Reland "Allow AudioSendStream to reconfig AudioNetworkAdaptor"
BUG=b/63898232, b/64053465
Originally Reviewed-on: https://chromium-review.googlesource.com/584707
Reverted-on: https://chromium-review.googlesource.com/586268
Change-Id: I212b0c1e81a6ccd73b051e6728e601a8641463b8
Reviewed-on: https://chromium-review.googlesource.com/586328
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Michael T <tschumim@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19153}diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc
index 376af81..f1395ab 100644
--- a/webrtc/audio/audio_send_stream.cc
+++ b/webrtc/audio/audio_send_stream.cc
@@ -487,7 +487,17 @@
bool AudioSendStream::ReconfigureSendCodec(AudioSendStream* stream,
const Config& new_config) {
const auto& old_config = stream->config_;
- if (new_config.send_codec_spec == old_config.send_codec_spec) {
+
+ if (!new_config.send_codec_spec) {
+ // We cannot de-configure a send codec. So we will do nothing.
+ // By design, the send codec should have not been configured.
+ RTC_DCHECK(!old_config.send_codec_spec);
+ return true;
+ }
+
+ if (new_config.send_codec_spec == old_config.send_codec_spec &&
+ new_config.audio_network_adaptor_config ==
+ old_config.audio_network_adaptor_config) {
return true;
}
@@ -501,9 +511,6 @@
return SetupSendCodec(stream, new_config);
}
- // Should never move a stream from fully configured to unconfigured.
- RTC_CHECK(new_config.send_codec_spec);
-
const rtc::Optional<int>& new_target_bitrate_bps =
new_config.send_codec_spec->target_bitrate_bps;
// If a bitrate has been specified for the codec, use it over the
diff --git a/webrtc/audio/audio_send_stream_unittest.cc b/webrtc/audio/audio_send_stream_unittest.cc
index 12fad6f..bec696b 100644
--- a/webrtc/audio/audio_send_stream_unittest.cc
+++ b/webrtc/audio/audio_send_stream_unittest.cc
@@ -43,6 +43,7 @@
using testing::Ne;
using testing::Invoke;
using testing::Return;
+using testing::StrEq;
const int kChannelId = 1;
const uint32_t kSsrc = 1234;
@@ -141,7 +142,8 @@
&packet_router_)),
fake_transport_(&packet_router_, send_side_cc_.get()),
bitrate_allocator_(&limit_observer_),
- worker_queue_("ConfigHelper_worker_queue") {
+ worker_queue_("ConfigHelper_worker_queue"),
+ audio_encoder_(nullptr) {
using testing::Invoke;
EXPECT_CALL(voice_engine_,
@@ -206,7 +208,6 @@
}
void SetupDefaultChannelProxy(bool audio_bwe_enabled) {
- using testing::StrEq;
channel_proxy_ = new testing::StrictMock<MockVoEChannelProxy>();
EXPECT_CALL(*channel_proxy_, GetRtpRtcp(_, _))
.WillRepeatedly(Invoke(
@@ -250,10 +251,25 @@
void SetupMockForSetupSendCodec(bool expect_set_encoder_call) {
if (expect_set_encoder_call) {
EXPECT_CALL(*channel_proxy_, SetEncoderForMock(_, _))
- .WillOnce(Return(true));
+ .WillOnce(Invoke(
+ [this](int payload_type, std::unique_ptr<AudioEncoder>* encoder) {
+ this->audio_encoder_ = std::move(*encoder);
+ return true;
+ }));
}
}
+ void SetupMockForModifyEncoder() {
+ // Let ModifyEncoder to invoke mock audio encoder.
+ EXPECT_CALL(*channel_proxy_, ModifyEncoder(_))
+ .WillRepeatedly(Invoke(
+ [this](rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)>
+ modifier) {
+ if (this->audio_encoder_)
+ modifier(&this->audio_encoder_);
+ }));
+ }
+
RtcpRttStats* rtcp_rtt_stats() { return &rtcp_rtt_stats_; }
void SetupMockForSendTelephoneEvent() {
@@ -329,6 +345,7 @@
// |worker_queue| is defined last to ensure all pending tasks are cancelled
// and deleted before any other members.
rtc::TaskQueue worker_queue_;
+ std::unique_ptr<AudioEncoder> audio_encoder_;
};
} // namespace
@@ -458,19 +475,27 @@
EXPECT_FALSE(send_stream.GetStats().typing_noise_detected);
}
-TEST(AudioSendStreamTest, SendCodecAppliesNetworkAdaptor) {
+TEST(AudioSendStreamTest, SendCodecAppliesAudioNetworkAdaptor) {
ConfigHelper helper(false, true);
auto stream_config = helper.config();
stream_config.send_codec_spec =
rtc::Optional<AudioSendStream::Config::SendCodecSpec>({0, kOpusFormat});
+ const std::string kAnaConfigString = "abcde";
+ const std::string kAnaReconfigString = "12345";
+
stream_config.audio_network_adaptor_config =
- rtc::Optional<std::string>("abced");
+ rtc::Optional<std::string>(kAnaConfigString);
EXPECT_CALL(helper.mock_encoder_factory(), MakeAudioEncoderMock(_, _, _))
- .WillOnce(Invoke([](int payload_type, const SdpAudioFormat& format,
- std::unique_ptr<AudioEncoder>* return_value) {
+ .WillOnce(Invoke([&kAnaConfigString, &kAnaReconfigString](
+ int payload_type, const SdpAudioFormat& format,
+ std::unique_ptr<AudioEncoder>* return_value) {
auto mock_encoder = SetupAudioEncoderMock(payload_type, format);
- EXPECT_CALL(*mock_encoder.get(), EnableAudioNetworkAdaptor(_, _))
+ EXPECT_CALL(*mock_encoder,
+ EnableAudioNetworkAdaptor(StrEq(kAnaConfigString), _))
+ .WillOnce(Return(true));
+ EXPECT_CALL(*mock_encoder,
+ EnableAudioNetworkAdaptor(StrEq(kAnaReconfigString), _))
.WillOnce(Return(true));
*return_value = std::move(mock_encoder);
}));
@@ -479,6 +504,12 @@
stream_config, helper.audio_state(), helper.worker_queue(),
helper.transport(), helper.bitrate_allocator(), helper.event_log(),
helper.rtcp_rtt_stats(), rtc::Optional<RtpState>());
+
+ stream_config.audio_network_adaptor_config =
+ rtc::Optional<std::string>(kAnaReconfigString);
+
+ helper.SetupMockForModifyEncoder();
+ send_stream.Reconfigure(stream_config);
}
// VAD is applied when codec is mono and the CNG frequency matches the codec