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Issue 3013033002: Remove VoEFile (Closed)
Patch Set: rebase Created 3 years, 3 months ago
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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 "voice_engine/coder.h"
12
13 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
14 #include "common_types.h" // NOLINT(build/include)
15 #include "modules/audio_coding/codecs/audio_format_conversion.h"
16 #include "modules/include/module_common_types.h"
17
18 namespace webrtc {
19 namespace {
20 AudioCodingModule::Config GetAcmConfig(uint32_t id) {
21 AudioCodingModule::Config config;
22 // This class does not handle muted output.
23 config.neteq_config.enable_muted_state = false;
24 config.id = id;
25 config.decoder_factory = CreateBuiltinAudioDecoderFactory();
26 return config;
27 }
28 } // namespace
29
30 AudioCoder::AudioCoder(uint32_t instance_id)
31 : acm_(AudioCodingModule::Create(GetAcmConfig(instance_id))),
32 receive_codec_(),
33 encode_timestamp_(0),
34 encoded_data_(nullptr),
35 encoded_length_in_bytes_(0),
36 decode_timestamp_(0) {
37 acm_->InitializeReceiver();
38 acm_->RegisterTransportCallback(this);
39 }
40
41 AudioCoder::~AudioCoder() {}
42
43 int32_t AudioCoder::SetEncodeCodec(const CodecInst& codec_inst) {
44 const bool success = codec_manager_.RegisterEncoder(codec_inst) &&
45 codec_manager_.MakeEncoder(&rent_a_codec_, acm_.get());
46 return success ? 0 : -1;
47 }
48
49 int32_t AudioCoder::SetDecodeCodec(const CodecInst& codec_inst) {
50 if (!acm_->RegisterReceiveCodec(codec_inst.pltype,
51 CodecInstToSdp(codec_inst))) {
52 return -1;
53 }
54 memcpy(&receive_codec_, &codec_inst, sizeof(CodecInst));
55 return 0;
56 }
57
58 int32_t AudioCoder::Decode(AudioFrame* decoded_audio,
59 uint32_t samp_freq_hz,
60 const int8_t* incoming_payload,
61 size_t payload_length) {
62 if (payload_length > 0) {
63 const uint8_t payload_type = receive_codec_.pltype;
64 decode_timestamp_ += receive_codec_.pacsize;
65 if (acm_->IncomingPayload((const uint8_t*)incoming_payload, payload_length,
66 payload_type, decode_timestamp_) == -1) {
67 return -1;
68 }
69 }
70 bool muted;
71 int32_t ret =
72 acm_->PlayoutData10Ms((uint16_t)samp_freq_hz, decoded_audio, &muted);
73 RTC_DCHECK(!muted);
74 return ret;
75 }
76
77 int32_t AudioCoder::PlayoutData(AudioFrame* decoded_audio,
78 uint16_t samp_freq_hz) {
79 bool muted;
80 int32_t ret = acm_->PlayoutData10Ms(samp_freq_hz, decoded_audio, &muted);
81 RTC_DCHECK(!muted);
82 return ret;
83 }
84
85 int32_t AudioCoder::Encode(const AudioFrame& audio,
86 int8_t* encoded_data,
87 size_t* encoded_length_in_bytes) {
88 // Fake a timestamp in case audio doesn't contain a correct timestamp.
89 // Make a local copy of the audio frame since audio is const
90 AudioFrame audio_frame;
91 audio_frame.CopyFrom(audio);
92 audio_frame.timestamp_ = encode_timestamp_;
93 encode_timestamp_ += static_cast<uint32_t>(audio_frame.samples_per_channel_);
94
95 // For any codec with a frame size that is longer than 10 ms the encoded
96 // length in bytes should be zero until a a full frame has been encoded.
97 encoded_length_in_bytes_ = 0;
98 encoded_data_ = encoded_data;
99 if (acm_->Add10MsData((AudioFrame&)audio_frame) == -1) {
100 return -1;
101 }
102
103 *encoded_length_in_bytes = encoded_length_in_bytes_;
104 return 0;
105 }
106
107 int32_t AudioCoder::SendData(FrameType /* frame_type */,
108 uint8_t /* payload_type */,
109 uint32_t /* time_stamp */,
110 const uint8_t* payload_data,
111 size_t payload_size,
112 const RTPFragmentationHeader* /* fragmentation*/) {
113 memcpy(encoded_data_, payload_data, sizeof(uint8_t) * payload_size);
114 encoded_length_in_bytes_ = payload_size;
115 return 0;
116 }
117
118 } // namespace webrtc
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