Split audio mixer into interface and implementation.
The AudioMixer is now split in a mixer and audio source interface part, which has moved to webrtc/api, and a default implementation part, which lies in webrtc/modules.
This change makes it possible to create other mixer implementations and is a first step to facilitate passing down a mixer from outside of WebRTC.
It will also create less build dependencies when the new mixer has replaced the old one.
NOTRY=True
TBR=henrik.lundin@webrtc.org
BUG=webrtc:6346
Review-Url: https://codereview.webrtc.org/2411313003
Cr-Commit-Position: refs/heads/master@{#14705}
diff --git a/webrtc/api/BUILD.gn b/webrtc/api/BUILD.gn
index 6de99f7..6424de9 100644
--- a/webrtc/api/BUILD.gn
+++ b/webrtc/api/BUILD.gn
@@ -321,6 +321,17 @@
]
}
+# GYP version: webrtc/api/api.gyp:audio_mixer_api
+rtc_source_set("audio_mixer_api") {
+ sources = [
+ "audio/audio_mixer.h",
+ ]
+
+ deps = [
+ "../base:rtc_base_approved",
+ ]
+}
+
if (rtc_include_tests) {
config("peerconnection_unittests_config") {
# The warnings below are enabled by default. Since GN orders compiler flags
diff --git a/webrtc/api/api.gyp b/webrtc/api/api.gyp
index c431e9e..8a7fe5a 100644
--- a/webrtc/api/api.gyp
+++ b/webrtc/api/api.gyp
@@ -232,5 +232,16 @@
'stats/rtcstatsreport.h',
],
}, # target rtc_stats_api
+ {
+ # GN version: webrtc/api:audio_mixer_api
+ 'target_name': 'audio_mixer_api',
+ 'type': 'static_library',
+ 'dependencies': [
+ '<(webrtc_root)/base/base.gyp:rtc_base_approved',
+ ],
+ 'sources': [
+ 'audio/audio_mixer.h',
+ ],
+ }, # target rtc_stats_api
], # targets
}
diff --git a/webrtc/api/audio/audio_mixer.h b/webrtc/api/audio/audio_mixer.h
new file mode 100644
index 0000000..960adbb
--- /dev/null
+++ b/webrtc/api/audio/audio_mixer.h
@@ -0,0 +1,82 @@
+/*
+ * Copyright (c) 2011 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 WEBRTC_API_AUDIO_AUDIO_MIXER_H_
+#define WEBRTC_API_AUDIO_AUDIO_MIXER_H_
+
+#include <memory>
+
+#include "webrtc/base/refcount.h"
+#include "webrtc/modules/include/module_common_types.h"
+
+namespace webrtc {
+
+// WORK IN PROGRESS
+// This class is under development and is not yet intended for for use outside
+// of WebRtc/Libjingle.
+class AudioMixer : public rtc::RefCountInterface {
+ public:
+ // A callback class that all mixer participants must inherit from/implement.
+ class Source {
+ public:
+ enum class AudioFrameInfo {
+ kNormal, // The samples in audio_frame are valid and should be used.
+ kMuted, // The samples in audio_frame should not be used, but
+ // should be implicitly interpreted as zero. Other
+ // fields in audio_frame may be read and should
+ // contain meaningful values.
+ kError, // The audio_frame will not be used.
+ };
+
+ struct AudioFrameWithInfo {
+ AudioFrame* audio_frame;
+ AudioFrameInfo audio_frame_info;
+ };
+
+ // The implementation of GetAudioFrameWithInfo should update
+ // audio_frame with new audio every time it's called. Implementing
+ // classes are allowed to return the same AudioFrame pointer on
+ // different calls. The pointer must stay valid until the next
+ // mixing call or until this audio source is disconnected from the
+ // mixer. The mixer may modify the contents of the passed
+ // AudioFrame pointer at any time until the next call to
+ // GetAudioFrameWithInfo, or until the source is removed from the
+ // mixer.
+ virtual AudioFrameWithInfo GetAudioFrameWithInfo(int sample_rate_hz) = 0;
+
+ // A way for a mixer implementation to distinguish participants.
+ virtual int Ssrc() = 0;
+
+ protected:
+ virtual ~Source() {}
+ };
+
+ // Since the mixer is reference counted, the destructor may be
+ // called from any thread.
+ ~AudioMixer() override {}
+
+ // Returns true if adding/removing was successful. A source is never
+ // added twice and removal is never attempted if a source has not
+ // been successfully added to the mixer. Addition and removal can
+ // happen on different threads.
+ virtual bool AddSource(Source* audio_source) = 0;
+ virtual bool RemoveSource(Source* audio_source) = 0;
+
+ // Performs mixing by asking registered audio sources for audio. The
+ // mixed result is placed in the provided AudioFrame. Will only be
+ // called from a single thread. The rate and channels arguments
+ // specify the rate and number of channels of the mix result.
+ virtual void Mix(int sample_rate_hz,
+ size_t number_of_channels,
+ AudioFrame* audio_frame_for_mixing) = 0;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_API_AUDIO_AUDIO_MIXER_H_