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| 1 /* | |
| 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | |
| 3 * | |
| 4 * Use of this source code is governed by a BSD-style license | |
| 5 * that can be found in the LICENSE file in the root of the source | |
| 6 * tree. An additional intellectual property rights grant can be found | |
| 7 * in the file PATENTS. All contributing project authors may | |
| 8 * be found in the AUTHORS file in the root of the source tree. | |
| 9 */ | |
| 10 | |
| 11 #include "webrtc/common_types.h" | |
| 12 #include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h" | |
| 13 #include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h" | |
| 14 #include "webrtc/modules/include/module_common_types.h" | |
| 15 #include "webrtc/modules/utility/source/coder.h" | |
| 16 | |
| 17 namespace webrtc { | |
| 18 namespace { | |
| 19 AudioCodingModule::Config GetAcmConfig(uint32_t id) { | |
| 20 AudioCodingModule::Config config; | |
| 21 // This class does not handle muted output. | |
| 22 config.neteq_config.enable_muted_state = false; | |
| 23 config.id = id; | |
| 24 config.decoder_factory = CreateBuiltinAudioDecoderFactory(); | |
| 25 return config; | |
| 26 } | |
| 27 } // namespace | |
| 28 | |
| 29 AudioCoder::AudioCoder(uint32_t instance_id) | |
| 30 : acm_(AudioCodingModule::Create(GetAcmConfig(instance_id))), | |
| 31 receive_codec_(), | |
| 32 encode_timestamp_(0), | |
| 33 encoded_data_(nullptr), | |
| 34 encoded_length_in_bytes_(0), | |
| 35 decode_timestamp_(0) { | |
| 36 acm_->InitializeReceiver(); | |
| 37 acm_->RegisterTransportCallback(this); | |
| 38 } | |
| 39 | |
| 40 AudioCoder::~AudioCoder() {} | |
| 41 | |
| 42 int32_t AudioCoder::SetEncodeCodec(const CodecInst& codec_inst) { | |
| 43 const bool success = codec_manager_.RegisterEncoder(codec_inst) && | |
| 44 codec_manager_.MakeEncoder(&rent_a_codec_, acm_.get()); | |
| 45 return success ? 0 : -1; | |
| 46 } | |
| 47 | |
| 48 int32_t AudioCoder::SetDecodeCodec(const CodecInst& codec_inst) { | |
| 49 if (!acm_->RegisterReceiveCodec(codec_inst.pltype, | |
| 50 CodecInstToSdp(codec_inst))) { | |
| 51 return -1; | |
| 52 } | |
| 53 memcpy(&receive_codec_, &codec_inst, sizeof(CodecInst)); | |
| 54 return 0; | |
| 55 } | |
| 56 | |
| 57 int32_t AudioCoder::Decode(AudioFrame* decoded_audio, | |
| 58 uint32_t samp_freq_hz, | |
| 59 const int8_t* incoming_payload, | |
| 60 size_t payload_length) { | |
| 61 if (payload_length > 0) { | |
| 62 const uint8_t payload_type = receive_codec_.pltype; | |
| 63 decode_timestamp_ += receive_codec_.pacsize; | |
| 64 if (acm_->IncomingPayload((const uint8_t*)incoming_payload, payload_length, | |
| 65 payload_type, decode_timestamp_) == -1) { | |
| 66 return -1; | |
| 67 } | |
| 68 } | |
| 69 bool muted; | |
| 70 int32_t ret = | |
| 71 acm_->PlayoutData10Ms((uint16_t)samp_freq_hz, decoded_audio, &muted); | |
| 72 RTC_DCHECK(!muted); | |
| 73 return ret; | |
| 74 } | |
| 75 | |
| 76 int32_t AudioCoder::PlayoutData(AudioFrame* decoded_audio, | |
| 77 uint16_t samp_freq_hz) { | |
| 78 bool muted; | |
| 79 int32_t ret = acm_->PlayoutData10Ms(samp_freq_hz, decoded_audio, &muted); | |
| 80 RTC_DCHECK(!muted); | |
| 81 return ret; | |
| 82 } | |
| 83 | |
| 84 int32_t AudioCoder::Encode(const AudioFrame& audio, | |
| 85 int8_t* encoded_data, | |
| 86 size_t* encoded_length_in_bytes) { | |
| 87 // Fake a timestamp in case audio doesn't contain a correct timestamp. | |
| 88 // Make a local copy of the audio frame since audio is const | |
| 89 AudioFrame audio_frame; | |
| 90 audio_frame.CopyFrom(audio); | |
| 91 audio_frame.timestamp_ = encode_timestamp_; | |
| 92 encode_timestamp_ += static_cast<uint32_t>(audio_frame.samples_per_channel_); | |
| 93 | |
| 94 // For any codec with a frame size that is longer than 10 ms the encoded | |
| 95 // length in bytes should be zero until a a full frame has been encoded. | |
| 96 encoded_length_in_bytes_ = 0; | |
| 97 if (acm_->Add10MsData((AudioFrame&)audio_frame) == -1) { | |
| 98 return -1; | |
| 99 } | |
| 100 encoded_data_ = encoded_data; | |
| 101 *encoded_length_in_bytes = encoded_length_in_bytes_; | |
| 102 return 0; | |
| 103 } | |
| 104 | |
| 105 int32_t AudioCoder::SendData(FrameType /* frame_type */, | |
| 106 uint8_t /* payload_type */, | |
| 107 uint32_t /* time_stamp */, | |
| 108 const uint8_t* payload_data, | |
| 109 size_t payload_size, | |
| 110 const RTPFragmentationHeader* /* fragmentation*/) { | |
| 111 memcpy(encoded_data_, payload_data, sizeof(uint8_t) * payload_size); | |
| 112 encoded_length_in_bytes_ = payload_size; | |
| 113 return 0; | |
| 114 } | |
| 115 | |
| 116 } // namespace webrtc | |
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Chromium Code Reviews
Issue 2622493002:
Move FilePlayer and FileRecorder to Voice Engine (Closed)
