Index: webrtc/audio/audio_send_stream_unittest.cc |
diff --git a/webrtc/audio/audio_send_stream_unittest.cc b/webrtc/audio/audio_send_stream_unittest.cc |
index 9ae0a879990db1e859426b389783691ac67fa0e0..8d0b2482d3538d895809eed24fe432130472e6a2 100644 |
--- a/webrtc/audio/audio_send_stream_unittest.cc |
+++ b/webrtc/audio/audio_send_stream_unittest.cc |
@@ -9,6 +9,7 @@ |
*/ |
#include <string> |
+#include <utility> |
#include <vector> |
#include "webrtc/audio/audio_send_stream.h" |
@@ -16,6 +17,7 @@ |
#include "webrtc/audio/conversion.h" |
#include "webrtc/base/task_queue.h" |
#include "webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h" |
+#include "webrtc/modules/audio_coding/codecs/builtin_audio_encoder_factory.h" |
#include "webrtc/modules/audio_mixer/audio_mixer_impl.h" |
#include "webrtc/modules/audio_processing/include/mock_audio_processing.h" |
#include "webrtc/modules/congestion_controller/include/congestion_controller.h" |
@@ -55,6 +57,8 @@ const int kTelephoneEventPayloadFrequency = 65432; |
const int kTelephoneEventCode = 45; |
const int kTelephoneEventDuration = 6789; |
const CodecInst kIsacCodec = {103, "isac", 16000, 320, 1, 32000}; |
+const int kIsacPayloadType = 103; |
+const SdpAudioFormat kIsacFormat = {"isac", 16000, 1}; |
kwiberg-webrtc
2017/03/01 12:26:47
The first one can be constexpr.
It makes me sad t
ossu
2017/03/20 18:19:48
Acknowledged.
|
class MockLimitObserver : public BitrateAllocator::LimitObserver { |
public: |
@@ -64,7 +68,7 @@ class MockLimitObserver : public BitrateAllocator::LimitObserver { |
}; |
struct ConfigHelper { |
- explicit ConfigHelper(bool audio_bwe_enabled) |
+ explicit ConfigHelper(bool audio_bwe_enabled, bool expect_set_encoder_call) |
kwiberg-webrtc
2017/03/01 12:26:47
Drop "explicit".
ossu
2017/03/02 01:30:28
Sure. Was in a rush!
|
: simulated_clock_(123456), |
stream_config_(nullptr), |
congestion_controller_(&simulated_clock_, |
@@ -96,7 +100,7 @@ struct ConfigHelper { |
return channel_proxy_; |
})); |
- SetupMockForSetupSendCodec(); |
+ SetupMockForSetupSendCodec(expect_set_encoder_call); |
stream_config_.voe_channel_id = kChannelId; |
stream_config_.rtp.ssrc = kSsrc; |
@@ -112,7 +116,10 @@ struct ConfigHelper { |
} |
// Use ISAC as default codec so as to prevent unnecessary |voice_engine_| |
// calls from the default ctor behavior. |
- stream_config_.send_codec_spec.codec_inst = kIsacCodec; |
+ stream_config_.send_codec_spec.payload_type = kIsacPayloadType; |
+ stream_config_.send_codec_spec.format = kIsacFormat; |
+ // TODO(ossu): Mock this factory. |
+ stream_config_.encoder_factory = CreateBuiltinAudioEncoderFactory(); |
stream_config_.min_bitrate_bps = 10000; |
stream_config_.max_bitrate_bps = 65000; |
} |
@@ -169,17 +176,19 @@ struct ConfigHelper { |
.Times(1); // Destructor resets the rtt stats. |
} |
- void SetupMockForSetupSendCodec() { |
- EXPECT_CALL(*channel_proxy_, SetVADStatus(false)) |
- .WillOnce(Return(true)); |
- EXPECT_CALL(*channel_proxy_, SetCodecFECStatus(false)) |
- .WillOnce(Return(true)); |
- EXPECT_CALL(*channel_proxy_, DisableAudioNetworkAdaptor()); |
- // Let |GetSendCodec| return false for the first time to indicate that no |
- // send codec has been set. |
- EXPECT_CALL(*channel_proxy_, GetSendCodec(_)).WillOnce(Return(false)); |
- EXPECT_CALL(*channel_proxy_, SetSendCodec(_)).WillOnce(Return(true)); |
+ void SetupMockForSetupSendCodec(bool expect_set_encoder_call) { |
+ if (expect_set_encoder_call) { |
+ EXPECT_CALL(*channel_proxy_, SetEncoderForMock(_, _)) |
+ .WillOnce(Return(true)); |
+ } |
+ // These should no longer be called. |
+ EXPECT_CALL(*channel_proxy_, SetVADStatus(_)).Times(0); |
the sun
2017/03/16 08:48:19
Well, or you could just remove those methods right
ossu
2017/03/20 18:19:48
Cleaned up these, and probably a few other, unused
|
+ EXPECT_CALL(*channel_proxy_, GetSendCodec(_)).Times(0); |
+ EXPECT_CALL(*channel_proxy_, SetSendCodec(_)).Times(0); |
+ EXPECT_CALL(*channel_proxy_, SetCodecFECStatus(false)).Times(0); |
+ EXPECT_CALL(*channel_proxy_, DisableAudioNetworkAdaptor()).Times(0); |
} |
+ |
RtcpRttStats* rtcp_rtt_stats() { return &rtcp_rtt_stats_; } |
void SetupMockForSendTelephoneEvent() { |
@@ -263,14 +272,11 @@ TEST(AudioSendStreamTest, ConfigToString) { |
config.max_bitrate_bps = 34000; |
config.send_codec_spec.nack_enabled = true; |
config.send_codec_spec.transport_cc_enabled = false; |
- config.send_codec_spec.enable_codec_fec = true; |
- config.send_codec_spec.enable_opus_dtx = false; |
- config.send_codec_spec.opus_max_playback_rate = 32000; |
config.send_codec_spec.cng_payload_type = 42; |
- config.send_codec_spec.cng_plfreq = 56; |
- config.send_codec_spec.min_ptime_ms = 20; |
- config.send_codec_spec.max_ptime_ms = 60; |
- config.send_codec_spec.codec_inst = kIsacCodec; |
+ config.send_codec_spec.payload_type = kIsacPayloadType; |
+ config.send_codec_spec.format = kIsacFormat; |
+ // TODO(ossu): Maybe mock this? |
+ config.encoder_factory = CreateBuiltinAudioEncoderFactory(); |
config.rtp.extensions.push_back( |
RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId)); |
EXPECT_EQ( |
@@ -279,15 +285,14 @@ TEST(AudioSendStreamTest, ConfigToString) { |
"{rtp_history_ms: 0}, c_name: foo_name}, send_transport: nullptr, " |
"voe_channel_id: 1, min_bitrate_bps: 12000, max_bitrate_bps: 34000, " |
"send_codec_spec: {nack_enabled: true, transport_cc_enabled: false, " |
- "enable_codec_fec: true, enable_opus_dtx: false, opus_max_playback_rate: " |
- "32000, cng_payload_type: 42, cng_plfreq: 56, min_ptime: 20, max_ptime: " |
- "60, codec_inst: {pltype: 103, plname: \"isac\", plfreq: 16000, pacsize: " |
- "320, channels: 1, rate: 32000}}}", |
+ "cng_payload_type: 42, payload_type: 103, " |
+ "format: {name: isac, clockrate_hz: 16000, num_channels: 1, " |
+ "parameters: {}}}}", |
config.ToString()); |
} |
TEST(AudioSendStreamTest, ConstructDestruct) { |
- ConfigHelper helper(false); |
+ ConfigHelper helper(false, true); |
internal::AudioSendStream send_stream( |
helper.config(), helper.audio_state(), helper.worker_queue(), |
helper.packet_router(), helper.congestion_controller(), |
@@ -295,7 +300,7 @@ TEST(AudioSendStreamTest, ConstructDestruct) { |
} |
TEST(AudioSendStreamTest, SendTelephoneEvent) { |
- ConfigHelper helper(false); |
+ ConfigHelper helper(false, true); |
internal::AudioSendStream send_stream( |
helper.config(), helper.audio_state(), helper.worker_queue(), |
helper.packet_router(), helper.congestion_controller(), |
@@ -307,7 +312,7 @@ TEST(AudioSendStreamTest, SendTelephoneEvent) { |
} |
TEST(AudioSendStreamTest, SetMuted) { |
- ConfigHelper helper(false); |
+ ConfigHelper helper(false, true); |
internal::AudioSendStream send_stream( |
helper.config(), helper.audio_state(), helper.worker_queue(), |
helper.packet_router(), helper.congestion_controller(), |
@@ -317,7 +322,7 @@ TEST(AudioSendStreamTest, SetMuted) { |
} |
TEST(AudioSendStreamTest, AudioBweCorrectObjectsOnChannelProxy) { |
- ConfigHelper helper(true); |
+ ConfigHelper helper(true, true); |
internal::AudioSendStream send_stream( |
helper.config(), helper.audio_state(), helper.worker_queue(), |
helper.packet_router(), helper.congestion_controller(), |
@@ -325,7 +330,7 @@ TEST(AudioSendStreamTest, AudioBweCorrectObjectsOnChannelProxy) { |
} |
TEST(AudioSendStreamTest, NoAudioBweCorrectObjectsOnChannelProxy) { |
- ConfigHelper helper(false); |
+ ConfigHelper helper(false, true); |
internal::AudioSendStream send_stream( |
helper.config(), helper.audio_state(), helper.worker_queue(), |
helper.packet_router(), helper.congestion_controller(), |
@@ -333,7 +338,7 @@ TEST(AudioSendStreamTest, NoAudioBweCorrectObjectsOnChannelProxy) { |
} |
TEST(AudioSendStreamTest, GetStats) { |
- ConfigHelper helper(false); |
+ ConfigHelper helper(false, true); |
internal::AudioSendStream send_stream( |
helper.config(), helper.audio_state(), helper.worker_queue(), |
helper.packet_router(), helper.congestion_controller(), |
@@ -364,7 +369,7 @@ TEST(AudioSendStreamTest, GetStats) { |
} |
TEST(AudioSendStreamTest, GetStatsTypingNoiseDetected) { |
- ConfigHelper helper(false); |
+ ConfigHelper helper(false, true); |
internal::AudioSendStream send_stream( |
helper.config(), helper.audio_state(), helper.worker_queue(), |
helper.packet_router(), helper.congestion_controller(), |
@@ -382,40 +387,13 @@ TEST(AudioSendStreamTest, GetStatsTypingNoiseDetected) { |
EXPECT_FALSE(send_stream.GetStats().typing_noise_detected); |
} |
-TEST(AudioSendStreamTest, SendCodecAppliesConfigParams) { |
- ConfigHelper helper(false); |
+TEST(AudioSendStreamTest, SendCodecAppliesNetworkAdaptor) { |
+ ConfigHelper helper(false, true); |
auto stream_config = helper.config(); |
- const CodecInst kOpusCodec = {111, "opus", 48000, 960, 2, 64000}; |
- stream_config.send_codec_spec.codec_inst = kOpusCodec; |
- stream_config.send_codec_spec.enable_codec_fec = true; |
- stream_config.send_codec_spec.enable_opus_dtx = true; |
- stream_config.send_codec_spec.opus_max_playback_rate = 12345; |
- stream_config.send_codec_spec.cng_plfreq = 16000; |
- stream_config.send_codec_spec.cng_payload_type = 105; |
- stream_config.send_codec_spec.min_ptime_ms = 10; |
- stream_config.send_codec_spec.max_ptime_ms = 60; |
+ const webrtc::SdpAudioFormat kOpusFormat = {"opus", 48000, 2}; |
+ stream_config.send_codec_spec.format = kOpusFormat; |
stream_config.audio_network_adaptor_config = |
rtc::Optional<std::string>("abced"); |
- EXPECT_CALL(*helper.channel_proxy(), SetCodecFECStatus(true)) |
- .WillOnce(Return(true)); |
- EXPECT_CALL( |
- *helper.channel_proxy(), |
- SetOpusDtx(stream_config.send_codec_spec.enable_opus_dtx)) |
- .WillOnce(Return(true)); |
- EXPECT_CALL( |
- *helper.channel_proxy(), |
- SetOpusMaxPlaybackRate( |
- stream_config.send_codec_spec.opus_max_playback_rate)) |
- .WillOnce(Return(true)); |
- EXPECT_CALL(*helper.channel_proxy(), |
- SetSendCNPayloadType( |
- stream_config.send_codec_spec.cng_payload_type, |
- webrtc::kFreq16000Hz)) |
- .WillOnce(Return(true)); |
- EXPECT_CALL( |
- *helper.channel_proxy(), |
- SetReceiverFrameLengthRange(stream_config.send_codec_spec.min_ptime_ms, |
- stream_config.send_codec_spec.max_ptime_ms)); |
EXPECT_CALL( |
*helper.channel_proxy(), |
EnableAudioNetworkAdaptor(*stream_config.audio_network_adaptor_config)); |
@@ -426,24 +404,37 @@ TEST(AudioSendStreamTest, SendCodecAppliesConfigParams) { |
} |
// VAD is applied when codec is mono and the CNG frequency matches the codec |
-// sample rate. |
+// clock rate. |
TEST(AudioSendStreamTest, SendCodecCanApplyVad) { |
- ConfigHelper helper(false); |
+ ConfigHelper helper(false, false); |
auto stream_config = helper.config(); |
- const CodecInst kG722Codec = {9, "g722", 8000, 160, 1, 16000}; |
- stream_config.send_codec_spec.codec_inst = kG722Codec; |
- stream_config.send_codec_spec.cng_plfreq = 8000; |
+ stream_config.send_codec_spec.payload_type = 9; |
+ stream_config.send_codec_spec.format = {"g722", 8000, 1}; |
stream_config.send_codec_spec.cng_payload_type = 105; |
- EXPECT_CALL(*helper.channel_proxy(), SetVADStatus(true)) |
- .WillOnce(Return(true)); |
+ |
+ using ::testing::Invoke; |
+ std::unique_ptr<AudioEncoder> stolen_encoder; |
+ EXPECT_CALL(*helper.channel_proxy(), SetEncoderForMock(_, _)) |
+ .WillOnce(Invoke([&stolen_encoder]( |
+ int payload_type, std::unique_ptr<AudioEncoder>& encoder) { |
+ stolen_encoder = std::move(encoder); |
+ return true; |
+ })); |
+ |
internal::AudioSendStream send_stream( |
stream_config, helper.audio_state(), helper.worker_queue(), |
helper.packet_router(), helper.congestion_controller(), |
helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats()); |
+ |
+ // We cannot truly determine if the encoder created is an AudioEncoderCng. It |
+ // is the only reasonable implementation that will return something from |
+ // ReclaimContainedEncoders, though. |
+ ASSERT_TRUE(stolen_encoder); |
+ EXPECT_FALSE(stolen_encoder->ReclaimContainedEncoders().empty()); |
} |
TEST(AudioSendStreamTest, DoesNotPassHigherBitrateThanMaxBitrate) { |
- ConfigHelper helper(false); |
+ ConfigHelper helper(false, true); |
internal::AudioSendStream send_stream( |
helper.config(), helper.audio_state(), helper.worker_queue(), |
helper.packet_router(), helper.congestion_controller(), |
@@ -455,7 +446,7 @@ TEST(AudioSendStreamTest, DoesNotPassHigherBitrateThanMaxBitrate) { |
} |
TEST(AudioSendStreamTest, ProbingIntervalOnBitrateUpdated) { |
- ConfigHelper helper(false); |
+ ConfigHelper helper(false, true); |
internal::AudioSendStream send_stream( |
helper.config(), helper.audio_state(), helper.worker_queue(), |
helper.packet_router(), helper.congestion_controller(), |