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Adopt absl::string_view in modules/audio_processing/ Bug: webrtc:13579 Change-Id: Idb05a64cfd16aed68d40cd427a6b516caa5e2077 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269387 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Commit-Queue: Ali Tofigh <alito@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37800} 3 年前
Adopt absl::string_view in modules/audio_processing/ Bug: webrtc:13579 Change-Id: Idb05a64cfd16aed68d40cd427a6b516caa5e2077 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269387 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Commit-Queue: Ali Tofigh <alito@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37800} 3 年前
Adopt absl::string_view in modules/audio_processing/ Bug: webrtc:13579 Change-Id: Idb05a64cfd16aed68d40cd427a6b516caa5e2077 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269387 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Commit-Queue: Ali Tofigh <alito@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37800} 3 年前
Refactor AecDump not to rely on QueuedTask Bug: webrtc:14245 Change-Id: Ib41765652745a247da2ae6c2ca6be714de927ca7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268185 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37542} 3 年前
Remove more traces of keyboard mic support from APM The 6-parameter Initialize method is removed. The has_keyboard parameter in the StreamConfig constructor is removed together with the underlying member and helper functions. Bug: chromium:1271981, b/217349489 Change-Id: I7259a114a395f74f735a9c06510c0fc0f0f008e3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250221 Reviewed-by: Sam Zackrisson <saza@google.com> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Auto-Submit: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35908} 4 年前
Ensured that all files in APM are using the webrtc namespace This CL adds namespaces to those files remaining within APM that do not have any such. BUG=webrtc:5298 Change-Id: I710b3d2a3644bea9d4bdffef0d77883b30303338 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171111 Commit-Queue: Per Åhgren <peah@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30850} 6 年前
APM: add AudioProcessingImpl::capture_::applied_input_volume(_changed) The recommended_stream_analog_level() getter is used to retrieve both the applied and the recommended input volume. This behavior is error-prone since the caller must know what is returned based on the point in the code (namely, before/after the AGC has changed the last applied input volume into a recommended level). This CL is a first step to make clarity on which input volume is handled in different parts of APM. Next in the pipeline: make recommended_stream_analog_level() a trivial getter that always returns the recommended level. Main changes: - When recommended_stream_analog_level() is called but set_stream_analog_level() is not called, APM logs an error and returns a fall-back volume (which should not be applied since, when set_stream_analog_level() is not called, no external input volume is expected to be present - When APM is used without calling the *_stream_analog_level() methods (e.g., when the caller does not provide any input volume), the recorded AEC dumps won't store Stream::applied_input_level Other changes: - Removed AudioProcessingImpl::capture_::prev_analog_mic_level - Removed redundant code in GainController2 around detecting input volume changes (already done by APM) - Adapted the audioproc_f and unpack_aecdump tools - Data dumps clean-up: the applied and the recommended input volumes are now recorded in an AGC implementation agnostic way Bug: webrtc:7494, b/241923537 Change-Id: I3cb4a731fd9f3dc19bf6ac679b7ed8c969ea283b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271544 Reviewed-by: Per Åhgren <peah@webrtc.org> Reviewed-by: Hanna Silen <silen@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38054} 3 年前
Refactor AecDump not to rely on QueuedTask Bug: webrtc:14245 Change-Id: Ib41765652745a247da2ae6c2ca6be714de927ca7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268185 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37542} 3 年前
Reformat the WebRTC code base Running clang-format with chromium's style guide. The goal is n-fold: * providing consistency and readability (that's what code guidelines are for) * preventing noise with presubmit checks and git cl format * building on the previous point: making it easier to automatically fix format issues * you name it Please consider using git-hyper-blame to ignore this commit. Bug: webrtc:9340 Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87 Reviewed-on: https://webrtc-review.googlesource.com/81185 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23660}7 年前
In common_audio/ and modules/audio_* replace mock macros with unified MOCK_METHOD macro Bug: webrtc:11564 Change-Id: Ib0ffce4de50a13b018926f6ea2865a2ec2fb2ec7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175621 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31333} 5 年前
Adopt absl::string_view in modules/audio_processing/ Bug: webrtc:13579 Change-Id: Idb05a64cfd16aed68d40cd427a6b516caa5e2077 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269387 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Commit-Queue: Ali Tofigh <alito@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37800} 3 年前