Delete AudioDecoder method IncomingPacket

Only the ISAC codec had an non-trivial implementation, for its unused
adaptive mode. This cl deletes that implementation, and the call
from NetEq, and the interface method.

Bug: webrtc:10098
Change-Id: Iaf7667e0ae867fc9d64286dff4c01a8ce0b6e2a4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153882
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29279}

api/audio_codecs/audio_decoder.cc

@@ -135,14 +135,6 @@ size_t AudioDecoder::DecodePlc(size_t num_frames, int16_t* decoded) {
 void AudioDecoder::GeneratePlc(size_t /*requested_samples_per_channel*/,
                                rtc::BufferT<int16_t>* /*concealment_audio*/) {}
 
-int AudioDecoder::IncomingPacket(const uint8_t* payload,
-                                 size_t payload_len,
-                                 uint16_t rtp_sequence_number,
-                                 uint32_t rtp_timestamp,
-                                 uint32_t arrival_timestamp) {
-  return 0;
-}
-
 int AudioDecoder::ErrorCode() {
   return 0;
 }

api/audio_codecs/audio_decoder.h

@@ -143,13 +143,6 @@ class AudioDecoder {
   // Resets the decoder state (empty buffers etc.).
   virtual void Reset() = 0;
 
-  // Notifies the decoder of an incoming packet to NetEQ.
-  virtual int IncomingPacket(const uint8_t* payload,
-                             size_t payload_len,
-                             uint16_t rtp_sequence_number,
-                             uint32_t rtp_timestamp,
-                             uint32_t arrival_timestamp);
-
   // Returns the last error code from the decoder.
   virtual int ErrorCode();
 

modules/audio_coding/acm2/audio_coding_module_unittest.cc

@@ -973,10 +973,6 @@ TEST_F(AcmReceiverBitExactnessOldApi, 48kHzOutputExternalDecoder) {
           fact_(CreateBuiltinAudioDecoderFactory()) {
       // Set expectations on the mock decoder and also delegate the calls to
       // the real decoder.
-      EXPECT_CALL(*mock_decoder_, IncomingPacket(_, _, _, _, _))
-          .Times(AtLeast(1))
-          .WillRepeatedly(
-              Invoke(&pcmu_decoder_, &AudioDecoderPcmU::IncomingPacket));
       EXPECT_CALL(*mock_decoder_, SampleRateHz())
           .Times(AtLeast(1))
           .WillRepeatedly(

modules/audio_coding/codecs/isac/audio_decoder_isac_t.h

@@ -33,11 +33,6 @@ class AudioDecoderIsacT final : public AudioDecoder {
   bool HasDecodePlc() const override;
   size_t DecodePlc(size_t num_frames, int16_t* decoded) override;
   void Reset() override;
-  int IncomingPacket(const uint8_t* payload,
-                     size_t payload_len,
-                     uint16_t rtp_sequence_number,
-                     uint32_t rtp_timestamp,
-                     uint32_t arrival_timestamp) override;
   int ErrorCode() override;
   int SampleRateHz() const override;
   size_t Channels() const override;

modules/audio_coding/codecs/isac/audio_decoder_isac_t_impl.h

@@ -64,18 +64,6 @@ void AudioDecoderIsacT<T>::Reset() {
   T::DecoderInit(isac_state_);
 }
 
-template <typename T>
-int AudioDecoderIsacT<T>::IncomingPacket(const uint8_t* payload,
-                                         size_t payload_len,
-                                         uint16_t rtp_sequence_number,
-                                         uint32_t rtp_timestamp,
-                                         uint32_t arrival_timestamp) {
-  int ret = T::UpdateBwEstimate(isac_state_, payload, payload_len,
-                                rtp_sequence_number, rtp_timestamp,
-                                arrival_timestamp);
-  return ret;
-}
-
 template <typename T>
 int AudioDecoderIsacT<T>::ErrorCode() {
   return T::GetErrorCode(isac_state_);

modules/audio_coding/codecs/isac/fix/source/isac_fix_type.h

@@ -90,15 +90,6 @@ class IsacFix {
                                              int bottleneck_bits_per_second) {
     WebRtcIsacfix_SetInitialBweBottleneck(inst, bottleneck_bits_per_second);
   }
-  static inline int16_t UpdateBwEstimate(instance_type* inst,
-                                         const uint8_t* encoded,
-                                         size_t packet_size,
-                                         uint16_t rtp_seq_number,
-                                         uint32_t send_ts,
-                                         uint32_t arr_ts) {
-    return WebRtcIsacfix_UpdateBwEstimate(inst, encoded, packet_size,
-                                          rtp_seq_number, send_ts, arr_ts);
-  }
   static inline int16_t SetMaxPayloadSize(instance_type* inst,
                                           int16_t max_payload_size_bytes) {
     return WebRtcIsacfix_SetMaxPayloadSize(inst, max_payload_size_bytes);

modules/audio_coding/codecs/isac/main/source/isac_float_type.h

@@ -87,15 +87,6 @@ struct IsacFloat {
                                              int bottleneck_bits_per_second) {
     WebRtcIsac_SetInitialBweBottleneck(inst, bottleneck_bits_per_second);
   }
-  static inline int16_t UpdateBwEstimate(instance_type* inst,
-                                         const uint8_t* encoded,
-                                         size_t packet_size,
-                                         uint16_t rtp_seq_number,
-                                         uint32_t send_ts,
-                                         uint32_t arr_ts) {
-    return WebRtcIsac_UpdateBwEstimate(inst, encoded, packet_size,
-                                       rtp_seq_number, send_ts, arr_ts);
-  }
   static inline int16_t SetMaxPayloadSize(instance_type* inst,
                                           int16_t max_payload_size_bytes) {
     return WebRtcIsac_SetMaxPayloadSize(inst, max_payload_size_bytes);

modules/audio_coding/neteq/neteq_impl.cc

@@ -594,19 +594,6 @@ int NetEqImpl::InsertPacketInternal(const RTPHeader& rtp_header,
     }
   }
 
-  // Update bandwidth estimate, if the packet is not comfort noise.
-  if (!packet_list.empty() &&
-      !decoder_database_->IsComfortNoise(main_payload_type)) {
-    // The list can be empty here if we got nothing but DTMF payloads.
-    AudioDecoder* decoder = decoder_database_->GetDecoder(main_payload_type);
-    RTC_DCHECK(decoder);  // Should always get a valid object, since we have
-                          // already checked that the payload types are known.
-    decoder->IncomingPacket(packet_list.front().payload.data(),
-                            packet_list.front().payload.size(),
-                            packet_list.front().sequence_number,
-                            packet_list.front().timestamp, receive_timestamp);
-  }
-
   PacketList parsed_packet_list;
   while (!packet_list.empty()) {
     Packet& packet = packet_list.front();

modules/audio_coding/neteq/neteq_impl_unittest.cc

@@ -335,15 +335,6 @@ TEST_F(NetEqImplTest, InsertPacket) {
         std::unique_ptr<MockAudioDecoder> mock_decoder(new MockAudioDecoder);
         EXPECT_CALL(*mock_decoder, Channels()).WillRepeatedly(Return(1));
         EXPECT_CALL(*mock_decoder, SampleRateHz()).WillRepeatedly(Return(8000));
-        // BWE update function called with first packet.
-        EXPECT_CALL(*mock_decoder,
-                    IncomingPacket(_, kPayloadLength, kFirstSequenceNumber,
-                                   kFirstTimestamp, kFirstReceiveTime));
-        // BWE update function called with second packet.
-        EXPECT_CALL(
-            *mock_decoder,
-            IncomingPacket(_, kPayloadLength, kFirstSequenceNumber + 1,
-                           kFirstTimestamp + 160, kFirstReceiveTime + 155));
         EXPECT_CALL(*mock_decoder, Die()).Times(1);  // Called when deleted.
 
         *dec = std::move(mock_decoder);
@@ -595,8 +586,6 @@ TEST_F(NetEqImplTest, ReorderedPacket) {
   EXPECT_CALL(mock_decoder, SampleRateHz())
       .WillRepeatedly(Return(kSampleRateHz));
   EXPECT_CALL(mock_decoder, Channels()).WillRepeatedly(Return(1));
-  EXPECT_CALL(mock_decoder, IncomingPacket(_, kPayloadLengthBytes, _, _, _))
-      .WillRepeatedly(Return(0));
   EXPECT_CALL(mock_decoder, PacketDuration(_, kPayloadLengthBytes))
       .WillRepeatedly(Return(rtc::checked_cast<int>(kPayloadLengthSamples)));
   int16_t dummy_output[kPayloadLengthSamples] = {0};
@@ -837,8 +826,6 @@ TEST_F(NetEqImplTest, CodecInternalCng) {
   EXPECT_CALL(mock_decoder, SampleRateHz())
       .WillRepeatedly(Return(kSampleRateKhz * 1000));
   EXPECT_CALL(mock_decoder, Channels()).WillRepeatedly(Return(1));
-  EXPECT_CALL(mock_decoder, IncomingPacket(_, kPayloadLengthBytes, _, _, _))
-      .WillRepeatedly(Return(0));
   EXPECT_CALL(mock_decoder, PacketDuration(_, kPayloadLengthBytes))
       .WillRepeatedly(Return(rtc::checked_cast<int>(kPayloadLengthSamples)));
   // Packed duration when asking the decoder for more CNG data (without a new
@@ -1112,8 +1099,6 @@ TEST_F(NetEqImplTest, DecodedPayloadTooShort) {
   EXPECT_CALL(mock_decoder, SampleRateHz())
       .WillRepeatedly(Return(kSampleRateHz));
   EXPECT_CALL(mock_decoder, Channels()).WillRepeatedly(Return(1));
-  EXPECT_CALL(mock_decoder, IncomingPacket(_, kPayloadLengthBytes, _, _, _))
-      .WillRepeatedly(Return(0));
   EXPECT_CALL(mock_decoder, PacketDuration(_, _))
       .WillRepeatedly(Return(rtc::checked_cast<int>(kPayloadLengthSamples)));
   int16_t dummy_output[kPayloadLengthSamples] = {0};
@@ -1181,8 +1166,6 @@ TEST_F(NetEqImplTest, DecodingError) {
   EXPECT_CALL(mock_decoder, SampleRateHz())
       .WillRepeatedly(Return(kSampleRateHz));
   EXPECT_CALL(mock_decoder, Channels()).WillRepeatedly(Return(1));
-  EXPECT_CALL(mock_decoder, IncomingPacket(_, kPayloadLengthBytes, _, _, _))
-      .WillRepeatedly(Return(0));
   EXPECT_CALL(mock_decoder, PacketDuration(_, _))
       .WillRepeatedly(Return(rtc::checked_cast<int>(kFrameLengthSamples)));
   EXPECT_CALL(mock_decoder, ErrorCode()).WillOnce(Return(kDecoderErrorCode));
@@ -1297,8 +1280,6 @@ TEST_F(NetEqImplTest, DecodingErrorDuringInternalCng) {
   EXPECT_CALL(mock_decoder, SampleRateHz())
       .WillRepeatedly(Return(kSampleRateHz));
   EXPECT_CALL(mock_decoder, Channels()).WillRepeatedly(Return(1));
-  EXPECT_CALL(mock_decoder, IncomingPacket(_, kPayloadLengthBytes, _, _, _))
-      .WillRepeatedly(Return(0));
   EXPECT_CALL(mock_decoder, PacketDuration(_, _))
       .WillRepeatedly(Return(rtc::checked_cast<int>(kFrameLengthSamples)));
   EXPECT_CALL(mock_decoder, ErrorCode()).WillOnce(Return(kDecoderErrorCode));

pc/peer_connection_end_to_end_unittest.cc

@@ -240,15 +240,6 @@ std::unique_ptr<webrtc::AudioDecoder> CreateForwardingMockDecoder(
   EXPECT_CALL(*mock_decoder, HasDecodePlc()).WillRepeatedly(Invoke([dec] {
     return dec->HasDecodePlc();
   }));
-  EXPECT_CALL(*mock_decoder, IncomingPacket(_, _, _, _, _))
-      .Times(AtLeast(1))
-      .WillRepeatedly(Invoke([dec](const uint8_t* payload, size_t payload_len,
-                                   uint16_t rtp_sequence_number,
-                                   uint32_t rtp_timestamp,
-                                   uint32_t arrival_timestamp) {
-        return dec->IncomingPacket(payload, payload_len, rtp_sequence_number,
-                                   rtp_timestamp, arrival_timestamp);
-      }));
   EXPECT_CALL(*mock_decoder, PacketDuration(_, _))
       .Times(AtLeast(1))
       .WillRepeatedly(Invoke([dec](const uint8_t* encoded, size_t encoded_len) {

test/fuzzers/BUILD.gn

@@ -296,16 +296,6 @@ webrtc_fuzzer_test("audio_decoder_isac_fuzzer") {
   ]
 }
 
-webrtc_fuzzer_test("audio_decoder_isac_incoming_packet_fuzzer") {
-  sources = [
-    "audio_decoder_isac_incoming_packet_fuzzer.cc",
-  ]
-  deps = [
-    ":audio_decoder_fuzzer",
-    "../../modules/audio_coding:isac",
-  ]
-}
-
 webrtc_fuzzer_test("audio_decoder_isacfix_fuzzer") {
   sources = [
     "audio_decoder_isacfix_fuzzer.cc",

test/fuzzers/audio_decoder_fuzzer.cc

@@ -74,31 +74,4 @@ void FuzzAudioDecoder(DecoderFunctionType decode_type,
   }
 }
 
-// This function is similar to FuzzAudioDecoder, but also reads fuzzed data into
-// RTP header values. The fuzzed data and values are sent to the decoder's
-// IncomingPacket method.
-void FuzzAudioDecoderIncomingPacket(const uint8_t* data,
-                                    size_t size,
-                                    AudioDecoder* decoder) {
-  const uint8_t* data_ptr = data;
-  size_t remaining_size = size;
-  size_t packet_len;
-  while (ParseInt<size_t, 2>(&data_ptr, &remaining_size, &packet_len)) {
-    uint16_t rtp_sequence_number;
-    if (!ParseInt(&data_ptr, &remaining_size, &rtp_sequence_number))
-      break;
-    uint32_t rtp_timestamp;
-    if (!ParseInt(&data_ptr, &remaining_size, &rtp_timestamp))
-      break;
-    uint32_t arrival_timestamp;
-    if (!ParseInt(&data_ptr, &remaining_size, &arrival_timestamp))
-      break;
-    if (remaining_size < packet_len)
-      break;
-    decoder->IncomingPacket(data_ptr, packet_len, rtp_sequence_number,
-                            rtp_timestamp, arrival_timestamp);
-    data_ptr += packet_len;
-    remaining_size -= packet_len;
-  }
-}
 }  // namespace webrtc

test/fuzzers/audio_decoder_fuzzer.h

@@ -31,9 +31,6 @@ void FuzzAudioDecoder(DecoderFunctionType decode_type,
                       size_t max_decoded_bytes,
                       int16_t* decoded);
 
-void FuzzAudioDecoderIncomingPacket(const uint8_t* data,
-                                    size_t size,
-                                    AudioDecoder* decoder);
 }  // namespace webrtc
 
 #endif  // TEST_FUZZERS_AUDIO_DECODER_FUZZER_H_

test/fuzzers/audio_decoder_isac_incoming_packet_fuzzer.cc

@@ -1,24 +0,0 @@
-/*
- *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h"
-#include "test/fuzzers/audio_decoder_fuzzer.h"
-
-namespace webrtc {
-void FuzzOneInput(const uint8_t* data, size_t size) {
-  if (size > 20000) {
-    return;
-  }
-  AudioDecoderIsacFloatImpl::Config c;
-  c.sample_rate_hz = 16000;
-  AudioDecoderIsacFloatImpl dec(c);
-  FuzzAudioDecoderIncomingPacket(data, size, &dec);
-}
-}  // namespace webrtc

test/mock_audio_decoder.h

@@ -26,8 +26,6 @@ class MockAudioDecoder : public AudioDecoder {
   MOCK_CONST_METHOD0(HasDecodePlc, bool());
   MOCK_METHOD2(DecodePlc, size_t(size_t, int16_t*));
   MOCK_METHOD0(Reset, void());
-  MOCK_METHOD5(IncomingPacket,
-               int(const uint8_t*, size_t, uint16_t, uint32_t, uint32_t));
   MOCK_METHOD0(ErrorCode, int());
   MOCK_CONST_METHOD2(PacketDuration, int(const uint8_t*, size_t));
   MOCK_CONST_METHOD0(Channels, size_t());