Implement AudioEncoder::GetFrameLengthRange() for all audio encoders.

The WebRTC-SendSideBwe-WithOverhead field trial requires audio
encoders to properly implement the
AudioEncoder::GetFrameLengthRange() function. Thic CL implements
the function for all audio encoders in WebRTC in preparation for
making that function pure virtual in the interface.


Bug: webrtc:11427
Change-Id: Ieab6b6c72c62af6ac9525a20fcb39bd477079551
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171503
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30890}
diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index 8efc221..ceee0c0 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -101,8 +101,10 @@
   deps = [
     ":webrtc_cng",
     "../../api/audio_codecs:audio_codecs_api",
+    "../../api/units:time_delta",
     "../../common_audio",
     "../../rtc_base:checks",
+    "//third_party/abseil-cpp/absl/types:optional",
   ]
 }
 
@@ -116,6 +118,7 @@
   deps = [
     "../../api:array_view",
     "../../api/audio_codecs:audio_codecs_api",
+    "../../api/units:time_delta",
     "../../common_audio",
     "../../rtc_base:checks",
     "../../rtc_base:rtc_base_approved",
@@ -137,8 +140,10 @@
     ":legacy_encoded_audio_frame",
     "../../api:array_view",
     "../../api/audio_codecs:audio_codecs_api",
+    "../../api/units:time_delta",
     "../../rtc_base:checks",
     "../../rtc_base:rtc_base_approved",
+    "//third_party/abseil-cpp/absl/types:optional",
   ]
   public_deps = [ ":g711_c" ]  # no-presubmit-check TODO(webrtc:8603)
 }
@@ -167,8 +172,10 @@
     "../../api:array_view",
     "../../api/audio_codecs:audio_codecs_api",
     "../../api/audio_codecs/g722:audio_encoder_g722_config",
+    "../../api/units:time_delta",
     "../../rtc_base:checks",
     "../../rtc_base:rtc_base_approved",
+    "//third_party/abseil-cpp/absl/types:optional",
   ]
   public_deps = [ ":g722_c" ]  # no-presubmit-check TODO(webrtc:8603)
 }
@@ -197,9 +204,11 @@
     "../../api:array_view",
     "../../api/audio_codecs:audio_codecs_api",
     "../../api/audio_codecs/ilbc:audio_encoder_ilbc_config",
+    "../../api/units:time_delta",
     "../../common_audio",
     "../../rtc_base:checks",
     "../../rtc_base:rtc_base_approved",
+    "//third_party/abseil-cpp/absl/types:optional",
   ]
   public_deps = [ ":ilbc_c" ]  # no-presubmit-check TODO(webrtc:8603)
 }
@@ -372,6 +381,7 @@
     ":isac_bwinfo",
     "../../api:scoped_refptr",
     "../../api/audio_codecs:audio_codecs_api",
+    "../../api/units:time_delta",
     "../../rtc_base:checks",
     "../../rtc_base:rtc_base_approved",
     "//third_party/abseil-cpp/absl/types:optional",
@@ -771,6 +781,7 @@
     "../../api/audio_codecs:audio_codecs_api",
     "../../api/audio_codecs/opus:audio_decoder_opus_config",
     "../../api/audio_codecs/opus:audio_encoder_opus_config",
+    "../../api/units:time_delta",
     "../../rtc_base:checks",
     "../../rtc_base:logging",
     "../../rtc_base:macromagic",
diff --git a/modules/audio_coding/codecs/cng/audio_encoder_cng.cc b/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
index 86d3f38..600cb0c 100644
--- a/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
+++ b/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
@@ -14,6 +14,8 @@
 #include <memory>
 #include <utility>
 
+#include "absl/types/optional.h"
+#include "api/units/time_delta.h"
 #include "modules/audio_coding/codecs/cng/webrtc_cng.h"
 #include "rtc_base/checks.h"
 
@@ -55,6 +57,8 @@
   void OnReceivedUplinkBandwidth(
       int target_audio_bitrate_bps,
       absl::optional<int64_t> bwe_period_ms) override;
+  absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
+      const override;
 
  private:
   EncodedInfo EncodePassive(size_t frames_to_encode, rtc::Buffer* encoded);
@@ -225,6 +229,11 @@
                                              bwe_period_ms);
 }
 
+absl::optional<std::pair<TimeDelta, TimeDelta>>
+AudioEncoderCng::GetFrameLengthRange() const {
+  return speech_encoder_->GetFrameLengthRange();
+}
+
 AudioEncoder::EncodedInfo AudioEncoderCng::EncodePassive(
     size_t frames_to_encode,
     rtc::Buffer* encoded) {
diff --git a/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc b/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
index 085deb1..547fedd 100644
--- a/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
+++ b/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
@@ -21,8 +21,11 @@
 #include "test/testsupport/rtc_expect_death.h"
 
 using ::testing::_;
+using ::testing::Eq;
 using ::testing::InSequence;
 using ::testing::Invoke;
+using ::testing::Not;
+using ::testing::Optional;
 using ::testing::Return;
 using ::testing::SetArgPointee;
 
@@ -233,6 +236,15 @@
   cng_->OnReceivedUplinkPacketLossFraction(0.5);
 }
 
+TEST_F(AudioEncoderCngTest, CheckGetFrameLengthRangePropagation) {
+  CreateCng(MakeCngConfig());
+  auto expected_range =
+      std::make_pair(TimeDelta::Millis(20), TimeDelta::Millis(20));
+  EXPECT_CALL(*mock_encoder_, GetFrameLengthRange())
+      .WillRepeatedly(Return(absl::make_optional(expected_range)));
+  EXPECT_THAT(cng_->GetFrameLengthRange(), Optional(Eq(expected_range)));
+}
+
 TEST_F(AudioEncoderCngTest, EncodeCallsVad) {
   EXPECT_CALL(*mock_encoder_, Num10MsFramesInNextPacket())
       .WillRepeatedly(Return(1U));
diff --git a/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc b/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
index fd560e4..65e2da4 100644
--- a/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
+++ b/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
@@ -89,6 +89,12 @@
   speech_buffer_.clear();
 }
 
+absl::optional<std::pair<TimeDelta, TimeDelta>>
+AudioEncoderPcm::GetFrameLengthRange() const {
+  return {{TimeDelta::Millis(num_10ms_frames_per_packet_ * 10),
+           TimeDelta::Millis(num_10ms_frames_per_packet_ * 10)}};
+}
+
 size_t AudioEncoderPcmA::EncodeCall(const int16_t* audio,
                                     size_t input_len,
                                     uint8_t* encoded) {
diff --git a/modules/audio_coding/codecs/g711/audio_encoder_pcm.h b/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
index e41c2a3..c4413f5 100644
--- a/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
+++ b/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
@@ -11,9 +11,12 @@
 #ifndef MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_
 #define MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_
 
+#include <utility>
 #include <vector>
 
+#include "absl/types/optional.h"
 #include "api/audio_codecs/audio_encoder.h"
+#include "api/units/time_delta.h"
 #include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
@@ -41,6 +44,8 @@
   size_t Max10MsFramesInAPacket() const override;
   int GetTargetBitrate() const override;
   void Reset() override;
+  absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
+      const override;
 
  protected:
   AudioEncoderPcm(const Config& config, int sample_rate_hz);
diff --git a/modules/audio_coding/codecs/g722/audio_encoder_g722.cc b/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
index d293163..b7d34ba 100644
--- a/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
+++ b/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
@@ -79,6 +79,12 @@
     RTC_CHECK_EQ(0, WebRtcG722_EncoderInit(encoders_[i].encoder));
 }
 
+absl::optional<std::pair<TimeDelta, TimeDelta>>
+AudioEncoderG722Impl::GetFrameLengthRange() const {
+  return {{TimeDelta::Millis(num_10ms_frames_per_packet_ * 10),
+           TimeDelta::Millis(num_10ms_frames_per_packet_ * 10)}};
+}
+
 AudioEncoder::EncodedInfo AudioEncoderG722Impl::EncodeImpl(
     uint32_t rtp_timestamp,
     rtc::ArrayView<const int16_t> audio,
diff --git a/modules/audio_coding/codecs/g722/audio_encoder_g722.h b/modules/audio_coding/codecs/g722/audio_encoder_g722.h
index cf45fb5..c836503 100644
--- a/modules/audio_coding/codecs/g722/audio_encoder_g722.h
+++ b/modules/audio_coding/codecs/g722/audio_encoder_g722.h
@@ -12,9 +12,12 @@
 #define MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_
 
 #include <memory>
+#include <utility>
 
+#include "absl/types/optional.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/audio_codecs/g722/audio_encoder_g722_config.h"
+#include "api/units/time_delta.h"
 #include "modules/audio_coding/codecs/g722/g722_interface.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/constructor_magic.h"
@@ -33,6 +36,8 @@
   size_t Max10MsFramesInAPacket() const override;
   int GetTargetBitrate() const override;
   void Reset() override;
+  absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
+      const override;
 
  protected:
   EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
diff --git a/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc b/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
index ac9214b..032de20 100644
--- a/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
+++ b/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
@@ -127,6 +127,12 @@
   num_10ms_frames_buffered_ = 0;
 }
 
+absl::optional<std::pair<TimeDelta, TimeDelta>>
+AudioEncoderIlbcImpl::GetFrameLengthRange() const {
+  return {{TimeDelta::Millis(num_10ms_frames_per_packet_ * 10),
+           TimeDelta::Millis(num_10ms_frames_per_packet_ * 10)}};
+}
+
 size_t AudioEncoderIlbcImpl::RequiredOutputSizeBytes() const {
   switch (num_10ms_frames_per_packet_) {
     case 2:
diff --git a/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h b/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h
index ada613b..fe3e329 100644
--- a/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h
+++ b/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h
@@ -11,8 +11,12 @@
 #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
 #define MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
 
+#include <utility>
+
+#include "absl/types/optional.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/audio_codecs/ilbc/audio_encoder_ilbc_config.h"
+#include "api/units/time_delta.h"
 #include "modules/audio_coding/codecs/ilbc/ilbc.h"
 #include "rtc_base/constructor_magic.h"
 
@@ -32,6 +36,8 @@
                          rtc::ArrayView<const int16_t> audio,
                          rtc::Buffer* encoded) override;
   void Reset() override;
+  absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
+      const override;
 
  private:
   size_t RequiredOutputSizeBytes() const;
diff --git a/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h b/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
index c7b595107..a3b8e76 100644
--- a/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
+++ b/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
@@ -11,10 +11,13 @@
 #ifndef MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_
 #define MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_
 
+#include <utility>
 #include <vector>
 
+#include "absl/types/optional.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/scoped_refptr.h"
+#include "api/units/time_delta.h"
 #include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
@@ -49,6 +52,8 @@
                          rtc::ArrayView<const int16_t> audio,
                          rtc::Buffer* encoded) override;
   void Reset() override;
+  absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
+      const override;
 
  private:
   // This value is taken from STREAM_SIZE_MAX_60 for iSAC float (60 ms) and
diff --git a/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h b/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
index 27a02b5..9ddb943 100644
--- a/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
+++ b/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
@@ -120,6 +120,13 @@
 }
 
 template <typename T>
+absl::optional<std::pair<TimeDelta, TimeDelta>>
+AudioEncoderIsacT<T>::GetFrameLengthRange() const {
+  return {{TimeDelta::Millis(config_.frame_size_ms),
+           TimeDelta::Millis(config_.frame_size_ms)}};
+}
+
+template <typename T>
 void AudioEncoderIsacT<T>::RecreateEncoderInstance(const Config& config) {
   RTC_CHECK(config.IsOk());
   packet_in_progress_ = false;
diff --git a/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.cc b/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.cc
index 0614a0b..1feef3d 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.cc
+++ b/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.cc
@@ -164,6 +164,12 @@
   RTC_CHECK(RecreateEncoderInstance(config_));
 }
 
+absl::optional<std::pair<TimeDelta, TimeDelta>>
+AudioEncoderMultiChannelOpusImpl::GetFrameLengthRange() const {
+  return {{TimeDelta::Millis(config_.frame_size_ms),
+           TimeDelta::Millis(config_.frame_size_ms)}};
+}
+
 // If the given config is OK, recreate the Opus encoder instance with those
 // settings, save the config, and return true. Otherwise, do nothing and return
 // false.
diff --git a/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.h b/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.h
index 593068c..eadb4a6 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.h
+++ b/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.h
@@ -12,12 +12,14 @@
 #define MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_MULTI_CHANNEL_OPUS_IMPL_H_
 
 #include <memory>
+#include <utility>
 #include <vector>
 
 #include "absl/types/optional.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/audio_codecs/audio_format.h"
 #include "api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.h"
+#include "api/units/time_delta.h"
 #include "modules/audio_coding/codecs/opus/opus_interface.h"
 #include "rtc_base/constructor_magic.h"
 
@@ -44,6 +46,8 @@
   int GetTargetBitrate() const override;
 
   void Reset() override;
+  absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
+      const override;
 
  protected:
   EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
diff --git a/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc b/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
index 05aaca1..e75806a 100644
--- a/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
+++ b/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
@@ -132,4 +132,9 @@
                                              bwe_period_ms);
 }
 
+absl::optional<std::pair<TimeDelta, TimeDelta>>
+AudioEncoderCopyRed::GetFrameLengthRange() const {
+  return speech_encoder_->GetFrameLengthRange();
+}
+
 }  // namespace webrtc
diff --git a/modules/audio_coding/codecs/red/audio_encoder_copy_red.h b/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
index 2dc13dd..c6e829e 100644
--- a/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
+++ b/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
@@ -15,10 +15,12 @@
 #include <stdint.h>
 
 #include <memory>
+#include <utility>
 
 #include "absl/types/optional.h"
 #include "api/array_view.h"
 #include "api/audio_codecs/audio_encoder.h"
+#include "api/units/time_delta.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/constructor_magic.h"
 
@@ -60,6 +62,8 @@
   void OnReceivedUplinkBandwidth(
       int target_audio_bitrate_bps,
       absl::optional<int64_t> bwe_period_ms) override;
+  absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
+      const override;
 
  protected:
   EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
diff --git a/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc b/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
index 648a88d..e20515a 100644
--- a/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
+++ b/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
@@ -20,9 +20,12 @@
 #include "test/testsupport/rtc_expect_death.h"
 
 using ::testing::_;
+using ::testing::Eq;
 using ::testing::InSequence;
 using ::testing::Invoke;
 using ::testing::MockFunction;
+using ::testing::Not;
+using ::testing::Optional;
 using ::testing::Return;
 using ::testing::SetArgPointee;
 
@@ -107,6 +110,14 @@
   red_->OnReceivedUplinkPacketLossFraction(0.5);
 }
 
+TEST_F(AudioEncoderCopyRedTest, CheckGetFrameLengthRangePropagation) {
+  auto expected_range =
+      std::make_pair(TimeDelta::Millis(20), TimeDelta::Millis(20));
+  EXPECT_CALL(*mock_encoder_, GetFrameLengthRange())
+      .WillRepeatedly(Return(absl::make_optional(expected_range)));
+  EXPECT_THAT(red_->GetFrameLengthRange(), Optional(Eq(expected_range)));
+}
+
 // Checks that the an Encode() call is immediately propagated to the speech
 // encoder.
 TEST_F(AudioEncoderCopyRedTest, CheckImmediateEncode) {