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Pass absolute capture time from WebRtcVoiceEngine to ACM.

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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}
This commit is contained in:
Minyue Li
2020-02-18 15:45:41 +01:00
committed by Commit Bot
parent 2272f20a0a
commit dea73ee8f9
4 changed files with 41 additions and 12 deletions
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24
modules/audio_coding/acm2/audio_coding_module.cc
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@ -109,7 +109,6 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
// 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 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
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(
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 @@ int32_t AudioCodingModuleImpl::Encode(const InputData& input_data) {
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::RegisterTransportCallback(
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 @@ int AudioCodingModuleImpl::Add10MsDataInternal(const AudioFrame& audio_frame,
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