| Index: webrtc/media/engine/webrtcvoiceengine.cc
|
| diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
|
| index eb9b4bb3569ce01f175d1f80988f0aefd21a4820..65457b05982fa70b9236a06708a212203e80bd16 100644
|
| --- a/webrtc/media/engine/webrtcvoiceengine.cc
|
| +++ b/webrtc/media/engine/webrtcvoiceengine.cc
|
| @@ -33,6 +33,7 @@
|
| #include "webrtc/media/base/audiosource.h"
|
| #include "webrtc/media/base/mediaconstants.h"
|
| #include "webrtc/media/base/streamparams.h"
|
| +#include "webrtc/media/engine/payload_type_mapper.h"
|
| #include "webrtc/media/engine/webrtcmediaengine.h"
|
| #include "webrtc/media/engine/webrtcvoe.h"
|
| #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
|
| @@ -249,7 +250,7 @@ class WebRtcVoiceCodecs final {
|
| public:
|
| // TODO(solenberg): Do this filtering once off-line, add a simple AudioCodec
|
| // list and add a test which verifies VoE supports the listed codecs.
|
| - static std::vector<AudioCodec> SupportedCodecs() {
|
| + static std::vector<AudioCodec> SupportedSendCodecs() {
|
| std::vector<AudioCodec> result;
|
| // Iterate first over our preferred codecs list, so that the results are
|
| // added in order of preference.
|
| @@ -512,13 +513,20 @@ WebRtcVoiceEngine::WebRtcVoiceEngine(
|
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| LOG(LS_INFO) << "WebRtcVoiceEngine::WebRtcVoiceEngine";
|
| RTC_DCHECK(voe_wrapper);
|
| + RTC_DCHECK(decoder_factory);
|
|
|
| signal_thread_checker_.DetachFromThread();
|
|
|
| // Load our audio codec list.
|
| - LOG(LS_INFO) << "Supported codecs in order of preference:";
|
| - codecs_ = WebRtcVoiceCodecs::SupportedCodecs();
|
| - for (const AudioCodec& codec : codecs_) {
|
| + LOG(LS_INFO) << "Supported send codecs in order of preference:";
|
| + send_codecs_ = WebRtcVoiceCodecs::SupportedSendCodecs();
|
| + for (const AudioCodec& codec : send_codecs_) {
|
| + LOG(LS_INFO) << ToString(codec);
|
| + }
|
| +
|
| + LOG(LS_INFO) << "Supported recv codecs in order of preference:";
|
| + recv_codecs_ = CollectRecvCodecs();
|
| + for (const AudioCodec& codec : recv_codecs_) {
|
| LOG(LS_INFO) << ToString(codec);
|
| }
|
|
|
| @@ -906,12 +914,12 @@ int WebRtcVoiceEngine::GetInputLevel() {
|
|
|
| const std::vector<AudioCodec>& WebRtcVoiceEngine::send_codecs() const {
|
| RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
|
| - return codecs_;
|
| + return send_codecs_;
|
| }
|
|
|
| const std::vector<AudioCodec>& WebRtcVoiceEngine::recv_codecs() const {
|
| RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
|
| - return codecs_;
|
| + return recv_codecs_;
|
| }
|
|
|
| RtpCapabilities WebRtcVoiceEngine::GetCapabilities() const {
|
| @@ -1072,6 +1080,64 @@ webrtc::AudioDeviceModule* WebRtcVoiceEngine::adm() {
|
| return adm_;
|
| }
|
|
|
| +AudioCodecs WebRtcVoiceEngine::CollectRecvCodecs() const {
|
| + PayloadTypeMapper mapper;
|
| + AudioCodecs out;
|
| + std::vector<webrtc::SdpAudioFormat> formats =
|
| + decoder_factory_->GetSupportedFormats();
|
| + std::map<int, bool> generate_cn = {{ 8000, false },
|
| + { 16000, false },
|
| + { 32000, false }};
|
| +
|
| + auto map_format = [&mapper, &out] (const webrtc::SdpAudioFormat& format) {
|
| + rtc::Optional<AudioCodec> opt_codec = mapper.ToAudioCodec(format);
|
| + if (opt_codec) {
|
| + out.push_back(*opt_codec);
|
| + } else {
|
| + LOG(LS_ERROR) << "Unable to assign payload type to format: " << format;
|
| + }
|
| + };
|
| +
|
| + for (const auto& format : formats) {
|
| + auto opt_codec = mapper.ToAudioCodec(format);
|
| + if (opt_codec) {
|
| + auto& codec = *opt_codec;
|
| +
|
| + if (IsCodec(codec, kOpusCodecName)) {
|
| + // TODO(ossu): Should we set this specifically for just this codec?
|
| + // If not, this can be replaced with a call to map_format.
|
| + codec.AddFeedbackParam(
|
| + FeedbackParam(kRtcpFbParamTransportCc, kParamValueEmpty));
|
| + }
|
| + out.push_back(codec);
|
| +
|
| + // TODO(ossu): We should get more than just a format from the factory, so
|
| + // we
|
| + // can determine if a format should be used with CN or not. For now,
|
| + // generate a CN entry for each supported clock rate also used by a format
|
| + // supported by the factory.
|
| + auto cn_entry = generate_cn.find(format.clockrate_hz);
|
| + if (cn_entry != generate_cn.end() /* && format.allow_comfort_noise */) {
|
| + cn_entry->second = true;
|
| + }
|
| + } else {
|
| + LOG(LS_ERROR) << "Unable to assign payload type to format: " << format;
|
| + }
|
| + }
|
| +
|
| + // Add CN codecs
|
| + for (const auto& cn : generate_cn) {
|
| + if (cn.second) {
|
| + map_format({kCnCodecName, cn.first, 1});
|
| + }
|
| + }
|
| +
|
| + // Add telephone-event codec
|
| + map_format({kDtmfCodecName, 8000, 1});
|
| +
|
| + return out;
|
| +}
|
| +
|
| class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
|
| : public AudioSource::Sink {
|
| public:
|
|
|