zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Pass absolute capture time from WebRtcVoiceEngine to ACM.

Browse Source 
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
Download Patch File Download Diff File Expand all files Collapse all files

2
api/audio/audio_frame.cc
View File

@ -41,6 +41,7 @@ void AudioFrame::ResetWithoutMuting() {
vad_activity_ = kVadUnknown;
profile_timestamp_ms_ = 0;
packet_infos_ = RtpPacketInfos();
absolute_capture_timestamp_ms_ = absl::nullopt;
}
void AudioFrame::UpdateFrame(uint32_t timestamp,
@ -86,6 +87,7 @@ void AudioFrame::CopyFrom(const AudioFrame& src) {
vad_activity_ = src.vad_activity_;
num_channels_ = src.num_channels_;
channel_layout_ = src.channel_layout_;
absolute_capture_timestamp_ms_ = src.absolute_capture_timestamp_ms();
const size_t length = samples_per_channel_ * num_channels_;
RTC_CHECK_LE(length, kMaxDataSizeSamples);

19
api/audio/audio_frame.h
View File

@ -104,6 +104,15 @@ class AudioFrame {
ChannelLayout channel_layout() const { return channel_layout_; }
int sample_rate_hz() const { return sample_rate_hz_; }
void set_absolute_capture_timestamp_ms(
int64_t absolute_capture_time_stamp_ms) {
absolute_capture_timestamp_ms_ = absolute_capture_time_stamp_ms;
}
absl::optional<int64_t> absolute_capture_timestamp_ms() const {
return absolute_capture_timestamp_ms_;
}
// RTP timestamp of the first sample in the AudioFrame.
uint32_t timestamp_ = 0;
// Time since the first frame in milliseconds.
@ -121,8 +130,8 @@ class AudioFrame {
// Monotonically increasing timestamp intended for profiling of audio frames.
// Typically used for measuring elapsed time between two different points in
// the audio path. No lock is used to save resources and we are thread safe
// by design. Also, absl::optional is not used since it will cause a "complex
// class/struct needs an explicit out-of-line destructor" build error.
// by design.
// TODO(nisse@webrtc.org): consider using absl::optional.
int64_t profile_timestamp_ms_ = 0;
// Information about packets used to assemble this audio frame. This is needed
@ -150,6 +159,12 @@ class AudioFrame {
int16_t data_[kMaxDataSizeSamples];
bool muted_ = true;
// Absolute capture timestamp when this audio frame was originally captured.
// This is only valid for audio frames captured on this machine. The absolute
// capture timestamp of a received frame is found in |packet_infos_|.
// This timestamp MUST be based on the same clock as rtc::TimeMillis().
absl::optional<int64_t> absolute_capture_timestamp_ms_;
RTC_DISALLOW_COPY_AND_ASSIGN(AudioFrame);
};

8
media/engine/webrtc_voice_engine.cc
View File

@ -880,8 +880,12 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
audio_frame->timestamp_, static_cast<const int16_t*>(audio_data),
number_of_frames, sample_rate, audio_frame->speech_type_,
audio_frame->vad_activity_, number_of_channels);
// TODO(bugs.webrtc.org/10739): pass absolute_capture_timestamp_ms to
// stream_.
// TODO(bugs.webrtc.org/10739): add dcheck that
// |absolute_capture_timestamp_ms| always receives a value.
if (absolute_capture_timestamp_ms) {
audio_frame->set_absolute_capture_timestamp_ms(
*absolute_capture_timestamp_ms);
}
stream_->SendAudioData(std::move(audio_frame));
}

24
modules/audio_coding/acm2/audio_coding_module.cc
View File

@ -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