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..9532e6bd6a571d1d6a60f030f6da9380e61247f9 100644
--- a/webrtc/audio/audio_send_stream_unittest.cc
+++ b/webrtc/audio/audio_send_stream_unittest.cc
@@ -16,6 +16,7 @@
 #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 +56,8 @@ 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};
 
 class MockLimitObserver : public BitrateAllocator::LimitObserver {
  public:
@@ -112,7 +115,8 @@ 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.format = kIsacFormat;
     stream_config_.min_bitrate_bps = 10000;
     stream_config_.max_bitrate_bps = 65000;
   }
@@ -175,10 +179,11 @@ struct ConfigHelper {
     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));
+    EXPECT_CALL(*channel_proxy_, SetSendFormat(_, _, _)).WillOnce(Return(true));
+    // These should no longer be called.
+    EXPECT_CALL(*channel_proxy_, GetSendCodec(_))
+        .Times(0);

[kwiberg-webrtc, 2017/02/21 23:35:03]

Won't clang-format remove this line break?

+    EXPECT_CALL(*channel_proxy_, SetSendCodec(_)).Times(0);
   }
   RtcpRttStats* rtcp_rtt_stats() { return &rtcp_rtt_stats_; }
 
@@ -263,14 +268,12 @@ 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.format = kIsacFormat;
+  // TODO(ossu): Maybe mock this?
+  config.encoder_factory = CreateBuiltinAudioEncoderFactory();
   config.rtp.extensions.push_back(
       RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId));
   EXPECT_EQ(
@@ -279,10 +282,9 @@ 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, cng_plfreq: 56, payload_type: 103, "
+      "format: {name: isac, clockrate_hz: 16000, num_channels: 1, "
+      "parameters: {}}}}",
       config.ToString());
 }
 
@@ -385,28 +387,12 @@ TEST(AudioSendStreamTest, GetStatsTypingNoiseDetected) {
 TEST(AudioSendStreamTest, SendCodecAppliesConfigParams) {
   ConfigHelper helper(false);
   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;
+  const webrtc::SdpAudioFormat kOpusFormat = {"opus", 48000, 2};

[kwiberg-webrtc, 2017/02/21 23:35:03]

Too bad we can't make this constexpr. Remind me to talk about a nice small
vector optimization I learned about when we're both at the office at the same
time...

+  stream_config.send_codec_spec.format.format = kOpusFormat;
   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;
   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,
@@ -414,10 +400,6 @@ TEST(AudioSendStreamTest, SendCodecAppliesConfigParams) {
       .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));
   internal::AudioSendStream send_stream(
       stream_config, helper.audio_state(), helper.worker_queue(),
@@ -430,8 +412,9 @@ TEST(AudioSendStreamTest, SendCodecAppliesConfigParams) {
 TEST(AudioSendStreamTest, SendCodecCanApplyVad) {
   ConfigHelper helper(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.payload_type = 9;
+  stream_config.send_codec_spec.format.format = {"g722", 8000, 1};
+  stream_config.send_codec_spec.format.info = {16000, 1, 64000};
   stream_config.send_codec_spec.cng_plfreq = 8000;
   stream_config.send_codec_spec.cng_payload_type = 105;
   EXPECT_CALL(*helper.channel_proxy(), SetVADStatus(true))