Bug 1860685 - Cherry-pick upstream libwebrtc commit 10e5724fe9 r=mjf

Upstream commit: https://webrtc.googlesource.com/src/+/10e5724fe9
       Delete deprecated variants of RTPSenderAudio::SendAudio

       Bug: webrtc:13757
       Change-Id: I402a31c847ca7ffe0ef20a0046959ec50c60e3ac
       Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319582
       Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
       Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
       Reviewed-by: Åsa Persson <asapersson@webrtc.org>
       Cr-Commit-Position: refs/heads/main@{#40740}

Differential Revision: https://phabricator.services.mozilla.com/D192395

third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc

@@ -141,35 +141,6 @@ bool RTPSenderAudio::MarkerBit(AudioFrameType frame_type, int8_t payload_type) {
   return marker_bit;
 }
 
-bool RTPSenderAudio::SendAudio(AudioFrameType frame_type,
-                               int8_t payload_type,
-                               uint32_t rtp_timestamp,
-                               const uint8_t* payload_data,
-                               size_t payload_size) {
-  return SendAudio({.type = frame_type,
-                    .payload{payload_data, payload_size},
-                    .payload_id = payload_type,
-                    .rtp_timestamp = rtp_timestamp});
-}
-
-bool RTPSenderAudio::SendAudio(AudioFrameType frame_type,
-                               int8_t payload_type,
-                               uint32_t rtp_timestamp,
-                               const uint8_t* payload_data,
-                               size_t payload_size,
-                               int64_t absolute_capture_timestamp_ms) {
-  RtpAudioFrame frame = {
-      .type = frame_type,
-      .payload{payload_data, payload_size},
-      .payload_id = payload_type,
-      .rtp_timestamp = rtp_timestamp,
-  };
-  if (absolute_capture_timestamp_ms > 0) {
-    frame.capture_time = Timestamp::Millis(absolute_capture_timestamp_ms);
-  }
-  return SendAudio(frame);
-}
-
 bool RTPSenderAudio::SendAudio(const RtpAudioFrame& frame) {
   RTC_DCHECK_GE(frame.payload_id, 0);
   RTC_DCHECK_LE(frame.payload_id, 127);
@@ -182,12 +153,10 @@ bool RTPSenderAudio::SendAudio(const RtpAudioFrame& frame) {
   // Alternatively, a source MAY decide to use a different spacing for event
   // updates, with a value of 50 ms RECOMMENDED.
   constexpr int kDtmfIntervalTimeMs = 50;
-  uint8_t audio_level_dbov = 0;
   uint32_t dtmf_payload_freq = 0;
   absl::optional<uint32_t> encoder_rtp_timestamp_frequency;
   {
     MutexLock lock(&send_audio_mutex_);
-    audio_level_dbov = audio_level_dbov_;
     dtmf_payload_freq = dtmf_payload_freq_;
     encoder_rtp_timestamp_frequency = encoder_rtp_timestamp_frequency_;
   }
@@ -278,10 +247,10 @@ bool RTPSenderAudio::SendAudio(const RtpAudioFrame& frame) {
   packet->SetPayloadType(frame.payload_id);
   packet->SetTimestamp(frame.rtp_timestamp);
   packet->set_capture_time(clock_->CurrentTime());
-  // Update audio level extension, if included.
+  // Set audio level extension, if included.
   packet->SetExtension<AudioLevel>(
       frame.type == AudioFrameType::kAudioFrameSpeech,
-      frame.audio_level_dbov.value_or(audio_level_dbov));
+      frame.audio_level_dbov.value_or(127));
 
   if (frame.capture_time.has_value()) {
     // Send absolute capture time periodically in order to optimize and save
@@ -326,16 +295,6 @@ bool RTPSenderAudio::SendAudio(const RtpAudioFrame& frame) {
   return true;
 }
 
-// Audio level magnitude and voice activity flag are set for each RTP packet
-int32_t RTPSenderAudio::SetAudioLevel(uint8_t level_dbov) {
-  if (level_dbov > 127) {
-    return -1;
-  }
-  MutexLock lock(&send_audio_mutex_);
-  audio_level_dbov_ = level_dbov;
-  return 0;
-}
-
 // Send a TelephoneEvent tone using RFC 2833 (4733)
 int32_t RTPSenderAudio::SendTelephoneEvent(uint8_t key,
                                            uint16_t time_ms,

third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio.h

@@ -64,26 +64,6 @@ class RTPSenderAudio {
   };
   bool SendAudio(const RtpAudioFrame& frame);
 
-  [[deprecated]] bool SendAudio(AudioFrameType frame_type,
-                                int8_t payload_type,
-                                uint32_t rtp_timestamp,
-                                const uint8_t* payload_data,
-                                size_t payload_size);
-
-  // `absolute_capture_timestamp_ms` and `Clock::CurrentTime`
-  // should be using the same epoch.
-  [[deprecated]] bool SendAudio(AudioFrameType frame_type,
-                                int8_t payload_type,
-                                uint32_t rtp_timestamp,
-                                const uint8_t* payload_data,
-                                size_t payload_size,
-                                int64_t absolute_capture_timestamp_ms);
-
-  // Store the audio level in dBov for
-  // header-extension-for-audio-level-indication.
-  // Valid range is [0,127]. Actual value is negative.
-  [[deprecated]] int32_t SetAudioLevel(uint8_t level_dbov);
-
   // Send a DTMF tone using RFC 2833 (4733)
   int32_t SendTelephoneEvent(uint8_t key, uint16_t time_ms, uint8_t level);
 
@@ -122,9 +102,6 @@ class RTPSenderAudio {
   int8_t cngfb_payload_type_ RTC_GUARDED_BY(send_audio_mutex_) = -1;
   int8_t last_payload_type_ RTC_GUARDED_BY(send_audio_mutex_) = -1;
 
-  // Audio level indication.
-  // (https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/)
-  uint8_t audio_level_dbov_ RTC_GUARDED_BY(send_audio_mutex_) = 127;
   OneTimeEvent first_packet_sent_;
 
   absl::optional<uint32_t> encoder_rtp_timestamp_frequency_

third_party/libwebrtc/moz-patch-stack/10e5724fe9.no-op-cherry-pick-msg

@@ -0,0 +1 @@
+We cherry-picked this in bug 1860685