commit dea73ee8f9c0707aed35b9edb928216877f72091
author Minyue Li <minyue@webrtc.org> Tue Feb 18 15:45:41 2020 +0100
committer Commit Bot <commit-bot@chromium.org> Tue Feb 18 16:54:48 2020 +0000
tree 76971cd333537db65921386e472377d957c1c542
parent 2272f20a0a14e705adaa21c773b43d594a5a2114

Pass absolute capture time from WebRtcVoiceEngine to ACM.

Bug: webrtc:10739
Change-Id: I6f264cb89ce340db642db3ef7dfc2b5d459f749e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167211
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Chen Xing <chxg@google.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30547}

modules/audio_coding/acm2/audio_coding_module.cc

diff --git a/modules/audio_coding/acm2/audio_coding_module.cc b/modules/audio_coding/acm2/audio_coding_module.cc
index f3dd5b1..e28be18 100644
--- a/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/modules/audio_coding/acm2/audio_coding_module.cc
@@ -109,7 +109,6 @@
     // If a re-mix is required (up or down), this buffer will store a re-mixed
     // version of the input.
     std::vector<int16_t> buffer;
-    int64_t absolute_capture_timestamp_ms;
   };
 
   InputData input_data_ RTC_GUARDED_BY(acm_crit_sect_);
@@ -132,7 +131,11 @@
 
   int Add10MsDataInternal(const AudioFrame& audio_frame, InputData* input_data)
       RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
-  int Encode(const InputData& input_data)
+
+  // TODO(bugs.webrtc.org/10739): change |absolute_capture_timestamp_ms| to
+  // int64_t when it always receives a valid value.
+  int Encode(const InputData& input_data,
+             absl::optional<int64_t> absolute_capture_timestamp_ms)
       RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
 
   int InitializeReceiverSafe() RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
@@ -231,7 +234,11 @@
 
 AudioCodingModuleImpl::~AudioCodingModuleImpl() = default;
 
-int32_t AudioCodingModuleImpl::Encode(const InputData& input_data) {
+int32_t AudioCodingModuleImpl::Encode(
+    const InputData& input_data,
+    absl::optional<int64_t> absolute_capture_timestamp_ms) {
+  // TODO(bugs.webrtc.org/10739): add dcheck that
+  // |audio_frame.absolute_capture_timestamp_ms()| always has a value.
   AudioEncoder::EncodedInfo encoded_info;
   uint8_t previous_pltype;
 
@@ -304,7 +311,7 @@
       packetization_callback_->SendData(
           frame_type, encoded_info.payload_type, encoded_info.encoded_timestamp,
           encode_buffer_.data(), encode_buffer_.size(),
-          input_data.absolute_capture_timestamp_ms);
+          absolute_capture_timestamp_ms.value_or(-1));
     }
 
     if (vad_callback_) {
@@ -339,7 +346,11 @@
 int AudioCodingModuleImpl::Add10MsData(const AudioFrame& audio_frame) {
   rtc::CritScope lock(&acm_crit_sect_);
   int r = Add10MsDataInternal(audio_frame, &input_data_);
-  return r < 0 ? r : Encode(input_data_);
+  // TODO(bugs.webrtc.org/10739): add dcheck that
+  // |audio_frame.absolute_capture_timestamp_ms()| always has a value.
+  return r < 0
+             ? r
+             : Encode(input_data_, audio_frame.absolute_capture_timestamp_ms());
 }
 
 int AudioCodingModuleImpl::Add10MsDataInternal(const AudioFrame& audio_frame,
@@ -394,9 +405,6 @@
   input_data->input_timestamp = ptr_frame->timestamp_;
   input_data->length_per_channel = ptr_frame->samples_per_channel_;
   input_data->audio_channel = current_num_channels;
-  // TODO(bugs.webrtc.org/10739): Assign from a corresponding field in
-  // audio_frame when it is added in AudioFrame.
-  input_data->absolute_capture_timestamp_ms = 0;
 
   if (!same_num_channels) {
     // Remixes the input frame to the output data and in the process resize the