Move AudioFrame to its own header file and target in api/.
This breaks the dependency api:audio_mixer_api --> modules:module_api,
and allows peerconnectioninterface.h to include audio_mixer.h, without
introducing a dependency cycle.
In addition, un-inline all AudioFrame methods, moving implementations
to audio_frame.cc, and replace assert by RTC_CHECK_*.
Bug: webrtc:7504
Change-Id: I11e3d3d22716e9b98976bf830103fbb06e7bbb77
Reviewed-on: https://webrtc-review.googlesource.com/51860
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22016}diff --git a/api/BUILD.gn b/api/BUILD.gn
index 63bc99c..3952cab 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -92,6 +92,7 @@
deps = [
":array_view",
+ ":audio_mixer_api",
":audio_options_api",
":optional",
":peerconnection_and_implicit_call_api",
@@ -188,6 +189,21 @@
]
}
+rtc_source_set("audio_frame_api") {
+ visibility = [ "*" ]
+ sources = [
+ "audio/audio_frame.cc",
+ "audio/audio_frame.h",
+ ]
+
+ deps = [
+ "../:typedefs",
+ "../rtc_base:checks",
+ "../rtc_base:deprecation",
+ "../rtc_base:rtc_base_approved",
+ ]
+}
+
rtc_source_set("audio_mixer_api") {
visibility = [ "*" ]
sources = [
@@ -195,7 +211,7 @@
]
deps = [
- "../modules:module_api",
+ ":audio_frame_api",
"../rtc_base:rtc_base_approved",
]
}
diff --git a/api/audio/audio_frame.cc b/api/audio/audio_frame.cc
new file mode 100644
index 0000000..3dc510d
--- /dev/null
+++ b/api/audio/audio_frame.cc
@@ -0,0 +1,183 @@
+/*
+ * Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio/audio_frame.h"
+
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/timeutils.h"
+
+namespace webrtc {
+
+AudioFrame::AudioFrame() {
+ // Visual Studio doesn't like this in the class definition.
+ static_assert(sizeof(data_) == kMaxDataSizeBytes, "kMaxDataSizeBytes");
+}
+
+void AudioFrame::Reset() {
+ ResetWithoutMuting();
+ muted_ = true;
+}
+
+void AudioFrame::ResetWithoutMuting() {
+ // TODO(wu): Zero is a valid value for |timestamp_|. We should initialize
+ // to an invalid value, or add a new member to indicate invalidity.
+ timestamp_ = 0;
+ elapsed_time_ms_ = -1;
+ ntp_time_ms_ = -1;
+ samples_per_channel_ = 0;
+ sample_rate_hz_ = 0;
+ num_channels_ = 0;
+ speech_type_ = kUndefined;
+ vad_activity_ = kVadUnknown;
+ profile_timestamp_ms_ = 0;
+}
+
+void AudioFrame::UpdateFrame(uint32_t timestamp,
+ const int16_t* data,
+ size_t samples_per_channel,
+ int sample_rate_hz,
+ SpeechType speech_type,
+ VADActivity vad_activity,
+ size_t num_channels) {
+ timestamp_ = timestamp;
+ samples_per_channel_ = samples_per_channel;
+ sample_rate_hz_ = sample_rate_hz;
+ speech_type_ = speech_type;
+ vad_activity_ = vad_activity;
+ num_channels_ = num_channels;
+
+ const size_t length = samples_per_channel * num_channels;
+ RTC_CHECK_LE(length, kMaxDataSizeSamples);
+ if (data != nullptr) {
+ memcpy(data_, data, sizeof(int16_t) * length);
+ muted_ = false;
+ } else {
+ muted_ = true;
+ }
+}
+
+void AudioFrame::CopyFrom(const AudioFrame& src) {
+ if (this == &src) return;
+
+ timestamp_ = src.timestamp_;
+ elapsed_time_ms_ = src.elapsed_time_ms_;
+ ntp_time_ms_ = src.ntp_time_ms_;
+ muted_ = src.muted();
+ samples_per_channel_ = src.samples_per_channel_;
+ sample_rate_hz_ = src.sample_rate_hz_;
+ speech_type_ = src.speech_type_;
+ vad_activity_ = src.vad_activity_;
+ num_channels_ = src.num_channels_;
+
+ const size_t length = samples_per_channel_ * num_channels_;
+ RTC_CHECK_LE(length, kMaxDataSizeSamples);
+ if (!src.muted()) {
+ memcpy(data_, src.data(), sizeof(int16_t) * length);
+ muted_ = false;
+ }
+}
+
+void AudioFrame::UpdateProfileTimeStamp() {
+ profile_timestamp_ms_ = rtc::TimeMillis();
+}
+
+int64_t AudioFrame::ElapsedProfileTimeMs() const {
+ if (profile_timestamp_ms_ == 0) {
+ // Profiling has not been activated.
+ return -1;
+ }
+ return rtc::TimeSince(profile_timestamp_ms_);
+}
+
+const int16_t* AudioFrame::data() const {
+ return muted_ ? empty_data() : data_;
+}
+
+// TODO(henrik.lundin) Can we skip zeroing the buffer?
+// See https://bugs.chromium.org/p/webrtc/issues/detail?id=5647.
+int16_t* AudioFrame::mutable_data() {
+ if (muted_) {
+ memset(data_, 0, kMaxDataSizeBytes);
+ muted_ = false;
+ }
+ return data_;
+}
+
+void AudioFrame::Mute() {
+ muted_ = true;
+}
+
+bool AudioFrame::muted() const { return muted_; }
+
+AudioFrame& AudioFrame::operator>>=(const int rhs) {
+ RTC_CHECK_GT(num_channels_, 0);
+ RTC_CHECK_LT(num_channels_, 3);
+ if ((num_channels_ > 2) || (num_channels_ < 1)) return *this;
+ if (muted_) return *this;
+
+ for (size_t i = 0; i < samples_per_channel_ * num_channels_; i++) {
+ data_[i] = static_cast<int16_t>(data_[i] >> rhs);
+ }
+ return *this;
+}
+
+AudioFrame& AudioFrame::operator+=(const AudioFrame& rhs) {
+ // Sanity check
+ RTC_CHECK_GT(num_channels_, 0);
+ RTC_CHECK_LT(num_channels_, 3);
+ if ((num_channels_ > 2) || (num_channels_ < 1)) return *this;
+ if (num_channels_ != rhs.num_channels_) return *this;
+
+ bool noPrevData = muted_;
+ if (samples_per_channel_ != rhs.samples_per_channel_) {
+ if (samples_per_channel_ == 0) {
+ // special case we have no data to start with
+ samples_per_channel_ = rhs.samples_per_channel_;
+ noPrevData = true;
+ } else {
+ return *this;
+ }
+ }
+
+ if ((vad_activity_ == kVadActive) || rhs.vad_activity_ == kVadActive) {
+ vad_activity_ = kVadActive;
+ } else if (vad_activity_ == kVadUnknown || rhs.vad_activity_ == kVadUnknown) {
+ vad_activity_ = kVadUnknown;
+ }
+
+ if (speech_type_ != rhs.speech_type_) speech_type_ = kUndefined;
+
+ if (!rhs.muted()) {
+ muted_ = false;
+ if (noPrevData) {
+ memcpy(data_, rhs.data(),
+ sizeof(int16_t) * rhs.samples_per_channel_ * num_channels_);
+ } else {
+ // IMPROVEMENT this can be done very fast in assembly
+ for (size_t i = 0; i < samples_per_channel_ * num_channels_; i++) {
+ int32_t wrap_guard =
+ static_cast<int32_t>(data_[i]) + static_cast<int32_t>(rhs.data_[i]);
+ data_[i] = rtc::saturated_cast<int16_t>(wrap_guard);
+ }
+ }
+ }
+
+ return *this;
+}
+
+// static
+const int16_t* AudioFrame::empty_data() {
+ static const int16_t kEmptyData[kMaxDataSizeSamples] = {0};
+ static_assert(sizeof(kEmptyData) == kMaxDataSizeBytes, "kMaxDataSizeBytes");
+ return kEmptyData;
+}
+
+} // namespace webrtc
diff --git a/api/audio/audio_frame.h b/api/audio/audio_frame.h
new file mode 100644
index 0000000..5cb2019
--- /dev/null
+++ b/api/audio/audio_frame.h
@@ -0,0 +1,152 @@
+/*
+ * Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_AUDIO_AUDIO_FRAME_H_
+#define API_AUDIO_AUDIO_FRAME_H_
+
+#include <stdint.h>
+#include <stdlib.h>
+
+#include "rtc_base/constructormagic.h"
+#include "rtc_base/deprecation.h"
+#include "typedefs.h" // NOLINT(build/include)
+
+namespace webrtc {
+
+/* This class holds up to 60 ms of super-wideband (32 kHz) stereo audio. It
+ * allows for adding and subtracting frames while keeping track of the resulting
+ * states.
+ *
+ * Notes
+ * - This is a de-facto api, not designed for external use. The AudioFrame class
+ * is in need of overhaul or even replacement, and anyone depending on it
+ * should be prepared for that.
+ * - The total number of samples is samples_per_channel_ * num_channels_.
+ * - Stereo data is interleaved starting with the left channel.
+ */
+class AudioFrame {
+ public:
+ // Using constexpr here causes linker errors unless the variable also has an
+ // out-of-class definition, which is impractical in this header-only class.
+ // (This makes no sense because it compiles as an enum value, which we most
+ // certainly cannot take the address of, just fine.) C++17 introduces inline
+ // variables which should allow us to switch to constexpr and keep this a
+ // header-only class.
+ enum : size_t {
+ // Stereo, 32 kHz, 60 ms (2 * 32 * 60)
+ kMaxDataSizeSamples = 3840,
+ kMaxDataSizeBytes = kMaxDataSizeSamples * sizeof(int16_t),
+ };
+
+ enum VADActivity {
+ kVadActive = 0,
+ kVadPassive = 1,
+ kVadUnknown = 2
+ };
+ enum SpeechType {
+ kNormalSpeech = 0,
+ kPLC = 1,
+ kCNG = 2,
+ kPLCCNG = 3,
+ kUndefined = 4
+ };
+
+ AudioFrame();
+
+ // Resets all members to their default state.
+ void Reset();
+ // Same as Reset(), but leaves mute state unchanged. Muting a frame requires
+ // the buffer to be zeroed on the next call to mutable_data(). Callers
+ // intending to write to the buffer immediately after Reset() can instead use
+ // ResetWithoutMuting() to skip this wasteful zeroing.
+ void ResetWithoutMuting();
+
+ // TODO(solenberg): Remove once downstream users of AudioFrame have updated.
+ RTC_DEPRECATED
+ void UpdateFrame(int id, uint32_t timestamp, const int16_t* data,
+ size_t samples_per_channel, int sample_rate_hz,
+ SpeechType speech_type, VADActivity vad_activity,
+ size_t num_channels = 1) {
+ RTC_UNUSED(id);
+ UpdateFrame(timestamp, data, samples_per_channel, sample_rate_hz,
+ speech_type, vad_activity, num_channels);
+ }
+
+ void UpdateFrame(uint32_t timestamp, const int16_t* data,
+ size_t samples_per_channel, int sample_rate_hz,
+ SpeechType speech_type, VADActivity vad_activity,
+ size_t num_channels = 1);
+
+ void CopyFrom(const AudioFrame& src);
+
+ // Sets a wall-time clock timestamp in milliseconds to be used for profiling
+ // of time between two points in the audio chain.
+ // Example:
+ // t0: UpdateProfileTimeStamp()
+ // t1: ElapsedProfileTimeMs() => t1 - t0 [msec]
+ void UpdateProfileTimeStamp();
+ // Returns the time difference between now and when UpdateProfileTimeStamp()
+ // was last called. Returns -1 if UpdateProfileTimeStamp() has not yet been
+ // called.
+ int64_t ElapsedProfileTimeMs() const;
+
+ // data() returns a zeroed static buffer if the frame is muted.
+ // mutable_frame() always returns a non-static buffer; the first call to
+ // mutable_frame() zeros the non-static buffer and marks the frame unmuted.
+ const int16_t* data() const;
+ int16_t* mutable_data();
+
+ // Prefer to mute frames using AudioFrameOperations::Mute.
+ void Mute();
+ // Frame is muted by default.
+ bool muted() const;
+
+ // These methods are deprecated. Use the functions in
+ // webrtc/audio/utility instead. These methods will exists for a
+ // short period of time until webrtc clients have updated. See
+ // webrtc:6548 for details.
+ RTC_DEPRECATED AudioFrame& operator>>=(const int rhs);
+ RTC_DEPRECATED AudioFrame& operator+=(const AudioFrame& rhs);
+
+ // RTP timestamp of the first sample in the AudioFrame.
+ uint32_t timestamp_ = 0;
+ // Time since the first frame in milliseconds.
+ // -1 represents an uninitialized value.
+ int64_t elapsed_time_ms_ = -1;
+ // NTP time of the estimated capture time in local timebase in milliseconds.
+ // -1 represents an uninitialized value.
+ int64_t ntp_time_ms_ = -1;
+ size_t samples_per_channel_ = 0;
+ int sample_rate_hz_ = 0;
+ size_t num_channels_ = 0;
+ SpeechType speech_type_ = kUndefined;
+ VADActivity vad_activity_ = kVadUnknown;
+ // 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, rtc::Optional is not used since it will cause a "complex
+ // class/struct needs an explicit out-of-line destructor" build error.
+ int64_t profile_timestamp_ms_ = 0;
+
+ private:
+ // A permamently zeroed out buffer to represent muted frames. This is a
+ // header-only class, so the only way to avoid creating a separate empty
+ // buffer per translation unit is to wrap a static in an inline function.
+ static const int16_t* empty_data();
+
+ int16_t data_[kMaxDataSizeSamples];
+ bool muted_ = true;
+
+ RTC_DISALLOW_COPY_AND_ASSIGN(AudioFrame);
+};
+
+} // namespace webrtc
+
+#endif // API_AUDIO_AUDIO_FRAME_H_
diff --git a/api/audio/audio_mixer.h b/api/audio/audio_mixer.h
index 63b8b8f..14eefc1 100644
--- a/api/audio/audio_mixer.h
+++ b/api/audio/audio_mixer.h
@@ -13,7 +13,7 @@
#include <memory>
-#include "modules/include/module_common_types.h"
+#include "api/audio/audio_frame.h"
#include "rtc_base/refcount.h"
namespace webrtc {
diff --git a/api/peerconnectioninterface.h b/api/peerconnectioninterface.h
index 4a36af9..22514ac 100644
--- a/api/peerconnectioninterface.h
+++ b/api/peerconnectioninterface.h
@@ -76,6 +76,7 @@
#include <utility>
#include <vector>
+#include "api/audio/audio_mixer.h"
#include "api/audio_codecs/audio_decoder_factory.h"
#include "api/audio_codecs/audio_encoder_factory.h"
#include "api/audio_options.h"
diff --git a/modules/BUILD.gn b/modules/BUILD.gn
index 4a77b21..a45fd348 100644
--- a/modules/BUILD.gn
+++ b/modules/BUILD.gn
@@ -51,6 +51,7 @@
":module_api_public",
"..:webrtc_common",
"../:typedefs",
+ "../api:audio_frame_api",
"../api:libjingle_peerconnection_api",
"../api:optional",
"../api:video_frame_api",
diff --git a/modules/include/module_common_types.h b/modules/include/module_common_types.h
index 5860b59..0842370 100644
--- a/modules/include/module_common_types.h
+++ b/modules/include/module_common_types.h
@@ -18,6 +18,9 @@
#include <limits>
#include "api/optional.h"
+// TODO(bugs.webrtc.org/7504): Included here because users of this header expect
+// it to declare AudioFrame. Delete as soon as all known users are updated.
+#include "api/audio/audio_frame.h"
#include "api/rtp_headers.h"
#include "api/video/video_rotation.h"
#include "common_types.h" // NOLINT(build/include)
@@ -288,289 +291,6 @@
virtual ~CallStatsObserver() {}
};
-/* This class holds up to 60 ms of super-wideband (32 kHz) stereo audio. It
- * allows for adding and subtracting frames while keeping track of the resulting
- * states.
- *
- * Notes
- * - The total number of samples is samples_per_channel_ * num_channels_
- * - Stereo data is interleaved starting with the left channel.
- */
-class AudioFrame {
- public:
- // Using constexpr here causes linker errors unless the variable also has an
- // out-of-class definition, which is impractical in this header-only class.
- // (This makes no sense because it compiles as an enum value, which we most
- // certainly cannot take the address of, just fine.) C++17 introduces inline
- // variables which should allow us to switch to constexpr and keep this a
- // header-only class.
- enum : size_t {
- // Stereo, 32 kHz, 60 ms (2 * 32 * 60)
- kMaxDataSizeSamples = 3840,
- kMaxDataSizeBytes = kMaxDataSizeSamples * sizeof(int16_t),
- };
-
- enum VADActivity {
- kVadActive = 0,
- kVadPassive = 1,
- kVadUnknown = 2
- };
- enum SpeechType {
- kNormalSpeech = 0,
- kPLC = 1,
- kCNG = 2,
- kPLCCNG = 3,
- kUndefined = 4
- };
-
- AudioFrame();
-
- // Resets all members to their default state.
- void Reset();
- // Same as Reset(), but leaves mute state unchanged. Muting a frame requires
- // the buffer to be zeroed on the next call to mutable_data(). Callers
- // intending to write to the buffer immediately after Reset() can instead use
- // ResetWithoutMuting() to skip this wasteful zeroing.
- void ResetWithoutMuting();
-
- // TODO(solenberg): Remove once downstream users of AudioFrame have updated.
- RTC_DEPRECATED
- void UpdateFrame(int id, uint32_t timestamp, const int16_t* data,
- size_t samples_per_channel, int sample_rate_hz,
- SpeechType speech_type, VADActivity vad_activity,
- size_t num_channels = 1) {
- RTC_UNUSED(id);
- UpdateFrame(timestamp, data, samples_per_channel, sample_rate_hz,
- speech_type, vad_activity, num_channels);
- }
-
- void UpdateFrame(uint32_t timestamp, const int16_t* data,
- size_t samples_per_channel, int sample_rate_hz,
- SpeechType speech_type, VADActivity vad_activity,
- size_t num_channels = 1);
-
- void CopyFrom(const AudioFrame& src);
-
- // Sets a wall-time clock timestamp in milliseconds to be used for profiling
- // of time between two points in the audio chain.
- // Example:
- // t0: UpdateProfileTimeStamp()
- // t1: ElapsedProfileTimeMs() => t1 - t0 [msec]
- void UpdateProfileTimeStamp();
- // Returns the time difference between now and when UpdateProfileTimeStamp()
- // was last called. Returns -1 if UpdateProfileTimeStamp() has not yet been
- // called.
- int64_t ElapsedProfileTimeMs() const;
-
- // data() returns a zeroed static buffer if the frame is muted.
- // mutable_frame() always returns a non-static buffer; the first call to
- // mutable_frame() zeros the non-static buffer and marks the frame unmuted.
- const int16_t* data() const;
- int16_t* mutable_data();
-
- // Prefer to mute frames using AudioFrameOperations::Mute.
- void Mute();
- // Frame is muted by default.
- bool muted() const;
-
- // These methods are deprecated. Use the functions in
- // webrtc/audio/utility instead. These methods will exists for a
- // short period of time until webrtc clients have updated. See
- // webrtc:6548 for details.
- RTC_DEPRECATED AudioFrame& operator>>=(const int rhs);
- RTC_DEPRECATED AudioFrame& operator+=(const AudioFrame& rhs);
-
- // RTP timestamp of the first sample in the AudioFrame.
- uint32_t timestamp_ = 0;
- // Time since the first frame in milliseconds.
- // -1 represents an uninitialized value.
- int64_t elapsed_time_ms_ = -1;
- // NTP time of the estimated capture time in local timebase in milliseconds.
- // -1 represents an uninitialized value.
- int64_t ntp_time_ms_ = -1;
- size_t samples_per_channel_ = 0;
- int sample_rate_hz_ = 0;
- size_t num_channels_ = 0;
- SpeechType speech_type_ = kUndefined;
- VADActivity vad_activity_ = kVadUnknown;
- // 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, rtc::Optional is not used since it will cause a "complex
- // class/struct needs an explicit out-of-line destructor" build error.
- int64_t profile_timestamp_ms_ = 0;
-
- private:
- // A permamently zeroed out buffer to represent muted frames. This is a
- // header-only class, so the only way to avoid creating a separate empty
- // buffer per translation unit is to wrap a static in an inline function.
- static const int16_t* empty_data() {
- static const int16_t kEmptyData[kMaxDataSizeSamples] = {0};
- static_assert(sizeof(kEmptyData) == kMaxDataSizeBytes, "kMaxDataSizeBytes");
- return kEmptyData;
- }
-
- int16_t data_[kMaxDataSizeSamples];
- bool muted_ = true;
-
- RTC_DISALLOW_COPY_AND_ASSIGN(AudioFrame);
-};
-
-inline AudioFrame::AudioFrame() {
- // Visual Studio doesn't like this in the class definition.
- static_assert(sizeof(data_) == kMaxDataSizeBytes, "kMaxDataSizeBytes");
-}
-
-inline void AudioFrame::Reset() {
- ResetWithoutMuting();
- muted_ = true;
-}
-
-inline void AudioFrame::ResetWithoutMuting() {
- // TODO(wu): Zero is a valid value for |timestamp_|. We should initialize
- // to an invalid value, or add a new member to indicate invalidity.
- timestamp_ = 0;
- elapsed_time_ms_ = -1;
- ntp_time_ms_ = -1;
- samples_per_channel_ = 0;
- sample_rate_hz_ = 0;
- num_channels_ = 0;
- speech_type_ = kUndefined;
- vad_activity_ = kVadUnknown;
- profile_timestamp_ms_ = 0;
-}
-
-inline void AudioFrame::UpdateFrame(uint32_t timestamp,
- const int16_t* data,
- size_t samples_per_channel,
- int sample_rate_hz,
- SpeechType speech_type,
- VADActivity vad_activity,
- size_t num_channels) {
- timestamp_ = timestamp;
- samples_per_channel_ = samples_per_channel;
- sample_rate_hz_ = sample_rate_hz;
- speech_type_ = speech_type;
- vad_activity_ = vad_activity;
- num_channels_ = num_channels;
-
- const size_t length = samples_per_channel * num_channels;
- assert(length <= kMaxDataSizeSamples);
- if (data != nullptr) {
- memcpy(data_, data, sizeof(int16_t) * length);
- muted_ = false;
- } else {
- muted_ = true;
- }
-}
-
-inline void AudioFrame::CopyFrom(const AudioFrame& src) {
- if (this == &src) return;
-
- timestamp_ = src.timestamp_;
- elapsed_time_ms_ = src.elapsed_time_ms_;
- ntp_time_ms_ = src.ntp_time_ms_;
- muted_ = src.muted();
- samples_per_channel_ = src.samples_per_channel_;
- sample_rate_hz_ = src.sample_rate_hz_;
- speech_type_ = src.speech_type_;
- vad_activity_ = src.vad_activity_;
- num_channels_ = src.num_channels_;
-
- const size_t length = samples_per_channel_ * num_channels_;
- assert(length <= kMaxDataSizeSamples);
- if (!src.muted()) {
- memcpy(data_, src.data(), sizeof(int16_t) * length);
- muted_ = false;
- }
-}
-
-inline void AudioFrame::UpdateProfileTimeStamp() {
- profile_timestamp_ms_ = rtc::TimeMillis();
-}
-
-inline int64_t AudioFrame::ElapsedProfileTimeMs() const {
- if (profile_timestamp_ms_ == 0) {
- // Profiling has not been activated.
- return -1;
- }
- return rtc::TimeSince(profile_timestamp_ms_);
-}
-
-inline const int16_t* AudioFrame::data() const {
- return muted_ ? empty_data() : data_;
-}
-
-// TODO(henrik.lundin) Can we skip zeroing the buffer?
-// See https://bugs.chromium.org/p/webrtc/issues/detail?id=5647.
-inline int16_t* AudioFrame::mutable_data() {
- if (muted_) {
- memset(data_, 0, kMaxDataSizeBytes);
- muted_ = false;
- }
- return data_;
-}
-
-inline void AudioFrame::Mute() {
- muted_ = true;
-}
-
-inline bool AudioFrame::muted() const { return muted_; }
-
-inline AudioFrame& AudioFrame::operator>>=(const int rhs) {
- assert((num_channels_ > 0) && (num_channels_ < 3));
- if ((num_channels_ > 2) || (num_channels_ < 1)) return *this;
- if (muted_) return *this;
-
- for (size_t i = 0; i < samples_per_channel_ * num_channels_; i++) {
- data_[i] = static_cast<int16_t>(data_[i] >> rhs);
- }
- return *this;
-}
-
-inline AudioFrame& AudioFrame::operator+=(const AudioFrame& rhs) {
- // Sanity check
- assert((num_channels_ > 0) && (num_channels_ < 3));
- if ((num_channels_ > 2) || (num_channels_ < 1)) return *this;
- if (num_channels_ != rhs.num_channels_) return *this;
-
- bool noPrevData = muted_;
- if (samples_per_channel_ != rhs.samples_per_channel_) {
- if (samples_per_channel_ == 0) {
- // special case we have no data to start with
- samples_per_channel_ = rhs.samples_per_channel_;
- noPrevData = true;
- } else {
- return *this;
- }
- }
-
- if ((vad_activity_ == kVadActive) || rhs.vad_activity_ == kVadActive) {
- vad_activity_ = kVadActive;
- } else if (vad_activity_ == kVadUnknown || rhs.vad_activity_ == kVadUnknown) {
- vad_activity_ = kVadUnknown;
- }
-
- if (speech_type_ != rhs.speech_type_) speech_type_ = kUndefined;
-
- if (!rhs.muted()) {
- muted_ = false;
- if (noPrevData) {
- memcpy(data_, rhs.data(),
- sizeof(int16_t) * rhs.samples_per_channel_ * num_channels_);
- } else {
- // IMPROVEMENT this can be done very fast in assembly
- for (size_t i = 0; i < samples_per_channel_ * num_channels_; i++) {
- int32_t wrap_guard =
- static_cast<int32_t>(data_[i]) + static_cast<int32_t>(rhs.data_[i]);
- data_[i] = rtc::saturated_cast<int16_t>(wrap_guard);
- }
- }
- }
-
- return *this;
-}
-
struct PacedPacketInfo {
PacedPacketInfo() {}
PacedPacketInfo(int probe_cluster_id,
