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Issue 2245413002: Revert of Move FilePlayer and FileRecorder to Voice Engine (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 4 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 "webrtc/voice_engine/coder.h"
12
13 #include "webrtc/common_types.h"
14 #include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
15 #include "webrtc/modules/include/module_common_types.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, [&] {
50 return rent_a_codec_.RentIsacDecoder(codec_inst.plfreq);
51 }) == -1) {
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 if (acm_->Add10MsData((AudioFrame&)audio_frame) == -1) {
99 return -1;
100 }
101 encoded_data_ = encoded_data;
102 encoded_length_in_bytes = encoded_length_in_bytes_;
103 return 0;
104 }
105
106 int32_t AudioCoder::SendData(FrameType /* frame_type */,
107 uint8_t /* payload_type */,
108 uint32_t /* time_stamp */,
109 const uint8_t* payload_data,
110 size_t payload_size,
111 const RTPFragmentationHeader* /* fragmentation*/) {
112 memcpy(encoded_data_, payload_data, sizeof(uint8_t) * payload_size);
113 encoded_length_in_bytes_ = payload_size;
114 return 0;
115 }
116
117 } // namespace webrtc
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