diff --git a/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc b/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc index c366295bfb..17d49a9b4d 100644 --- a/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc +++ b/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc @@ -578,7 +578,9 @@ ACMGenericCodec* ACMCodecDB::CreateCodecInstance(const CodecInst& codec_inst, // All we have support for right now. if (!STR_CASE_CMP(codec_inst.plname, "ISAC")) { #if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) - return new ACMISAC(kISAC, enable_red); + return new ACMGenericCodecWrapper(codec_inst, cng_pt_nb, cng_pt_wb, + cng_pt_swb, cng_pt_fb, enable_red, + red_payload_type); #endif } else if (!STR_CASE_CMP(codec_inst.plname, "PCMU") || !STR_CASE_CMP(codec_inst.plname, "PCMA")) { diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc index fc1191a45f..1837be6656 100644 --- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc +++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc @@ -270,7 +270,6 @@ int32_t AudioCodingModuleImpl::Process() { FrameType frame_type = kAudioFrameSpeech; uint8_t current_payload_type = 0; bool has_data_to_send = false; - bool red_active = false; RTPFragmentationHeader my_fragmentation; // Keep the scope of the ACM critical section limited. @@ -295,43 +294,39 @@ int32_t AudioCodingModuleImpl::Process() { } else { switch (encoding_type) { case kNoEncoding: { - current_payload_type = previous_pltype_; + FATAL() << "This case is no longer valid."; frame_type = kFrameEmpty; length_bytes = 0; break; } case kActiveNormalEncoded: case kPassiveNormalEncoded: { - current_payload_type = static_cast(send_codec_inst_.pltype); frame_type = kAudioFrameSpeech; break; } case kPassiveDTXNB: { - current_payload_type = cng_nb_pltype_; frame_type = kAudioFrameCN; is_first_red_ = true; break; } case kPassiveDTXWB: { - current_payload_type = cng_wb_pltype_; frame_type = kAudioFrameCN; is_first_red_ = true; break; } case kPassiveDTXSWB: { - current_payload_type = cng_swb_pltype_; frame_type = kAudioFrameCN; is_first_red_ = true; break; } case kPassiveDTXFB: { - current_payload_type = cng_fb_pltype_; frame_type = kAudioFrameCN; is_first_red_ = true; break; } } has_data_to_send = true; + current_payload_type = encoded_info.payload_type; previous_pltype_ = current_payload_type; ConvertEncodedInfoToFragmentationHeader(encoded_info, &my_fragmentation); @@ -348,8 +343,9 @@ int32_t AudioCodingModuleImpl::Process() { // have been switched to the new AudioEncoder interface. if ((codecs_[current_send_codec_idx_]->ExternalRedNeeded()) && ((encoding_type == kActiveNormalEncoded) || - (encoding_type == kPassiveNormalEncoded))) { + (encoding_type == kPassiveNormalEncoded))) { DCHECK(encoded_info.redundant.empty()); + FATAL() << "Don't go here!"; // RED is enabled within this scope. // // Note that, a special solution exists for iSAC since it is the only @@ -389,7 +385,6 @@ int32_t AudioCodingModuleImpl::Process() { // // Hence, even if every second packet is dropped, perfect // reconstruction is possible. - red_active = true; has_data_to_send = false; // Skip the following part for the first packet in a RED session. @@ -457,7 +452,7 @@ int32_t AudioCodingModuleImpl::Process() { CriticalSectionScoped lock(callback_crit_sect_); if (packetization_callback_ != NULL) { - if (red_active || my_fragmentation.fragmentationVectorSize > 0) { + if (my_fragmentation.fragmentationVectorSize > 0) { // Callback with payload data, including redundant data (RED). packetization_callback_->SendData(frame_type, current_payload_type, rtp_timestamp, stream, length_bytes, diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc index 4ee53391e7..5185c1209a 100644 --- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc +++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc @@ -809,15 +809,14 @@ TEST_F(AcmSenderBitExactness, DISABLED_ON_ANDROID(IsacSwb30ms)) { ASSERT_NO_FATAL_FAILURE( SetUpTest(acm2::ACMCodecDB::kISACSWB, 1, 104, 960, 960)); Run(AcmReceiverBitExactness::PlatformChecksum( - "98d960600eb4ddb3fcbe11f5057ddfd7", + "2b3c387d06f00b7b7aad4c9be56fb83d", "", - "2f6dfe142f735f1d96f6bd86d2526f42"), + "5683b58da0fbf2063c7adc2e6bfb3fb8"), AcmReceiverBitExactness::PlatformChecksum( - "cc9d2d86a71d6f99f97680a5c27e2762", + "bcc2041e7744c7ebd9f701866856849c", "", - "7b214fc3a5e33d68bf30e77969371f31"), - 33, - test::AcmReceiveTest::kMonoOutput); + "ce86106a93419aefb063097108ec94ab"), + 33, test::AcmReceiveTest::kMonoOutput); } TEST_F(AcmSenderBitExactness, Pcm16_8000khz_10ms) { diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc index 8cff055da1..56f4096431 100644 --- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc +++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc @@ -824,15 +824,14 @@ TEST_F(AcmSenderBitExactnessOldApi, MAYBE_IsacWb60ms) { TEST_F(AcmSenderBitExactnessOldApi, DISABLED_ON_ANDROID(IsacSwb30ms)) { ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 32000, 1, 104, 960, 960)); Run(AcmReceiverBitExactnessOldApi::PlatformChecksum( - "98d960600eb4ddb3fcbe11f5057ddfd7", + "2b3c387d06f00b7b7aad4c9be56fb83d", "", - "2f6dfe142f735f1d96f6bd86d2526f42"), + "5683b58da0fbf2063c7adc2e6bfb3fb8"), AcmReceiverBitExactnessOldApi::PlatformChecksum( - "cc9d2d86a71d6f99f97680a5c27e2762", + "bcc2041e7744c7ebd9f701866856849c", "", - "7b214fc3a5e33d68bf30e77969371f31"), - 33, - test::AcmReceiveTestOldApi::kMonoOutput); + "ce86106a93419aefb063097108ec94ab"), + 33, test::AcmReceiveTestOldApi::kMonoOutput); } TEST_F(AcmSenderBitExactnessOldApi, Pcm16_8000khz_10ms) {