WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType
This removes one more place where we were unable to handle codecs not
in the built-in set.
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2686043006
Cr-Commit-Position: refs/heads/master@{#17370}
diff --git a/webrtc/modules/audio_coding/acm2/acm_receiver.cc b/webrtc/modules/audio_coding/acm2/acm_receiver.cc
index 21dbc74..4980d27 100644
--- a/webrtc/modules/audio_coding/acm2/acm_receiver.cc
+++ b/webrtc/modules/audio_coding/acm2/acm_receiver.cc
@@ -179,6 +179,10 @@
return 0;
}
+void AcmReceiver::SetCodecs(const std::map<int, SdpAudioFormat>& codecs) {
+ neteq_->SetCodecs(codecs);
+}
+
int32_t AcmReceiver::AddCodec(int acm_codec_id,
uint8_t payload_type,
size_t channels,
diff --git a/webrtc/modules/audio_coding/acm2/acm_receiver.h b/webrtc/modules/audio_coding/acm2/acm_receiver.h
index 63ed43d..8f0963f 100644
--- a/webrtc/modules/audio_coding/acm2/acm_receiver.h
+++ b/webrtc/modules/audio_coding/acm2/acm_receiver.h
@@ -79,6 +79,9 @@
//
int GetAudio(int desired_freq_hz, AudioFrame* audio_frame, bool* muted);
+ // Replace the current set of decoders with the specified set.
+ void SetCodecs(const std::map<int, SdpAudioFormat>& codecs);
+
//
// Adds a new codec to the NetEq codec database.
//
diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
index daeea35..bc814f6 100644
--- a/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
@@ -121,6 +121,8 @@
// Get current playout frequency.
int PlayoutFrequency() const override;
+ void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs) override;
+
bool RegisterReceiveCodec(int rtp_payload_type,
const SdpAudioFormat& audio_format) override;
@@ -318,16 +320,6 @@
webrtc::AudioEncoder::CodecType::kMaxLoggedAudioCodecTypes));
}
-// TODO(turajs): the same functionality is used in NetEq. If both classes
-// need them, make it a static function in ACMCodecDB.
-bool IsCodecRED(const CodecInst& codec) {
- return (STR_CASE_CMP(codec.plname, "RED") == 0);
-}
-
-bool IsCodecCN(const CodecInst& codec) {
- return (STR_CASE_CMP(codec.plname, "CN") == 0);
-}
-
// Stereo-to-mono can be used as in-place.
int DownMix(const AudioFrame& frame,
size_t length_out_buff,
@@ -956,19 +948,6 @@
receiver_.SetMaximumDelay(0);
receiver_.FlushBuffers();
- // Register RED and CN.
- auto db = acm2::RentACodec::Database();
- for (size_t i = 0; i < db.size(); i++) {
- if (IsCodecRED(db[i]) || IsCodecCN(db[i])) {
- if (receiver_.AddCodec(static_cast<int>(i),
- static_cast<uint8_t>(db[i].pltype), 1,
- db[i].plfreq, nullptr, db[i].plname) < 0) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
- "Cannot register master codec.");
- return -1;
- }
- }
- }
receiver_initialized_ = true;
return 0;
}
@@ -987,6 +966,12 @@
return receiver_.last_output_sample_rate_hz();
}
+void AudioCodingModuleImpl::SetReceiveCodecs(
+ const std::map<int, SdpAudioFormat>& codecs) {
+ rtc::CritScope lock(&acm_crit_sect_);
+ receiver_.SetCodecs(codecs);
+}
+
bool AudioCodingModuleImpl::RegisterReceiveCodec(
int rtp_payload_type,
const SdpAudioFormat& audio_format) {