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 17d49a9b4d..c366295bfb 100644 --- a/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc +++ b/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc @@ -578,9 +578,7 @@ 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 ACMGenericCodecWrapper(codec_inst, cng_pt_nb, cng_pt_wb, - cng_pt_swb, cng_pt_fb, enable_red, - red_payload_type); + return new ACMISAC(kISAC, enable_red); #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 8d0f3181d1..fc1191a45f 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 @@ -270,6 +270,7 @@ 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. @@ -301,32 +302,36 @@ int32_t AudioCodingModuleImpl::Process() { } 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); @@ -343,9 +348,8 @@ 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 @@ -385,6 +389,7 @@ 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. @@ -452,7 +457,7 @@ int32_t AudioCodingModuleImpl::Process() { CriticalSectionScoped lock(callback_crit_sect_); if (packetization_callback_ != NULL) { - if (my_fragmentation.fragmentationVectorSize > 0) { + if (red_active || 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 5185c1209a..4ee53391e7 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 @@ -809,14 +809,15 @@ TEST_F(AcmSenderBitExactness, DISABLED_ON_ANDROID(IsacSwb30ms)) { ASSERT_NO_FATAL_FAILURE( SetUpTest(acm2::ACMCodecDB::kISACSWB, 1, 104, 960, 960)); Run(AcmReceiverBitExactness::PlatformChecksum( - "2b3c387d06f00b7b7aad4c9be56fb83d", + "98d960600eb4ddb3fcbe11f5057ddfd7", "", - "5683b58da0fbf2063c7adc2e6bfb3fb8"), + "2f6dfe142f735f1d96f6bd86d2526f42"), AcmReceiverBitExactness::PlatformChecksum( - "bcc2041e7744c7ebd9f701866856849c", + "cc9d2d86a71d6f99f97680a5c27e2762", "", - "ce86106a93419aefb063097108ec94ab"), - 33, test::AcmReceiveTest::kMonoOutput); + "7b214fc3a5e33d68bf30e77969371f31"), + 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 329edd3362..1501d037c2 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 @@ -932,14 +932,15 @@ 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( - "2b3c387d06f00b7b7aad4c9be56fb83d", + "98d960600eb4ddb3fcbe11f5057ddfd7", "", - "5683b58da0fbf2063c7adc2e6bfb3fb8"), + "2f6dfe142f735f1d96f6bd86d2526f42"), AcmReceiverBitExactnessOldApi::PlatformChecksum( - "bcc2041e7744c7ebd9f701866856849c", + "cc9d2d86a71d6f99f97680a5c27e2762", "", - "ce86106a93419aefb063097108ec94ab"), - 33, test::AcmReceiveTestOldApi::kMonoOutput); + "7b214fc3a5e33d68bf30e77969371f31"), + 33, + test::AcmReceiveTestOldApi::kMonoOutput); } TEST_F(AcmSenderBitExactnessOldApi, Pcm16_8000khz_10ms) {