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-rw-r--r--content/renderer/media/webrtc_audio_device_impl.cc132
-rw-r--r--content/renderer/media/webrtc_audio_device_impl.h7
-rw-r--r--media/audio/sample_rates.cc23
-rw-r--r--media/audio/sample_rates.h32
-rw-r--r--media/base/audio_decoder_config.cc43
-rw-r--r--media/media.gyp2
6 files changed, 195 insertions, 44 deletions
diff --git a/content/renderer/media/webrtc_audio_device_impl.cc b/content/renderer/media/webrtc_audio_device_impl.cc
index a215eb4..616e263 100644
--- a/content/renderer/media/webrtc_audio_device_impl.cc
+++ b/content/renderer/media/webrtc_audio_device_impl.cc
@@ -5,11 +5,14 @@
#include "content/renderer/media/webrtc_audio_device_impl.h"
#include "base/bind.h"
+#include "base/metrics/histogram.h"
#include "base/string_util.h"
#include "base/win/windows_version.h"
#include "content/renderer/media/audio_hardware.h"
#include "content/renderer/render_thread_impl.h"
#include "media/audio/audio_util.h"
+#include "media/audio/audio_parameters.h"
+#include "media/audio/sample_rates.h"
using media::AudioParameters;
@@ -31,6 +34,94 @@ static int kValidInputRates[] = {48000};
static int kValidOutputRates[] = {48000};
#endif
+namespace {
+
+// Helper enum used for histogramming buffer sizes expressed in number of
+// audio frames. This enumerator covers all supported sizes for all platforms.
+// Example: k480 <=> 480 audio frames <=> 10ms@48kHz.
+// TODO(henrika): can be moved to the media namespace if more clients need it.
+enum AudioFramesPerBuffer {
+ k160,
+ k320,
+ k440, // WebRTC works internally with 440 audio frames at 44.1kHz.
+ k480,
+ k640,
+ k880,
+ k960,
+ k1440,
+ k1920,
+ kUnexpectedAudioBufferSize // Must always be last!
+};
+
+enum HistogramDirection {
+ kAudioOutput,
+ kAudioInput
+};
+
+} // anonymous namespace
+
+// Helper method to convert integral values to their respective enum values
+// above, or kUnexpectedAudioBufferSize if no match exists.
+static AudioFramesPerBuffer AsAudioFramesPerBuffer(int frames_per_buffer) {
+ switch (frames_per_buffer) {
+ case 160: return k160;
+ case 320: return k320;
+ case 440: return k440;
+ case 480: return k480;
+ case 640: return k640;
+ case 880: return k880;
+ case 960: return k960;
+ case 1440: return k1440;
+ case 1920: return k1920;
+ }
+ return kUnexpectedAudioBufferSize;
+}
+
+// Helper method which adds histogram data to be uploaded as part of an
+// UMA logging event. Names: "WebRTC.Audio[Output|Input]SampleRate".
+static void AddHistogramSampleRate(HistogramDirection dir, int param) {
+ media::AudioSampleRate asr = media::AsAudioSampleRate(param);
+ if (asr != media::kUnexpectedAudioSampleRate) {
+ if (dir == kAudioOutput) {
+ UMA_HISTOGRAM_ENUMERATION("WebRTC.AudioOutputSampleRate",
+ asr, media::kUnexpectedAudioSampleRate);
+ } else {
+ UMA_HISTOGRAM_ENUMERATION("WebRTC.AudioInputSampleRate",
+ asr, media::kUnexpectedAudioSampleRate);
+ }
+ } else {
+ // Report unexpected sample rates using a unique histogram name.
+ if (dir == kAudioOutput) {
+ UMA_HISTOGRAM_COUNTS("WebRTC.AudioOutputSampleRateUnexpected", param);
+ } else {
+ UMA_HISTOGRAM_COUNTS("WebRTC.AudioInputSampleRateUnexpected", param);
+ }
+ }
+}
+
+// Helper method which adds histogram data to be uploaded as part of an
+// UMA logging event. Names: "WebRTC.Audio[Output|Input]FramesPerBuffer".
+static void AddHistogramFramesPerBuffer(HistogramDirection dir, int param) {
+ AudioFramesPerBuffer afpb = AsAudioFramesPerBuffer(param);
+ if (afpb != kUnexpectedAudioBufferSize) {
+ if (dir == kAudioOutput) {
+ UMA_HISTOGRAM_ENUMERATION("WebRTC.AudioOutputFramesPerBuffer",
+ afpb, kUnexpectedAudioBufferSize);
+ } else {
+ UMA_HISTOGRAM_ENUMERATION("WebRTC.AudioInputFramesPerBuffer",
+ afpb, kUnexpectedAudioBufferSize);
+ }
+ } else {
+ // Report unexpected sample rates using a unique histogram name.
+ if (dir == kAudioOutput) {
+ UMA_HISTOGRAM_COUNTS("WebRTC.AudioOutputFramesPerBufferUnexpected",
+ param);
+ } else {
+ UMA_HISTOGRAM_COUNTS("WebRTC.AudioInputFramesPerBufferUnexpected", param);
+ }
+ }
+}
+
WebRtcAudioDeviceImpl::WebRtcAudioDeviceImpl()
: ref_count_(0),
render_loop_(base::MessageLoopProxy::current()),
@@ -327,6 +418,7 @@ int32_t WebRtcAudioDeviceImpl::Init() {
// This request is based on a synchronous IPC message.
int out_sample_rate = audio_hardware::GetOutputSampleRate();
DVLOG(1) << "Audio output hardware sample rate: " << out_sample_rate;
+ AddHistogramSampleRate(kAudioOutput, out_sample_rate);
// Verify that the reported output hardware sample rate is supported
// on the current platform.
@@ -342,6 +434,7 @@ int32_t WebRtcAudioDeviceImpl::Init() {
// This request is based on a synchronous IPC message.
int in_sample_rate = audio_hardware::GetInputSampleRate();
DVLOG(1) << "Audio input hardware sample rate: " << in_sample_rate;
+ AddHistogramSampleRate(kAudioInput, in_sample_rate);
// Verify that the reported input hardware sample rate is supported
// on the current platform.
@@ -355,11 +448,10 @@ int32_t WebRtcAudioDeviceImpl::Init() {
// Ask the browser for the default number of audio input channels.
// This request is based on a synchronous IPC message.
- ChannelLayout input_channel_layout =
- audio_hardware::GetInputChannelLayout();
- DVLOG(1) << "Audio input hardware channels: " << input_channel_layout;
-
+ ChannelLayout in_channel_layout = audio_hardware::GetInputChannelLayout();
+ DVLOG(1) << "Audio input hardware channels: " << in_channel_layout;
ChannelLayout out_channel_layout = CHANNEL_LAYOUT_MONO;
+
AudioParameters::Format in_format = AudioParameters::AUDIO_PCM_LINEAR;
int in_buffer_size = 0;
int out_buffer_size = 0;
@@ -451,7 +543,7 @@ int32_t WebRtcAudioDeviceImpl::Init() {
}
// Linux
#elif defined(OS_LINUX) || defined(OS_OPENBSD)
- input_channel_layout = CHANNEL_LAYOUT_STEREO;
+ in_channel_layout = CHANNEL_LAYOUT_STEREO;
out_channel_layout = CHANNEL_LAYOUT_MONO;
// Based on tests using the current ALSA implementation in Chrome, we have
@@ -473,13 +565,20 @@ int32_t WebRtcAudioDeviceImpl::Init() {
out_sample_rate, 16, out_buffer_size);
input_audio_parameters_.Reset(
- in_format, input_channel_layout, in_sample_rate,
+ in_format, in_channel_layout, in_sample_rate,
16, in_buffer_size);
// Create and configure the audio capturing client.
audio_input_device_ = new AudioInputDevice(
input_audio_parameters_, this, this);
+ UMA_HISTOGRAM_ENUMERATION("WebRTC.AudioOutputChannelLayout",
+ out_channel_layout, CHANNEL_LAYOUT_MAX);
+ UMA_HISTOGRAM_ENUMERATION("WebRTC.AudioInputChannelLayout",
+ in_channel_layout, CHANNEL_LAYOUT_MAX);
+ AddHistogramFramesPerBuffer(kAudioOutput, out_buffer_size);
+ AddHistogramFramesPerBuffer(kAudioInput, in_buffer_size);
+
// Create and configure the audio rendering client.
audio_output_device_ = new AudioDevice(output_audio_parameters_, this);
@@ -632,11 +731,15 @@ int32_t WebRtcAudioDeviceImpl::StartPlayout() {
if (!audio_transport_callback_) {
return -1;
}
+
if (playing_) {
// webrtc::VoiceEngine assumes that it is OK to call Start() twice and
// that the call is ignored the second time.
return 0;
}
+
+ start_render_time_ = base::Time::Now();
+
audio_output_device_->Start();
playing_ = true;
return 0;
@@ -648,6 +751,14 @@ int32_t WebRtcAudioDeviceImpl::StopPlayout() {
// webrtc::VoiceEngine assumes that it is OK to call Stop() just in case.
return 0;
}
+
+ // Add histogram data to be uploaded as part of an UMA logging event.
+ // This histogram keeps track of total playout times.
+ if (!start_render_time_.is_null()) {
+ base::TimeDelta render_time = base::Time::Now() - start_render_time_;
+ UMA_HISTOGRAM_LONG_TIMES("WebRTC.AudioRenderTime", render_time);
+ }
+
audio_output_device_->Stop();
playing_ = false;
return 0;
@@ -678,6 +789,8 @@ int32_t WebRtcAudioDeviceImpl::StartRecording() {
return 0;
}
+ start_capture_time_ = base::Time::Now();
+
// Specify the session_id which is mapped to a certain device.
audio_input_device_->SetDevice(session_id_);
audio_input_device_->Start();
@@ -696,6 +809,13 @@ int32_t WebRtcAudioDeviceImpl::StopRecording() {
}
}
+ // Add histogram data to be uploaded as part of an UMA logging event.
+ // This histogram keeps track of total recording times.
+ if (!start_capture_time_.is_null()) {
+ base::TimeDelta capture_time = base::Time::Now() - start_capture_time_;
+ UMA_HISTOGRAM_LONG_TIMES("WebRTC.AudioCaptureTime", capture_time);
+ }
+
audio_input_device_->Stop();
base::AutoLock auto_lock(lock_);
diff --git a/content/renderer/media/webrtc_audio_device_impl.h b/content/renderer/media/webrtc_audio_device_impl.h
index 5d948ca..6d4442a 100644
--- a/content/renderer/media/webrtc_audio_device_impl.h
+++ b/content/renderer/media/webrtc_audio_device_impl.h
@@ -128,8 +128,7 @@
// The adaptive analog mode of the AGC is always enabled for desktop platforms
// in WebRTC.
//
-// Before recording starts, the ADM sets an AGC state in the
-// AudioInputDevice by calling AudioInputDevice::SetAutomaticGainControl(true).
+// Before recording starts, the ADM enables AGC on the AudioInputDevice.
//
// A capture session with AGC is started up as follows (simplified):
//
@@ -450,6 +449,10 @@ class CONTENT_EXPORT WebRtcAudioDeviceImpl
// Local copy of the current Automatic Gain Control state.
bool agc_is_enabled_;
+ // Used for histograms of total recording and playout times.
+ base::Time start_capture_time_;
+ base::Time start_render_time_;
+
DISALLOW_COPY_AND_ASSIGN(WebRtcAudioDeviceImpl);
};
diff --git a/media/audio/sample_rates.cc b/media/audio/sample_rates.cc
new file mode 100644
index 0000000..4bb75a4
--- /dev/null
+++ b/media/audio/sample_rates.cc
@@ -0,0 +1,23 @@
+// Copyright (c) 2012 The Chromium Authors. All rights reserved.
+// Use of this source code is governed by a BSD-style license that can be
+// found in the LICENSE file.
+
+#include "media/audio/sample_rates.h"
+
+namespace media {
+
+AudioSampleRate AsAudioSampleRate(int sample_rate) {
+ switch (sample_rate) {
+ case 8000: return k8000Hz;
+ case 16000: return k16000Hz;
+ case 32000: return k32000Hz;
+ case 48000: return k48000Hz;
+ case 96000: return k96000Hz;
+ case 11025: return k11025Hz;
+ case 22050: return k22050Hz;
+ case 44100: return k44100Hz;
+ }
+ return kUnexpectedAudioSampleRate;
+}
+
+} // namespace media
diff --git a/media/audio/sample_rates.h b/media/audio/sample_rates.h
new file mode 100644
index 0000000..9d47a13
--- /dev/null
+++ b/media/audio/sample_rates.h
@@ -0,0 +1,32 @@
+// Copyright (c) 2012 The Chromium Authors. All rights reserved.
+// Use of this source code is governed by a BSD-style license that can be
+// found in the LICENSE file.
+
+#ifndef MEDIA_AUDIO_SAMPLE_RATES_H_
+#define MEDIA_AUDIO_SAMPLE_RATES_H_
+
+#include "media/base/media_export.h"
+
+namespace media {
+
+// Enumeration used for histogramming sample rates into distinct buckets.
+enum AudioSampleRate {
+ // Do not change the order of these values.
+ k8000Hz = 0,
+ k16000Hz = 1,
+ k32000Hz = 2,
+ k48000Hz = 3,
+ k96000Hz = 4,
+ k11025Hz = 5,
+ k22050Hz = 6,
+ k44100Hz = 7,
+ kUnexpectedAudioSampleRate // Must always be last!
+};
+
+// Helper method to convert integral values to their respective enum values,
+// or kUnexpectedAudioSampleRate if no match exists.
+MEDIA_EXPORT AudioSampleRate AsAudioSampleRate(int sample_rate);
+
+} // namespace media
+
+#endif // MEDIA_AUDIO_SAMPLE_RATES_H_
diff --git a/media/base/audio_decoder_config.cc b/media/base/audio_decoder_config.cc
index 38e0136..5d15e7f 100644
--- a/media/base/audio_decoder_config.cc
+++ b/media/base/audio_decoder_config.cc
@@ -6,6 +6,7 @@
#include "base/logging.h"
#include "base/metrics/histogram.h"
+#include "media/audio/sample_rates.h"
#include "media/base/limits.h"
namespace media {
@@ -28,36 +29,6 @@ AudioDecoderConfig::AudioDecoderConfig(AudioCodec codec,
extra_data, extra_data_size, true);
}
-// Helper enum used only for histogramming samples-per-second. Put
-// commonly-used rates here to get accurate reporting. Uncommon rates are
-// reported in a separate, bucketized, histogram.
-enum AudioSamplesPerSecond {
- k8000Hz,
- k16000Hz,
- k32000Hz,
- k48000Hz,
- k96000Hz,
- k11025Hz,
- k22050Hz,
- k44100Hz,
- kUnexpected // Must always be last!
-};
-
-// Helper method to convert integral values to their respective enum values
-// above, or kUnexpected if no match exists.
-static AudioSamplesPerSecond AsAudioSamplesPerSecond(int samples_per_second) {
- switch (samples_per_second) {
- case 8000: return k8000Hz;
- case 16000: return k16000Hz;
- case 32000: return k32000Hz;
- case 48000: return k48000Hz;
- case 11025: return k11025Hz;
- case 22050: return k22050Hz;
- case 44100: return k44100Hz;
- default: return kUnexpected;
- }
-}
-
void AudioDecoderConfig::Initialize(AudioCodec codec,
int bits_per_channel,
ChannelLayout channel_layout,
@@ -73,12 +44,12 @@ void AudioDecoderConfig::Initialize(AudioCodec codec,
// any values over 32 and even that is huge.
UMA_HISTOGRAM_ENUMERATION("Media.AudioBitsPerChannel", bits_per_channel,
40);
- UMA_HISTOGRAM_ENUMERATION(
- "Media.AudioChannelLayout", channel_layout, CHANNEL_LAYOUT_MAX);
- AudioSamplesPerSecond asps = AsAudioSamplesPerSecond(samples_per_second);
- if (asps != kUnexpected) {
- UMA_HISTOGRAM_ENUMERATION("Media.AudioSamplesPerSecond", asps,
- kUnexpected);
+ UMA_HISTOGRAM_ENUMERATION("Media.AudioChannelLayout", channel_layout,
+ CHANNEL_LAYOUT_MAX);
+ AudioSampleRate asr = media::AsAudioSampleRate(samples_per_second);
+ if (asr != kUnexpectedAudioSampleRate) {
+ UMA_HISTOGRAM_ENUMERATION("Media.AudioSamplesPerSecond", asr,
+ kUnexpectedAudioSampleRate);
} else {
UMA_HISTOGRAM_COUNTS(
"Media.AudioSamplesPerSecondUnexpected", samples_per_second);
diff --git a/media/media.gyp b/media/media.gyp
index 3bd7cc4..673f51e 100644
--- a/media/media.gyp
+++ b/media/media.gyp
@@ -97,6 +97,8 @@
'audio/null_audio_sink.h',
'audio/pulse/pulse_output.cc',
'audio/pulse/pulse_output.h',
+ 'audio/sample_rates.cc',
+ 'audio/sample_rates.h',
'audio/simple_sources.cc',
'audio/simple_sources.h',
'audio/win/audio_low_latency_input_win.cc',
generated by cgit v1.1 at 2025-01-18 00:34:04 +0000