Move ownership of voe::Channel into Audio[Receive|Send]Stream.
* VoEBase contains only stub methods (until downstream code is
updated).
* voe::Channel and ChannelProxy classes remain, but are now created
internally to the streams. As a result,
internal::Audio[Receive|Send]Stream can have a ChannelProxy injected
for testing.
* Stream classes share Call::module_process_thread_ for their RtpRtcp
modules, rather than using a separate thread shared only among audio
streams.
* voe::Channel instances use Call::worker_queue_ for encoding packets,
rather than having a separate queue for audio (send) streams.
Bug: webrtc:4690
Change-Id: I8059ef224ad13aa0a6ded2cafc52599c7f64d68d
Reviewed-on: https://webrtc-review.googlesource.com/34640
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21578}diff --git a/audio/audio_send_stream_unittest.cc b/audio/audio_send_stream_unittest.cc
index 5afdffe..4644bf4 100644
--- a/audio/audio_send_stream_unittest.cc
+++ b/audio/audio_send_stream_unittest.cc
@@ -33,7 +33,6 @@
#include "test/mock_audio_encoder.h"
#include "test/mock_audio_encoder_factory.h"
#include "test/mock_voe_channel_proxy.h"
-#include "test/mock_voice_engine.h"
namespace webrtc {
namespace test {
@@ -46,7 +45,6 @@
using testing::Return;
using testing::StrEq;
-const int kChannelId = 1;
const uint32_t kSsrc = 1234;
const char* kCName = "foo_name";
const int kAudioLevelId = 2;
@@ -141,7 +139,6 @@
using testing::Invoke;
AudioState::Config config;
- config.voice_engine = &voice_engine_;
config.audio_mixer = AudioMixerImpl::Create();
config.audio_processing = audio_processing_;
config.audio_device_module =
@@ -149,19 +146,12 @@
audio_state_ = AudioState::Create(config);
SetupDefaultChannelProxy(audio_bwe_enabled);
-
- EXPECT_CALL(voice_engine_, ChannelProxyFactory(kChannelId))
- .WillOnce(Invoke([this](int channel_id) {
- return channel_proxy_;
- }));
-
SetupMockForSetupSendCodec(expect_set_encoder_call);
- // Use ISAC as default codec so as to prevent unnecessary |voice_engine_|
+ // Use ISAC as default codec so as to prevent unnecessary |channel_proxy_|
// calls from the default ctor behavior.
stream_config_.send_codec_spec =
AudioSendStream::Config::SendCodecSpec(kIsacPayloadType, kIsacFormat);
- stream_config_.voe_channel_id = kChannelId;
stream_config_.rtp.ssrc = kSsrc;
stream_config_.rtp.nack.rtp_history_ms = 200;
stream_config_.rtp.c_name = kCName;
@@ -175,18 +165,27 @@
stream_config_.max_bitrate_bps = 65000;
}
+ std::unique_ptr<internal::AudioSendStream> CreateAudioSendStream() {
+ return std::unique_ptr<internal::AudioSendStream>(
+ new internal::AudioSendStream(
+ stream_config_,
+ audio_state_,
+ &worker_queue_,
+ &fake_transport_,
+ &bitrate_allocator_,
+ &event_log_,
+ &rtcp_rtt_stats_,
+ rtc::nullopt,
+ std::unique_ptr<voe::ChannelProxy>(channel_proxy_)));
+ }
+
AudioSendStream::Config& config() { return stream_config_; }
MockAudioEncoderFactory& mock_encoder_factory() {
return *static_cast<MockAudioEncoderFactory*>(
stream_config_.encoder_factory.get());
}
- rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; }
MockVoEChannelProxy* channel_proxy() { return channel_proxy_; }
RtpTransportControllerSendInterface* transport() { return &fake_transport_; }
- BitrateAllocator* bitrate_allocator() { return &bitrate_allocator_; }
- rtc::TaskQueue* worker_queue() { return &worker_queue_; }
- RtcEventLog* event_log() { return &event_log_; }
- MockVoiceEngine* voice_engine() { return &voice_engine_; }
static void AddBweToConfig(AudioSendStream::Config* config) {
config->rtp.extensions.push_back(
@@ -255,8 +254,6 @@
}));
}
- RtcpRttStats* rtcp_rtt_stats() { return &rtcp_rtt_stats_; }
-
void SetupMockForSendTelephoneEvent() {
EXPECT_TRUE(channel_proxy_);
EXPECT_CALL(*channel_proxy_,
@@ -305,7 +302,6 @@
}
private:
- testing::StrictMock<MockVoiceEngine> voice_engine_;
rtc::scoped_refptr<AudioState> audio_state_;
AudioSendStream::Config stream_config_;
testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr;
@@ -332,7 +328,6 @@
AudioSendStream::Config config(nullptr);
config.rtp.ssrc = kSsrc;
config.rtp.c_name = kCName;
- config.voe_channel_id = kChannelId;
config.min_bitrate_bps = 12000;
config.max_bitrate_bps = 34000;
config.send_codec_spec =
@@ -347,7 +342,7 @@
"{rtp: {ssrc: 1234, extensions: [{uri: "
"urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 2}], nack: "
"{rtp_history_ms: 0}, c_name: foo_name}, send_transport: null, "
- "voe_channel_id: 1, min_bitrate_bps: 12000, max_bitrate_bps: 34000, "
+ "min_bitrate_bps: 12000, max_bitrate_bps: 34000, "
"send_codec_spec: {nack_enabled: true, transport_cc_enabled: false, "
"cng_payload_type: 42, payload_type: 103, "
"format: {name: isac, clockrate_hz: 16000, num_channels: 1, "
@@ -357,58 +352,40 @@
TEST(AudioSendStreamTest, ConstructDestruct) {
ConfigHelper helper(false, true);
- internal::AudioSendStream send_stream(
- helper.config(), helper.audio_state(), helper.worker_queue(),
- helper.transport(), helper.bitrate_allocator(), helper.event_log(),
- helper.rtcp_rtt_stats(), rtc::nullopt);
+ auto send_stream = helper.CreateAudioSendStream();
}
TEST(AudioSendStreamTest, SendTelephoneEvent) {
ConfigHelper helper(false, true);
- internal::AudioSendStream send_stream(
- helper.config(), helper.audio_state(), helper.worker_queue(),
- helper.transport(), helper.bitrate_allocator(), helper.event_log(),
- helper.rtcp_rtt_stats(), rtc::nullopt);
+ auto send_stream = helper.CreateAudioSendStream();
helper.SetupMockForSendTelephoneEvent();
- EXPECT_TRUE(send_stream.SendTelephoneEvent(kTelephoneEventPayloadType,
+ EXPECT_TRUE(send_stream->SendTelephoneEvent(kTelephoneEventPayloadType,
kTelephoneEventPayloadFrequency, kTelephoneEventCode,
kTelephoneEventDuration));
}
TEST(AudioSendStreamTest, SetMuted) {
ConfigHelper helper(false, true);
- internal::AudioSendStream send_stream(
- helper.config(), helper.audio_state(), helper.worker_queue(),
- helper.transport(), helper.bitrate_allocator(), helper.event_log(),
- helper.rtcp_rtt_stats(), rtc::nullopt);
+ auto send_stream = helper.CreateAudioSendStream();
EXPECT_CALL(*helper.channel_proxy(), SetInputMute(true));
- send_stream.SetMuted(true);
+ send_stream->SetMuted(true);
}
TEST(AudioSendStreamTest, AudioBweCorrectObjectsOnChannelProxy) {
ConfigHelper helper(true, true);
- internal::AudioSendStream send_stream(
- helper.config(), helper.audio_state(), helper.worker_queue(),
- helper.transport(), helper.bitrate_allocator(), helper.event_log(),
- helper.rtcp_rtt_stats(), rtc::nullopt);
+ auto send_stream = helper.CreateAudioSendStream();
}
TEST(AudioSendStreamTest, NoAudioBweCorrectObjectsOnChannelProxy) {
ConfigHelper helper(false, true);
- internal::AudioSendStream send_stream(
- helper.config(), helper.audio_state(), helper.worker_queue(),
- helper.transport(), helper.bitrate_allocator(), helper.event_log(),
- helper.rtcp_rtt_stats(), rtc::nullopt);
+ auto send_stream = helper.CreateAudioSendStream();
}
TEST(AudioSendStreamTest, GetStats) {
ConfigHelper helper(false, true);
- internal::AudioSendStream send_stream(
- helper.config(), helper.audio_state(), helper.worker_queue(),
- helper.transport(), helper.bitrate_allocator(), helper.event_log(),
- helper.rtcp_rtt_stats(), rtc::nullopt);
+ auto send_stream = helper.CreateAudioSendStream();
helper.SetupMockForGetStats();
- AudioSendStream::Stats stats = send_stream.GetStats(true);
+ AudioSendStream::Stats stats = send_stream->GetStats(true);
EXPECT_EQ(kSsrc, stats.local_ssrc);
EXPECT_EQ(static_cast<int64_t>(kCallStats.bytesSent), stats.bytes_sent);
EXPECT_EQ(kCallStats.packetsSent, stats.packets_sent);
@@ -441,13 +418,12 @@
TEST(AudioSendStreamTest, SendCodecAppliesAudioNetworkAdaptor) {
ConfigHelper helper(false, true);
- auto stream_config = helper.config();
- stream_config.send_codec_spec =
+ helper.config().send_codec_spec =
AudioSendStream::Config::SendCodecSpec(0, kOpusFormat);
const std::string kAnaConfigString = "abcde";
const std::string kAnaReconfigString = "12345";
- stream_config.audio_network_adaptor_config = kAnaConfigString;
+ helper.config().audio_network_adaptor_config = kAnaConfigString;
EXPECT_CALL(helper.mock_encoder_factory(), MakeAudioEncoderMock(_, _, _))
.WillOnce(Invoke([&kAnaConfigString, &kAnaReconfigString](
@@ -463,25 +439,22 @@
*return_value = std::move(mock_encoder);
}));
- internal::AudioSendStream send_stream(
- stream_config, helper.audio_state(), helper.worker_queue(),
- helper.transport(), helper.bitrate_allocator(), helper.event_log(),
- helper.rtcp_rtt_stats(), rtc::nullopt);
+ auto send_stream = helper.CreateAudioSendStream();
+ auto stream_config = helper.config();
stream_config.audio_network_adaptor_config = kAnaReconfigString;
helper.SetupMockForModifyEncoder();
- send_stream.Reconfigure(stream_config);
+ send_stream->Reconfigure(stream_config);
}
// VAD is applied when codec is mono and the CNG frequency matches the codec
// clock rate.
TEST(AudioSendStreamTest, SendCodecCanApplyVad) {
ConfigHelper helper(false, false);
- auto stream_config = helper.config();
- stream_config.send_codec_spec =
+ helper.config().send_codec_spec =
AudioSendStream::Config::SendCodecSpec(9, kG722Format);
- stream_config.send_codec_spec->cng_payload_type = 105;
+ helper.config().send_codec_spec->cng_payload_type = 105;
using ::testing::Invoke;
std::unique_ptr<AudioEncoder> stolen_encoder;
EXPECT_CALL(*helper.channel_proxy(), SetEncoderForMock(_, _))
@@ -492,10 +465,7 @@
return true;
}));
- internal::AudioSendStream send_stream(
- stream_config, helper.audio_state(), helper.worker_queue(),
- helper.transport(), helper.bitrate_allocator(), helper.event_log(),
- helper.rtcp_rtt_stats(), rtc::nullopt);
+ auto send_stream = helper.CreateAudioSendStream();
// We cannot truly determine if the encoder created is an AudioEncoderCng. It
// is the only reasonable implementation that will return something from
@@ -506,24 +476,18 @@
TEST(AudioSendStreamTest, DoesNotPassHigherBitrateThanMaxBitrate) {
ConfigHelper helper(false, true);
- internal::AudioSendStream send_stream(
- helper.config(), helper.audio_state(), helper.worker_queue(),
- helper.transport(), helper.bitrate_allocator(), helper.event_log(),
- helper.rtcp_rtt_stats(), rtc::nullopt);
+ auto send_stream = helper.CreateAudioSendStream();
EXPECT_CALL(*helper.channel_proxy(),
SetBitrate(helper.config().max_bitrate_bps, _));
- send_stream.OnBitrateUpdated(helper.config().max_bitrate_bps + 5000, 0.0, 50,
- 6000);
+ send_stream->OnBitrateUpdated(helper.config().max_bitrate_bps + 5000, 0.0, 50,
+ 6000);
}
TEST(AudioSendStreamTest, ProbingIntervalOnBitrateUpdated) {
ConfigHelper helper(false, true);
- internal::AudioSendStream send_stream(
- helper.config(), helper.audio_state(), helper.worker_queue(),
- helper.transport(), helper.bitrate_allocator(), helper.event_log(),
- helper.rtcp_rtt_stats(), rtc::nullopt);
+ auto send_stream = helper.CreateAudioSendStream();
EXPECT_CALL(*helper.channel_proxy(), SetBitrate(_, 5000));
- send_stream.OnBitrateUpdated(50000, 0.0, 50, 5000);
+ send_stream->OnBitrateUpdated(50000, 0.0, 50, 5000);
}
// Test that AudioSendStream doesn't recreate the encoder unnecessarily.
@@ -537,23 +501,16 @@
EXPECT_CALL(*helper.channel_proxy(), SetEncoderForMock(_, _))
.WillOnce(Return(true));
- auto stream_config = helper.config();
- stream_config.send_codec_spec =
+ helper.config().send_codec_spec =
AudioSendStream::Config::SendCodecSpec(9, kG722Format);
- stream_config.send_codec_spec->cng_payload_type = 105;
- internal::AudioSendStream send_stream(
- stream_config, helper.audio_state(), helper.worker_queue(),
- helper.transport(), helper.bitrate_allocator(), helper.event_log(),
- helper.rtcp_rtt_stats(), rtc::nullopt);
- send_stream.Reconfigure(stream_config);
+ helper.config().send_codec_spec->cng_payload_type = 105;
+ auto send_stream = helper.CreateAudioSendStream();
+ send_stream->Reconfigure(helper.config());
}
TEST(AudioSendStreamTest, ReconfigureTransportCcResetsFirst) {
ConfigHelper helper(false, true);
- internal::AudioSendStream send_stream(
- helper.config(), helper.audio_state(), helper.worker_queue(),
- helper.transport(), helper.bitrate_allocator(), helper.event_log(),
- helper.rtcp_rtt_stats(), rtc::nullopt);
+ auto send_stream = helper.CreateAudioSendStream();
auto new_config = helper.config();
ConfigHelper::AddBweToConfig(&new_config);
EXPECT_CALL(*helper.channel_proxy(),
@@ -567,7 +524,7 @@
helper.transport(), Ne(nullptr)))
.Times(1);
}
- send_stream.Reconfigure(new_config);
+ send_stream->Reconfigure(new_config);
}
} // namespace test
} // namespace webrtc