diff --git a/webrtc/call/call_perf_tests.cc b/webrtc/call/call_perf_tests.cc index 43d7aa5e06..5a60acee5e 100644 --- a/webrtc/call/call_perf_tests.cc +++ b/webrtc/call/call_perf_tests.cc @@ -40,9 +40,6 @@ #include "webrtc/test/testsupport/fileutils.h" #include "webrtc/test/testsupport/perf_test.h" #include "webrtc/voice_engine/include/voe_base.h" -#include "webrtc/voice_engine/include/voe_codec.h" -#include "webrtc/voice_engine/include/voe_rtp_rtcp.h" -#include "webrtc/voice_engine/include/voe_video_sync.h" using webrtc::test::DriftingClock; using webrtc::test::FakeAudioDevice; @@ -152,7 +149,6 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec, metrics::Reset(); VoiceEngine* voice_engine = VoiceEngine::Create(); VoEBase* voe_base = VoEBase::GetInterface(voice_engine); - VoECodec* voe_codec = VoECodec::GetInterface(voice_engine); const std::string audio_filename = test::ResourcePath("voice_engine/audio_long16", "pcm"); ASSERT_STRNE("", audio_filename.c_str()); @@ -226,12 +222,11 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec, AudioSendStream::Config audio_send_config(&audio_send_transport); audio_send_config.voe_channel_id = send_channel_id; audio_send_config.rtp.ssrc = kAudioSendSsrc; + audio_send_config.send_codec_spec.codec_inst = + CodecInst{103, "ISAC", 16000, 480, 1, 32000}; AudioSendStream* audio_send_stream = sender_call_->CreateAudioSendStream(audio_send_config); - CodecInst isac = {103, "ISAC", 16000, 480, 1, 32000}; - EXPECT_EQ(0, voe_codec->SetSendCodec(send_channel_id, isac)); - video_send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs; if (fec == FecMode::kOn) { video_send_config_.rtp.ulpfec.red_payload_type = kRedPayloadType; @@ -297,7 +292,6 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec, voe_base->DeleteChannel(send_channel_id); voe_base->DeleteChannel(recv_channel_id); voe_base->Release(); - voe_codec->Release(); DestroyCalls(); diff --git a/webrtc/test/call_test.cc b/webrtc/test/call_test.cc index 57aca89dc7..23a82bf696 100644 --- a/webrtc/test/call_test.cc +++ b/webrtc/test/call_test.cc @@ -13,7 +13,6 @@ #include "webrtc/test/call_test.h" #include "webrtc/test/testsupport/fileutils.h" #include "webrtc/voice_engine/include/voe_base.h" -#include "webrtc/voice_engine/include/voe_codec.h" namespace webrtc { namespace test { @@ -201,6 +200,8 @@ void CallTest::CreateSendConfig(size_t num_video_streams, audio_send_config_ = AudioSendStream::Config(send_transport); audio_send_config_.voe_channel_id = voe_send_.channel_id; audio_send_config_.rtp.ssrc = kAudioSendSsrc; + audio_send_config_.send_codec_spec.codec_inst = + CodecInst{kAudioSendPayloadType, "ISAC", 16000, 480, 1, 32000}; } } @@ -227,9 +228,9 @@ void CallTest::CreateMatchingReceiveConfigs(Transport* rtcp_send_transport) { } } - RTC_DCHECK(num_audio_streams_ <= 1); + RTC_DCHECK_GE(1u, num_audio_streams_); if (num_audio_streams_ == 1) { - RTC_DCHECK(voe_send_.channel_id >= 0); + RTC_DCHECK_LE(0, voe_send_.channel_id); AudioReceiveStream::Config audio_config; audio_config.rtp.local_ssrc = kReceiverLocalAudioSsrc; audio_config.rtcp_send_transport = rtcp_send_transport; @@ -291,8 +292,6 @@ void CallTest::CreateAudioStreams() { audio_receive_streams_.push_back( receiver_call_->CreateAudioReceiveStream(audio_receive_configs_[i])); } - CodecInst isac = {kAudioSendPayloadType, "ISAC", 16000, 480, 1, 32000}; - EXPECT_EQ(0, voe_send_.codec->SetSendCodec(voe_send_.channel_id, isac)); } void CallTest::DestroyStreams() { @@ -316,7 +315,6 @@ void CallTest::CreateVoiceEngines() { CreateFakeAudioDevices(); voe_send_.voice_engine = VoiceEngine::Create(); voe_send_.base = VoEBase::GetInterface(voe_send_.voice_engine); - voe_send_.codec = VoECodec::GetInterface(voe_send_.voice_engine); EXPECT_EQ(0, voe_send_.base->Init(fake_send_audio_device_.get(), nullptr, decoder_factory_)); VoEBase::ChannelConfig config; @@ -326,7 +324,6 @@ void CallTest::CreateVoiceEngines() { voe_recv_.voice_engine = VoiceEngine::Create(); voe_recv_.base = VoEBase::GetInterface(voe_recv_.voice_engine); - voe_recv_.codec = VoECodec::GetInterface(voe_recv_.voice_engine); EXPECT_EQ(0, voe_recv_.base->Init(fake_recv_audio_device_.get(), nullptr, decoder_factory_)); voe_recv_.channel_id = voe_recv_.base->CreateChannel(); @@ -338,15 +335,11 @@ void CallTest::DestroyVoiceEngines() { voe_recv_.channel_id = -1; voe_recv_.base->Release(); voe_recv_.base = nullptr; - voe_recv_.codec->Release(); - voe_recv_.codec = nullptr; voe_send_.base->DeleteChannel(voe_send_.channel_id); voe_send_.channel_id = -1; voe_send_.base->Release(); voe_send_.base = nullptr; - voe_send_.codec->Release(); - voe_send_.codec = nullptr; VoiceEngine::Delete(voe_send_.voice_engine); voe_send_.voice_engine = nullptr; diff --git a/webrtc/test/call_test.h b/webrtc/test/call_test.h index ff84782602..74a54517a1 100644 --- a/webrtc/test/call_test.h +++ b/webrtc/test/call_test.h @@ -26,7 +26,6 @@ namespace webrtc { class VoEBase; -class VoECodec; namespace test { @@ -123,12 +122,10 @@ class CallTest : public ::testing::Test { VoiceEngineState() : voice_engine(nullptr), base(nullptr), - codec(nullptr), channel_id(-1) {} VoiceEngine* voice_engine; VoEBase* base; - VoECodec* codec; int channel_id; }; diff --git a/webrtc/video/video_quality_test.cc b/webrtc/video/video_quality_test.cc index b45cfde1ab..c5db0534c6 100644 --- a/webrtc/video/video_quality_test.cc +++ b/webrtc/video/video_quality_test.cc @@ -37,7 +37,6 @@ #include "webrtc/test/vcm_capturer.h" #include "webrtc/test/video_renderer.h" #include "webrtc/voice_engine/include/voe_base.h" -#include "webrtc/voice_engine/include/voe_codec.h" namespace { @@ -54,13 +53,11 @@ struct VoiceEngineState { VoiceEngineState() : voice_engine(nullptr), base(nullptr), - codec(nullptr), send_channel_id(-1), receive_channel_id(-1) {} webrtc::VoiceEngine* voice_engine; webrtc::VoEBase* base; - webrtc::VoECodec* codec; int send_channel_id; int receive_channel_id; }; @@ -70,7 +67,6 @@ void CreateVoiceEngine(VoiceEngineState* voe, decoder_factory) { voe->voice_engine = webrtc::VoiceEngine::Create(); voe->base = webrtc::VoEBase::GetInterface(voe->voice_engine); - voe->codec = webrtc::VoECodec::GetInterface(voe->voice_engine); EXPECT_EQ(0, voe->base->Init(nullptr, nullptr, decoder_factory)); webrtc::VoEBase::ChannelConfig config; config.enable_voice_pacing = true; @@ -87,8 +83,6 @@ void DestroyVoiceEngine(VoiceEngineState* voe) { voe->receive_channel_id = -1; voe->base->Release(); voe->base = nullptr; - voe->codec->Release(); - voe->codec = nullptr; webrtc::VoiceEngine::Delete(voe->voice_engine); voe->voice_engine = nullptr; @@ -1341,6 +1335,8 @@ void VideoQualityTest::RunWithRenderers(const Params& params) { audio_send_config_.min_bitrate_kbps = kOpusMinBitrate / 1000; audio_send_config_.max_bitrate_kbps = kOpusBitrateFb / 1000; } + audio_send_config_.send_codec_spec.codec_inst = + CodecInst{120, "OPUS", 48000, 960, 2, 64000}; audio_send_stream_ = call->CreateAudioSendStream(audio_send_config_); @@ -1356,9 +1352,6 @@ void VideoQualityTest::RunWithRenderers(const Params& params) { audio_config.sync_group = kSyncGroup; audio_receive_stream = call->CreateAudioReceiveStream(audio_config); - - const CodecInst kOpusInst = {120, "OPUS", 48000, 960, 2, 64000}; - EXPECT_EQ(0, voe.codec->SetSendCodec(voe.send_channel_id, kOpusInst)); } StartEncodedFrameLogs(video_receive_stream);