Injectable audio encoders: WebRtcVoiceEngine and company

webrtc/audio/audio_send_stream_unittest.cc

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 that the second one can't be.

[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 now...

[ossu, 2017/03/20 18:19:48]

Cleaned up these, and probably a few other, unused functions from ChannelProxy.

+    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(),