Remove AudioCodingModule::IncomingPayload
This method is no longer in use.
Bug: webrtc:3520
Change-Id: Ie1419fa95e6ef482e9adc1ed7af57af2c3510a65
Reviewed-on: https://webrtc-review.googlesource.com/4667
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20047}diff --git a/modules/audio_coding/acm2/audio_coding_module.cc b/modules/audio_coding/acm2/audio_coding_module.cc
index 2778610..3320d1b 100644
--- a/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/modules/audio_coding/acm2/audio_coding_module.cc
@@ -149,13 +149,6 @@
const size_t payload_length,
const WebRtcRTPHeader& rtp_info) override;
- // Incoming payloads, without rtp-info, the rtp-info will be created in ACM.
- // One usage for this API is when pre-encoded files are pushed in ACM.
- int IncomingPayload(const uint8_t* incoming_payload,
- const size_t payload_length,
- uint8_t payload_type,
- uint32_t timestamp) override;
-
// Minimum playout delay.
int SetMinimumPlayoutDelay(int time_ms) override;
@@ -291,14 +284,6 @@
// This is to keep track of CN instances where we can send DTMFs.
uint8_t previous_pltype_ RTC_GUARDED_BY(acm_crit_sect_);
- // Used when payloads are pushed into ACM without any RTP info
- // One example is when pre-encoded bit-stream is pushed from
- // a file.
- // IMPORTANT: this variable is only used in IncomingPayload(), therefore,
- // no lock acquired when interacting with this variable. If it is going to
- // be used in other methods, locks need to be taken.
- std::unique_ptr<WebRtcRTPHeader> aux_rtp_header_;
-
bool receiver_initialized_ RTC_GUARDED_BY(acm_crit_sect_);
AudioFrame preprocess_frame_ RTC_GUARDED_BY(acm_crit_sect_);
@@ -1147,35 +1132,6 @@
return 0;
}
-// TODO(kwiberg): Remove this method, and have callers call IncomingPacket
-// instead. The translation logic and state belong with them, not with
-// AudioCodingModuleImpl.
-int AudioCodingModuleImpl::IncomingPayload(const uint8_t* incoming_payload,
- size_t payload_length,
- uint8_t payload_type,
- uint32_t timestamp) {
- // We are not acquiring any lock when interacting with |aux_rtp_header_| no
- // other method uses this member variable.
- if (!aux_rtp_header_) {
- // This is the first time that we are using |dummy_rtp_header_|
- // so we have to create it.
- aux_rtp_header_.reset(new WebRtcRTPHeader);
- aux_rtp_header_->header.payloadType = payload_type;
- // Don't matter in this case.
- aux_rtp_header_->header.ssrc = 0;
- aux_rtp_header_->header.markerBit = false;
- // Start with random numbers.
- aux_rtp_header_->header.sequenceNumber = 0x1234; // Arbitrary.
- aux_rtp_header_->type.Audio.channel = 1;
- }
-
- aux_rtp_header_->header.timestamp = timestamp;
- IncomingPacket(incoming_payload, payload_length, *aux_rtp_header_);
- // Get ready for the next payload.
- aux_rtp_header_->header.sequenceNumber++;
- return 0;
-}
-
int AudioCodingModuleImpl::SetOpusApplication(OpusApplicationMode application) {
rtc::CritScope lock(&acm_crit_sect_);
if (!HaveValidEncoder("SetOpusApplication")) {
diff --git a/modules/audio_coding/include/audio_coding_module.h b/modules/audio_coding/include/audio_coding_module.h
index 2013cd7..41756fb 100644
--- a/modules/audio_coding/include/audio_coding_module.h
+++ b/modules/audio_coding/include/audio_coding_module.h
@@ -588,35 +588,6 @@
const WebRtcRTPHeader& rtp_info) = 0;
///////////////////////////////////////////////////////////////////////////
- // int32_t IncomingPayload()
- // Call this API to push incoming payloads when there is no rtp-info.
- // The rtp-info will be created in ACM. One usage for this API is when
- // pre-encoded files are pushed in ACM
- //
- // Inputs:
- // -incoming_payload : received payload.
- // -payload_len_byte : the length, in bytes, of the received payload.
- // -payload_type : the payload-type. This specifies which codec has
- // to be used to decode the payload.
- // -timestamp : send timestamp of the payload. ACM starts with
- // a random value and increment it by the
- // packet-size, which is given when the codec in
- // question is registered by RegisterReceiveCodec().
- // Therefore, it is essential to have the timestamp
- // if the frame-size differ from the registered
- // value or if the incoming payload contains DTX
- // packets.
- //
- // Return value:
- // -1 if failed to push in the payload
- // 0 if payload is successfully pushed in.
- //
- virtual int32_t IncomingPayload(const uint8_t* incoming_payload,
- const size_t payload_len_byte,
- const uint8_t payload_type,
- const uint32_t timestamp = 0) = 0;
-
- ///////////////////////////////////////////////////////////////////////////
// int SetMinimumPlayoutDelay()
// Set a minimum for the playout delay, used for lip-sync. NetEq maintains
// such a delay unless channel condition yields to a higher delay.
