api/array_view.h"
#include "api/environment/environment.h"
#include "api/fec_controller.h"
-#include "api/field_trials_view.h"
#include "api/media_types.h"
#include "api/rtc_error.h"
#include "api/rtc_event_log/rtc_event_log.h"
@@ -284,9 +283,6 @@ class Call final : public webrtc::Call,
int FeedbackAccordingToRfc8888Count() override;
int FeedbackAccordingToTransportCcCount() override;
- const FieldTrialsView& trials() const override;
- const Environment& env() const override;
-
TaskQueueBase* network_thread() const override;
TaskQueueBase* worker_thread() const override;
@@ -1036,7 +1032,7 @@ webrtc::VideoReceiveStreamInterface* Call::CreateVideoReceiveStream(
VideoReceiveStream2* receive_stream = new VideoReceiveStream2(
env_, this, num_cpu_cores_, transport_send_->packet_router(),
std::move(configuration), call_stats_.get(),
- std::make_unique<VCMTiming>(&env_.clock(), trials()),
+ std::make_unique<VCMTiming>(&env_.clock(), env_.field_trials()),
&nack_periodic_processor_, decode_sync_.get());
// TODO(bugs.webrtc.org/11993): Set this up asynchronously on the network
// thread.
@@ -1177,14 +1173,6 @@ int Call::FeedbackAccordingToTransportCcCount() {
return transport_send_->ReceivedTransportCcFeedbackCount();
}
-const FieldTrialsView& Call::trials() const {
- return env_.field_trials();
-}
-
-const Environment& Call::env() const {
- return env_;
-}
-
TaskQueueBase* Call::network_thread() const {
return network_thread_;
}
diff --git a/third_party/libwebrtc/call/call.h b/third_party/libwebrtc/call/call.h
@@ -16,9 +16,7 @@
#include "absl/strings/string_view.h"
#include "api/adaptation/resource.h"
-#include "api/environment/environment.h"
#include "api/fec_controller.h"
-#include "api/field_trials_view.h"
#include "api/media_types.h"
#include "api/rtp_headers.h"
#include "api/rtp_parameters.h"
@@ -155,14 +153,10 @@ class Call {
virtual int FeedbackAccordingToRfc8888Count() = 0;
virtual int FeedbackAccordingToTransportCcCount() = 0;
- // TODO(bugs.webrtc.org/440271885): Use env() instead.
- virtual const FieldTrialsView& trials() const = 0;
- virtual const Environment& env() const = 0;
-
virtual TaskQueueBase* network_thread() const = 0;
virtual TaskQueueBase* worker_thread() const = 0;
- virtual ~Call() {}
+ virtual ~Call() = default;
};
} // namespace webrtc
diff --git a/third_party/libwebrtc/media/base/fake_media_engine.cc b/third_party/libwebrtc/media/base/fake_media_engine.cc
@@ -572,7 +572,8 @@ scoped_refptr<AudioState> FakeVoiceEngine::GetAudioState() const {
return scoped_refptr<AudioState>();
}
std::unique_ptr<VoiceMediaSendChannelInterface>
-FakeVoiceEngine::CreateSendChannel(Call* call,
+FakeVoiceEngine::CreateSendChannel(const Environment& /*env*/,
+ Call* call,
const MediaConfig& /* config */,
const AudioOptions& options,
const CryptoOptions& /* crypto_options */,
@@ -583,7 +584,8 @@ FakeVoiceEngine::CreateSendChannel(Call* call,
return ch;
}
std::unique_ptr<VoiceMediaReceiveChannelInterface>
-FakeVoiceEngine::CreateReceiveChannel(Call* call,
+FakeVoiceEngine::CreateReceiveChannel(const Environment& /*env*/,
+ Call* call,
const MediaConfig& /* config */,
const AudioOptions& options,
const CryptoOptions& /* crypto_options */,
diff --git a/third_party/libwebrtc/media/base/fake_media_engine.h b/third_party/libwebrtc/media/base/fake_media_engine.h
@@ -790,12 +790,14 @@ class FakeVoiceEngine : public VoiceEngineInterface {
scoped_refptr<AudioState> GetAudioState() const override;
std::unique_ptr<VoiceMediaSendChannelInterface> CreateSendChannel(
+ const Environment& env,
Call* call,
const MediaConfig& config,
const AudioOptions& options,
const CryptoOptions& crypto_options,
AudioCodecPairId codec_pair_id) override;
std::unique_ptr<VoiceMediaReceiveChannelInterface> CreateReceiveChannel(
+ const Environment& env,
Call* call,
const MediaConfig& config,
const AudioOptions& options,
diff --git a/third_party/libwebrtc/media/base/media_engine.h b/third_party/libwebrtc/media/base/media_engine.h
@@ -97,18 +97,20 @@ class VoiceEngineInterface : public RtpHeaderExtensionQueryInterface {
virtual scoped_refptr<AudioState> GetAudioState() const = 0;
virtual std::unique_ptr<VoiceMediaSendChannelInterface> CreateSendChannel(
- Call* /* call */,
- const MediaConfig& /* config */,
- const AudioOptions& /* options */,
- const CryptoOptions& /* crypto_options */,
- AudioCodecPairId /* codec_pair_id */) = 0;
+ const Environment& env,
+ Call* call,
+ const MediaConfig& config,
+ const AudioOptions& options,
+ const CryptoOptions& crypto_options,
+ AudioCodecPairId codec_pair_id) = 0;
virtual std::unique_ptr<VoiceMediaReceiveChannelInterface>
- CreateReceiveChannel(Call* /* call */,
- const MediaConfig& /* config */,
- const AudioOptions& /* options */,
- const CryptoOptions& /* crypto_options */,
- AudioCodecPairId /* codec_pair_id */) = 0;
+ CreateReceiveChannel(const Environment& env,
+ Call* call,
+ const MediaConfig& config,
+ const AudioOptions& options,
+ const CryptoOptions& crypto_options,
+ AudioCodecPairId codec_pair_id) = 0;
// Legacy: Retrieve list of supported codecs.
// + protection codecs, and assigns PT numbers that may have to be
diff --git a/third_party/libwebrtc/media/engine/fake_webrtc_call.h b/third_party/libwebrtc/media/engine/fake_webrtc_call.h
@@ -37,7 +37,6 @@
#include "api/audio_codecs/audio_format.h"
#include "api/crypto/frame_decryptor_interface.h"
#include "api/environment/environment.h"
-#include "api/field_trials_view.h"
#include "api/frame_transformer_interface.h"
#include "api/media_types.h"
#include "api/rtp_headers.h"
@@ -432,8 +431,6 @@ class FakeCall final : public Call, public PacketReceiver {
void SetClientBitratePreferences(
const BitrateSettings& /* preferences */) override {}
- const FieldTrialsView& trials() const override { return env_.field_trials(); }
- const Environment& env() const override { return env_; }
void SetPreferredRtcpCcAckType(
RtcpFeedbackType preferred_rtcp_cc_ack_type) override {}
int FeedbackAccordingToRfc8888Count() { return 0; }
diff --git a/third_party/libwebrtc/media/engine/webrtc_video_engine.cc b/third_party/libwebrtc/media/engine/webrtc_video_engine.cc
@@ -1802,8 +1802,8 @@ WebRtcVideoSendChannel::WebRtcVideoSendStream::WebRtcVideoSendStream(
codec_settings_list),
rtp_parameters_(CreateRtpParametersWithEncodings(sp)),
sending_(false),
- disable_automatic_resize_(
- call->trials().IsEnabled("WebRTC-Video-DisableAutomaticResize")) {
+ disable_automatic_resize_(env_.field_trials().IsEnabled(
+ "WebRTC-Video-DisableAutomaticResize")) {
// Maximum packet size may come in RtpConfig from external transport, for
// example from QuicTransportInterface implementation, so do not exceed
// given max_packet_size.
diff --git a/third_party/libwebrtc/media/engine/webrtc_voice_engine.cc b/third_party/libwebrtc/media/engine/webrtc_voice_engine.cc
@@ -577,23 +577,25 @@ scoped_refptr<AudioState> WebRtcVoiceEngine::GetAudioState() const {
}
std::unique_ptr<VoiceMediaSendChannelInterface>
-WebRtcVoiceEngine::CreateSendChannel(Call* call,
+WebRtcVoiceEngine::CreateSendChannel(const Environment& env,
+ Call* call,
const MediaConfig& config,
const AudioOptions& options,
const CryptoOptions& crypto_options,
AudioCodecPairId codec_pair_id) {
return std::make_unique<WebRtcVoiceSendChannel>(
- this, config, options, crypto_options, call, codec_pair_id);
+ env, this, config, options, crypto_options, call, codec_pair_id);
}
std::unique_ptr<VoiceMediaReceiveChannelInterface>
-WebRtcVoiceEngine::CreateReceiveChannel(Call* call,
+WebRtcVoiceEngine::CreateReceiveChannel(const Environment& env,
+ Call* call,
const MediaConfig& config,
const AudioOptions& options,
const CryptoOptions& crypto_options,
AudioCodecPairId codec_pair_id) {
return std::make_unique<WebRtcVoiceReceiveChannel>(
- this, config, options, crypto_options, call, codec_pair_id);
+ env, this, config, options, crypto_options, call, codec_pair_id);
}
void WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
@@ -833,6 +835,7 @@ AudioState* WebRtcVoiceEngine::audio_state() {
class WebRtcVoiceSendChannel::WebRtcAudioSendStream : public AudioSource::Sink {
public:
WebRtcAudioSendStream(
+ const Environment& env,
uint32_t ssrc,
const std::string& mid,
const std::string& c_name,
@@ -851,7 +854,7 @@ class WebRtcVoiceSendChannel::WebRtcAudioSendStream : public AudioSource::Sink {
const std::optional<AudioCodecPairId> codec_pair_id,
scoped_refptr<FrameEncryptorInterface> frame_encryptor,
const CryptoOptions& crypto_options)
- : env_(call->env()),
+ : env_(env),
adaptive_ptime_config_(env_.field_trials()),
call_(call),
config_(send_transport),
@@ -1272,6 +1275,7 @@ class WebRtcVoiceSendChannel::WebRtcAudioSendStream : public AudioSource::Sink {
};
WebRtcVoiceSendChannel::WebRtcVoiceSendChannel(
+ const Environment& env,
WebRtcVoiceEngine* engine,
const MediaConfig& config,
const AudioOptions& options,
@@ -1279,7 +1283,7 @@ WebRtcVoiceSendChannel::WebRtcVoiceSendChannel(
Call* call,
AudioCodecPairId codec_pair_id)
: MediaChannelUtil(call->network_thread(), config.enable_dscp),
- env_(call->env()),
+ env_(env),
worker_thread_(call->worker_thread()),
engine_(engine),
call_(call),
@@ -1628,7 +1632,7 @@ bool WebRtcVoiceSendChannel::AddSendStream(const StreamParams& sp) {
std::optional<std::string> audio_network_adaptor_config =
GetAudioNetworkAdaptorConfig(options_);
WebRtcAudioSendStream* stream = new WebRtcAudioSendStream(
- ssrc, mid_, sp.cname, sp.id, send_codec_spec_, ExtmapAllowMixed(),
+ env_, ssrc, mid_, sp.cname, sp.id, send_codec_spec_, ExtmapAllowMixed(),
send_rtp_extensions_, rtcp_cc_ack_type_, max_send_bitrate_bps_,
audio_config_.rtcp_report_interval_ms, audio_network_adaptor_config,
call_, transport(), engine()->encoder_factory_, codec_pair_id_, nullptr,
@@ -2108,6 +2112,7 @@ class WebRtcVoiceReceiveChannel::WebRtcAudioReceiveStream {
};
WebRtcVoiceReceiveChannel::WebRtcVoiceReceiveChannel(
+ const Environment& env,
WebRtcVoiceEngine* engine,
const MediaConfig& config,
const AudioOptions& options,
@@ -2115,7 +2120,7 @@ WebRtcVoiceReceiveChannel::WebRtcVoiceReceiveChannel(
Call* call,
AudioCodecPairId codec_pair_id)
: MediaChannelUtil(call->network_thread(), config.enable_dscp),
- env_(call->env()),
+ env_(env),
worker_thread_(call->worker_thread()),
engine_(engine),
call_(call),
diff --git a/third_party/libwebrtc/media/engine/webrtc_voice_engine.h b/third_party/libwebrtc/media/engine/webrtc_voice_engine.h
@@ -96,6 +96,7 @@ class WebRtcVoiceEngine final : public VoiceEngineInterface {
scoped_refptr<AudioState> GetAudioState() const override;
std::unique_ptr<VoiceMediaSendChannelInterface> CreateSendChannel(
+ const Environment& env,
Call* call,
const MediaConfig& config,
const AudioOptions& options,
@@ -103,6 +104,7 @@ class WebRtcVoiceEngine final : public VoiceEngineInterface {
AudioCodecPairId codec_pair_id) override;
std::unique_ptr<VoiceMediaReceiveChannelInterface> CreateReceiveChannel(
+ const Environment& env,
Call* call,
const MediaConfig& config,
const AudioOptions& options,
@@ -176,7 +178,8 @@ class WebRtcVoiceEngine final : public VoiceEngineInterface {
class WebRtcVoiceSendChannel final : public MediaChannelUtil,
public VoiceMediaSendChannelInterface {
public:
- WebRtcVoiceSendChannel(WebRtcVoiceEngine* engine,
+ WebRtcVoiceSendChannel(const Environment& env,
+ WebRtcVoiceEngine* engine,
const MediaConfig& config,
const AudioOptions& options,
const CryptoOptions& crypto_options,
@@ -285,7 +288,7 @@ class WebRtcVoiceSendChannel final : public MediaChannelUtil,
bool SetMaxSendBitrate(int bps);
void SetupRecording();
- const Environment& env_;
+ const Environment env_;
TaskQueueBase* const worker_thread_;
ScopedTaskSafety task_safety_;
SequenceChecker network_thread_checker_{SequenceChecker::kDetached};
@@ -335,7 +338,8 @@ class WebRtcVoiceReceiveChannel final
: public MediaChannelUtil,
public VoiceMediaReceiveChannelInterface {
public:
- WebRtcVoiceReceiveChannel(WebRtcVoiceEngine* engine,
+ WebRtcVoiceReceiveChannel(const Environment& env,
+ WebRtcVoiceEngine* engine,
const MediaConfig& config,
const AudioOptions& options,
const CryptoOptions& crypto_options,
@@ -434,7 +438,7 @@ class WebRtcVoiceReceiveChannel final
// unsignaled anymore (i.e. it is now removed, or signaled), and return true.
bool MaybeDeregisterUnsignaledRecvStream(uint32_t ssrc);
- const Environment& env_;
+ const Environment env_;
TaskQueueBase* const worker_thread_;
ScopedTaskSafety task_safety_;
SequenceChecker network_thread_checker_{SequenceChecker::kDetached};
diff --git a/third_party/libwebrtc/media/engine/webrtc_voice_engine_unittest.cc b/third_party/libwebrtc/media/engine/webrtc_voice_engine_unittest.cc
@@ -299,10 +299,10 @@ class WebRtcVoiceEngineTestFake : public ::testing::TestWithParam<bool> {
bool SetupChannel() {
send_channel_ = engine_->CreateSendChannel(
- &call_, webrtc::MediaConfig(), webrtc::AudioOptions(),
+ env_, &call_, webrtc::MediaConfig(), webrtc::AudioOptions(),
webrtc::CryptoOptions(), webrtc::AudioCodecPairId::Create());
receive_channel_ = engine_->CreateReceiveChannel(
- &call_, webrtc::MediaConfig(), webrtc::AudioOptions(),
+ env_, &call_, webrtc::MediaConfig(), webrtc::AudioOptions(),
webrtc::CryptoOptions(), webrtc::AudioCodecPairId::Create());
send_channel_->SetSsrcListChangedCallback(
[receive_channel =
@@ -3225,7 +3225,7 @@ TEST_P(WebRtcVoiceEngineTestFake, InitRecordingOnSend) {
std::unique_ptr<webrtc::VoiceMediaSendChannelInterface> send_channel(
engine_->CreateSendChannel(
- &call_, webrtc::MediaConfig(), webrtc::AudioOptions(),
+ env_, &call_, webrtc::MediaConfig(), webrtc::AudioOptions(),
webrtc::CryptoOptions(), webrtc::AudioCodecPairId::Create()));
send_channel->SetSend(true);
@@ -3240,7 +3240,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SkipInitRecordingOnSend) {
options.init_recording_on_send = false;
std::unique_ptr<webrtc::VoiceMediaSendChannelInterface> send_channel(
- engine_->CreateSendChannel(&call_, webrtc::MediaConfig(), options,
+ engine_->CreateSendChannel(env_, &call_, webrtc::MediaConfig(), options,
webrtc::CryptoOptions(),
webrtc::AudioCodecPairId::Create()));
@@ -3267,11 +3267,11 @@ TEST_P(WebRtcVoiceEngineTestFake, SetOptionOverridesViaChannels) {
std::unique_ptr<webrtc::VoiceMediaSendChannelInterface> send_channel1(
engine_->CreateSendChannel(
- &call_, webrtc::MediaConfig(), webrtc::AudioOptions(),
+ env_, &call_, webrtc::MediaConfig(), webrtc::AudioOptions(),
webrtc::CryptoOptions(), webrtc::AudioCodecPairId::Create()));
std::unique_ptr<webrtc::VoiceMediaSendChannelInterface> send_channel2(
engine_->CreateSendChannel(
- &call_, webrtc::MediaConfig(), webrtc::AudioOptions(),
+ env_, &call_, webrtc::MediaConfig(), webrtc::AudioOptions(),
webrtc::CryptoOptions(), webrtc::AudioCodecPairId::Create()));
// Have to add a stream to make SetSend work.
@@ -3387,18 +3387,18 @@ TEST_P(WebRtcVoiceEngineTestFake, TestSetDscpOptions) {
std::unique_ptr<webrtc::VoiceMediaSendChannelInterface> channel;
webrtc::RtpParameters parameters;
- channel = engine_->CreateSendChannel(&call_, config, webrtc::AudioOptions(),
- webrtc::CryptoOptions(),
- webrtc::AudioCodecPairId::Create());
+ channel = engine_->CreateSendChannel(
+ env_, &call_, config, webrtc::AudioOptions(), webrtc::CryptoOptions(),
+ webrtc::AudioCodecPairId::Create());
channel->SetInterface(&network_interface);
// Default value when DSCP is disabled should be DSCP_DEFAULT.
EXPECT_EQ(webrtc::DSCP_DEFAULT, network_interface.dscp());
channel->SetInterface(nullptr);
config.enable_dscp = true;
- channel = engine_->CreateSendChannel(&call_, config, webrtc::AudioOptions(),
- webrtc::CryptoOptions(),
- webrtc::AudioCodecPairId::Create());
+ channel = engine_->CreateSendChannel(
+ env_, &call_, config, webrtc::AudioOptions(), webrtc::CryptoOptions(),
+ webrtc::AudioCodecPairId::Create());
channel->SetInterface(&network_interface);
EXPECT_EQ(webrtc::DSCP_DEFAULT, network_interface.dscp());
@@ -3427,9 +3427,9 @@ TEST_P(WebRtcVoiceEngineTestFake, TestSetDscpOptions) {
// Verify that setting the option to false resets the
// DiffServCodePoint.
config.enable_dscp = false;
- channel = engine_->CreateSendChannel(&call_, config, webrtc::AudioOptions(),
- webrtc::CryptoOptions(),
- webrtc::AudioCodecPairId::Create());
+ channel = engine_->CreateSendChannel(
+ env_, &call_, config, webrtc::AudioOptions(), webrtc::CryptoOptions(),
+ webrtc::AudioCodecPairId::Create());
channel->SetInterface(&network_interface);
// Default value when DSCP is disabled should be DSCP_DEFAULT.
EXPECT_EQ(webrtc::DSCP_DEFAULT, network_interface.dscp());
@@ -3831,12 +3831,12 @@ TEST(WebRtcVoiceEngineTest, StartupShutdown) {
std::unique_ptr<Call> call = Call::Create(CallConfig(env));
std::unique_ptr<webrtc::VoiceMediaSendChannelInterface> send_channel =
engine.CreateSendChannel(
- call.get(), webrtc::MediaConfig(), webrtc::AudioOptions(),
+ env, call.get(), webrtc::MediaConfig(), webrtc::AudioOptions(),
webrtc::CryptoOptions(), webrtc::AudioCodecPairId::Create());
EXPECT_TRUE(send_channel);
std::unique_ptr<webrtc::VoiceMediaReceiveChannelInterface> receive_channel =
engine.CreateReceiveChannel(
- call.get(), webrtc::MediaConfig(), webrtc::AudioOptions(),
+ env, call.get(), webrtc::MediaConfig(), webrtc::AudioOptions(),
webrtc::CryptoOptions(), webrtc::AudioCodecPairId::Create());
EXPECT_TRUE(receive_channel);
}
@@ -3860,12 +3860,12 @@ TEST(WebRtcVoiceEngineTest, StartupShutdownWithExternalADM) {
std::unique_ptr<Call> call = Call::Create(CallConfig(env));
std::unique_ptr<webrtc::VoiceMediaSendChannelInterface> send_channel =
engine.CreateSendChannel(
- call.get(), webrtc::MediaConfig(), webrtc::AudioOptions(),
+ env, call.get(), webrtc::MediaConfig(), webrtc::AudioOptions(),
webrtc::CryptoOptions(), webrtc::AudioCodecPairId::Create());
EXPECT_TRUE(send_channel);
std::unique_ptr<webrtc::VoiceMediaReceiveChannelInterface>
receive_channel = engine.CreateReceiveChannel(
- call.get(), webrtc::MediaConfig(), webrtc::AudioOptions(),
+ env, call.get(), webrtc::MediaConfig(), webrtc::AudioOptions(),
webrtc::CryptoOptions(), webrtc::AudioCodecPairId::Create());
EXPECT_TRUE(receive_channel);
}
@@ -3944,7 +3944,7 @@ TEST(WebRtcVoiceEngineTest, Has32Channels) {
while (channels.size() < 32) {
std::unique_ptr<webrtc::VoiceMediaSendChannelInterface> channel =
engine.CreateSendChannel(
- call.get(), webrtc::MediaConfig(), webrtc::AudioOptions(),
+ env, call.get(), webrtc::MediaConfig(), webrtc::AudioOptions(),
webrtc::CryptoOptions(), webrtc::AudioCodecPairId::Create());
if (!channel)
break;
@@ -3977,7 +3977,7 @@ TEST(WebRtcVoiceEngineTest, SetRecvCodecs) {
engine.Init();
std::unique_ptr<Call> call = Call::Create(CallConfig(env));
webrtc::WebRtcVoiceReceiveChannel channel(
- &engine, webrtc::MediaConfig(), webrtc::AudioOptions(),
+ env, &engine, webrtc::MediaConfig(), webrtc::AudioOptions(),
webrtc::CryptoOptions(), call.get(),
webrtc::AudioCodecPairId::Create());
webrtc::AudioReceiverParameters parameters;
@@ -4006,7 +4006,7 @@ TEST(WebRtcVoiceEngineTest, SetRtpSendParametersMaxBitrate) {
}
std::unique_ptr<Call> call = Call::Create(std::move(call_config));
webrtc::WebRtcVoiceSendChannel channel(
- &engine, webrtc::MediaConfig(), webrtc::AudioOptions(),
+ env, &engine, webrtc::MediaConfig(), webrtc::AudioOptions(),
webrtc::CryptoOptions(), call.get(), webrtc::AudioCodecPairId::Create());
{
webrtc::AudioSenderParameter params;
diff --git a/third_party/libwebrtc/moz-patch-stack/s0119.patch b/third_party/libwebrtc/moz-patch-stack/s0119.patch
@@ -12,10 +12,10 @@ Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/19164e58323d59c1f
3 files changed, 7 insertions(+), 22 deletions(-)
diff --git a/call/call.cc b/call/call.cc
-index 01c6cc7be6..7b9ef6b5a9 100644
+index 0b756c03a1..8cbffd0374 100644
--- a/call/call.cc
+++ b/call/call.cc
-@@ -517,19 +517,6 @@ class Call final : public webrtc::Call,
+@@ -513,19 +513,6 @@ class Call final : public webrtc::Call,
};
} // namespace internal
@@ -36,10 +36,10 @@ index 01c6cc7be6..7b9ef6b5a9 100644
std::unique_ptr<RtpTransportControllerSendInterface> transport_send;
if (config.rtp_transport_controller_send_factory != nullptr) {
diff --git a/call/call.h b/call/call.h
-index f100da4d15..4361577be4 100644
+index 5e3d3de221..aa9da48ebe 100644
--- a/call/call.h
+++ b/call/call.h
-@@ -27,6 +27,7 @@
+@@ -25,6 +25,7 @@
#include "api/transport/bitrate_settings.h"
#include "call/audio_receive_stream.h"
#include "call/audio_send_stream.h"
@@ -47,7 +47,7 @@ index f100da4d15..4361577be4 100644
#include "call/call_config.h"
#include "call/flexfec_receive_stream.h"
#include "call/packet_receiver.h"
-@@ -52,15 +53,7 @@ namespace webrtc {
+@@ -50,15 +51,7 @@ namespace webrtc {
class Call {
public:
diff --git a/third_party/libwebrtc/pc/rtp_sender_receiver_unittest.cc b/third_party/libwebrtc/pc/rtp_sender_receiver_unittest.cc
@@ -114,13 +114,13 @@ class RtpSenderReceiverTest
// Create the channels, discard the result; we get them later.
// Fake media channels are owned by the media engine.
voice_media_send_channel_ = media_engine_->voice().CreateSendChannel(
- &fake_call_, MediaConfig(), AudioOptions(), CryptoOptions(),
+ env_, &fake_call_, MediaConfig(), AudioOptions(), CryptoOptions(),
AudioCodecPairId::Create());
video_media_send_channel_ = media_engine_->video().CreateSendChannel(
env_, &fake_call_, MediaConfig(), VideoOptions(), CryptoOptions(),
video_bitrate_allocator_factory_.get());
voice_media_receive_channel_ = media_engine_->voice().CreateReceiveChannel(
- &fake_call_, MediaConfig(), AudioOptions(), CryptoOptions(),
+ env_, &fake_call_, MediaConfig(), AudioOptions(), CryptoOptions(),
AudioCodecPairId::Create());
video_media_receive_channel_ = media_engine_->video().CreateReceiveChannel(
env_, &fake_call_, MediaConfig(), VideoOptions(), CryptoOptions());
diff --git a/third_party/libwebrtc/pc/rtp_transceiver.cc b/third_party/libwebrtc/pc/rtp_transceiver.cc
@@ -234,11 +234,11 @@ RTCError RtpTransceiver::CreateChannel(
std::unique_ptr<VoiceMediaSendChannelInterface> media_send_channel =
media_engine()->voice().CreateSendChannel(
- call_ptr, media_config, audio_options, crypto_options,
+ env_, call_ptr, media_config, audio_options, crypto_options,
codec_pair_id);
std::unique_ptr<VoiceMediaReceiveChannelInterface> media_receive_channel =
media_engine()->voice().CreateReceiveChannel(
- call_ptr, media_config, audio_options, crypto_options,
+ env_, call_ptr, media_config, audio_options, crypto_options,
codec_pair_id);
// Note that this is safe because both sending and
// receiving channels will be deleted at the same time.
