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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 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 | 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 | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
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22 #include "webrtc/modules/audio_coding/acm2/acm_resampler.h" | 22 #include "webrtc/modules/audio_coding/acm2/acm_resampler.h" |
23 #include "webrtc/modules/audio_coding/acm2/call_statistics.h" | 23 #include "webrtc/modules/audio_coding/acm2/call_statistics.h" |
24 #include "webrtc/system_wrappers/include/logging.h" | 24 #include "webrtc/system_wrappers/include/logging.h" |
25 #include "webrtc/system_wrappers/include/metrics.h" | 25 #include "webrtc/system_wrappers/include/metrics.h" |
26 #include "webrtc/system_wrappers/include/rw_lock_wrapper.h" | 26 #include "webrtc/system_wrappers/include/rw_lock_wrapper.h" |
27 #include "webrtc/system_wrappers/include/trace.h" | 27 #include "webrtc/system_wrappers/include/trace.h" |
28 #include "webrtc/typedefs.h" | 28 #include "webrtc/typedefs.h" |
29 | 29 |
30 namespace webrtc { | 30 namespace webrtc { |
31 | 31 |
32 namespace { | |
33 | |
34 struct CodecNameIDPair { | |
hlundin-webrtc
2016/05/12 12:23:39
The style guide says you should write CodecNameIdP
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35 const char* name; | |
36 size_t id; | |
37 }; | |
kwiberg-webrtc
2016/05/12 13:01:51
Please document. Alternatively, move this definiti
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38 | |
39 | |
40 // Ensures that every element of passed array har .id field less than | |
hlundin-webrtc
2016/05/12 12:23:39
har?
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41 // kMaxCodecNames and that count starts from '1' with 1-step | |
42 // increments | |
43 template<int N> | |
44 constexpr bool idSLessThanMaxCodecNames(const CodecNameIDPair (&array) [N], | |
hlundin-webrtc
2016/05/12 12:23:39
Name formatting...
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45 size_t index) { | |
46 return (index >= N) || | |
47 (array[index].id < acm2::AudioCodingModuleImpl::kMaxAudioCodecNames && | |
48 array[index].id == index + 1 && | |
49 idSLessThanMaxCodecNames(array, index + 1)); | |
50 } | |
kwiberg-webrtc
2016/05/12 13:01:51
This is a lot of hard-to-read code, and it doesn't
aleloi
2016/05/12 13:25:46
I claim it does check that ID:s are different and
kwiberg-webrtc
2016/05/12 23:30:21
Oh, I see. I didn't read it carefully, just enough
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51 | |
52 // Translates the name of a codec returned by GetCodecName() into | |
kwiberg-webrtc
2016/05/12 13:01:51
That's no longer where the names come from.
aleloi
2016/05/12 13:25:47
Acknowledged.
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53 // one of the codec IDs in the table 'histogram_id' below. | |
54 size_t CodecNameToHistogramCodecType(const char* codec_name) { | |
55 constexpr CodecNameIDPair histogram_id[] = { | |
kwiberg-webrtc
2016/05/12 13:01:51
static constexpr. Otherwise the compiler might put
aleloi
2016/05/12 13:25:46
Done.
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56 {"Opus", 1}, | |
kwiberg-webrtc
2016/05/12 13:01:51
Add a comment before the first entry explaining wh
aleloi
2016/05/12 13:25:47
Done.
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57 {"iSAC", 2}, | |
58 {"g711A", 3}, | |
59 {"g711U", 4}, | |
60 {"g722", 5}, | |
61 {"iLBC", 6} | |
kwiberg-webrtc
2016/05/12 13:01:51
Use a trailing comma here too. Otherwise, the diff
aleloi
2016/05/12 13:25:47
I didn't know that was allowed in C++! Always lear
kwiberg-webrtc
2016/05/12 23:30:21
I've never actually checked what the standard says
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62 }; | |
63 | |
64 static_assert(idSLessThanMaxCodecNames(histogram_id, 0), | |
65 "Cannot guarantee that codec ID does not exceed boundary."); | |
66 | |
67 for (const auto & codec : histogram_id) { | |
kwiberg-webrtc
2016/05/12 13:01:51
No space between auto and &. (Did you run git cl f
aleloi
2016/05/12 13:25:47
Nope, forgot!
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68 if (codec_name != nullptr && strcmp(codec.name, codec_name) == 0) | |
ossu
2016/05/12 12:50:14
You should check codec_name before entering the lo
kwiberg-webrtc
2016/05/12 13:01:51
Pointers are true iff they're not null, so just
aleloi
2016/05/12 13:25:46
Done.
aleloi
2016/05/12 13:25:47
I followed ossu's recommendation and moved the che
kwiberg-webrtc
2016/05/12 23:30:21
OK. You're not alone in inexplicably preferring th
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69 return codec.id; | |
70 } | |
71 return 0; | |
72 } | |
73 | |
74 // Adds a codec usage sample to the histogram. | |
75 void UpdateCodecTypeHistogram(size_t codec_type) { | |
76 RTC_HISTOGRAM_ENUMERATION("WebRTC.Audio.Encoder.CodecType", | |
77 codec_type, | |
78 acm2::AudioCodingModuleImpl::kMaxAudioCodecNames); | |
79 } | |
80 } // namespace | |
81 | |
82 | |
32 namespace acm2 { | 83 namespace acm2 { |
33 | 84 |
34 struct EncoderFactory { | 85 struct EncoderFactory { |
35 AudioEncoder* external_speech_encoder = nullptr; | 86 AudioEncoder* external_speech_encoder = nullptr; |
36 CodecManager codec_manager; | 87 CodecManager codec_manager; |
37 RentACodec rent_a_codec; | 88 RentACodec rent_a_codec; |
38 }; | 89 }; |
39 | 90 |
40 namespace { | 91 namespace { |
41 | 92 |
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178 expected_in_ts_(0xD87F3F9F), | 229 expected_in_ts_(0xD87F3F9F), |
179 receiver_(config), | 230 receiver_(config), |
180 bitrate_logger_("WebRTC.Audio.TargetBitrateInKbps"), | 231 bitrate_logger_("WebRTC.Audio.TargetBitrateInKbps"), |
181 encoder_factory_(new EncoderFactory), | 232 encoder_factory_(new EncoderFactory), |
182 encoder_stack_(nullptr), | 233 encoder_stack_(nullptr), |
183 previous_pltype_(255), | 234 previous_pltype_(255), |
184 receiver_initialized_(false), | 235 receiver_initialized_(false), |
185 first_10ms_data_(false), | 236 first_10ms_data_(false), |
186 first_frame_(true), | 237 first_frame_(true), |
187 packetization_callback_(NULL), | 238 packetization_callback_(NULL), |
188 vad_callback_(NULL) { | 239 vad_callback_(NULL), |
240 codec_histogram_bins_log(), | |
kwiberg-webrtc
2016/05/12 13:01:51
Don't you need to fill this with zeros?
aleloi
2016/05/12 13:25:47
It gets filled with zeroes with codec_histogram_bi
kwiberg-webrtc
2016/05/12 23:30:21
Right. Thanks!
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241 number_of_consecutive_empty_packets(0) { | |
189 if (InitializeReceiverSafe() < 0) { | 242 if (InitializeReceiverSafe() < 0) { |
190 WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, | 243 WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, |
191 "Cannot initialize receiver"); | 244 "Cannot initialize receiver"); |
192 } | 245 } |
193 WEBRTC_TRACE(webrtc::kTraceMemory, webrtc::kTraceAudioCoding, id_, "Created"); | 246 WEBRTC_TRACE(webrtc::kTraceMemory, webrtc::kTraceAudioCoding, id_, "Created"); |
194 } | 247 } |
195 | 248 |
196 AudioCodingModuleImpl::~AudioCodingModuleImpl() = default; | 249 AudioCodingModuleImpl::~AudioCodingModuleImpl() = default; |
197 | 250 |
198 int32_t AudioCodingModuleImpl::Encode(const InputData& input_data) { | 251 int32_t AudioCodingModuleImpl::Encode(const InputData& input_data) { |
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224 input_data.length_per_channel), | 277 input_data.length_per_channel), |
225 &encode_buffer_); | 278 &encode_buffer_); |
226 | 279 |
227 bitrate_logger_.MaybeLog(encoder_stack_->GetTargetBitrate() / 1000); | 280 bitrate_logger_.MaybeLog(encoder_stack_->GetTargetBitrate() / 1000); |
228 if (encode_buffer_.size() == 0 && !encoded_info.send_even_if_empty) { | 281 if (encode_buffer_.size() == 0 && !encoded_info.send_even_if_empty) { |
229 // Not enough data. | 282 // Not enough data. |
230 return 0; | 283 return 0; |
231 } | 284 } |
232 previous_pltype = previous_pltype_; // Read it while we have the critsect. | 285 previous_pltype = previous_pltype_; // Read it while we have the critsect. |
233 | 286 |
287 // Log codec type to histogram once every 500 packets. | |
288 if (encoded_info.encoded_bytes == 0) { | |
289 number_of_consecutive_empty_packets += 1; | |
ossu
2016/05/12 12:50:14
I'd prefer ++number_of_consecutive_empty_packets.
aleloi
2016/05/12 13:25:47
Done.
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290 } | |
291 else { | |
ossu
2016/05/12 12:50:14
else on the same line as the closing curly seems t
aleloi
2016/05/12 13:25:47
Done.
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292 size_t codec_type = | |
ossu
2016/05/12 12:50:14
So the codec id is looked up based on its name onc
kwiberg-webrtc
2016/05/12 13:01:51
I'm guessing we'd need quite a lot of entries befo
aleloi
2016/05/12 13:25:47
We only do trivial operations in the lookup. A map
kwiberg-webrtc
2016/05/12 23:30:21
Yes. std::map will also do a lot of cache-unfriend
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293 CodecNameToHistogramCodecType(encoded_info.encoder_name); | |
294 codec_histogram_bins_log[codec_type] += 1; | |
295 number_of_consecutive_empty_packets = 0; | |
kwiberg-webrtc
2016/05/12 13:01:51
Here you zero number_of_consecutive_empty_packets
aleloi
2016/05/12 13:25:47
Acknowledged.
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296 if (codec_histogram_bins_log[codec_type] >= 500) { | |
297 codec_histogram_bins_log[codec_type] -= 500; | |
298 UpdateCodecTypeHistogram(codec_type); | |
299 } | |
300 } | |
301 | |
234 RTPFragmentationHeader my_fragmentation; | 302 RTPFragmentationHeader my_fragmentation; |
235 ConvertEncodedInfoToFragmentationHeader(encoded_info, &my_fragmentation); | 303 ConvertEncodedInfoToFragmentationHeader(encoded_info, &my_fragmentation); |
236 FrameType frame_type; | 304 FrameType frame_type; |
237 if (encode_buffer_.size() == 0 && encoded_info.send_even_if_empty) { | 305 if (encode_buffer_.size() == 0 && encoded_info.send_even_if_empty) { |
238 frame_type = kEmptyFrame; | 306 frame_type = kEmptyFrame; |
239 encoded_info.payload_type = previous_pltype; | 307 encoded_info.payload_type = previous_pltype; |
240 } else { | 308 } else { |
241 RTC_DCHECK_GT(encode_buffer_.size(), 0u); | 309 RTC_DCHECK_GT(encode_buffer_.size(), 0u); |
242 frame_type = encoded_info.speech ? kAudioFrameSpeech : kAudioFrameCN; | 310 frame_type = encoded_info.speech ? kAudioFrameSpeech : kAudioFrameCN; |
243 } | 311 } |
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938 return receiver_.LeastRequiredDelayMs(); | 1006 return receiver_.LeastRequiredDelayMs(); |
939 } | 1007 } |
940 | 1008 |
941 void AudioCodingModuleImpl::GetDecodingCallStatistics( | 1009 void AudioCodingModuleImpl::GetDecodingCallStatistics( |
942 AudioDecodingCallStats* call_stats) const { | 1010 AudioDecodingCallStats* call_stats) const { |
943 receiver_.GetDecodingCallStatistics(call_stats); | 1011 receiver_.GetDecodingCallStatistics(call_stats); |
944 } | 1012 } |
945 | 1013 |
946 } // namespace acm2 | 1014 } // namespace acm2 |
947 } // namespace webrtc | 1015 } // namespace webrtc |
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