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Side by Side Diff: webrtc/audio/audio_send_stream.cc

Issue 2452643002: Revert of Clean up logging in AudioSendStream::SetupSendCodec(). (Closed)
Patch Set: Created 4 years, 1 month ago
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1 /* 1 /*
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2015 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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28 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h" 28 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
29 #include "webrtc/voice_engine/include/voe_volume_control.h" 29 #include "webrtc/voice_engine/include/voe_volume_control.h"
30 #include "webrtc/voice_engine/voice_engine_impl.h" 30 #include "webrtc/voice_engine/voice_engine_impl.h"
31 31
32 namespace webrtc { 32 namespace webrtc {
33 33
34 namespace { 34 namespace {
35 35
36 constexpr char kOpusCodecName[] = "opus"; 36 constexpr char kOpusCodecName[] = "opus";
37 37
38 // TODO(minyue): Remove |LOG_RTCERR2|.
39 #define LOG_RTCERR2(func, a1, a2, err) \
40 LOG(LS_WARNING) << "" << #func << "(" << a1 << ", " << a2 \
41 << ") failed, err=" << err
42
43 // TODO(minyue): Remove |LOG_RTCERR3|.
44 #define LOG_RTCERR3(func, a1, a2, a3, err) \
45 LOG(LS_WARNING) << "" << #func << "(" << a1 << ", " << a2 << ", " << a3 \
46 << ") failed, err=" << err
47
48 std::string ToString(const webrtc::CodecInst& codec) {
49 std::stringstream ss;
50 ss << codec.plname << "/" << codec.plfreq << "/" << codec.channels << " ("
51 << codec.pltype << ")";
52 return ss.str();
53 }
54
38 bool IsCodec(const webrtc::CodecInst& codec, const char* ref_name) { 55 bool IsCodec(const webrtc::CodecInst& codec, const char* ref_name) {
39 return (_stricmp(codec.plname, ref_name) == 0); 56 return (_stricmp(codec.plname, ref_name) == 0);
40 } 57 }
58
41 } // namespace 59 } // namespace
42 60
61 std::string AudioSendStream::Config::Rtp::ToString() const {
62 std::stringstream ss;
63 ss << "{ssrc: " << ssrc;
64 ss << ", extensions: [";
65 for (size_t i = 0; i < extensions.size(); ++i) {
66 ss << extensions[i].ToString();
67 if (i != extensions.size() - 1) {
68 ss << ", ";
69 }
70 }
71 ss << ']';
72 ss << ", nack: " << nack.ToString();
73 ss << ", c_name: " << c_name;
74 ss << '}';
75 return ss.str();
76 }
77
78 std::string AudioSendStream::Config::ToString() const {
79 std::stringstream ss;
80 ss << "{rtp: " << rtp.ToString();
81 ss << ", voe_channel_id: " << voe_channel_id;
82 // TODO(solenberg): Encoder config.
83 ss << ", cng_payload_type: " << send_codec_spec.cng_payload_type;
84 ss << '}';
85 return ss.str();
86 }
87
43 namespace internal { 88 namespace internal {
44 AudioSendStream::AudioSendStream( 89 AudioSendStream::AudioSendStream(
45 const webrtc::AudioSendStream::Config& config, 90 const webrtc::AudioSendStream::Config& config,
46 const rtc::scoped_refptr<webrtc::AudioState>& audio_state, 91 const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
47 rtc::TaskQueue* worker_queue, 92 rtc::TaskQueue* worker_queue,
48 CongestionController* congestion_controller, 93 CongestionController* congestion_controller,
49 BitrateAllocator* bitrate_allocator, 94 BitrateAllocator* bitrate_allocator,
50 RtcEventLog* event_log) 95 RtcEventLog* event_log)
51 : worker_queue_(worker_queue), 96 : worker_queue_(worker_queue),
52 config_(config), 97 config_(config),
(...skipping 228 matching lines...) Expand 10 before | Expand all | Expand 10 after
281 ScopedVoEInterface<VoECodec> codec(voice_engine()); 326 ScopedVoEInterface<VoECodec> codec(voice_engine());
282 327
283 const int channel = config_.voe_channel_id; 328 const int channel = config_.voe_channel_id;
284 329
285 // Disable VAD and FEC unless we know the other side wants them. 330 // Disable VAD and FEC unless we know the other side wants them.
286 codec->SetVADStatus(channel, false); 331 codec->SetVADStatus(channel, false);
287 codec->SetFECStatus(channel, false); 332 codec->SetFECStatus(channel, false);
288 333
289 const auto& send_codec_spec = config_.send_codec_spec; 334 const auto& send_codec_spec = config_.send_codec_spec;
290 335
291 // We set the codec first, since the below extra configuration is only applied 336 // Set the codec immediately, since SetVADStatus() depends on whether
292 // to the "current" codec. 337 // the current codec is mono or stereo.
338 LOG(LS_INFO) << "Send channel " << channel << " selected voice codec "
339 << ToString(send_codec_spec.codec_inst)
340 << ", bitrate=" << send_codec_spec.codec_inst.rate;
293 341
294 // If codec is already configured, we do not it again. 342 // If codec is already configured, we do not it again.
295 // TODO(minyue): check if this check is really needed, or can we move it into 343 // TODO(minyue): check if this check is really needed, or can we move it into
296 // |codec->SetSendCodec|. 344 // |codec->SetSendCodec|.
297 webrtc::CodecInst current_codec = {0}; 345 webrtc::CodecInst current_codec = {0};
298 if (codec->GetSendCodec(channel, current_codec) != 0 || 346 if (codec->GetSendCodec(channel, current_codec) != 0 ||
299 (send_codec_spec.codec_inst != current_codec)) { 347 (send_codec_spec.codec_inst != current_codec)) {
300 if (codec->SetSendCodec(channel, send_codec_spec.codec_inst) == -1) { 348 if (codec->SetSendCodec(channel, send_codec_spec.codec_inst) == -1) {
301 LOG(LS_WARNING) << "SetSendCodec() failed: " << base->LastError(); 349 LOG_RTCERR2(SetSendCodec, channel, ToString(send_codec_spec.codec_inst),
350 base->LastError());
302 return false; 351 return false;
303 } 352 }
304 } 353 }
305 354
306 // Codec internal FEC. Treat any failure as fatal internal error. 355 // FEC should be enabled after SetSendCodec.
307 if (send_codec_spec.enable_codec_fec) { 356 if (send_codec_spec.enable_codec_fec) {
308 if (codec->SetFECStatus(channel, true) != 0) { 357 LOG(LS_INFO) << "Attempt to enable codec internal FEC on channel "
309 LOG(LS_WARNING) << "SetFECStatus() failed: " << base->LastError(); 358 << channel;
359 if (codec->SetFECStatus(channel, true) == -1) {
360 // Enable codec internal FEC. Treat any failure as fatal internal error.
361 LOG_RTCERR2(SetFECStatus, channel, true, base->LastError());
310 return false; 362 return false;
311 } 363 }
312 } 364 }
313 365
314 // DTX and maxplaybackrate are only set if current codec is Opus.
315 if (IsCodec(send_codec_spec.codec_inst, kOpusCodecName)) { 366 if (IsCodec(send_codec_spec.codec_inst, kOpusCodecName)) {
316 if (codec->SetOpusDtx(channel, send_codec_spec.enable_opus_dtx) != 0) { 367 // DTX and maxplaybackrate should be set after SetSendCodec. Because current
317 LOG(LS_WARNING) << "SetOpusDtx() failed: " << base->LastError(); 368 // send codec has to be Opus.
369
370 // Set Opus internal DTX.
371 LOG(LS_INFO) << "Attempt to "
372 << (send_codec_spec.enable_opus_dtx ? "enable" : "disable")
373 << " Opus DTX on channel " << channel;
374 if (codec->SetOpusDtx(channel, send_codec_spec.enable_opus_dtx)) {
375 LOG_RTCERR2(SetOpusDtx, channel, send_codec_spec.enable_opus_dtx,
376 base->LastError());
318 return false; 377 return false;
319 } 378 }
320 379
321 // If opus_max_playback_rate <= 0, the default maximum playback rate 380 // If opus_max_playback_rate <= 0, the default maximum playback rate
322 // (48 kHz) will be used. 381 // (48 kHz) will be used.
323 if (send_codec_spec.opus_max_playback_rate > 0) { 382 if (send_codec_spec.opus_max_playback_rate > 0) {
383 LOG(LS_INFO) << "Attempt to set maximum playback rate to "
384 << send_codec_spec.opus_max_playback_rate
385 << " Hz on channel " << channel;
324 if (codec->SetOpusMaxPlaybackRate( 386 if (codec->SetOpusMaxPlaybackRate(
325 channel, send_codec_spec.opus_max_playback_rate) != 0) { 387 channel, send_codec_spec.opus_max_playback_rate) == -1) {
326 LOG(LS_WARNING) << "SetOpusMaxPlaybackRate() failed: " 388 LOG_RTCERR2(SetOpusMaxPlaybackRate, channel,
327 << base->LastError(); 389 send_codec_spec.opus_max_playback_rate, base->LastError());
328 return false; 390 return false;
329 } 391 }
330 } 392 }
331 } 393 }
332 394
333 // Set the CN payloadtype and the VAD status. 395 // Set the CN payloadtype and the VAD status.
334 if (send_codec_spec.cng_payload_type != -1) { 396 if (send_codec_spec.cng_payload_type != -1) {
335 // The CN payload type for 8000 Hz clockrate is fixed at 13. 397 // The CN payload type for 8000 Hz clockrate is fixed at 13.
336 if (send_codec_spec.cng_plfreq != 8000) { 398 if (send_codec_spec.cng_plfreq != 8000) {
337 webrtc::PayloadFrequencies cn_freq; 399 webrtc::PayloadFrequencies cn_freq;
338 switch (send_codec_spec.cng_plfreq) { 400 switch (send_codec_spec.cng_plfreq) {
339 case 16000: 401 case 16000:
340 cn_freq = webrtc::kFreq16000Hz; 402 cn_freq = webrtc::kFreq16000Hz;
341 break; 403 break;
342 case 32000: 404 case 32000:
343 cn_freq = webrtc::kFreq32000Hz; 405 cn_freq = webrtc::kFreq32000Hz;
344 break; 406 break;
345 default: 407 default:
346 RTC_NOTREACHED(); 408 RTC_NOTREACHED();
347 return false; 409 return false;
348 } 410 }
349 if (codec->SetSendCNPayloadType(channel, send_codec_spec.cng_payload_type, 411 if (codec->SetSendCNPayloadType(channel, send_codec_spec.cng_payload_type,
350 cn_freq) != 0) { 412 cn_freq) == -1) {
351 LOG(LS_WARNING) << "SetSendCNPayloadType() failed: " 413 LOG_RTCERR3(SetSendCNPayloadType, channel,
352 << base->LastError(); 414 send_codec_spec.cng_payload_type, cn_freq,
415 base->LastError());
416
353 // TODO(ajm): This failure condition will be removed from VoE. 417 // TODO(ajm): This failure condition will be removed from VoE.
354 // Restore the return here when we update to a new enough webrtc. 418 // Restore the return here when we update to a new enough webrtc.
355 // 419 //
356 // Not returning false because the SetSendCNPayloadType will fail if 420 // Not returning false because the SetSendCNPayloadType will fail if
357 // the channel is already sending. 421 // the channel is already sending.
358 // This can happen if the remote description is applied twice, for 422 // This can happen if the remote description is applied twice, for
359 // example in the case of ROAP on top of JSEP, where both side will 423 // example in the case of ROAP on top of JSEP, where both side will
360 // send the offer. 424 // send the offer.
361 } 425 }
362 } 426 }
363 427
364 // Only turn on VAD if we have a CN payload type that matches the 428 // Only turn on VAD if we have a CN payload type that matches the
365 // clockrate for the codec we are going to use. 429 // clockrate for the codec we are going to use.
366 if (send_codec_spec.cng_plfreq == send_codec_spec.codec_inst.plfreq && 430 if (send_codec_spec.cng_plfreq == send_codec_spec.codec_inst.plfreq &&
367 send_codec_spec.codec_inst.channels == 1) { 431 send_codec_spec.codec_inst.channels == 1) {
368 // TODO(minyue): If CN frequency == 48000 Hz is allowed, consider the 432 // TODO(minyue): If CN frequency == 48000 Hz is allowed, consider the
369 // interaction between VAD and Opus FEC. 433 // interaction between VAD and Opus FEC.
370 if (codec->SetVADStatus(channel, true) != 0) { 434 LOG(LS_INFO) << "Enabling VAD";
371 LOG(LS_WARNING) << "SetVADStatus() failed: " << base->LastError(); 435 if (codec->SetVADStatus(channel, true) == -1) {
436 LOG_RTCERR2(SetVADStatus, channel, true, base->LastError());
372 return false; 437 return false;
373 } 438 }
374 } 439 }
375 } 440 }
376 return true; 441 return true;
377 } 442 }
378 443
379 } // namespace internal 444 } // namespace internal
380 } // namespace webrtc 445 } // namespace webrtc
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