diff options
Diffstat (limited to 'servers/audio_server.cpp')
| -rw-r--r-- | servers/audio_server.cpp | 37 |
1 files changed, 27 insertions, 10 deletions
diff --git a/servers/audio_server.cpp b/servers/audio_server.cpp index 511a8b3122..b46befd502 100644 --- a/servers/audio_server.cpp +++ b/servers/audio_server.cpp @@ -122,8 +122,10 @@ int AudioDriver::_get_configured_mix_rate() { // In the case of invalid mix rate, let's default to a sensible value.. if (mix_rate <= 0) { +#ifndef WEB_ENABLED WARN_PRINT(vformat("Invalid mix rate of %d, consider reassigning setting \'%s\'. \nDefaulting mix rate to value %d.", mix_rate, audio_driver_setting, AudioDriverManager::DEFAULT_MIX_RATE)); +#endif mix_rate = AudioDriverManager::DEFAULT_MIX_RATE; } @@ -199,8 +201,8 @@ int AudioDriverManager::get_driver_count() { void AudioDriverManager::initialize(int p_driver) { GLOBAL_DEF_RST("audio/driver/enable_input", false); - GLOBAL_DEF_RST("audio/driver/mix_rate", DEFAULT_MIX_RATE); - GLOBAL_DEF_RST("audio/driver/mix_rate.web", 0); // Safer default output_latency for web (use browser default). + GLOBAL_DEF_RST(PropertyInfo(Variant::INT, "audio/driver/mix_rate", PROPERTY_HINT_RANGE, "11025,192000,1,or_greater,suffix:Hz"), DEFAULT_MIX_RATE); + GLOBAL_DEF_RST(PropertyInfo(Variant::INT, "audio/driver/mix_rate.web", PROPERTY_HINT_RANGE, "0,192000,1,or_greater,suffix:Hz"), 0); // Safer default output_latency for web (use browser default). int failed_driver = -1; @@ -271,7 +273,14 @@ void AudioServer::_driver_process(int p_frames, int32_t *p_buffer) { //master master, send to output int cs = master->channels.size(); + + // Take away 1 from the stride, as we are manually incrementing by 1 for stereo. + uintptr_t stride_minus_one = (cs * 2) - 1; + for (int k = 0; k < cs; k++) { + // The destination start for data will be the same in all cases. + int32_t *dest = &p_buffer[from_buf * (cs * 2) + (k * 2)]; + if (master->channels[k].active) { const AudioFrame *buf = master->channels[k].buffer.ptr(); @@ -279,18 +288,25 @@ void AudioServer::_driver_process(int p_frames, int32_t *p_buffer) { float l = CLAMP(buf[from + j].l, -1.0, 1.0); int32_t vl = l * ((1 << 20) - 1); int32_t vl2 = (vl < 0 ? -1 : 1) * (ABS(vl) << 11); - p_buffer[(from_buf + j) * (cs * 2) + k * 2 + 0] = vl2; + *dest = vl2; + dest++; float r = CLAMP(buf[from + j].r, -1.0, 1.0); int32_t vr = r * ((1 << 20) - 1); int32_t vr2 = (vr < 0 ? -1 : 1) * (ABS(vr) << 11); - p_buffer[(from_buf + j) * (cs * 2) + k * 2 + 1] = vr2; + *dest = vr2; + dest += stride_minus_one; } } else { + // Bizarrely, profiling indicates that detecting the common case of cs == 1, + // k == 0, and using memset is SLOWER than setting them individually. + // Perhaps it gets optimized to a faster instruction than memset. for (int j = 0; j < to_copy; j++) { - p_buffer[(from_buf + j) * (cs * 2) + k * 2 + 0] = 0; - p_buffer[(from_buf + j) * (cs * 2) + k * 2 + 1] = 0; + *dest = 0; + dest++; + *dest = 0; + dest += stride_minus_one; } } } @@ -1118,7 +1134,7 @@ float AudioServer::get_playback_speed_scale() const { return playback_speed_scale; } -void AudioServer::start_playback_stream(Ref<AudioStreamPlayback> p_playback, StringName p_bus, Vector<AudioFrame> p_volume_db_vector, float p_start_time, float p_pitch_scale) { +void AudioServer::start_playback_stream(Ref<AudioStreamPlayback> p_playback, const StringName &p_bus, Vector<AudioFrame> p_volume_db_vector, float p_start_time, float p_pitch_scale) { ERR_FAIL_COND(p_playback.is_null()); HashMap<StringName, Vector<AudioFrame>> map; @@ -1127,7 +1143,7 @@ void AudioServer::start_playback_stream(Ref<AudioStreamPlayback> p_playback, Str start_playback_stream(p_playback, map, p_start_time, p_pitch_scale); } -void AudioServer::start_playback_stream(Ref<AudioStreamPlayback> p_playback, HashMap<StringName, Vector<AudioFrame>> p_bus_volumes, float p_start_time, float p_pitch_scale, float p_highshelf_gain, float p_attenuation_cutoff_hz) { +void AudioServer::start_playback_stream(Ref<AudioStreamPlayback> p_playback, const HashMap<StringName, Vector<AudioFrame>> &p_bus_volumes, float p_start_time, float p_pitch_scale, float p_highshelf_gain, float p_attenuation_cutoff_hz) { ERR_FAIL_COND(p_playback.is_null()); AudioStreamPlaybackListNode *playback_node = new AudioStreamPlaybackListNode(); @@ -1147,6 +1163,7 @@ void AudioServer::start_playback_stream(Ref<AudioStreamPlayback> p_playback, Has for (int channel_idx = 0; channel_idx < MAX_CHANNELS_PER_BUS; channel_idx++) { new_bus_details->volume[idx][channel_idx] = pair.value[channel_idx]; } + idx++; } playback_node->bus_details = new_bus_details; playback_node->prev_bus_details = new AudioStreamPlaybackBusDetails(); @@ -1185,7 +1202,7 @@ void AudioServer::stop_playback_stream(Ref<AudioStreamPlayback> p_playback) { } while (!playback_node->state.compare_exchange_strong(old_state, new_state)); } -void AudioServer::set_playback_bus_exclusive(Ref<AudioStreamPlayback> p_playback, StringName p_bus, Vector<AudioFrame> p_volumes) { +void AudioServer::set_playback_bus_exclusive(Ref<AudioStreamPlayback> p_playback, const StringName &p_bus, Vector<AudioFrame> p_volumes) { ERR_FAIL_COND(p_volumes.size() != MAX_CHANNELS_PER_BUS); HashMap<StringName, Vector<AudioFrame>> map; @@ -1194,7 +1211,7 @@ void AudioServer::set_playback_bus_exclusive(Ref<AudioStreamPlayback> p_playback set_playback_bus_volumes_linear(p_playback, map); } -void AudioServer::set_playback_bus_volumes_linear(Ref<AudioStreamPlayback> p_playback, HashMap<StringName, Vector<AudioFrame>> p_bus_volumes) { +void AudioServer::set_playback_bus_volumes_linear(Ref<AudioStreamPlayback> p_playback, const HashMap<StringName, Vector<AudioFrame>> &p_bus_volumes) { ERR_FAIL_COND(p_bus_volumes.size() > MAX_BUSES_PER_PLAYBACK); AudioStreamPlaybackListNode *playback_node = _find_playback_list_node(p_playback); |