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Issue 2037623002: Move FilePlayer and FileRecorder to Voice Engine (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@remove0
Patch Set: GN build fix Created 4 years, 6 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/common_types.h"
12 #include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
13 #include "webrtc/modules/include/module_common_types.h"
14 #include "webrtc/modules/utility/source/coder.h"
15
16 namespace webrtc {
17 namespace {
18 AudioCodingModule::Config GetAcmConfig(uint32_t id) {
19 AudioCodingModule::Config config;
20 // This class does not handle muted output.
21 config.neteq_config.enable_muted_state = false;
22 config.id = id;
23 config.decoder_factory = CreateBuiltinAudioDecoderFactory();
24 return config;
25 }
26 } // namespace
27
28 AudioCoder::AudioCoder(uint32_t instance_id)
29 : acm_(AudioCodingModule::Create(GetAcmConfig(instance_id))),
30 receive_codec_(),
31 encode_timestamp_(0),
32 encoded_data_(nullptr),
33 encoded_length_in_bytes_(0),
34 decode_timestamp_(0) {
35 acm_->InitializeReceiver();
36 acm_->RegisterTransportCallback(this);
37 }
38
39 AudioCoder::~AudioCoder() {}
40
41 int32_t AudioCoder::SetEncodeCodec(const CodecInst& codec_inst) {
42 const bool success = codec_manager_.RegisterEncoder(codec_inst) &&
43 codec_manager_.MakeEncoder(&rent_a_codec_, acm_.get());
44 return success ? 0 : -1;
45 }
46
47 int32_t AudioCoder::SetDecodeCodec(const CodecInst& codec_inst) {
48 if (acm_->RegisterReceiveCodec(codec_inst, [&] {
49 return rent_a_codec_.RentIsacDecoder(codec_inst.plfreq);
50 }) == -1) {
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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